From bd6d417743d941c3e5eabb21abbcac9737f11061 Mon Sep 17 00:00:00 2001 From: Mike Arthur Date: Tue, 18 Aug 2009 20:37:49 +0100 Subject: ASoC: Add WM8711 CODEC driver The WM8711 or WM8711L (WM8711/L) is a low power stereo DAC with an integrated headphone driver. The WM8711/L is designed specifically for portable MP3 audio and speech players. The WM8711/L is also ideal for MD, CD machines and DAT players. Signed-off-by: Mike Arthur Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8711.c | 685 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8711.h | 42 +++ 4 files changed, 733 insertions(+) create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a01cb0..663840e67766 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8580 if I2C + select SND_SOC_WM8711 if I2C select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI @@ -120,6 +121,9 @@ config SND_SOC_WM8510 config SND_SOC_WM8580 tristate +config SND_SOC_WM8711 + tristate + config SND_SOC_WM8728 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f2653803ede8..19950e998b6b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -17,6 +17,7 @@ snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o +snd-soc-wm8711-objs := wm8711.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o +obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c new file mode 100644 index 000000000000..84ead3f9293f --- /dev/null +++ b/sound/soc/codecs/wm8711.c @@ -0,0 +1,685 @@ +/* + * wm8711.c -- WM8711 ALSA SoC Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur + * + * Based on wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8711.h" + +#define AUDIO_NAME "wm8711" +#define WM8711_VERSION "0.3" + +/* codec private data */ +struct wm8711_priv { + unsigned int sysclk; +}; + +/* + * wm8711 register cache + * We can't read the WM8711 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { + 0x0079, 0x0079, 0x000a, 0x0008, + 0x009f, 0x000a, 0x0000, 0x0000 +}; + +/* + * read wm8711 register cache + */ +static inline unsigned int wm8711_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8711_RESET) + return 0; + if (reg >= WM8711_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8711 register cache + */ +static inline void wm8711_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8711_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8711 register space + */ +static int wm8711_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8753 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8711_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8711_reset(c) wm8711_write(c, WM8711_RESET, 0) + +static const struct snd_kcontrol_new wm8711_snd_controls[] = { + +SOC_DOUBLE_R("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, + 0, 127, 0), +SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, + 7, 1, 0), + +}; + +/* add non dapm controls */ +static int wm8711_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8711_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8711_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Output Mixer */ +static const struct snd_kcontrol_new wm8711_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1, + &wm8711_output_mixer_controls[0], + ARRAY_SIZE(wm8711_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, +}; + +static int wm8711_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets, + ARRAY_SIZE(wm8711_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return 0; +} + +static int wm8711_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8711_priv *wm8711 = codec->private_data; + u16 iface = wm8711_read_reg_cache(codec, WM8711_IFACE) & 0xfffc; + int i = get_coeff(wm8711->sysclk, params_rate(params)); + u16 srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + wm8711_write(codec, WM8711_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + } + + wm8711_write(codec, WM8711_IFACE, iface); + return 0; +} + +static int wm8711_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* set active */ + wm8711_write(codec, WM8711_ACTIVE, 0x0001); + + return 0; +} + +static void wm8711_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + wm8711_write(codec, WM8711_ACTIVE, 0x0); + } +} + +static int wm8711_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8711_read_reg_cache(codec, WM8711_APDIGI) & 0xfff7; + + if (mute) + wm8711_write(codec, WM8711_APDIGI, mute_reg | 0x8); + else + wm8711_write(codec, WM8711_APDIGI, mute_reg); + + return 0; +} + +static int wm8711_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8711_priv *wm8711 = codec->private_data; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + wm8711->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + wm8711_write(codec, WM8711_IFACE, iface); + return 0; +} + + +static int wm8711_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = wm8711_read_reg_cache(codec, WM8711_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + wm8711_write(codec, WM8711_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + wm8711_write(codec, WM8711_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + wm8711_write(codec, WM8711_ACTIVE, 0x0); + wm8711_write(codec, WM8711_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8711_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8711_ops = { + .prepare = wm8711_pcm_prepare, + .hw_params = wm8711_hw_params, + .shutdown = wm8711_shutdown, + .digital_mute = wm8711_mute, + .set_sysclk = wm8711_set_dai_sysclk, + .set_fmt = wm8711_set_dai_fmt, +}; + +struct snd_soc_dai wm8711_dai = { + .name = "WM8711", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8711_RATES, + .formats = WM8711_FORMATS,}, + .ops = &wm8711_ops, +}; +EXPORT_SYMBOL_GPL(wm8711_dai); + +static int wm8711_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8711_write(codec, WM8711_ACTIVE, 0x0); + wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8711_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8711_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the WM8711 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8711_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + int reg, ret = 0; + + codec->name = "WM8711"; + codec->owner = THIS_MODULE; + codec->read = wm8711_read_reg_cache; + codec->write = wm8711_write; + codec->set_bias_level = wm8711_set_bias_level; + codec->dai = &wm8711_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8711_reg); + codec->reg_cache = kmemdup(wm8711_reg, sizeof(wm8711_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8711_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8711: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* set the update bits */ + reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V); + wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100); + reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V); + wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100); + + wm8711_add_controls(codec); + wm8711_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8711: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8711_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8711 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +#define I2C_DRIVERID_WM8711 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8711_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8711_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8711_socdev; + struct wm8711_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->card->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + + i2c_set_clientdata(i2c, codec); + + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + pr_err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8711_init(socdev); + if (ret < 0) { + pr_err("failed to initialise WM8711\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8711_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8711_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8711_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8711_i2c_driver = { + .driver = { + .name = "WM8711 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8711, + .attach_adapter = wm8711_i2c_attach, + .detach_client = wm8711_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8711", + .driver = &wm8711_i2c_driver, +}; +#endif + +static int wm8711_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8711_setup_data *setup; + struct snd_soc_codec *codec; + struct wm8711_priv *wm8711; + int ret = 0; + + pr_info("WM8711 Audio Codec %s", WM8711_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + if (wm8711 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = wm8711; + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8711_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8711_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8711_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec->control_data) + wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8711_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8711 = { + .probe = wm8711_probe, + .remove = wm8711_remove, + .suspend = wm8711_suspend, + .resume = wm8711_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); + +static int __init wm8711_modinit(void) +{ + return snd_soc_register_dai(&wm8711_dai); +} +module_init(wm8711_modinit); + +static void __exit wm8711_exit(void) +{ + snd_soc_unregister_dai(&wm8711_dai); +} +module_exit(wm8711_exit); + +MODULE_DESCRIPTION("ASoC WM8711 driver"); +MODULE_AUTHOR("Mike Arthur"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8711.h b/sound/soc/codecs/wm8711.h new file mode 100644 index 000000000000..381e84a43816 --- /dev/null +++ b/sound/soc/codecs/wm8711.h @@ -0,0 +1,42 @@ +/* + * wm8711.h -- WM8711 Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur + * + * Based on wm8731.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8711_H +#define _WM8711_H + +/* WM8711 register space */ + +#define WM8711_LOUT1V 0x02 +#define WM8711_ROUT1V 0x03 +#define WM8711_APANA 0x04 +#define WM8711_APDIGI 0x05 +#define WM8711_PWR 0x06 +#define WM8711_IFACE 0x07 +#define WM8711_SRATE 0x08 +#define WM8711_ACTIVE 0x09 +#define WM8711_RESET 0x0f + +#define WM8711_CACHEREGNUM 8 + +#define WM8711_SYSCLK 0 +#define WM8711_DAI 0 + +struct wm8711_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8711_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8711; + +#endif -- cgit v1.2.2 From 318b0b8d90326aee6a66c994432eee95c0a9aaea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Aug 2009 20:57:33 +0100 Subject: ASoC: Update WM8711 to driver model registration method Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 297 ++++++++++++++++++++-------------------------- 1 file changed, 129 insertions(+), 168 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 84ead3f9293f..812283e27603 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -28,11 +28,12 @@ #include "wm8711.h" -#define AUDIO_NAME "wm8711" -#define WM8711_VERSION "0.3" +static struct snd_soc_codec *wm8711_codec; /* codec private data */ struct wm8711_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8711_CACHEREGNUM]; unsigned int sysclk; }; @@ -442,241 +443,201 @@ static int wm8711_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8711 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8711_init(struct snd_soc_device *socdev) +static int wm8711_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; - int reg, ret = 0; - - codec->name = "WM8711"; - codec->owner = THIS_MODULE; - codec->read = wm8711_read_reg_cache; - codec->write = wm8711_write; - codec->set_bias_level = wm8711_set_bias_level; - codec->dai = &wm8711_dai; - codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8711_reg); - codec->reg_cache = kmemdup(wm8711_reg, sizeof(wm8711_reg), GFP_KERNEL); + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; - if (codec->reg_cache == NULL) - return -ENOMEM; + if (wm8711_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } - wm8711_reset(codec); + socdev->card->codec = wm8711_codec; + codec = wm8711_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "wm8711: failed to create pcms\n"); + dev_err(codec->dev, "failed to create pcms: %d\n", ret); goto pcm_err; } - /* power on device */ - wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* set the update bits */ - reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V); - wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100); - reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V); - wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100); - - wm8711_add_controls(codec); + snd_soc_add_controls(codec, wm8711_snd_controls, + ARRAY_SIZE(wm8711_snd_controls)); wm8711_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "wm8711: failed to register card\n"); + dev_err(codec->dev, "failed to register card: %d\n", ret); goto card_err; } + return ret; card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); pcm_err: - kfree(codec->reg_cache); return ret; } -static struct snd_soc_device *wm8711_socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -/* - * WM8711 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ -#define I2C_DRIVERID_WM8711 0xfefe /* liam - need a proper id */ - -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; +/* power down chip */ +static int wm8711_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); -static struct i2c_driver wm8711_i2c_driver; -static struct i2c_client client_template; + return 0; +} -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ +struct snd_soc_codec_device soc_codec_dev_wm8711 = { + .probe = wm8711_probe, + .remove = wm8711_remove, + .suspend = wm8711_suspend, + .resume = wm8711_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); -static int wm8711_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8711_register(struct wm8711_priv *wm8711) { - struct snd_soc_device *socdev = wm8711_socdev; - struct wm8711_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec = socdev->card->codec; - struct i2c_client *i2c; int ret; + struct snd_soc_codec *codec = &wm8711->codec; + u16 reg; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; + if (wm8711_codec) { + dev_err(codec->dev, "Another WM8711 is registered\n"); + return -EINVAL; } - i2c_set_clientdata(i2c, codec); + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); - codec->control_data = i2c; + codec->private_data = wm8711; + codec->name = "WM8711"; + codec->owner = THIS_MODULE; + codec->read = wm8711_read_reg_cache; + codec->write = wm8711_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8711_set_bias_level; + codec->dai = &wm8711_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8711_CACHEREGNUM; + codec->reg_cache = &wm8711->reg_cache; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } + memcpy(codec->reg_cache, wm8711_reg, sizeof(wm8711_reg)); - ret = wm8711_init(socdev); + ret = wm8711_reset(codec); if (ret < 0) { - pr_err("failed to initialise WM8711\n"); - goto err; + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; } - return ret; -err: - kfree(codec); - kfree(i2c); - return ret; -} + wm8711_dai.dev = codec->dev; -static int wm8711_i2c_detach(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); + wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits */ + reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V); + wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100); + reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V); + wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100); + + wm8711_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8711_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } - i2c_detach_client(client); - kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8711_i2c_attach(struct i2c_adapter *adap) +static void wm8711_unregister(struct wm8711_priv *wm8711) { - return i2c_probe(adap, &addr_data, wm8711_codec_probe); + wm8711_set_bias_level(&wm8711->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8711_dai); + snd_soc_unregister_codec(&wm8711->codec); + kfree(wm8711); + wm8711_codec = NULL; } -/* corgi i2c codec control layer */ -static struct i2c_driver wm8711_i2c_driver = { - .driver = { - .name = "WM8711 I2C Codec", - .owner = THIS_MODULE, - }, - .id = I2C_DRIVERID_WM8711, - .attach_adapter = wm8711_i2c_attach, - .detach_client = wm8711_i2c_detach, - .command = NULL, -}; - -static struct i2c_client client_template = { - .name = "WM8711", - .driver = &wm8711_i2c_driver, -}; -#endif - -static int wm8711_probe(struct platform_device *pdev) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8711_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8711_setup_data *setup; - struct snd_soc_codec *codec; struct wm8711_priv *wm8711; - int ret = 0; - - pr_info("WM8711 Audio Codec %s", WM8711_VERSION); - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; + struct snd_soc_codec *codec; wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); - if (wm8711 == NULL) { - kfree(codec); + if (wm8711 == NULL) return -ENOMEM; - } - codec->private_data = wm8711; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - wm8711_socdev = socdev; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; - codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8711_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); - } -#else - /* Add other interfaces here */ -#endif - return ret; -} + codec = &wm8711->codec; + codec->hw_write = (hw_write_t)i2c_master_send; -/* power down chip */ -static int wm8711_remove(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; + i2c_set_clientdata(i2c, wm8711); + codec->control_data = i2c; - if (codec->control_data) - wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); + codec->dev = &i2c->dev; - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm8711_i2c_driver); -#endif - kfree(codec->private_data); - kfree(codec); + return wm8711_register(wm8711); +} +static __devexit int wm8711_i2c_remove(struct i2c_client *client) +{ + struct wm8711_priv *wm8711 = i2c_get_clientdata(client); + wm8711_unregister(wm8711); return 0; } -struct snd_soc_codec_device soc_codec_dev_wm8711 = { - .probe = wm8711_probe, - .remove = wm8711_remove, - .suspend = wm8711_suspend, - .resume = wm8711_resume, +static const struct i2c_device_id wm8711_i2c_id[] = { + { "wm8711", 0 }, + { } }; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); +MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); + +static struct i2c_driver wm8711_i2c_driver = { + .driver = { + .name = "WM8711 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8711_i2c_probe, + .remove = __devexit_p(wm8711_i2c_remove), + .id_table = wm8711_i2c_id, +}; +#endif static int __init wm8711_modinit(void) { - return snd_soc_register_dai(&wm8711_dai); + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8711_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n", + ret); + } +#endif + return 0; } module_init(wm8711_modinit); static void __exit wm8711_exit(void) { - snd_soc_unregister_dai(&wm8711_dai); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8711_i2c_driver); +#endif } module_exit(wm8711_exit); -- cgit v1.2.2 From d97d2e35b903b11dc6f7f8fcbe9a82fd8929e234 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Aug 2009 21:12:30 +0100 Subject: ASoC: Factor out WM8711 cache I/O Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 114 ++++++++++++++++------------------------------ 1 file changed, 38 insertions(+), 76 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 812283e27603..c7b1af89297b 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -48,55 +48,7 @@ static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { 0x009f, 0x000a, 0x0000, 0x0000 }; -/* - * read wm8711 register cache - */ -static inline unsigned int wm8711_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8711_RESET) - return 0; - if (reg >= WM8711_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8711 register cache - */ -static inline void wm8711_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8711_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the WM8711 register space - */ -static int wm8711_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8711_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8711_reset(c) wm8711_write(c, WM8711_RESET, 0) +#define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) static const struct snd_kcontrol_new wm8711_snd_controls[] = { @@ -224,12 +176,12 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8711_priv *wm8711 = codec->private_data; - u16 iface = wm8711_read_reg_cache(codec, WM8711_IFACE) & 0xfffc; + u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc; int i = get_coeff(wm8711->sysclk, params_rate(params)); u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | coeff_div[i].usb; - wm8711_write(codec, WM8711_SRATE, srate); + snd_soc_write(codec, WM8711_SRATE, srate); /* bit size */ switch (params_format(params)) { @@ -243,7 +195,7 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream, break; } - wm8711_write(codec, WM8711_IFACE, iface); + snd_soc_write(codec, WM8711_IFACE, iface); return 0; } @@ -253,7 +205,7 @@ static int wm8711_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; /* set active */ - wm8711_write(codec, WM8711_ACTIVE, 0x0001); + snd_soc_write(codec, WM8711_ACTIVE, 0x0001); return 0; } @@ -266,19 +218,19 @@ static void wm8711_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - wm8711_write(codec, WM8711_ACTIVE, 0x0); + snd_soc_write(codec, WM8711_ACTIVE, 0x0); } } static int wm8711_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8711_read_reg_cache(codec, WM8711_APDIGI) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8711_APDIGI) & 0xfff7; if (mute) - wm8711_write(codec, WM8711_APDIGI, mute_reg | 0x8); + snd_soc_write(codec, WM8711_APDIGI, mute_reg | 0x8); else - wm8711_write(codec, WM8711_APDIGI, mute_reg); + snd_soc_write(codec, WM8711_APDIGI, mute_reg); return 0; } @@ -356,7 +308,7 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - wm8711_write(codec, WM8711_IFACE, iface); + snd_soc_write(codec, WM8711_IFACE, iface); return 0; } @@ -364,20 +316,20 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8711_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = wm8711_read_reg_cache(codec, WM8711_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: - wm8711_write(codec, WM8711_PWR, reg); + snd_soc_write(codec, WM8711_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - wm8711_write(codec, WM8711_PWR, reg | 0x0040); + snd_soc_write(codec, WM8711_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - wm8711_write(codec, WM8711_ACTIVE, 0x0); - wm8711_write(codec, WM8711_PWR, 0xffff); + snd_soc_write(codec, WM8711_ACTIVE, 0x0); + snd_soc_write(codec, WM8711_PWR, 0xffff); break; } codec->bias_level = level; @@ -419,7 +371,7 @@ static int wm8711_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - wm8711_write(codec, WM8711_ACTIVE, 0x0); + snd_soc_write(codec, WM8711_ACTIVE, 0x0); wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -501,7 +453,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8711 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); -static int wm8711_register(struct wm8711_priv *wm8711) +static int wm8711_register(struct wm8711_priv *wm8711, + enum snd_soc_control_type control) { int ret; struct snd_soc_codec *codec = &wm8711->codec; @@ -519,8 +472,6 @@ static int wm8711_register(struct wm8711_priv *wm8711) codec->private_data = wm8711; codec->name = "WM8711"; codec->owner = THIS_MODULE; - codec->read = wm8711_read_reg_cache; - codec->write = wm8711_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8711_set_bias_level; codec->dai = &wm8711_dai; @@ -530,10 +481,16 @@ static int wm8711_register(struct wm8711_priv *wm8711) memcpy(codec->reg_cache, wm8711_reg, sizeof(wm8711_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8711_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err; } wm8711_dai.dev = codec->dev; @@ -541,27 +498,32 @@ static int wm8711_register(struct wm8711_priv *wm8711) wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V); - wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100); - reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V); - wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8711_LOUT1V); + snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8711_ROUT1V); + snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100); wm8711_codec = codec; ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm8711_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8711); + return ret; } static void wm8711_unregister(struct wm8711_priv *wm8711) @@ -592,7 +554,7 @@ static __devinit int wm8711_i2c_probe(struct i2c_client *i2c, codec->dev = &i2c->dev; - return wm8711_register(wm8711); + return wm8711_register(wm8711, SND_SOC_I2C); } static __devexit int wm8711_i2c_remove(struct i2c_client *client) -- cgit v1.2.2 From 08aff8cd7a8568588d460c4bf8875a492d430314 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Aug 2009 21:15:14 +0100 Subject: ASoC: Add SPI support to WM8711 Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8711.c | 66 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 67 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a5cfa78eb166..20ebf7437f98 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,7 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8523 if I2C select SND_SOC_WM8580 if I2C - select SND_SOC_WM8711 if I2C + select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index c7b1af89297b..1a7fca7d1ef9 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -535,6 +535,62 @@ static void wm8711_unregister(struct wm8711_priv *wm8711) wm8711_codec = NULL; } +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8711_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8711_priv *wm8711; + + wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + if (wm8711 == NULL) + return -ENOMEM; + + codec = &wm8711->codec; + codec->control_data = spi; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, wm8711); + + return wm8711_register(wm8711, SND_SOC_SPI); +} + +static int __devexit wm8711_spi_remove(struct spi_device *spi) +{ + struct wm8711_priv *wm8711 = dev_get_drvdata(&spi->dev); + + wm8711_unregister(wm8711); + + return 0; +} + +#ifdef CONFIG_PM +static int wm8711_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8711_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8711_spi_suspend NULL +#define wm8711_spi_resume NULL +#endif + +static struct spi_driver wm8711_spi_driver = { + .driver = { + .name = "wm8711", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8711_spi_probe, + .suspend = wm8711_spi_suspend, + .resume = wm8711_spi_resume, + .remove = __devexit_p(wm8711_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8711_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -590,6 +646,13 @@ static int __init wm8711_modinit(void) printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n", ret); } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8731_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n", + ret); + } #endif return 0; } @@ -600,6 +663,9 @@ static void __exit wm8711_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8711_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8731_spi_driver); +#endif } module_exit(wm8711_exit); -- cgit v1.2.2 From 431f7771774e8f37dde5acb3f7c4c5f6fa1109e3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Aug 2009 21:17:34 +0100 Subject: ASoC: WM8711 minor cleanups Coding style changes only. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 25 +++---------------------- 1 file changed, 3 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 1a7fca7d1ef9..f98c2bc32f9e 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -59,22 +59,6 @@ SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, }; -/* add non dapm controls */ -static int wm8711_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8711_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8711_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new wm8711_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0), @@ -336,11 +320,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define WM8711_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) +#define WM8711_RATES SNDRV_PCM_RATE_8000_96000 #define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -361,7 +341,8 @@ struct snd_soc_dai wm8711_dai = { .channels_min = 1, .channels_max = 2, .rates = WM8711_RATES, - .formats = WM8711_FORMATS,}, + .formats = WM8711_FORMATS, + }, .ops = &wm8711_ops, }; EXPORT_SYMBOL_GPL(wm8711_dai); -- cgit v1.2.2 From b5ab887e6dfa12c32ef39827da47d5d021320a3f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Aug 2009 21:29:31 +0100 Subject: ASoC: Add TLV information to WM8711 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index f98c2bc32f9e..ae083eb92fb7 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include "wm8711.h" @@ -50,10 +51,12 @@ static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { #define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); + static const struct snd_kcontrol_new wm8711_snd_controls[] = { -SOC_DOUBLE_R("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, - 0, 127, 0), +SOC_DOUBLE_R_TLV("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, + 0, 127, 0, out_tlv), SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, 7, 1, 0), -- cgit v1.2.2 From 85488037bb9b533b064be66412dbe1dbcd2734d9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 5 Sep 2009 18:52:16 +0100 Subject: ASoC: Add source argument to PLL configuration More and more devices feature PLLs and FLLs with the ability to select between multiple input clocks. In order to better support these devices a new argument, source, has been added to the set_pll() configuration API. Using set_clkdiv() is often difficult due to the need to stop the PLL/FLL before any reconfiguration can be done. Signed-off-by: Mark Brown --- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 3 ++- sound/soc/codecs/wm8510.c | 4 ++-- sound/soc/codecs/wm8580.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- sound/soc/codecs/wm8900.c | 4 ++-- sound/soc/codecs/wm8940.c | 4 ++-- sound/soc/codecs/wm8960.c | 4 ++-- sound/soc/codecs/wm8974.c | 4 ++-- sound/soc/codecs/wm8990.c | 4 ++-- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9713.c | 4 ++-- sound/soc/imx/mx27vis_wm8974.c | 2 +- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 ++-- sound/soc/pxa/zylonite.c | 5 +++-- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/soc-core.c | 8 +++++--- 20 files changed, 37 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 9eb610c2ba91..9df4c68ef000 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -268,7 +268,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - ret = snd_soc_dai_set_pll(codec_dai, 0, + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_get_rate(CODEC_CLK), pll_out); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 71c9c4bb2632..0ebd99b7493e 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1101,7 +1101,7 @@ static inline int fll_factors(struct _fll_div *fll_div, unsigned int input, } static int wm8350_set_fll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, + int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index b9ef4d915221..9cb8e50f0fbb 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1011,7 +1011,8 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, } static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - unsigned int freq_in, unsigned int freq_out) + int source, unsigned int freq_in, + unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8400_priv *wm8400 = codec->private_data; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 060d5d06ba95..5702435af81b 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -271,8 +271,8 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c78150..3be5c0b2552c 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -407,8 +407,8 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, return 0; } -static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { int offset; struct snd_soc_codec *codec = codec_dai->codec; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d80d414cfbbd..f60f3a02d1f8 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -723,8 +723,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; int offset; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 5e9c855c0036..882604ef768c 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -814,8 +814,8 @@ reenable: return 0; } -static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out); } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index da97aae475a2..914d788a2b76 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -536,8 +536,8 @@ static void pll_factors(unsigned int target, unsigned int source) } /* Untested at the moment */ -static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f59703be61c8..416fb3c17018 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -540,8 +540,8 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index d8a013ab3177..fa4d85bd048b 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -329,8 +329,8 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 2d702db4131d..f657e9a5fe26 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -972,8 +972,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { u16 reg; struct snd_soc_codec *codec = codec_dai->codec; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d9987999e92c..6b32a2852603 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -422,7 +422,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, return 0; } -static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, +static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, unsigned int Fref, unsigned int Fout) { struct snd_soc_codec *codec = dai->codec; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index abed37acf787..ca3d449ed89e 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -800,8 +800,8 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; return wm9713_set_pll(codec, pll_id, freq_in, freq_out); diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c index e4dcb539108a..0267d2d91685 100644 --- a/sound/soc/imx/mx27vis_wm8974.c +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -157,7 +157,7 @@ static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, /* codec PLL input is 25 MHz */ - ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, + ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, 25000000, pll_out); if (ret < 0) { printk(KERN_ERR "Error when setting PLL input\n"); diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 9f7c61e23daf..4c8d99a8d386 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -213,7 +213,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, return ret; /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 5b9ed6464789..57f201c94ca8 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -305,8 +305,8 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, /* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 9a386b4c4ed1..dd678ae24398 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -74,7 +74,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_codec *codec) { if (clk_pout) - snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0); + snd_soc_dai_set_pll(&codec->dai[0], 0, 0, + clk_get_rate(pout), 0); snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, ARRAY_SIZE(zylonite_dapm_widgets)); @@ -128,7 +129,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 0c52e36ddd87..6ddd1b3b16b3 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -119,7 +119,7 @@ static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream, return ret; /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, iis_clkrate / 4, pll_out); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 906709e6dd5f..16009eba9cba 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -137,7 +137,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, return ret; /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, iis_clkrate / 4, pll_out); if (ret < 0) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad2a97e..05fdc8023da4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2197,16 +2197,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); * snd_soc_dai_set_pll - configure DAI PLL. * @dai: DAI * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL * @freq_in: PLL input clock frequency in Hz * @freq_out: requested PLL output clock frequency in Hz * * Configures and enables PLL to generate output clock based on input clock. */ -int snd_soc_dai_set_pll(struct snd_soc_dai *dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) { if (dai->ops && dai->ops->set_pll) - return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); + return dai->ops->set_pll(dai, pll_id, source, + freq_in, freq_out); else return -EINVAL; } -- cgit v1.2.2 From 341c9b84bc01040bd5c75140303e32f6b10098f3 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Mon, 7 Sep 2009 12:04:37 +0900 Subject: ASoC: Factor out I2C 8 bit address 8 bit data I/O This patch is for the AK4671 codec driver using this format. Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c8ceddc2a26c..404231ee8780 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -77,6 +77,35 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data, #define snd_soc_7_9_spi_write NULL #endif +static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -151,6 +180,7 @@ static struct { unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read }, + { 8, 8, snd_soc_8_8_write, NULL, snd_soc_8_8_read, NULL }, { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read, snd_soc_8_16_read_i2c }, }; -- cgit v1.2.2 From 215edda3adf502ccdf3a358ab35b616e7abd25ff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Sep 2009 18:59:05 +0100 Subject: ASoC: Allow per-route connectedness checks for supplies Some chips with complex internal supply (particularly clocking) arragements may have multiple options for some of the supply connections. Since these don't affect user-visible audio routing the expectation would be that they would be managed automatically by one of the drivers. Support these users by allowing routes to have a connected function which is queried before the connectedness of the path is checked as normal. Currently this is only done for supplies, other widgets could be supported but are not currently since the expectation for them is that audio routing will be under the control of userspace. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d8b08ef8731..37f7adeae323 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -718,6 +718,10 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + if (path->connected && + !path->connected(path->source, path->sink)) + continue; + if (path->sink && path->sink->power_check && path->sink->power_check(path->sink)) { power = 1; @@ -1136,6 +1140,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->sname); list_for_each_entry(p, &w->sources, list_sink) { + if (p->connected && !p->connected(w, p->sink)) + continue; + if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, " in %s %s\n", @@ -1143,6 +1150,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, p->source->name); } list_for_each_entry(p, &w->sinks, list_source) { + if (p->connected && !p->connected(w, p->sink)) + continue; + if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, " out %s %s\n", @@ -1385,10 +1395,13 @@ int snd_soc_dapm_sync(struct snd_soc_codec *codec) EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, - const char *sink, const char *control, const char *source) + const struct snd_soc_dapm_route *route) { struct snd_soc_dapm_path *path; struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; + const char *sink = route->sink; + const char *control = route->control; + const char *source = route->source; int ret = 0; /* find src and dest widgets */ @@ -1412,6 +1425,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, path->source = wsource; path->sink = wsink; + path->connected = route->connected; INIT_LIST_HEAD(&path->list); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); @@ -1512,8 +1526,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, int i, ret; for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(codec, route->sink, - route->control, route->source); + ret = snd_soc_dapm_add_route(codec, route); if (ret < 0) { printk(KERN_ERR "Failed to add route %s->%s\n", route->source, -- cgit v1.2.2 From 2312fd8f6b252b7d3c1d74b20c75b7bff98bab65 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Thu, 10 Sep 2009 00:12:43 +0900 Subject: ASoC: AK4671: add ak4671 codec driver The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier, Receiver-Amplifier and Headphone-Amplifier. The datasheet for the ak4671 can find at the following url: http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4671.c | 825 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 +++++++++ 4 files changed, 987 insertions(+) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93af3b0..a2bb659ec184 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -96,6 +97,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_AK4671 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28486ba..13f7b4f2a152 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c new file mode 100644 index 000000000000..b61214d1c5de --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,825 @@ +/* + * ak4671.c -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ak4671.h" + +static struct snd_soc_codec *ak4671_codec; + +/* codec private data */ +struct ak4671_priv { + struct snd_soc_codec codec; + u8 reg_cache[AK4671_CACHEREGNUM]; +}; + +/* ak4671 register cache & default register settings */ +static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { + 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + 0x02, /* AK4671_FORMAT_SELECT (0x03) */ + 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ + 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ + 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ + 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ + 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + 0x00, /* this register not used */ + 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ + 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ + 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ + 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ + 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ + 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ + 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ + 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ + 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ + 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ + 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ + 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ + 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ + 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ + 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ + 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ + 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ + 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ + 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +}; + +/* + * LOUT1/ROUT1 output volume control: + * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB) + */ +static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1); + +/* + * LOUT2/ROUT2 output volume control: + * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB) + */ +static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1); + +/* + * LOUT3/ROUT3 output volume control: + * from -6 to 3 dB in 3 dB steps + */ +static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0); + +/* + * Mic amp gain control: + * from -15 to 30 dB in 3 dB steps + * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not + * available + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new ak4671_snd_controls[] = { + /* Common playback gain controls */ + SOC_SINGLE_TLV("Line Output1 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv), + SOC_SINGLE_TLV("Headphone Output2 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv), + SOC_SINGLE_TLV("Line Output3 Playback Volume", + AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv), + + /* Common capture gain controls */ + SOC_DOUBLE_TLV("Mic Amp Capture Volume", + AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv), +}; + +/* event handlers */ +static int ak4671_out2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + } + + return 0; +} + +/* Output Mixers */ +static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +/* Input MUXs */ +static const char *ak4671_lin_mux_texts[] = + {"LIN1", "LIN2", "LIN3", "LIN4"}; +static const struct soc_enum ak4671_lin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, + ARRAY_SIZE(ak4671_lin_mux_texts), + ak4671_lin_mux_texts); +static const struct snd_kcontrol_new ak4671_lin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); + +static const char *ak4671_rin_mux_texts[] = + {"RIN1", "RIN2", "RIN3", "RIN4"}; +static const struct soc_enum ak4671_rin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, + ARRAY_SIZE(ak4671_rin_mux_texts), + ak4671_rin_mux_texts); +static const struct snd_kcontrol_new ak4671_rin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); + +static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("LIN4"), + SND_SOC_DAPM_INPUT("RIN4"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 6, 0), + SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 7, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 4, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 5, 0), + + /* PGA */ + SND_SOC_DAPM_PGA("LOUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("ROUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0, + &ak4671_lout1_mixer_controls[0], + ARRAY_SIZE(ak4671_lout1_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0, + &ak4671_rout1_mixer_controls[0], + ARRAY_SIZE(ak4671_rout1_mixer_controls)), + SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 0, 0, &ak4671_lout2_mixer_controls[0], + ARRAY_SIZE(ak4671_lout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 1, 0, &ak4671_rout2_mixer_controls[0], + ARRAY_SIZE(ak4671_rout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0, + &ak4671_lout3_mixer_controls[0], + ARRAY_SIZE(ak4671_lout3_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0, + &ak4671_rout3_mixer_controls[0], + ARRAY_SIZE(ak4671_rout3_mixer_controls)), + + /* Input MUXs */ + SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0, + &ak4671_lin_mux_control), + SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0, + &ak4671_rin_mux_control), + + /* Mic Power */ + SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0), + + /* Supply */ + SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DAC Left", "NULL", "PMPLL"}, + {"DAC Right", "NULL", "PMPLL"}, + {"ADC Left", "NULL", "PMPLL"}, + {"ADC Right", "NULL", "PMPLL"}, + + /* Outputs */ + {"LOUT1", "NULL", "LOUT1 Mixer"}, + {"ROUT1", "NULL", "ROUT1 Mixer"}, + {"LOUT2", "NULL", "LOUT2 Mix Amp"}, + {"ROUT2", "NULL", "ROUT2 Mix Amp"}, + {"LOUT3", "NULL", "LOUT3 Mixer"}, + {"ROUT3", "NULL", "ROUT3 Mixer"}, + + {"LOUT1 Mixer", "DACL", "DAC Left"}, + {"ROUT1 Mixer", "DACR", "DAC Right"}, + {"LOUT2 Mixer", "DACHL", "DAC Left"}, + {"ROUT2 Mixer", "DACHR", "DAC Right"}, + {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT3 Mixer", "DACSL", "DAC Left"}, + {"ROUT3 Mixer", "DACSR", "DAC Right"}, + + /* Inputs */ + {"LIN MUX", "LIN1", "LIN1"}, + {"LIN MUX", "LIN2", "LIN2"}, + {"LIN MUX", "LIN3", "LIN3"}, + {"LIN MUX", "LIN4", "LIN4"}, + + {"RIN MUX", "RIN1", "RIN1"}, + {"RIN MUX", "RIN2", "RIN2"}, + {"RIN MUX", "RIN3", "RIN3"}, + {"RIN MUX", "RIN4", "RIN4"}, + + {"LIN1", NULL, "Mic Bias"}, + {"RIN1", NULL, "Mic Bias"}, + {"LIN2", NULL, "Mic Bias"}, + {"RIN2", NULL, "Mic Bias"}, + + {"ADC Left", "NULL", "LIN MUX"}, + {"ADC Right", "NULL", "RIN MUX"}, + + /* Analog Loops */ + {"LIN1 Mixing Circuit", "NULL", "LIN1"}, + {"RIN1 Mixing Circuit", "NULL", "RIN1"}, + {"LIN2 Mixing Circuit", "NULL", "LIN2"}, + {"RIN2 Mixing Circuit", "NULL", "RIN2"}, + {"LIN3 Mixing Circuit", "NULL", "LIN3"}, + {"RIN3 Mixing Circuit", "NULL", "RIN3"}, + {"LIN4 Mixing Circuit", "NULL", "LIN4"}, + {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + + {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, +}; + +static int ak4671_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4671_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 fs; + + fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + fs &= ~AK4671_FS; + + switch (params_rate(params)) { + case 8000: + fs |= AK4671_FS_8KHZ; + break; + case 12000: + fs |= AK4671_FS_12KHZ; + break; + case 16000: + fs |= AK4671_FS_16KHZ; + break; + case 24000: + fs |= AK4671_FS_24KHZ; + break; + case 11025: + fs |= AK4671_FS_11_025KHZ; + break; + case 22050: + fs |= AK4671_FS_22_05KHZ; + break; + case 32000: + fs |= AK4671_FS_32KHZ; + break; + case 44100: + fs |= AK4671_FS_44_1KHZ; + break; + case 48000: + fs |= AK4671_FS_48KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs); + + return 0; +} + +static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u8 pll; + + pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + pll &= ~AK4671_PLL; + + switch (freq) { + case 11289600: + pll |= AK4671_PLL_11_2896MHZ; + break; + case 12000000: + pll |= AK4671_PLL_12MHZ; + break; + case 12288000: + pll |= AK4671_PLL_12_288MHZ; + break; + case 13000000: + pll |= AK4671_PLL_13MHZ; + break; + case 13500000: + pll |= AK4671_PLL_13_5MHZ; + break; + case 19200000: + pll |= AK4671_PLL_19_2MHZ; + break; + case 24000000: + pll |= AK4671_PLL_24MHZ; + break; + case 26000000: + pll |= AK4671_PLL_26MHZ; + break; + case 27000000: + pll |= AK4671_PLL_27MHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll); + + return 0; +} + +static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = snd_soc_read(codec, AK4671_FORMAT_SELECT); + format &= ~AK4671_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK4671_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK4671_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_A: + format |= AK4671_DIF_DSP_MODE; + format |= AK4671_BCKP; + format |= AK4671_MSBS; + break; + default: + return -EINVAL; + } + + /* set mode and format */ + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode); + snd_soc_write(codec, AK4671_FORMAT_SELECT, format); + + return 0; +} + +static int ak4671_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops ak4671_dai_ops = { + .hw_params = ak4671_hw_params, + .set_sysclk = ak4671_set_dai_sysclk, + .set_fmt = ak4671_set_dai_fmt, +}; + +struct snd_soc_dai ak4671_dai = { + .name = "AK4671", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .ops = &ak4671_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4671_dai); + +static int ak4671_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ak4671_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4671_codec; + codec = ak4671_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ak4671_snd_controls, + ARRAY_SIZE(ak4671_snd_controls)); + ak4671_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int ak4671_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4671; + codec->name = "AK4671"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ak4671_set_bias_level; + codec->dai = &ak4671_dai; + codec->num_dai = 1; + codec->reg_cache_size = AK4671_CACHEREGNUM; + codec->reg_cache = &ak4671->reg_cache; + + memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ak4671_dai.dev = codec->dev; + ak4671_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(ak4671); + return ret; +} + +static void ak4671_unregister(struct ak4671_priv *ak4671) +{ + ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ak4671_dai); + snd_soc_unregister_codec(&ak4671->codec); + kfree(ak4671); + ak4671_codec = NULL; +} + +static int __devinit ak4671_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct ak4671_priv *ak4671; + struct snd_soc_codec *codec; + + ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + if (ak4671 == NULL) + return -ENOMEM; + + codec = &ak4671->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(client, ak4671); + codec->control_data = client; + + codec->dev = &client->dev; + + return ak4671_register(ak4671, SND_SOC_I2C); +} + +static __devexit int ak4671_i2c_remove(struct i2c_client *client) +{ + struct ak4671_priv *ak4671 = i2c_get_clientdata(client); + + ak4671_unregister(ak4671); + + return 0; +} + +static const struct i2c_device_id ak4671_i2c_id[] = { + { "ak4671", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); + +static struct i2c_driver ak4671_i2c_driver = { + .driver = { + .name = "ak4671", + .owner = THIS_MODULE, + }, + .probe = ak4671_i2c_probe, + .remove = __devexit_p(ak4671_i2c_remove), + .id_table = ak4671_i2c_id, +}; + +static int __init ak4671_modinit(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_modinit); + +static void __exit ak4671_exit(void) +{ + i2c_del_driver(&ak4671_i2c_driver); +} +module_exit(ak4671_exit); + +MODULE_DESCRIPTION("ASoC AK4671 codec driver"); +MODULE_AUTHOR("Joonyoung Shim "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h new file mode 100644 index 000000000000..e2fad964e88b --- /dev/null +++ b/sound/soc/codecs/ak4671.h @@ -0,0 +1,156 @@ +/* + * ak4671.h -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _AK4671_H +#define _AK4671_H + +#define AK4671_AD_DA_POWER_MANAGEMENT 0x00 +#define AK4671_PLL_MODE_SELECT0 0x01 +#define AK4671_PLL_MODE_SELECT1 0x02 +#define AK4671_FORMAT_SELECT 0x03 +#define AK4671_MIC_SIGNAL_SELECT 0x04 +#define AK4671_MIC_AMP_GAIN 0x05 +#define AK4671_MIXING_POWER_MANAGEMENT0 0x06 +#define AK4671_MIXING_POWER_MANAGEMENT1 0x07 +#define AK4671_OUTPUT_VOLUME_CONTROL 0x08 +#define AK4671_LOUT1_SIGNAL_SELECT 0x09 +#define AK4671_ROUT1_SIGNAL_SELECT 0x0a +#define AK4671_LOUT2_SIGNAL_SELECT 0x0b +#define AK4671_ROUT2_SIGNAL_SELECT 0x0c +#define AK4671_LOUT3_SIGNAL_SELECT 0x0d +#define AK4671_ROUT3_SIGNAL_SELECT 0x0e +#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f +#define AK4671_LOUT2_POWER_MANAGERMENT 0x10 +#define AK4671_LOUT3_POWER_MANAGERMENT 0x11 +#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12 +#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13 +#define AK4671_ALC_REFERENCE_SELECT 0x14 +#define AK4671_DIGITAL_MIXING_CONTROL 0x15 +#define AK4671_ALC_TIMER_SELECT 0x16 +#define AK4671_ALC_MODE_CONTROL 0x17 +#define AK4671_MODE_CONTROL1 0x18 +#define AK4671_MODE_CONTROL2 0x19 +#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a +#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b +#define AK4671_SIDETONE_A_CONTROL 0x1c +#define AK4671_DIGITAL_FILTER_SELECT 0x1d +#define AK4671_FIL3_COEFFICIENT0 0x1e +#define AK4671_FIL3_COEFFICIENT1 0x1f +#define AK4671_FIL3_COEFFICIENT2 0x20 +#define AK4671_FIL3_COEFFICIENT3 0x21 +#define AK4671_EQ_COEFFICIENT0 0x22 +#define AK4671_EQ_COEFFICIENT1 0x23 +#define AK4671_EQ_COEFFICIENT2 0x24 +#define AK4671_EQ_COEFFICIENT3 0x25 +#define AK4671_EQ_COEFFICIENT4 0x26 +#define AK4671_EQ_COEFFICIENT5 0x27 +#define AK4671_FIL1_COEFFICIENT0 0x28 +#define AK4671_FIL1_COEFFICIENT1 0x29 +#define AK4671_FIL1_COEFFICIENT2 0x2a +#define AK4671_FIL1_COEFFICIENT3 0x2b +#define AK4671_FIL2_COEFFICIENT0 0x2c +#define AK4671_FIL2_COEFFICIENT1 0x2d +#define AK4671_FIL2_COEFFICIENT2 0x2e +#define AK4671_FIL2_COEFFICIENT3 0x2f +#define AK4671_DIGITAL_FILTER_SELECT2 0x30 +#define AK4671_E1_COEFFICIENT0 0x32 +#define AK4671_E1_COEFFICIENT1 0x33 +#define AK4671_E1_COEFFICIENT2 0x34 +#define AK4671_E1_COEFFICIENT3 0x35 +#define AK4671_E1_COEFFICIENT4 0x36 +#define AK4671_E1_COEFFICIENT5 0x37 +#define AK4671_E2_COEFFICIENT0 0x38 +#define AK4671_E2_COEFFICIENT1 0x39 +#define AK4671_E2_COEFFICIENT2 0x3a +#define AK4671_E2_COEFFICIENT3 0x3b +#define AK4671_E2_COEFFICIENT4 0x3c +#define AK4671_E2_COEFFICIENT5 0x3d +#define AK4671_E3_COEFFICIENT0 0x3e +#define AK4671_E3_COEFFICIENT1 0x3f +#define AK4671_E3_COEFFICIENT2 0x40 +#define AK4671_E3_COEFFICIENT3 0x41 +#define AK4671_E3_COEFFICIENT4 0x42 +#define AK4671_E3_COEFFICIENT5 0x43 +#define AK4671_E4_COEFFICIENT0 0x44 +#define AK4671_E4_COEFFICIENT1 0x45 +#define AK4671_E4_COEFFICIENT2 0x46 +#define AK4671_E4_COEFFICIENT3 0x47 +#define AK4671_E4_COEFFICIENT4 0x48 +#define AK4671_E4_COEFFICIENT5 0x49 +#define AK4671_E5_COEFFICIENT0 0x4a +#define AK4671_E5_COEFFICIENT1 0x4b +#define AK4671_E5_COEFFICIENT2 0x4c +#define AK4671_E5_COEFFICIENT3 0x4d +#define AK4671_E5_COEFFICIENT4 0x4e +#define AK4671_E5_COEFFICIENT5 0x4f +#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50 +#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51 +#define AK4671_EQ_CONTRO_10KHZ 0x52 +#define AK4671_PCM_IF_CONTROL0 0x53 +#define AK4671_PCM_IF_CONTROL1 0x54 +#define AK4671_PCM_IF_CONTROL2 0x55 +#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56 +#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57 +#define AK4671_SIDETONE_VOLUME_CONTROL 0x58 +#define AK4671_DIGITAL_MIXING_CONTROL2 0x59 +#define AK4671_SAR_ADC_CONTROL 0x5a + +#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) + +/* Bitfield Definitions */ + +/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ +#define AK4671_PMVCM 0x01 + +/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */ +#define AK4671_PLL 0x0f +#define AK4671_PLL_11_2896MHZ (4 << 0) +#define AK4671_PLL_12_288MHZ (5 << 0) +#define AK4671_PLL_12MHZ (6 << 0) +#define AK4671_PLL_24MHZ (7 << 0) +#define AK4671_PLL_19_2MHZ (8 << 0) +#define AK4671_PLL_13_5MHZ (12 << 0) +#define AK4671_PLL_27MHZ (13 << 0) +#define AK4671_PLL_13MHZ (14 << 0) +#define AK4671_PLL_26MHZ (15 << 0) +#define AK4671_FS 0xf0 +#define AK4671_FS_8KHZ (0 << 4) +#define AK4671_FS_12KHZ (1 << 4) +#define AK4671_FS_16KHZ (2 << 4) +#define AK4671_FS_24KHZ (3 << 4) +#define AK4671_FS_11_025KHZ (5 << 4) +#define AK4671_FS_22_05KHZ (7 << 4) +#define AK4671_FS_32KHZ (10 << 4) +#define AK4671_FS_48KHZ (11 << 4) +#define AK4671_FS_44_1KHZ (15 << 4) + +/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */ +#define AK4671_PMPLL 0x01 +#define AK4671_M_S 0x02 + +/* AK4671_FORMAT_SELECT (0x03) Fields */ +#define AK4671_DIF 0x03 +#define AK4671_DIF_DSP_MODE (0 << 0) +#define AK4671_DIF_MSB_MODE (2 << 0) +#define AK4671_DIF_I2S_MODE (3 << 0) +#define AK4671_BCKP 0x04 +#define AK4671_MSBS 0x08 +#define AK4671_SDOD 0x10 + +/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */ +#define AK4671_MUTEN 0x04 + +extern struct snd_soc_dai ak4671_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4671; + +#endif -- cgit v1.2.2 From 472df3cbae8da6a949f1392a11958b8d21383735 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Sat, 12 Sep 2009 01:16:29 +0800 Subject: ASoC: Provide API for reordering channels The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 05fdc8023da4..f5b356f8acfb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2252,6 +2252,30 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); +/** + * snd_soc_dai_set_channel_map - configure DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + if (dai->ops && dai->ops->set_channel_map) + return dai->ops->set_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); + /** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI -- cgit v1.2.2 From fd5ad654e665b5c30c8d755a106309c8ea9f3e7b Mon Sep 17 00:00:00 2001 From: Jassi Date: Tue, 15 Sep 2009 19:02:38 +0900 Subject: ASoC: S3C I2S LRCLK polarity option. 1) Explicitly set LRCLK polarity for I2S Vs LSM/MSB modes. 2) Convert from numerical to bit-field values for BCLK selection. 3) Use proper error checking for return value from clk_get Signed-off-by: Jassi Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 30 +++++++++++++++++++++++++++++- 1 file changed, 29 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index aa7af0b8d421..819c3c086d69 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -308,12 +308,15 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_MSB; break; case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_LSB; break; case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_IIS; break; default: @@ -463,6 +466,31 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case S3C_I2SV2_DIV_BCLK: + if (div > 3) { + /* convert value to bit field */ + + switch (div) { + case 16: + div = S3C2412_IISMOD_BCLK_16FS; + break; + + case 32: + div = S3C2412_IISMOD_BCLK_32FS; + break; + + case 24: + div = S3C2412_IISMOD_BCLK_24FS; + break; + + case 48: + div = S3C2412_IISMOD_BCLK_48FS; + break; + + default: + return -EINVAL; + } + } + reg = readl(i2s->regs + S3C2412_IISMOD); reg &= ~S3C2412_IISMOD_BCLK_MASK; writel(reg | div, i2s->regs + S3C2412_IISMOD); @@ -622,7 +650,7 @@ int s3c_i2sv2_probe(struct platform_device *pdev, } i2s->iis_pclk = clk_get(dev, "iis"); - if (i2s->iis_pclk == NULL) { + if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); iounmap(i2s->regs); return -ENOENT; -- cgit v1.2.2 From 08db48f1ee1adf8919484f731d4ad6b264cfc564 Mon Sep 17 00:00:00 2001 From: Barry Song Date: Tue, 15 Sep 2009 11:24:52 +0800 Subject: ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836 Signed-off-by: Barry Song Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad1836.c | 7 ++++++ sound/soc/blackfin/bf5xx-ad1938.c | 9 +++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 +++++--- sound/soc/blackfin/bf5xx-tdm.c | 45 +++++++++++++++++++++++++++++++------- sound/soc/blackfin/bf5xx-tdm.h | 11 ++++++++++ 5 files changed, 69 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e304b0f..0f45a3f56be8 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e91810c..2ef1e5013b8c 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e823bd18..a8c73cbbd685 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096badf09a5..600987d8a871 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,40 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + unsigned int slot; + unsigned int tx_mapped = 0, rx_mapped = 0; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_tdm.tx_map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_tdm.rx_map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +261,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +327,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d90cd4..04189a18c1ba 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define BFIN_TDM_DAI_MAX_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; + unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- cgit v1.2.2 From 9b95b166789d3ec57cea8cca0d42e602b8643ab0 Mon Sep 17 00:00:00 2001 From: Miguel Aguilar Date: Wed, 2 Sep 2009 15:33:59 -0600 Subject: ASoC: Davinci: Add audio codec support for DM365 EVM This patch enables tlv320aic3101 support on DM365 EVM and it was tested on DM365 EVM rev c. Note: this patch was created based on temp/asoc branch. Signed-off-by: Miguel Aguilar Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 4 ++-- sound/soc/davinci/davinci-evm.c | 7 ++++--- 2 files changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 4dfd4ad9d90e..047ee39418c0 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -13,9 +13,9 @@ config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f659405..7ccbe6684fc2 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -45,7 +45,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -176,7 +177,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci-evm audio machine driver */ +/* davinci dm6446, dm355 or dm365 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -243,7 +244,7 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm()) { + if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { -- cgit v1.2.2 From 8bb014895547eeeb9aa61a654f24e41e15919304 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Sep 2009 19:38:53 +0100 Subject: ASoC: Add S3C64xx IIS CDCLK source selection CDCLK can either be an output generated by the CPU, intended for use as the CODEC master clock, or an input (probably from the CODEC) providing a master clock for the IIS block. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 13 +++++++++++++ sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + 2 files changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 3c06c401d0fb..aaf452096be2 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -99,6 +99,19 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, iismod |= S3C64XX_IISMOD_IMS_SYSMUX; break; + case S3C64XX_CLKSRC_CDCLK: + switch (dir) { + case SND_SOC_CLOCK_IN: + iismod |= S3C64XX_IISMOD_CDCLKCON; + break; + case SND_SOC_CLOCK_OUT: + iismod &= ~S3C64XX_IISMOD_CDCLKCON; + break; + default: + return -EINVAL; + } + break; + default: return -EINVAL; } diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index 02148cee2613..abe7253b55fc 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -25,6 +25,7 @@ struct clk; #define S3C64XX_CLKSRC_PCLK (0) #define S3C64XX_CLKSRC_MUX (1) +#define S3C64XX_CLKSRC_CDCLK (2) extern struct snd_soc_dai s3c64xx_i2s_dai[]; -- cgit v1.2.2 From b1cd6b9ec7c749ddfad628c8c12659591ae195e6 Mon Sep 17 00:00:00 2001 From: Jassi Date: Fri, 18 Sep 2009 15:22:27 +0900 Subject: ASoC: Return correct codec clock in s3c64xx-i2s Instead of always returnig pointer to the 'audio-bus' clock, check which clock is used to generate internal clocks and then return it's pointer. Signed-off-by: Jassi Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index aaf452096be2..43fb253a3429 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -124,8 +124,12 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) { struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - return i2s->iis_cclk; + if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; } EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); -- cgit v1.2.2 From d0f5fa17aa63262685e43b798ca0830d89786235 Mon Sep 17 00:00:00 2001 From: jassi brar Date: Sat, 19 Sep 2009 09:46:06 +0900 Subject: ASoC: Support WM8580 based audio subsystem on SMDK64xx machines New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. [Added dependency on BROKEN since the IISv4 interface hasn't been merged yet, fixed the PLL API usage and removed the disabling of the PLL in the hw_free function since that'll break simultaneous playback and record -- broonie.] Signed-off-by: Jassi Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 9 ++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 273 ++++++++++++++++++++++++++++++++++++ 3 files changed, 284 insertions(+) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428fc1adb..d7912f1e4627 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,15 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + depends on BROKEN + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7dd3fc6..7790406f90b7 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 000000000000..482aaf10eff6 --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,273 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Assuming the CODEC driver evaluates it's rfs too from this call */ + ret = snd_soc_dai_set_pll(codec_dai, 0, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_enable_pin(codec, "Center/Sub"); + snd_soc_dapm_enable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar@samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- cgit v1.2.2 From d62ab3589462d406e98731799361f46095467882 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Sep 2009 04:21:47 -0700 Subject: ASoC: Convert soc-cache to use C99 style initialisers for the table Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 404231ee8780..d2505e8b06c9 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -179,10 +179,20 @@ static struct { unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { - { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read }, - { 8, 8, snd_soc_8_8_write, NULL, snd_soc_8_8_read, NULL }, - { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read, - snd_soc_8_16_read_i2c }, + { + .addr_bits = 7, .data_bits = 9, + .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, + .spi_write = snd_soc_7_9_spi_write + }, + { + .addr_bits = 8, .data_bits = 8, + .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, + }, + { + .addr_bits = 8, .data_bits = 16, + .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, + .i2c_read = snd_soc_8_16_read_i2c, + }, }; /** -- cgit v1.2.2 From 8f34692f63d66805b51ff408f4067748d3c1c3fd Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:36 +0200 Subject: ALSA: ak4620 support, codec regs listed in proc * complete support for ak4620 * codec regs listed in proc for all codecs/chips * adding total regs for each codec * fixing nb. of steps in input attenuation controls Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4xxx-adda.c | 136 ++++++++++++++++++++++++++++++++---------- sound/pci/ice1712/juli.c | 21 ------- 2 files changed, 104 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index ee47abab764e..1adb8a3c2b62 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -19,7 +19,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #include #include @@ -29,6 +29,7 @@ #include #include #include +#include MODULE_AUTHOR("Jaroslav Kysela , Takashi Iwai "); MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters"); @@ -52,26 +53,21 @@ EXPORT_SYMBOL(snd_akm4xxx_write); static void ak4524_reset(struct snd_akm4xxx *ak, int state) { unsigned int chip; - unsigned char reg, maxreg; + unsigned char reg; - if (ak->type == SND_AK4528) - maxreg = 0x06; - else - maxreg = 0x08; for (chip = 0; chip < ak->num_dacs/2; chip++) { snd_akm4xxx_write(ak, chip, 0x01, state ? 0x00 : 0x03); if (state) continue; /* DAC volumes */ - for (reg = 0x04; reg < maxreg; reg++) + for (reg = 0x04; reg < ak->total_regs; reg++) snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg)); } } /* reset procedure for AK4355 and AK4358 */ -static void ak435X_reset(struct snd_akm4xxx *ak, int state, - unsigned char total_regs) +static void ak435X_reset(struct snd_akm4xxx *ak, int state) { unsigned char reg; @@ -79,7 +75,7 @@ static void ak435X_reset(struct snd_akm4xxx *ak, int state, snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */ return; } - for (reg = 0x00; reg < total_regs; reg++) + for (reg = 0x00; reg < ak->total_regs; reg++) if (reg != 0x01) snd_akm4xxx_write(ak, 0, reg, snd_akm4xxx_get(ak, 0, reg)); @@ -91,12 +87,11 @@ static void ak4381_reset(struct snd_akm4xxx *ak, int state) { unsigned int chip; unsigned char reg; - for (chip = 0; chip < ak->num_dacs/2; chip++) { snd_akm4xxx_write(ak, chip, 0x00, state ? 0x0c : 0x0f); if (state) continue; - for (reg = 0x01; reg < 0x05; reg++) + for (reg = 0x01; reg < ak->total_regs; reg++) snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg)); } @@ -113,16 +108,17 @@ void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state) switch (ak->type) { case SND_AK4524: case SND_AK4528: + case SND_AK4620: ak4524_reset(ak, state); break; case SND_AK4529: /* FIXME: needed for ak4529? */ break; case SND_AK4355: - ak435X_reset(ak, state, 0x0b); + ak435X_reset(ak, state); break; case SND_AK4358: - ak435X_reset(ak, state, 0x10); + ak435X_reset(ak, state); break; case SND_AK4381: ak4381_reset(ak, state); @@ -139,7 +135,7 @@ EXPORT_SYMBOL(snd_akm4xxx_reset); * Volume conversion table for non-linear volumes * from -63.5dB (mute) to 0dB step 0.5dB * - * Used for AK4524 input/ouput attenuation, AK4528, and + * Used for AK4524/AK4620 input/ouput attenuation, AK4528, and * AK5365 input attenuation */ static const unsigned char vol_cvt_datt[128] = { @@ -259,8 +255,22 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) 0x00, 0x0f, /* 0: power-up, un-reset */ 0xff, 0xff }; + static const unsigned char inits_ak4620[] = { + 0x00, 0x07, /* 0: normal */ + 0x01, 0x00, /* 0: reset */ + 0x01, 0x02, /* 1: RSTAD */ + 0x01, 0x03, /* 1: RSTDA */ + 0x01, 0x0f, /* 1: normal */ + 0x02, 0x60, /* 2: 24bit I2S */ + 0x03, 0x01, /* 3: deemphasis off */ + 0x04, 0x00, /* 4: LIN muted */ + 0x05, 0x00, /* 5: RIN muted */ + 0x06, 0x00, /* 6: LOUT muted */ + 0x07, 0x00, /* 7: ROUT muted */ + 0xff, 0xff + }; - int chip, num_chips; + int chip; const unsigned char *ptr, *inits; unsigned char reg, data; @@ -270,42 +280,64 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) switch (ak->type) { case SND_AK4524: inits = inits_ak4524; - num_chips = ak->num_dacs / 2; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4524"; + ak->total_regs = 0x08; break; case SND_AK4528: inits = inits_ak4528; - num_chips = ak->num_dacs / 2; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4528"; + ak->total_regs = 0x06; break; case SND_AK4529: inits = inits_ak4529; - num_chips = 1; + ak->num_chips = 1; + ak->name = "ak4529"; + ak->total_regs = 0x0d; break; case SND_AK4355: inits = inits_ak4355; - num_chips = 1; + ak->num_chips = 1; + ak->name = "ak4355"; + ak->total_regs = 0x0b; break; case SND_AK4358: inits = inits_ak4358; - num_chips = 1; + ak->num_chips = 1; + ak->name = "ak4358"; + ak->total_regs = 0x10; break; case SND_AK4381: inits = inits_ak4381; - num_chips = ak->num_dacs / 2; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4381"; + ak->total_regs = 0x05; break; case SND_AK5365: /* FIXME: any init sequence? */ + ak->num_chips = 1; + ak->name = "ak5365"; + ak->total_regs = 0x08; return; + case SND_AK4620: + inits = inits_ak4620; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4620"; + ak->total_regs = 0x08; + break; default: snd_BUG(); return; } - for (chip = 0; chip < num_chips; chip++) { + for (chip = 0; chip < ak->num_chips; chip++) { ptr = inits; while (*ptr != 0xff) { reg = *ptr++; data = *ptr++; snd_akm4xxx_write(ak, chip, reg, data); + udelay(10); } } } @@ -688,6 +720,12 @@ static int build_dac_controls(struct snd_akm4xxx *ak) AK_COMPOSE(idx/2, (idx%2) + 3, 0, 255); knew.tlv.p = db_scale_linear; break; + case SND_AK4620: + /* register 6 & 7 */ + knew.private_value = + AK_COMPOSE(idx/2, (idx%2) + 6, 0, 255); + knew.tlv.p = db_scale_linear; + break; default: return -EINVAL; } @@ -704,10 +742,12 @@ static int build_dac_controls(struct snd_akm4xxx *ak) static int build_adc_controls(struct snd_akm4xxx *ak) { - int idx, err, mixer_ch, num_stereo; + int idx, err, mixer_ch, num_stereo, max_steps; struct snd_kcontrol_new knew; mixer_ch = 0; + if (ak->type == SND_AK4528) + return 0; /* no controls */ for (idx = 0; idx < ak->num_adcs;) { memset(&knew, 0, sizeof(knew)); if (! ak->adc_info || ! ak->adc_info[mixer_ch].name) { @@ -733,13 +773,12 @@ static int build_adc_controls(struct snd_akm4xxx *ak) } /* register 4 & 5 */ if (ak->type == SND_AK5365) - knew.private_value = - AK_COMPOSE(idx/2, (idx%2) + 4, 0, 151) | - AK_VOL_CVT | AK_IPGA; + max_steps = 152; else - knew.private_value = - AK_COMPOSE(idx/2, (idx%2) + 4, 0, 163) | - AK_VOL_CVT | AK_IPGA; + max_steps = 164; + knew.private_value = + AK_COMPOSE(idx/2, (idx%2) + 4, 0, max_steps) | + AK_VOL_CVT | AK_IPGA; knew.tlv.p = db_scale_vol_datt; err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak)); if (err < 0) @@ -808,6 +847,7 @@ static int build_deemphasis(struct snd_akm4xxx *ak, int num_emphs) switch (ak->type) { case SND_AK4524: case SND_AK4528: + case SND_AK4620: /* register 3 */ knew.private_value = AK_COMPOSE(idx, 3, 0, 0); break; @@ -834,6 +874,35 @@ static int build_deemphasis(struct snd_akm4xxx *ak, int num_emphs) return 0; } +#ifdef CONFIG_PROC_FS +static void proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_akm4xxx *ak = (struct snd_akm4xxx *)entry->private_data; + int reg, val, chip; + for (chip = 0; chip < ak->num_chips; chip++) { + for (reg = 0; reg < ak->total_regs; reg++) { + val = snd_akm4xxx_get(ak, chip, reg); + snd_iprintf(buffer, "chip %d: 0x%02x = 0x%02x\n", chip, + reg, val); + } + } +} + +static int proc_init(struct snd_akm4xxx *ak) +{ + struct snd_info_entry *entry; + int err; + err = snd_card_proc_new(ak->card, ak->name, &entry); + if (err < 0) + return err; + snd_info_set_text_ops(entry, ak, proc_regs_read); + return 0; +} +#else /* !CONFIG_PROC_FS */ +static int proc_init(struct snd_akm4xxx *ak) {} +#endif + int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) { int err, num_emphs; @@ -845,18 +914,21 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) err = build_adc_controls(ak); if (err < 0) return err; - if (ak->type == SND_AK4355 || ak->type == SND_AK4358) num_emphs = 1; + else if (ak->type == SND_AK4620) + num_emphs = 0; else num_emphs = ak->num_dacs / 2; err = build_deemphasis(ak, num_emphs); + if (err < 0) + return err; + err = proc_init(ak); if (err < 0) return err; return 0; } - EXPORT_SYMBOL(snd_akm4xxx_build_controls); static int __init alsa_akm4xxx_module_init(void) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index fd948bfd9aef..4789e8bfdc17 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -412,25 +412,6 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = { }, }; - -static void ak4358_proc_regs_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; - int reg, val; - for (reg = 0; reg <= 0xf; reg++) { - val = snd_akm4xxx_get(ice->akm, 0, reg); - snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); - } -} - -static void ak4358_proc_init(struct snd_ice1712 *ice) -{ - struct snd_info_entry *entry; - if (!snd_card_proc_new(ice->card, "ak4358_codec", &entry)) - snd_info_set_text_ops(entry, ice, ak4358_proc_regs_read); -} - static char *slave_vols[] __devinitdata = { PCM_VOLUME, MONITOR_AN_IN_VOLUME, @@ -496,8 +477,6 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) /* only capture SPDIF over AK4114 */ err = snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - - ak4358_proc_init(ice); if (err < 0) return err; return 0; -- cgit v1.2.2 From 42cfa276aebd28e5cc4350ff6c7d75f1cb84dd98 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:37 +0200 Subject: ALSA: ak4113 support * complete support for ak4113 * based on code for ak4114 and ak4117 Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/i2c/other/Makefile | 3 +- sound/i2c/other/ak4113.c | 639 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 641 insertions(+), 1 deletion(-) create mode 100644 sound/i2c/other/ak4113.c (limited to 'sound') diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile index 703d954238f4..2dad40f3f622 100644 --- a/sound/i2c/other/Makefile +++ b/sound/i2c/other/Makefile @@ -5,6 +5,7 @@ snd-ak4114-objs := ak4114.o snd-ak4117-objs := ak4117.o +snd-ak4113-objs := ak4113.o snd-ak4xxx-adda-objs := ak4xxx-adda.o snd-pt2258-objs := pt2258.o snd-tea575x-tuner-objs := tea575x-tuner.o @@ -12,5 +13,5 @@ snd-tea575x-tuner-objs := tea575x-tuner.o # Module Dependency obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o obj-$(CONFIG_SND_ICE1712) += snd-ak4xxx-adda.o -obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4xxx-adda.o snd-pt2258.o +obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4113.o snd-ak4xxx-adda.o snd-pt2258.o obj-$(CONFIG_SND_FM801_TEA575X) += snd-tea575x-tuner.o diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c new file mode 100644 index 000000000000..fff62cc8607c --- /dev/null +++ b/sound/i2c/other/ak4113.c @@ -0,0 +1,639 @@ +/* + * Routines for control of the AK4113 via I2C/4-wire serial interface + * IEC958 (S/PDIF) receiver by Asahi Kasei + * Copyright (c) by Jaroslav Kysela + * Copyright (c) by Pavel Hofman + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Pavel Hofman "); +MODULE_DESCRIPTION("AK4113 IEC958 (S/PDIF) receiver by Asahi Kasei"); +MODULE_LICENSE("GPL"); + +#define AK4113_ADDR 0x00 /* fixed address */ + +static void ak4113_stats(struct work_struct *work); +static void ak4113_init_regs(struct ak4113 *chip); + + +static void reg_write(struct ak4113 *ak4113, unsigned char reg, + unsigned char val) +{ + ak4113->write(ak4113->private_data, reg, val); + if (reg < sizeof(ak4113->regmap)) + ak4113->regmap[reg] = val; +} + +static inline unsigned char reg_read(struct ak4113 *ak4113, unsigned char reg) +{ + return ak4113->read(ak4113->private_data, reg); +} + +static void snd_ak4113_free(struct ak4113 *chip) +{ + chip->init = 1; /* don't schedule new work */ + mb(); + cancel_delayed_work(&chip->work); + flush_scheduled_work(); + kfree(chip); +} + +static int snd_ak4113_dev_free(struct snd_device *device) +{ + struct ak4113 *chip = device->device_data; + snd_ak4113_free(chip); + return 0; +} + +int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, + ak4113_write_t *write, const unsigned char pgm[5], + void *private_data, struct ak4113 **r_ak4113) +{ + struct ak4113 *chip; + int err = 0; + unsigned char reg; + static struct snd_device_ops ops = { + .dev_free = snd_ak4113_dev_free, + }; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + spin_lock_init(&chip->lock); + chip->card = card; + chip->read = read; + chip->write = write; + chip->private_data = private_data; + INIT_DELAYED_WORK(&chip->work, ak4113_stats); + + for (reg = 0; reg < AK4113_WRITABLE_REGS ; reg++) + chip->regmap[reg] = pgm[reg]; + ak4113_init_regs(chip); + + chip->rcs0 = reg_read(chip, AK4113_REG_RCS0) & ~(AK4113_QINT | + AK4113_CINT | AK4113_STC); + chip->rcs1 = reg_read(chip, AK4113_REG_RCS1); + chip->rcs2 = reg_read(chip, AK4113_REG_RCS2); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto __fail; + + if (r_ak4113) + *r_ak4113 = chip; + return 0; + +__fail: + snd_ak4113_free(chip); + return err < 0 ? err : -EIO; +} +EXPORT_SYMBOL_GPL(snd_ak4113_create); + +void snd_ak4113_reg_write(struct ak4113 *chip, unsigned char reg, + unsigned char mask, unsigned char val) +{ + if (reg >= AK4113_WRITABLE_REGS) + return; + reg_write(chip, reg, (chip->regmap[reg] & ~mask) | val); +} +EXPORT_SYMBOL_GPL(snd_ak4113_reg_write); + +static void ak4113_init_regs(struct ak4113 *chip) +{ + unsigned char old = chip->regmap[AK4113_REG_PWRDN], reg; + + /* bring the chip to reset state and powerdown state */ + reg_write(chip, AK4113_REG_PWRDN, old & ~(AK4113_RST|AK4113_PWN)); + udelay(200); + /* release reset, but leave powerdown */ + reg_write(chip, AK4113_REG_PWRDN, (old | AK4113_RST) & ~AK4113_PWN); + udelay(200); + for (reg = 1; reg < AK4113_WRITABLE_REGS; reg++) + reg_write(chip, reg, chip->regmap[reg]); + /* release powerdown, everything is initialized now */ + reg_write(chip, AK4113_REG_PWRDN, old | AK4113_RST | AK4113_PWN); +} + +void snd_ak4113_reinit(struct ak4113 *chip) +{ + chip->init = 1; + mb(); + flush_scheduled_work(); + ak4113_init_regs(chip); + /* bring up statistics / event queing */ + chip->init = 0; + if (chip->kctls[0]) + schedule_delayed_work(&chip->work, HZ / 10); +} +EXPORT_SYMBOL_GPL(snd_ak4113_reinit); + +static unsigned int external_rate(unsigned char rcs1) +{ + switch (rcs1 & (AK4113_FS0|AK4113_FS1|AK4113_FS2|AK4113_FS3)) { + case AK4113_FS_8000HZ: + return 8000; + case AK4113_FS_11025HZ: + return 11025; + case AK4113_FS_16000HZ: + return 16000; + case AK4113_FS_22050HZ: + return 22050; + case AK4113_FS_24000HZ: + return 24000; + case AK4113_FS_32000HZ: + return 32000; + case AK4113_FS_44100HZ: + return 44100; + case AK4113_FS_48000HZ: + return 48000; + case AK4113_FS_64000HZ: + return 64000; + case AK4113_FS_88200HZ: + return 88200; + case AK4113_FS_96000HZ: + return 96000; + case AK4113_FS_176400HZ: + return 176400; + case AK4113_FS_192000HZ: + return 192000; + default: + return 0; + } +} + +static int snd_ak4113_in_error_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = LONG_MAX; + return 0; +} + +static int snd_ak4113_in_error_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + long *ptr; + + spin_lock_irq(&chip->lock); + ptr = (long *)(((char *)chip) + kcontrol->private_value); + ucontrol->value.integer.value[0] = *ptr; + *ptr = 0; + spin_unlock_irq(&chip->lock); + return 0; +} + +#define snd_ak4113_in_bit_info snd_ctl_boolean_mono_info + +static int snd_ak4113_in_bit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned char reg = kcontrol->private_value & 0xff; + unsigned char bit = (kcontrol->private_value >> 8) & 0xff; + unsigned char inv = (kcontrol->private_value >> 31) & 1; + + ucontrol->value.integer.value[0] = + ((reg_read(chip, reg) & (1 << bit)) ? 1 : 0) ^ inv; + return 0; +} + +static int snd_ak4113_rx_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 5; + return 0; +} + +static int snd_ak4113_rx_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + (AK4113_IPS(chip->regmap[AK4113_REG_IO1])); + return 0; +} + +static int snd_ak4113_rx_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + int change; + u8 old_val; + + spin_lock_irq(&chip->lock); + old_val = chip->regmap[AK4113_REG_IO1]; + change = ucontrol->value.integer.value[0] != AK4113_IPS(old_val); + if (change) + reg_write(chip, AK4113_REG_IO1, + (old_val & (~AK4113_IPS(0xff))) | + (AK4113_IPS(ucontrol->value.integer.value[0]))); + spin_unlock_irq(&chip->lock); + return change; +} + +static int snd_ak4113_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 192000; + return 0; +} + +static int snd_ak4113_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = external_rate(reg_read(chip, + AK4113_REG_RCS1)); + return 0; +} + +static int snd_ak4113_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_ak4113_spdif_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned i; + + for (i = 0; i < AK4113_REG_RXCSB_SIZE; i++) + ucontrol->value.iec958.status[i] = reg_read(chip, + AK4113_REG_RXCSB0 + i); + return 0; +} + +static int snd_ak4113_spdif_mask_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_ak4113_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memset(ucontrol->value.iec958.status, 0xff, AK4113_REG_RXCSB_SIZE); + return 0; +} + +static int snd_ak4113_spdif_pinfo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xffff; + uinfo->count = 4; + return 0; +} + +static int snd_ak4113_spdif_pget(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned short tmp; + + ucontrol->value.integer.value[0] = 0xf8f2; + ucontrol->value.integer.value[1] = 0x4e1f; + tmp = reg_read(chip, AK4113_REG_Pc0) | + (reg_read(chip, AK4113_REG_Pc1) << 8); + ucontrol->value.integer.value[2] = tmp; + tmp = reg_read(chip, AK4113_REG_Pd0) | + (reg_read(chip, AK4113_REG_Pd1) << 8); + ucontrol->value.integer.value[3] = tmp; + return 0; +} + +static int snd_ak4113_spdif_qinfo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = AK4113_REG_QSUB_SIZE; + return 0; +} + +static int snd_ak4113_spdif_qget(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned i; + + for (i = 0; i < AK4113_REG_QSUB_SIZE; i++) + ucontrol->value.bytes.data[i] = reg_read(chip, + AK4113_REG_QSUB_ADDR + i); + return 0; +} + +/* Don't forget to change AK4113_CONTROLS define!!! */ +static struct snd_kcontrol_new snd_ak4113_iec958_controls[] = { +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Parity Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, parity_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 V-Bit Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, v_bit_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 C-CRC Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, ccrc_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Q-CRC Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, qcrc_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 External Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_rate_info, + .get = snd_ak4113_rate_get, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK), + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = snd_ak4113_spdif_mask_info, + .get = snd_ak4113_spdif_mask_get, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_spdif_info, + .get = snd_ak4113_spdif_get, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Preample Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_spdif_pinfo, + .get = snd_ak4113_spdif_pget, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Q-subcode Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_spdif_qinfo, + .get = snd_ak4113_spdif_qget, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Audio", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_bit_info, + .get = snd_ak4113_in_bit_get, + .private_value = (1<<31) | (1<<8) | AK4113_REG_RCS0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Non-PCM Bitstream", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_bit_info, + .get = snd_ak4113_in_bit_get, + .private_value = (0<<8) | AK4113_REG_RCS1, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 DTS Bitstream", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_bit_info, + .get = snd_ak4113_in_bit_get, + .private_value = (1<<8) | AK4113_REG_RCS1, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "AK4113 Input Select", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE, + .info = snd_ak4113_rx_info, + .get = snd_ak4113_rx_get, + .put = snd_ak4113_rx_put, +} +}; + +static void snd_ak4113_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct ak4113 *ak4113 = entry->private_data; + int reg, val; + /* all ak4113 registers 0x00 - 0x1c */ + for (reg = 0; reg < 0x1d; reg++) { + val = reg_read(ak4113, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void snd_ak4113_proc_init(struct ak4113 *ak4113) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ak4113->card, "ak4113", &entry)) + snd_info_set_text_ops(entry, ak4113, snd_ak4113_proc_regs_read); +} + +int snd_ak4113_build(struct ak4113 *ak4113, + struct snd_pcm_substream *cap_substream) +{ + struct snd_kcontrol *kctl; + unsigned int idx; + int err; + + if (snd_BUG_ON(!cap_substream)) + return -EINVAL; + ak4113->substream = cap_substream; + for (idx = 0; idx < AK4113_CONTROLS; idx++) { + kctl = snd_ctl_new1(&snd_ak4113_iec958_controls[idx], ak4113); + if (kctl == NULL) + return -ENOMEM; + kctl->id.device = cap_substream->pcm->device; + kctl->id.subdevice = cap_substream->number; + err = snd_ctl_add(ak4113->card, kctl); + if (err < 0) + return err; + ak4113->kctls[idx] = kctl; + } + snd_ak4113_proc_init(ak4113); + /* trigger workq */ + schedule_delayed_work(&ak4113->work, HZ / 10); + return 0; +} +EXPORT_SYMBOL_GPL(snd_ak4113_build); + +int snd_ak4113_external_rate(struct ak4113 *ak4113) +{ + unsigned char rcs1; + + rcs1 = reg_read(ak4113, AK4113_REG_RCS1); + return external_rate(rcs1); +} +EXPORT_SYMBOL_GPL(snd_ak4113_external_rate); + +int snd_ak4113_check_rate_and_errors(struct ak4113 *ak4113, unsigned int flags) +{ + struct snd_pcm_runtime *runtime = + ak4113->substream ? ak4113->substream->runtime : NULL; + unsigned long _flags; + int res = 0; + unsigned char rcs0, rcs1, rcs2; + unsigned char c0, c1; + + rcs1 = reg_read(ak4113, AK4113_REG_RCS1); + if (flags & AK4113_CHECK_NO_STAT) + goto __rate; + rcs0 = reg_read(ak4113, AK4113_REG_RCS0); + rcs2 = reg_read(ak4113, AK4113_REG_RCS2); + spin_lock_irqsave(&ak4113->lock, _flags); + if (rcs0 & AK4113_PAR) + ak4113->parity_errors++; + if (rcs0 & AK4113_V) + ak4113->v_bit_errors++; + if (rcs2 & AK4113_CCRC) + ak4113->ccrc_errors++; + if (rcs2 & AK4113_QCRC) + ak4113->qcrc_errors++; + c0 = (ak4113->rcs0 & (AK4113_QINT | AK4113_CINT | AK4113_STC | + AK4113_AUDION | AK4113_AUTO | AK4113_UNLCK)) ^ + (rcs0 & (AK4113_QINT | AK4113_CINT | AK4113_STC | + AK4113_AUDION | AK4113_AUTO | AK4113_UNLCK)); + c1 = (ak4113->rcs1 & (AK4113_DTSCD | AK4113_NPCM | AK4113_PEM | + AK4113_DAT | 0xf0)) ^ + (rcs1 & (AK4113_DTSCD | AK4113_NPCM | AK4113_PEM | + AK4113_DAT | 0xf0)); + ak4113->rcs0 = rcs0 & ~(AK4113_QINT | AK4113_CINT | AK4113_STC); + ak4113->rcs1 = rcs1; + ak4113->rcs2 = rcs2; + spin_unlock_irqrestore(&ak4113->lock, _flags); + + if (rcs0 & AK4113_PAR) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[0]->id); + if (rcs0 & AK4113_V) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[1]->id); + if (rcs2 & AK4113_CCRC) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[2]->id); + if (rcs2 & AK4113_QCRC) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[3]->id); + + /* rate change */ + if (c1 & 0xf0) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[4]->id); + + if ((c1 & AK4113_PEM) | (c0 & AK4113_CINT)) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[6]->id); + if (c0 & AK4113_QINT) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[8]->id); + + if (c0 & AK4113_AUDION) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[9]->id); + if (c1 & AK4113_NPCM) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[10]->id); + if (c1 & AK4113_DTSCD) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[11]->id); + + if (ak4113->change_callback && (c0 | c1) != 0) + ak4113->change_callback(ak4113, c0, c1); + +__rate: + /* compare rate */ + res = external_rate(rcs1); + if (!(flags & AK4113_CHECK_NO_RATE) && runtime && + (runtime->rate != res)) { + snd_pcm_stream_lock_irqsave(ak4113->substream, _flags); + if (snd_pcm_running(ak4113->substream)) { + /*printk(KERN_DEBUG "rate changed (%i <- %i)\n", + * runtime->rate, res); */ + snd_pcm_stop(ak4113->substream, + SNDRV_PCM_STATE_DRAINING); + wake_up(&runtime->sleep); + res = 1; + } + snd_pcm_stream_unlock_irqrestore(ak4113->substream, _flags); + } + return res; +} +EXPORT_SYMBOL_GPL(snd_ak4113_check_rate_and_errors); + +static void ak4113_stats(struct work_struct *work) +{ + struct ak4113 *chip = container_of(work, struct ak4113, work.work); + + if (!chip->init) + snd_ak4113_check_rate_and_errors(chip, chip->check_flags); + + schedule_delayed_work(&chip->work, HZ / 10); +} -- cgit v1.2.2 From 494703062b6e6ef5e72364aafc9bcbc172d53dea Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:38 +0200 Subject: ALSA: ice1724 - adding GPIO routines for mask and direction * get/set routines for GPIO mask and direction Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 12 ++++++++++++ sound/pci/ice1712/ice1712.h | 7 +++++++ sound/pci/ice1712/ice1724.c | 19 +++++++++++++++++++ 3 files changed, 38 insertions(+) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index cecf1ffeeaaa..56d8d67f1ac3 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -298,6 +298,16 @@ static void snd_ice1712_set_gpio_dir(struct snd_ice1712 *ice, unsigned int data) inb(ICEREG(ice, DATA)); /* dummy read for pci-posting */ } +static unsigned int snd_ice1712_get_gpio_dir(struct snd_ice1712 *ice) +{ + return snd_ice1712_read(ice, ICE1712_IREG_GPIO_DIRECTION); +} + +static unsigned int snd_ice1712_get_gpio_mask(struct snd_ice1712 *ice) +{ + return snd_ice1712_read(ice, ICE1712_IREG_GPIO_WRITE_MASK); +} + static void snd_ice1712_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) { snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, data); @@ -2557,7 +2567,9 @@ static int __devinit snd_ice1712_create(struct snd_card *card, mutex_init(&ice->i2c_mutex); mutex_init(&ice->open_mutex); ice->gpio.set_mask = snd_ice1712_set_gpio_mask; + ice->gpio.get_mask = snd_ice1712_get_gpio_mask; ice->gpio.set_dir = snd_ice1712_set_gpio_dir; + ice->gpio.get_dir = snd_ice1712_get_gpio_dir; ice->gpio.set_data = snd_ice1712_set_gpio_data; ice->gpio.get_data = snd_ice1712_get_gpio_data; diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 9da2dae64c5b..b31a59d0625c 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -359,7 +359,9 @@ struct snd_ice1712 { unsigned int saved[2]; /* for ewx_i2c */ /* operators */ void (*set_mask)(struct snd_ice1712 *ice, unsigned int data); + unsigned int (*get_mask)(struct snd_ice1712 *ice); void (*set_dir)(struct snd_ice1712 *ice, unsigned int data); + unsigned int (*get_dir)(struct snd_ice1712 *ice); void (*set_data)(struct snd_ice1712 *ice, unsigned int data); unsigned int (*get_data)(struct snd_ice1712 *ice); /* misc operators - move to another place? */ @@ -399,6 +401,11 @@ static inline void snd_ice1712_gpio_set_dir(struct snd_ice1712 *ice, unsigned in ice->gpio.set_dir(ice, bits); } +static inline unsigned int snd_ice1712_gpio_get_dir(struct snd_ice1712 *ice) +{ + return ice->gpio.get_dir(ice); +} + static inline void snd_ice1712_gpio_set_mask(struct snd_ice1712 *ice, unsigned int bits) { ice->gpio.set_mask(ice, bits); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index af6e00148621..2213beec009a 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -196,6 +196,12 @@ static void snd_vt1724_set_gpio_dir(struct snd_ice1712 *ice, unsigned int data) inw(ICEREG1724(ice, GPIO_DIRECTION)); /* dummy read for pci-posting */ } +/* get gpio direction 0 = read, 1 = write */ +static unsigned int snd_vt1724_get_gpio_dir(struct snd_ice1712 *ice) +{ + return inl(ICEREG1724(ice, GPIO_DIRECTION)); +} + /* set the gpio mask (0 = writable) */ static void snd_vt1724_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) { @@ -205,6 +211,17 @@ static void snd_vt1724_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) inw(ICEREG1724(ice, GPIO_WRITE_MASK)); /* dummy read for pci-posting */ } +static unsigned int snd_vt1724_get_gpio_mask(struct snd_ice1712 *ice) +{ + unsigned int mask; + if (!ice->vt1720) + mask = (unsigned int)inb(ICEREG1724(ice, GPIO_WRITE_MASK_22)); + else + mask = 0; + mask = (mask << 16) | inw(ICEREG1724(ice, GPIO_WRITE_MASK)); + return mask; +} + static void snd_vt1724_set_gpio_data(struct snd_ice1712 *ice, unsigned int data) { outw(data, ICEREG1724(ice, GPIO_DATA)); @@ -2434,7 +2451,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card, mutex_init(&ice->open_mutex); mutex_init(&ice->i2c_mutex); ice->gpio.set_mask = snd_vt1724_set_gpio_mask; + ice->gpio.get_mask = snd_vt1724_get_gpio_mask; ice->gpio.set_dir = snd_vt1724_set_gpio_dir; + ice->gpio.get_dir = snd_vt1724_get_gpio_dir; ice->gpio.set_data = snd_vt1724_set_gpio_data; ice->gpio.get_data = snd_vt1724_get_gpio_data; ice->card = card; -- cgit v1.2.2 From 6796d5a05f4d3caad17d2586b3e5776fda50ef82 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:39 +0200 Subject: ALSA: ice1724 - pro-rate-locking makes sense only for internal clock mode * pro-rate-locking applies to internal clock mode only * required rate and current rate are compared for internal clock mode only Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 23 ++++++++++++++++------- 1 file changed, 16 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 2213beec009a..514e15385f7a 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -118,9 +118,12 @@ static inline int stdclock_is_spdif_master(struct snd_ice1712 *ice) return (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER) ? 1 : 0; } +/* + * locking rate makes sense only for internal clock mode + */ static inline int is_pro_rate_locked(struct snd_ice1712 *ice) { - return ice->is_spdif_master(ice) || PRO_RATE_LOCKED; + return (!ice->is_spdif_master(ice)) && PRO_RATE_LOCKED; } /* @@ -668,16 +671,22 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, return -EBUSY; } if (!force && is_pro_rate_locked(ice)) { + /* comparing required and current rate - makes sense for + * internal clock only */ spin_unlock_irqrestore(&ice->reg_lock, flags); return (rate == ice->cur_rate) ? 0 : -EBUSY; } - old_rate = ice->get_rate(ice); - if (force || (old_rate != rate)) - ice->set_rate(ice, rate); - else if (rate == ice->cur_rate) { - spin_unlock_irqrestore(&ice->reg_lock, flags); - return 0; + if (force || !ice->is_spdif_master(ice)) { + /* force means the rate was switched by ucontrol, otherwise + * setting clock rate for internal clock mode */ + old_rate = ice->get_rate(ice); + if (force || (old_rate != rate)) + ice->set_rate(ice, rate); + else if (rate == ice->cur_rate) { + spin_unlock_irqrestore(&ice->reg_lock, flags); + return 0; + } } ice->cur_rate = rate; -- cgit v1.2.2 From 1ff97cb9dd9f53b33ce6710a4f861f43e70e8ca4 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:40 +0200 Subject: ALSA: ice1724 - Support for multiple external clock types * Support for customization of the external clock names * Adding hooks to playback_pro_open and capture_pro_open, allowing e.g. limiting available stream rates to a single value when the external clock rate is detected Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.h | 7 ++++-- sound/pci/ice1712/ice1724.c | 58 +++++++++++++++++++++++++++++++++++---------- sound/pci/ice1712/juli.c | 3 ++- 3 files changed, 52 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index b31a59d0625c..4615bca39e18 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -379,8 +379,11 @@ struct snd_ice1712 { unsigned int (*get_rate)(struct snd_ice1712 *ice); void (*set_rate)(struct snd_ice1712 *ice, unsigned int rate); unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate); - void (*set_spdif_clock)(struct snd_ice1712 *ice); - + int (*set_spdif_clock)(struct snd_ice1712 *ice, int type); + int (*get_spdif_master_type)(struct snd_ice1712 *ice); + char **ext_clock_names; + int ext_clock_count; + void (*pro_open)(struct snd_ice1712 *, struct snd_pcm_substream *); #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 514e15385f7a..3f11195b2631 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -104,6 +104,8 @@ static int PRO_RATE_LOCKED; static int PRO_RATE_RESET = 1; static unsigned int PRO_RATE_DEFAULT = 44100; +static char *ext_clock_names[1] = { "IEC958 In" }; + /* * Basic I/O */ @@ -1042,6 +1044,8 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->pro_open) + ice->pro_open(ice, substream); return 0; } @@ -1060,6 +1064,8 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->pro_open) + ice->pro_open(ice, substream); return 0; } @@ -1813,15 +1819,21 @@ static int snd_vt1724_pro_internal_clock_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + int hw_rates_count = ice->hw_rates->count; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = ice->hw_rates->count + 1; + + uinfo->value.enumerated.items = hw_rates_count + ice->ext_clock_count; + /* upper limit - keep at top */ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - if (uinfo->value.enumerated.item == uinfo->value.enumerated.items - 1) - strcpy(uinfo->value.enumerated.name, "IEC958 Input"); + if (uinfo->value.enumerated.item >= hw_rates_count) + /* ext_clock items */ + strcpy(uinfo->value.enumerated.name, + ice->ext_clock_names[ + uinfo->value.enumerated.item - hw_rates_count]); else + /* int clock items */ sprintf(uinfo->value.enumerated.name, "%d", ice->hw_rates->list[uinfo->value.enumerated.item]); return 0; @@ -1835,7 +1847,8 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol, spin_lock_irq(&ice->reg_lock); if (ice->is_spdif_master(ice)) { - ucontrol->value.enumerated.item[0] = ice->hw_rates->count; + ucontrol->value.enumerated.item[0] = ice->hw_rates->count + + ice->get_spdif_master_type(ice); } else { rate = ice->get_rate(ice); ucontrol->value.enumerated.item[0] = 0; @@ -1850,8 +1863,14 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol, return 0; } +static int stdclock_get_spdif_master_type(struct snd_ice1712 *ice) +{ + /* standard external clock - only single type - SPDIF IN */ + return 0; +} + /* setting clock to external - SPDIF */ -static void stdclock_set_spdif_clock(struct snd_ice1712 *ice) +static int stdclock_set_spdif_clock(struct snd_ice1712 *ice, int type) { unsigned char oval; unsigned char i2s_oval; @@ -1860,27 +1879,30 @@ static void stdclock_set_spdif_clock(struct snd_ice1712 *ice) /* setting 256fs */ i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); + return 0; } + static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int old_rate, new_rate; unsigned int item = ucontrol->value.enumerated.item[0]; - unsigned int spdif = ice->hw_rates->count; + unsigned int first_ext_clock = ice->hw_rates->count; - if (item > spdif) + if (item > first_ext_clock + ice->ext_clock_count - 1) return -EINVAL; + /* if rate = 0 => external clock */ spin_lock_irq(&ice->reg_lock); if (ice->is_spdif_master(ice)) old_rate = 0; else old_rate = ice->get_rate(ice); - if (item == spdif) { - /* switching to external clock via SPDIF */ - ice->set_spdif_clock(ice); + if (item >= first_ext_clock) { + /* switching to external clock */ + ice->set_spdif_clock(ice, item - first_ext_clock); new_rate = 0; } else { /* internal on-card clock */ @@ -1892,7 +1914,7 @@ static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol, } spin_unlock_irq(&ice->reg_lock); - /* the first reset to the SPDIF master mode? */ + /* the first switch to the ext. clock mode? */ if (old_rate != new_rate && !new_rate) { /* notify akm chips as well */ unsigned int i; @@ -2550,6 +2572,9 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, return err; } + /* field init before calling chip_init */ + ice->ext_clock_count = 0; + for (tbl = card_tables; *tbl; tbl++) { for (c = *tbl; c->subvendor; c++) { if (c->subvendor == ice->eeprom.subvendor) { @@ -2588,6 +2613,13 @@ __found: ice->set_mclk = stdclock_set_mclk; if (!ice->set_spdif_clock) ice->set_spdif_clock = stdclock_set_spdif_clock; + if (!ice->get_spdif_master_type) + ice->get_spdif_master_type = stdclock_get_spdif_master_type; + if (!ice->ext_clock_names) + ice->ext_clock_names = ext_clock_names; + if (!ice->ext_clock_count) + ice->ext_clock_count = ARRAY_SIZE(ext_clock_names); + if (!ice->hw_rates) set_std_hw_rates(ice); @@ -2747,7 +2779,7 @@ static int snd_vt1724_resume(struct pci_dev *pci) if (ice->pm_saved_is_spdif_master) { /* switching to external clock via SPDIF */ - ice->set_spdif_clock(ice); + ice->set_spdif_clock(ice, 0); } else { /* internal on-card clock */ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 4789e8bfdc17..4bed9633a4cd 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -529,13 +529,14 @@ static inline unsigned char juli_set_mclk(struct snd_ice1712 *ice, } /* setting clock to external - SPDIF */ -static void juli_set_spdif_clock(struct snd_ice1712 *ice) +static int juli_set_spdif_clock(struct snd_ice1712 *ice, int type) { unsigned int old; old = ice->gpio.get_data(ice); /* external clock (= 0), multiply 1x, 48kHz */ ice->gpio.set_data(ice, (old & ~GPIO_RATE_MASK) | GPIO_MULTI_1X | GPIO_FREQ_48KHZ); + return 0; } /* Called when ak4114 detects change in the input SPDIF stream */ -- cgit v1.2.2 From 6ef80706184be792499a4485a7957f2660b6a076 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:41 +0200 Subject: ALSA: ice1724 - Infrasonic Quartet support * three external clock types * all controls supported Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/Makefile | 2 +- sound/pci/ice1712/ice1724.c | 3 + sound/pci/ice1712/quartet.c | 1130 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/ice1712/quartet.h | 10 + 4 files changed, 1144 insertions(+), 1 deletion(-) create mode 100644 sound/pci/ice1712/quartet.c create mode 100644 sound/pci/ice1712/quartet.h (limited to 'sound') diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile index 536eae2ccf94..f7ce33f00ea5 100644 --- a/sound/pci/ice1712/Makefile +++ b/sound/pci/ice1712/Makefile @@ -5,7 +5,7 @@ snd-ice17xx-ak4xxx-objs := ak4xxx.o snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o -snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o +snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o # Toplevel Module Dependency obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3f11195b2631..3896fb931de1 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -53,6 +53,7 @@ #include "phase.h" #include "wtm.h" #include "se.h" +#include "quartet.h" MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)"); @@ -70,6 +71,7 @@ MODULE_SUPPORTED_DEVICE("{" PHASE_DEVICE_DESC WTM_DEVICE_DESC SE_DEVICE_DESC + QTET_DEVICE_DESC "{VIA,VT1720}," "{VIA,VT1724}," "{ICEnsemble,Generic ICE1724}," @@ -2184,6 +2186,7 @@ static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_phase_cards, snd_vt1724_wtm_cards, snd_vt1724_se_cards, + snd_vt1724_qtet_cards, NULL, }; diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c new file mode 100644 index 000000000000..1948632787e6 --- /dev/null +++ b/sound/pci/ice1712/quartet.c @@ -0,0 +1,1130 @@ +/* + * ALSA driver for ICEnsemble VT1724 (Envy24HT) + * + * Lowlevel functions for Infrasonic Quartet + * + * Copyright (c) 2009 Pavel Hofman + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ice1712.h" +#include "envy24ht.h" +#include +#include "quartet.h" + +struct qtet_spec { + struct ak4113 *ak4113; + unsigned int scr; /* system control register */ + unsigned int mcr; /* monitoring control register */ + unsigned int cpld; /* cpld register */ +}; + +struct qtet_kcontrol_private { + unsigned int bit; + void (*set_register)(struct snd_ice1712 *ice, unsigned int val); + unsigned int (*get_register)(struct snd_ice1712 *ice); + unsigned char *texts[2]; +}; + +enum { + IN12_SEL = 0, + IN34_SEL, + AIN34_SEL, + COAX_OUT, + IN12_MON12, + IN12_MON34, + IN34_MON12, + IN34_MON34, + OUT12_MON34, + OUT34_MON12, +}; + +static char *ext_clock_names[3] = {"IEC958 In", "Word Clock 1xFS", + "Word Clock 256xFS"}; + +/* chip address on I2C bus */ +#define AK4113_ADDR 0x26 /* S/PDIF receiver */ + +/* chip address on SPI bus */ +#define AK4620_ADDR 0x02 /* ADC/DAC */ + + +/* + * GPIO pins + */ + +/* GPIO0 - O - DATA0, def. 0 */ +#define GPIO_D0 (1<<0) +/* GPIO1 - I/O - DATA1, Jack Detect Input0 (0:present, 1:missing), def. 1 */ +#define GPIO_D1_JACKDTC0 (1<<1) +/* GPIO2 - I/O - DATA2, Jack Detect Input1 (0:present, 1:missing), def. 1 */ +#define GPIO_D2_JACKDTC1 (1<<2) +/* GPIO3 - I/O - DATA3, def. 1 */ +#define GPIO_D3 (1<<3) +/* GPIO4 - I/O - DATA4, SPI CDTO, def. 1 */ +#define GPIO_D4_SPI_CDTO (1<<4) +/* GPIO5 - I/O - DATA5, SPI CCLK, def. 1 */ +#define GPIO_D5_SPI_CCLK (1<<5) +/* GPIO6 - I/O - DATA6, Cable Detect Input (0:detected, 1:not detected */ +#define GPIO_D6_CD (1<<6) +/* GPIO7 - I/O - DATA7, Device Detect Input (0:detected, 1:not detected */ +#define GPIO_D7_DD (1<<7) +/* GPIO8 - O - CPLD Chip Select, def. 1 */ +#define GPIO_CPLD_CSN (1<<8) +/* GPIO9 - O - CPLD register read/write (0:write, 1:read), def. 0 */ +#define GPIO_CPLD_RW (1<<9) +/* GPIO10 - O - SPI Chip Select for CODEC#0, def. 1 */ +#define GPIO_SPI_CSN0 (1<<10) +/* GPIO11 - O - SPI Chip Select for CODEC#1, def. 1 */ +#define GPIO_SPI_CSN1 (1<<11) +/* GPIO12 - O - Ex. Register Output Enable (0:enable, 1:disable), def. 1, + * init 0 */ +#define GPIO_EX_GPIOE (1<<12) +/* GPIO13 - O - Ex. Register0 Chip Select for System Control Register, + * def. 1 */ +#define GPIO_SCR (1<<13) +/* GPIO14 - O - Ex. Register1 Chip Select for Monitor Control Register, + * def. 1 */ +#define GPIO_MCR (1<<14) + +#define GPIO_SPI_ALL (GPIO_D4_SPI_CDTO | GPIO_D5_SPI_CCLK |\ + GPIO_SPI_CSN0 | GPIO_SPI_CSN1) + +#define GPIO_DATA_MASK (GPIO_D0 | GPIO_D1_JACKDTC0 | \ + GPIO_D2_JACKDTC1 | GPIO_D3 | \ + GPIO_D4_SPI_CDTO | GPIO_D5_SPI_CCLK | \ + GPIO_D6_CD | GPIO_D7_DD) + +/* System Control Register GPIO_SCR data bits */ +/* Mic/Line select relay (0:line, 1:mic) */ +#define SCR_RELAY GPIO_D0 +/* Phantom power drive control (0:5V, 1:48V) */ +#define SCR_PHP_V GPIO_D1_JACKDTC0 +/* H/W mute control (0:Normal, 1:Mute) */ +#define SCR_MUTE GPIO_D2_JACKDTC1 +/* Phantom power control (0:Phantom on, 1:off) */ +#define SCR_PHP GPIO_D3 +/* Analog input 1/2 Source Select */ +#define SCR_AIN12_SEL0 GPIO_D4_SPI_CDTO +#define SCR_AIN12_SEL1 GPIO_D5_SPI_CCLK +/* Analog input 3/4 Source Select (0:line, 1:hi-z) */ +#define SCR_AIN34_SEL GPIO_D6_CD +/* Codec Power Down (0:power down, 1:normal) */ +#define SCR_CODEC_PDN GPIO_D7_DD + +#define SCR_AIN12_LINE (0) +#define SCR_AIN12_MIC (SCR_AIN12_SEL0) +#define SCR_AIN12_LOWCUT (SCR_AIN12_SEL1 | SCR_AIN12_SEL0) + +/* Monitor Control Register GPIO_MCR data bits */ +/* Input 1/2 to Monitor 1/2 (0:off, 1:on) */ +#define MCR_IN12_MON12 GPIO_D0 +/* Input 1/2 to Monitor 3/4 (0:off, 1:on) */ +#define MCR_IN12_MON34 GPIO_D1_JACKDTC0 +/* Input 3/4 to Monitor 1/2 (0:off, 1:on) */ +#define MCR_IN34_MON12 GPIO_D2_JACKDTC1 +/* Input 3/4 to Monitor 3/4 (0:off, 1:on) */ +#define MCR_IN34_MON34 GPIO_D3 +/* Output to Monitor 1/2 (0:off, 1:on) */ +#define MCR_OUT34_MON12 GPIO_D4_SPI_CDTO +/* Output to Monitor 3/4 (0:off, 1:on) */ +#define MCR_OUT12_MON34 GPIO_D5_SPI_CCLK + +/* CPLD Register DATA bits */ +/* Clock Rate Select */ +#define CPLD_CKS0 GPIO_D0 +#define CPLD_CKS1 GPIO_D1_JACKDTC0 +#define CPLD_CKS2 GPIO_D2_JACKDTC1 +/* Sync Source Select (0:Internal, 1:External) */ +#define CPLD_SYNC_SEL GPIO_D3 +/* Word Clock FS Select (0:FS, 1:256FS) */ +#define CPLD_WORD_SEL GPIO_D4_SPI_CDTO +/* Coaxial Output Source (IS-Link) (0:SPDIF, 1:I2S) */ +#define CPLD_COAX_OUT GPIO_D5_SPI_CCLK +/* Input 1/2 Source Select (0:Analog12, 1:An34) */ +#define CPLD_IN12_SEL GPIO_D6_CD +/* Input 3/4 Source Select (0:Analog34, 1:Digital In) */ +#define CPLD_IN34_SEL GPIO_D7_DD + +/* internal clock (CPLD_SYNC_SEL = 0) options */ +#define CPLD_CKS_44100HZ (0) +#define CPLD_CKS_48000HZ (CPLD_CKS0) +#define CPLD_CKS_88200HZ (CPLD_CKS1) +#define CPLD_CKS_96000HZ (CPLD_CKS1 | CPLD_CKS0) +#define CPLD_CKS_176400HZ (CPLD_CKS2) +#define CPLD_CKS_192000HZ (CPLD_CKS2 | CPLD_CKS0) + +#define CPLD_CKS_MASK (CPLD_CKS0 | CPLD_CKS1 | CPLD_CKS2) + +/* external clock (CPLD_SYNC_SEL = 1) options */ +/* external clock - SPDIF */ +#define CPLD_EXT_SPDIF (0 | CPLD_SYNC_SEL) +/* external clock - WordClock 1xfs */ +#define CPLD_EXT_WORDCLOCK_1FS (CPLD_CKS1 | CPLD_SYNC_SEL) +/* external clock - WordClock 256xfs */ +#define CPLD_EXT_WORDCLOCK_256FS (CPLD_CKS1 | CPLD_WORD_SEL |\ + CPLD_SYNC_SEL) + +#define EXT_SPDIF_TYPE 0 +#define EXT_WORDCLOCK_1FS_TYPE 1 +#define EXT_WORDCLOCK_256FS_TYPE 2 + +#define AK4620_DFS0 (1<<0) +#define AK4620_DFS1 (1<<1) +#define AK4620_CKS0 (1<<2) +#define AK4620_CKS1 (1<<3) +/* Clock and Format Control register */ +#define AK4620_DFS_REG 0x02 + +/* Deem and Volume Control register */ +#define AK4620_DEEMVOL_REG 0x03 +#define AK4620_SMUTE (1<<7) + +/* + * Conversion from int value to its binary form. Used for debugging. + * The output buffer must be allocated prior to calling the function. + */ +static char *get_binary(char *buffer, int value) +{ + int i, j, pos; + pos = 0; + for (i = 0; i < 4; ++i) { + for (j = 0; j < 8; ++j) { + if (value & (1 << (31-(i*8 + j)))) + buffer[pos] = '1'; + else + buffer[pos] = '0'; + pos++; + } + if (i < 3) { + buffer[pos] = ' '; + pos++; + } + } + buffer[pos] = '\0'; + return buffer; +} + +/* + * Initial setup of the conversion array GPIO <-> rate + */ +static unsigned int qtet_rates[] = { + 44100, 48000, 88200, + 96000, 176400, 192000, +}; + +static unsigned int cks_vals[] = { + CPLD_CKS_44100HZ, CPLD_CKS_48000HZ, CPLD_CKS_88200HZ, + CPLD_CKS_96000HZ, CPLD_CKS_176400HZ, CPLD_CKS_192000HZ, +}; + +static struct snd_pcm_hw_constraint_list qtet_rates_info = { + .count = ARRAY_SIZE(qtet_rates), + .list = qtet_rates, + .mask = 0, +}; + +static void qtet_ak4113_write(void *private_data, unsigned char reg, + unsigned char val) +{ + snd_vt1724_write_i2c((struct snd_ice1712 *)private_data, AK4113_ADDR, + reg, val); +} + +static unsigned char qtet_ak4113_read(void *private_data, unsigned char reg) +{ + return snd_vt1724_read_i2c((struct snd_ice1712 *)private_data, + AK4113_ADDR, reg); +} + + +/* + * AK4620 section + */ + +/* + * Write data to addr register of ak4620 + */ +static void qtet_akm_write(struct snd_akm4xxx *ak, int chip, + unsigned char addr, unsigned char data) +{ + unsigned int tmp, orig_dir; + int idx; + unsigned int addrdata; + struct snd_ice1712 *ice = ak->private_data[0]; + + if (snd_BUG_ON(chip < 0 || chip >= 4)) + return; + /*printk(KERN_DEBUG "Writing to AK4620: chip=%d, addr=0x%x, + data=0x%x\n", chip, addr, data);*/ + orig_dir = ice->gpio.get_dir(ice); + ice->gpio.set_dir(ice, orig_dir | GPIO_SPI_ALL); + /* set mask - only SPI bits */ + ice->gpio.set_mask(ice, ~GPIO_SPI_ALL); + + tmp = ice->gpio.get_data(ice); + /* high all */ + tmp |= GPIO_SPI_ALL; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* drop chip select */ + if (chip) + /* CODEC 1 */ + tmp &= ~GPIO_SPI_CSN1; + else + tmp &= ~GPIO_SPI_CSN0; + ice->gpio.set_data(ice, tmp); + udelay(100); + + /* build I2C address + data byte */ + addrdata = (AK4620_ADDR << 6) | 0x20 | (addr & 0x1f); + addrdata = (addrdata << 8) | data; + for (idx = 15; idx >= 0; idx--) { + /* drop clock */ + tmp &= ~GPIO_D5_SPI_CCLK; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* set data */ + if (addrdata & (1 << idx)) + tmp |= GPIO_D4_SPI_CDTO; + else + tmp &= ~GPIO_D4_SPI_CDTO; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* raise clock */ + tmp |= GPIO_D5_SPI_CCLK; + ice->gpio.set_data(ice, tmp); + udelay(100); + } + /* all back to 1 */ + tmp |= GPIO_SPI_ALL; + ice->gpio.set_data(ice, tmp); + udelay(100); + + /* return all gpios to non-writable */ + ice->gpio.set_mask(ice, 0xffffff); + /* restore GPIOs direction */ + ice->gpio.set_dir(ice, orig_dir); +} + +static void qtet_akm_set_regs(struct snd_akm4xxx *ak, unsigned char addr, + unsigned char mask, unsigned char value) +{ + unsigned char tmp; + int chip; + for (chip = 0; chip < ak->num_chips; chip++) { + tmp = snd_akm4xxx_get(ak, chip, addr); + /* clear the bits */ + tmp &= ~mask; + /* set the new bits */ + tmp |= value; + snd_akm4xxx_write(ak, chip, addr, tmp); + } +} + +/* + * change the rate of AK4620 + */ +static void qtet_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) +{ + unsigned char ak4620_dfs; + + if (rate == 0) /* no hint - S/PDIF input is master or the new spdif + input rate undetected, simply return */ + return; + + /* adjust DFS on codecs - see datasheet */ + if (rate > 108000) + ak4620_dfs = AK4620_DFS1 | AK4620_CKS1; + else if (rate > 54000) + ak4620_dfs = AK4620_DFS0 | AK4620_CKS0; + else + ak4620_dfs = 0; + + /* set new value */ + qtet_akm_set_regs(ak, AK4620_DFS_REG, AK4620_DFS0 | AK4620_DFS1 | + AK4620_CKS0 | AK4620_CKS1, ak4620_dfs); +} + +#define AK_CONTROL(xname, xch) { .name = xname, .num_channels = xch } + +#define PCM_12_PLAYBACK_VOLUME "PCM 1/2 Playback Volume" +#define PCM_34_PLAYBACK_VOLUME "PCM 3/4 Playback Volume" +#define PCM_12_CAPTURE_VOLUME "PCM 1/2 Capture Volume" +#define PCM_34_CAPTURE_VOLUME "PCM 3/4 Capture Volume" + +static const struct snd_akm4xxx_dac_channel qtet_dac[] = { + AK_CONTROL(PCM_12_PLAYBACK_VOLUME, 2), + AK_CONTROL(PCM_34_PLAYBACK_VOLUME, 2), +}; + +static const struct snd_akm4xxx_adc_channel qtet_adc[] = { + AK_CONTROL(PCM_12_CAPTURE_VOLUME, 2), + AK_CONTROL(PCM_34_CAPTURE_VOLUME, 2), +}; + +static struct snd_akm4xxx akm_qtet_dac __devinitdata = { + .type = SND_AK4620, + .num_dacs = 4, /* DAC1 - Output 12 + */ + .num_adcs = 4, /* ADC1 - Input 12 + */ + .ops = { + .write = qtet_akm_write, + .set_rate_val = qtet_akm_set_rate_val, + }, + .dac_info = qtet_dac, + .adc_info = qtet_adc, +}; + +/* Communication routines with the CPLD */ + + +/* Writes data to external register reg, both reg and data are + * GPIO representations */ +static void reg_write(struct snd_ice1712 *ice, unsigned int reg, + unsigned int data) +{ + unsigned int tmp; + + mutex_lock(&ice->gpio_mutex); + /* set direction of used GPIOs*/ + /* all outputs */ + tmp = 0x00ffff; + ice->gpio.set_dir(ice, tmp); + /* mask - writable bits */ + ice->gpio.set_mask(ice, ~(tmp)); + /* write the data */ + tmp = ice->gpio.get_data(ice); + tmp &= ~GPIO_DATA_MASK; + tmp |= data; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* drop output enable */ + tmp &= ~GPIO_EX_GPIOE; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* drop the register gpio */ + tmp &= ~reg; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* raise the register GPIO */ + tmp |= reg; + ice->gpio.set_data(ice, tmp); + udelay(100); + + /* raise all data gpios */ + tmp |= GPIO_DATA_MASK; + ice->gpio.set_data(ice, tmp); + /* mask - immutable bits */ + ice->gpio.set_mask(ice, 0xffffff); + /* outputs only 8-15 */ + ice->gpio.set_dir(ice, 0x00ff00); + mutex_unlock(&ice->gpio_mutex); +} + +static unsigned int get_scr(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + return spec->scr; +} + +static unsigned int get_mcr(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + return spec->mcr; +} + +static unsigned int get_cpld(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + return spec->cpld; +} + +static void set_scr(struct snd_ice1712 *ice, unsigned int val) +{ + struct qtet_spec *spec = ice->spec; + reg_write(ice, GPIO_SCR, val); + spec->scr = val; +} + +static void set_mcr(struct snd_ice1712 *ice, unsigned int val) +{ + struct qtet_spec *spec = ice->spec; + reg_write(ice, GPIO_MCR, val); + spec->mcr = val; +} + +static void set_cpld(struct snd_ice1712 *ice, unsigned int val) +{ + struct qtet_spec *spec = ice->spec; + reg_write(ice, GPIO_CPLD_CSN, val); + spec->cpld = val; +} +#ifdef CONFIG_PROC_FS +static void proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = entry->private_data; + char bin_buffer[36]; + + snd_iprintf(buffer, "SCR: %s\n", get_binary(bin_buffer, + get_scr(ice))); + snd_iprintf(buffer, "MCR: %s\n", get_binary(bin_buffer, + get_mcr(ice))); + snd_iprintf(buffer, "CPLD: %s\n", get_binary(bin_buffer, + get_cpld(ice))); +} + +static void proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "quartet", &entry)) + snd_info_set_text_ops(entry, ice, proc_regs_read); +} +#else /* !CONFIG_PROC_FS */ +static void proc_init(struct snd_ice1712 *ice) {} +#endif + +static int qtet_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val; + val = get_scr(ice) & SCR_MUTE; + ucontrol->value.integer.value[0] = (val) ? 0 : 1; + return 0; +} + +static int qtet_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new, smute; + old = get_scr(ice) & SCR_MUTE; + if (ucontrol->value.integer.value[0]) { + /* unmute */ + new = 0; + /* un-smuting DAC */ + smute = 0; + } else { + /* mute */ + new = SCR_MUTE; + /* smuting DAC */ + smute = AK4620_SMUTE; + } + if (old != new) { + struct snd_akm4xxx *ak = ice->akm; + set_scr(ice, (get_scr(ice) & ~SCR_MUTE) | new); + /* set smute */ + qtet_akm_set_regs(ak, AK4620_DEEMVOL_REG, AK4620_SMUTE, smute); + return 1; + } + /* no change */ + return 0; +} + +static int qtet_ain12_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[3] = {"Line In 1/2", "Mic", "Mic + Low-cut"}; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = ARRAY_SIZE(texts); + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int qtet_ain12_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val, result; + val = get_scr(ice) & (SCR_AIN12_SEL1 | SCR_AIN12_SEL0); + switch (val) { + case SCR_AIN12_LINE: + result = 0; + break; + case SCR_AIN12_MIC: + result = 1; + break; + case SCR_AIN12_LOWCUT: + result = 2; + break; + default: + /* BUG - no other combinations allowed */ + snd_BUG(); + result = 0; + } + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int qtet_ain12_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new, tmp, masked_old; + old = new = get_scr(ice); + masked_old = old & (SCR_AIN12_SEL1 | SCR_AIN12_SEL0); + tmp = ucontrol->value.integer.value[0]; + if (tmp == 2) + tmp = 3; /* binary 10 is not supported */ + tmp <<= 4; /* shifting to SCR_AIN12_SEL0 */ + if (tmp != masked_old) { + /* change requested */ + switch (tmp) { + case SCR_AIN12_LINE: + new = old & ~(SCR_AIN12_SEL1 | SCR_AIN12_SEL0); + set_scr(ice, new); + /* turn off relay */ + new &= ~SCR_RELAY; + set_scr(ice, new); + break; + case SCR_AIN12_MIC: + /* turn on relay */ + new = old | SCR_RELAY; + set_scr(ice, new); + new = (new & ~SCR_AIN12_SEL1) | SCR_AIN12_SEL0; + set_scr(ice, new); + break; + case SCR_AIN12_LOWCUT: + /* turn on relay */ + new = old | SCR_RELAY; + set_scr(ice, new); + new |= SCR_AIN12_SEL1 | SCR_AIN12_SEL0; + set_scr(ice, new); + break; + default: + snd_BUG(); + } + return 1; + } + /* no change */ + return 0; +} + +static int qtet_php_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val; + /* if phantom voltage =48V, phantom on */ + val = get_scr(ice) & SCR_PHP_V; + ucontrol->value.integer.value[0] = val ? 1 : 0; + return 0; +} + +static int qtet_php_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new; + old = new = get_scr(ice); + if (ucontrol->value.integer.value[0] /* phantom on requested */ + && (~old & SCR_PHP_V)) /* 0 = voltage 5V */ { + /* is off, turn on */ + /* turn voltage on first, = 1 */ + new = old | SCR_PHP_V; + set_scr(ice, new); + /* turn phantom on, = 0 */ + new &= ~SCR_PHP; + set_scr(ice, new); + } else if (!ucontrol->value.integer.value[0] && (old & SCR_PHP_V)) { + /* phantom off requested and 1 = voltage 48V */ + /* is on, turn off */ + /* turn voltage off first, = 0 */ + new = old & ~SCR_PHP_V; + set_scr(ice, new); + /* turn phantom off, = 1 */ + new |= SCR_PHP; + set_scr(ice, new); + } + if (old != new) + return 1; + /* no change */ + return 0; +} + +#define PRIV_SW(xid, xbit, xreg) [xid] = {.bit = xbit,\ + .set_register = set_##xreg,\ + .get_register = get_##xreg, } + + +#define PRIV_ENUM2(xid, xbit, xreg, xtext1, xtext2) [xid] = {.bit = xbit,\ + .set_register = set_##xreg,\ + .get_register = get_##xreg,\ + .texts = {xtext1, xtext2} } + +static struct qtet_kcontrol_private qtet_privates[] = { + PRIV_ENUM2(IN12_SEL, CPLD_IN12_SEL, cpld, "An In 1/2", "An In 3/4"), + PRIV_ENUM2(IN34_SEL, CPLD_IN34_SEL, cpld, "An In 3/4", "IEC958 In"), + PRIV_ENUM2(AIN34_SEL, SCR_AIN34_SEL, scr, "Line In 3/4", "Hi-Z"), + PRIV_ENUM2(COAX_OUT, CPLD_COAX_OUT, cpld, "IEC958", "I2S"), + PRIV_SW(IN12_MON12, MCR_IN12_MON12, mcr), + PRIV_SW(IN12_MON34, MCR_IN12_MON34, mcr), + PRIV_SW(IN34_MON12, MCR_IN34_MON12, mcr), + PRIV_SW(IN34_MON34, MCR_IN34_MON34, mcr), + PRIV_SW(OUT12_MON34, MCR_OUT12_MON34, mcr), + PRIV_SW(OUT34_MON12, MCR_OUT34_MON12, mcr), +}; + +static int qtet_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct qtet_kcontrol_private private = + qtet_privates[kcontrol->private_value]; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = ARRAY_SIZE(private.texts); + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + private.texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int qtet_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct qtet_kcontrol_private private = + qtet_privates[kcontrol->private_value]; + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = + (private.get_register(ice) & private.bit) ? 1 : 0; + return 0; +} + +static int qtet_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct qtet_kcontrol_private private = + qtet_privates[kcontrol->private_value]; + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new; + old = private.get_register(ice); + if (ucontrol->value.integer.value[0]) + new = old | private.bit; + else + new = old & ~private.bit; + if (old != new) { + private.set_register(ice, new); + return 1; + } + /* no change */ + return 0; +} + +#define qtet_sw_info snd_ctl_boolean_mono_info + +#define QTET_CONTROL(xname, xtype, xpriv) \ + {.iface = SNDRV_CTL_ELEM_IFACE_MIXER,\ + .name = xname,\ + .info = qtet_##xtype##_info,\ + .get = qtet_sw_get,\ + .put = qtet_sw_put,\ + .private_value = xpriv } + +static struct snd_kcontrol_new qtet_controls[] __devinitdata = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = qtet_sw_info, + .get = qtet_mute_get, + .put = qtet_mute_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Phantom Power", + .info = qtet_sw_info, + .get = qtet_php_get, + .put = qtet_php_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog In 1/2 Capture Switch", + .info = qtet_ain12_enum_info, + .get = qtet_ain12_sw_get, + .put = qtet_ain12_sw_put, + .private_value = 0 + }, + QTET_CONTROL("Analog In 3/4 Capture Switch", enum, AIN34_SEL), + QTET_CONTROL("PCM In 1/2 Capture Switch", enum, IN12_SEL), + QTET_CONTROL("PCM In 3/4 Capture Switch", enum, IN34_SEL), + QTET_CONTROL("Coax Output Source", enum, COAX_OUT), + QTET_CONTROL("Analog In 1/2 to Monitor 1/2", sw, IN12_MON12), + QTET_CONTROL("Analog In 1/2 to Monitor 3/4", sw, IN12_MON34), + QTET_CONTROL("Analog In 3/4 to Monitor 1/2", sw, IN34_MON12), + QTET_CONTROL("Analog In 3/4 to Monitor 3/4", sw, IN34_MON34), + QTET_CONTROL("Output 1/2 to Monitor 3/4", sw, OUT12_MON34), + QTET_CONTROL("Output 3/4 to Monitor 1/2", sw, OUT34_MON12), +}; + +static char *slave_vols[] __devinitdata = { + PCM_12_PLAYBACK_VOLUME, + PCM_34_PLAYBACK_VOLUME, + NULL +}; + +static __devinitdata +DECLARE_TLV_DB_SCALE(qtet_master_db_scale, -6350, 50, 1); + +static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card, + const char *name) +{ + struct snd_ctl_elem_id sid; + memset(&sid, 0, sizeof(sid)); + /* FIXME: strcpy is bad. */ + strcpy(sid.name, name); + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_find_id(card, &sid); +} + +static void __devinit add_slaves(struct snd_card *card, + struct snd_kcontrol *master, char **list) +{ + for (; *list; list++) { + struct snd_kcontrol *slave = ctl_find(card, *list); + if (slave) + snd_ctl_add_slave(master, slave); + } +} + +static int __devinit qtet_add_controls(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + int err, i; + struct snd_kcontrol *vmaster; + err = snd_ice1712_akm4xxx_build_controls(ice); + if (err < 0) + return err; + for (i = 0; i < ARRAY_SIZE(qtet_controls); i++) { + err = snd_ctl_add(ice->card, + snd_ctl_new1(&qtet_controls[i], ice)); + if (err < 0) + return err; + } + + /* Create virtual master control */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", + qtet_master_db_scale); + if (!vmaster) + return -ENOMEM; + add_slaves(ice->card, vmaster, slave_vols); + err = snd_ctl_add(ice->card, vmaster); + if (err < 0) + return err; + /* only capture SPDIF over AK4113 */ + err = snd_ak4113_build(spec->ak4113, + ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); + if (err < 0) + return err; + return 0; +} + +static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) +{ + /* CPLD_SYNC_SEL: 0 = internal, 1 = external (i.e. spdif master) */ + return (get_cpld(ice) & CPLD_SYNC_SEL) ? 1 : 0; +} + +static unsigned int qtet_get_rate(struct snd_ice1712 *ice) +{ + int i; + unsigned char result; + + result = get_cpld(ice) & CPLD_CKS_MASK; + for (i = 0; i < ARRAY_SIZE(cks_vals); i++) + if (cks_vals[i] == result) + return qtet_rates[i]; + return 0; +} + +static int get_cks_val(int rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(qtet_rates); i++) + if (qtet_rates[i] == rate) + return cks_vals[i]; + return 0; +} + +/* setting new rate */ +static void qtet_set_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned int new; + unsigned char val; + /* switching ice1724 to external clock - supplied by ext. circuits */ + val = inb(ICEMT1724(ice, RATE)); + outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); + + new = (get_cpld(ice) & ~CPLD_CKS_MASK) | get_cks_val(rate); + /* switch to internal clock, drop CPLD_SYNC_SEL */ + new &= ~CPLD_SYNC_SEL; + /* printk(KERN_DEBUG "QT - set_rate: old %x, new %x\n", + get_cpld(ice), new); */ + set_cpld(ice, new); +} + +static inline unsigned char qtet_set_mclk(struct snd_ice1712 *ice, + unsigned int rate) +{ + /* no change in master clock */ + return 0; +} + +/* setting clock to external - SPDIF */ +static int qtet_set_spdif_clock(struct snd_ice1712 *ice, int type) +{ + unsigned int old, new; + + old = new = get_cpld(ice); + new &= ~(CPLD_CKS_MASK | CPLD_WORD_SEL); + switch (type) { + case EXT_SPDIF_TYPE: + new |= CPLD_EXT_SPDIF; + break; + case EXT_WORDCLOCK_1FS_TYPE: + new |= CPLD_EXT_WORDCLOCK_1FS; + break; + case EXT_WORDCLOCK_256FS_TYPE: + new |= CPLD_EXT_WORDCLOCK_256FS; + break; + default: + snd_BUG(); + } + if (old != new) { + set_cpld(ice, new); + /* changed */ + return 1; + } + return 0; +} + +static int qtet_get_spdif_master_type(struct snd_ice1712 *ice) +{ + unsigned int val; + int result; + val = get_cpld(ice); + /* checking only rate/clock-related bits */ + val &= (CPLD_CKS_MASK | CPLD_WORD_SEL | CPLD_SYNC_SEL); + if (!(val & CPLD_SYNC_SEL)) { + /* switched to internal clock, is not any external type */ + result = -1; + } else { + switch (val) { + case (CPLD_EXT_SPDIF): + result = EXT_SPDIF_TYPE; + break; + case (CPLD_EXT_WORDCLOCK_1FS): + result = EXT_WORDCLOCK_1FS_TYPE; + break; + case (CPLD_EXT_WORDCLOCK_256FS): + result = EXT_WORDCLOCK_256FS_TYPE; + break; + default: + /* undefined combination of external clock setup */ + snd_BUG(); + result = 0; + } + } + return result; +} + +/* Called when ak4113 detects change in the input SPDIF stream */ +static void qtet_ak4113_change(struct ak4113 *ak4113, unsigned char c0, + unsigned char c1) +{ + struct snd_ice1712 *ice = ak4113->change_callback_private; + int rate; + if ((qtet_get_spdif_master_type(ice) == EXT_SPDIF_TYPE) && + c1) { + /* only for SPDIF master mode, rate was changed */ + rate = snd_ak4113_external_rate(ak4113); + /* printk(KERN_DEBUG "ak4113 - input rate changed to %d\n", + rate); */ + qtet_akm_set_rate_val(ice->akm, rate); + } +} + +/* + * If clock slaved to SPDIF-IN, setting runtime rate + * to the detected external rate + */ +static void qtet_spdif_in_open(struct snd_ice1712 *ice, + struct snd_pcm_substream *substream) +{ + struct qtet_spec *spec = ice->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int rate; + + if (qtet_get_spdif_master_type(ice) != EXT_SPDIF_TYPE) + /* not external SPDIF, no rate limitation */ + return; + /* only external SPDIF can detect incoming sample rate */ + rate = snd_ak4113_external_rate(spec->ak4113); + if (rate >= runtime->hw.rate_min && rate <= runtime->hw.rate_max) { + runtime->hw.rate_min = rate; + runtime->hw.rate_max = rate; + } +} + +/* + * initialize the chip + */ +static int __devinit qtet_init(struct snd_ice1712 *ice) +{ + static const unsigned char ak4113_init_vals[] = { + /* AK4113_REG_PWRDN */ AK4113_RST | AK4113_PWN | + AK4113_OCKS0 | AK4113_OCKS1, + /* AK4113_REQ_FORMAT */ AK4113_DIF_I24I2S | AK4113_VTX | + AK4113_DEM_OFF | AK4113_DEAU, + /* AK4113_REG_IO0 */ AK4113_OPS2 | AK4113_TXE | + AK4113_XTL_24_576M, + /* AK4113_REG_IO1 */ AK4113_EFH_1024LRCLK | AK4113_IPS(0), + /* AK4113_REG_INT0_MASK */ 0, + /* AK4113_REG_INT1_MASK */ 0, + /* AK4113_REG_DATDTS */ 0, + }; + int err; + struct qtet_spec *spec; + struct snd_akm4xxx *ak; + unsigned char val; + + /* switching ice1724 to external clock - supplied by ext. circuits */ + val = inb(ICEMT1724(ice, RATE)); + outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + /* qtet is clocked by Xilinx array */ + ice->hw_rates = &qtet_rates_info; + ice->is_spdif_master = qtet_is_spdif_master; + ice->get_rate = qtet_get_rate; + ice->set_rate = qtet_set_rate; + ice->set_mclk = qtet_set_mclk; + ice->set_spdif_clock = qtet_set_spdif_clock; + ice->get_spdif_master_type = qtet_get_spdif_master_type; + ice->ext_clock_names = ext_clock_names; + ice->ext_clock_count = ARRAY_SIZE(ext_clock_names); + /* since Qtet can detect correct SPDIF-in rate, all streams can be + * limited to this specific rate */ + ice->spdif.ops.open = ice->pro_open = qtet_spdif_in_open; + ice->spec = spec; + + /* Mute Off */ + /* SCR Initialize*/ + /* keep codec power down first */ + set_scr(ice, SCR_PHP); + udelay(1); + /* codec power up */ + set_scr(ice, SCR_PHP | SCR_CODEC_PDN); + + /* MCR Initialize */ + set_mcr(ice, 0); + + /* CPLD Initialize */ + set_cpld(ice, 0); + + + ice->num_total_dacs = 2; + ice->num_total_adcs = 2; + + ice->akm = kcalloc(2, sizeof(struct snd_akm4xxx), GFP_KERNEL); + ak = ice->akm; + if (!ak) + return -ENOMEM; + /* only one codec with two chips */ + ice->akm_codecs = 1; + err = snd_ice1712_akm4xxx_init(ak, &akm_qtet_dac, NULL, ice); + if (err < 0) + return err; + err = snd_ak4113_create(ice->card, + qtet_ak4113_read, + qtet_ak4113_write, + ak4113_init_vals, + ice, &spec->ak4113); + if (err < 0) + return err; + /* callback for codecs rate setting */ + spec->ak4113->change_callback = qtet_ak4113_change; + spec->ak4113->change_callback_private = ice; + /* AK41143 in Quartet can detect external rate correctly + * (i.e. check_flags = 0) */ + spec->ak4113->check_flags = 0; + + proc_init(ice); + + qtet_set_rate(ice, 44100); + return 0; +} + +static unsigned char qtet_eeprom[] __devinitdata = { + [ICE_EEP2_SYSCONF] = 0x28, /* clock 256(24MHz), mpu401, 1xADC, + 1xDACs, SPDIF in */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0x78, /* 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, in, out-ext */ + [ICE_EEP2_GPIO_DIR] = 0x00, /* 0-7 inputs, switched to output + only during output operations */ + [ICE_EEP2_GPIO_DIR1] = 0xff, /* 8-15 outputs */ + [ICE_EEP2_GPIO_DIR2] = 0x00, + [ICE_EEP2_GPIO_MASK] = 0xff, /* changed only for OUT operations */ + [ICE_EEP2_GPIO_MASK1] = 0x00, + [ICE_EEP2_GPIO_MASK2] = 0xff, + + [ICE_EEP2_GPIO_STATE] = 0x00, /* inputs */ + [ICE_EEP2_GPIO_STATE1] = 0x7d, /* all 1, but GPIO_CPLD_RW + and GPIO15 always zero */ + [ICE_EEP2_GPIO_STATE2] = 0x00, /* inputs */ +}; + +/* entry point */ +struct snd_ice1712_card_info snd_vt1724_qtet_cards[] __devinitdata = { + { + .subvendor = VT1724_SUBDEVICE_QTET, + .name = "Infrasonic Quartet", + .model = "quartet", + .chip_init = qtet_init, + .build_controls = qtet_add_controls, + .eeprom_size = sizeof(qtet_eeprom), + .eeprom_data = qtet_eeprom, + }, + { } /* terminator */ +}; diff --git a/sound/pci/ice1712/quartet.h b/sound/pci/ice1712/quartet.h new file mode 100644 index 000000000000..80809b72439a --- /dev/null +++ b/sound/pci/ice1712/quartet.h @@ -0,0 +1,10 @@ +#ifndef __SOUND_QTET_H +#define __SOUND_QTET_H + +#define QTET_DEVICE_DESC "{Infrasonic,Quartet}," + +#define VT1724_SUBDEVICE_QTET 0x30305349 /* Infrasonic Quartet */ + +extern struct snd_ice1712_card_info snd_vt1724_qtet_cards[]; + +#endif /* __SOUND_QTET_H */ -- cgit v1.2.2 From 4f272341c7a42a71586523f196b242bccde3be8c Mon Sep 17 00:00:00 2001 From: Tobias Hansen Date: Tue, 22 Sep 2009 16:52:08 +0200 Subject: ALSA: snd-usb-us122l: add support for US-144 Adds support for US-144 when attached on USB1.1. Unlike the US-122L it uses both USB interfaces 0 and 1. Signed-off-by: Tobias Hansen Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 75 ++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 67 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index fd44946ce4b3..6c7b64a23c13 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -66,6 +66,28 @@ static int us122l_create_usbmidi(struct snd_card *card) iface, &quirk); } +static int us144_create_usbmidi(struct snd_card *card) +{ + static struct snd_usb_midi_endpoint_info quirk_data = { + .out_ep = 4, + .in_ep = 3, + .out_cables = 0x001, + .in_cables = 0x001 + }; + static struct snd_usb_audio_quirk quirk = { + .vendor_name = "US144", + .product_name = NAME_ALLCAPS, + .ifnum = 0, + .type = QUIRK_MIDI_US122L, + .data = &quirk_data + }; + struct usb_device *dev = US122L(card)->chip.dev; + struct usb_interface *iface = usb_ifnum_to_if(dev, 0); + + return snd_usb_create_midi_interface(&US122L(card)->chip, + iface, &quirk); +} + /* * Wrapper for usb_control_msg(). * Allocates a temp buffer to prevent dmaing from/to the stack. @@ -171,6 +193,11 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) if (!us122l->first) us122l->first = file; + + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->chip.dev, 0); + usb_autopm_get_interface(iface); + } iface = usb_ifnum_to_if(us122l->chip.dev, 1); usb_autopm_get_interface(iface); return 0; @@ -179,8 +206,14 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file) { struct us122l *us122l = hw->private_data; - struct usb_interface *iface = usb_ifnum_to_if(us122l->chip.dev, 1); + struct usb_interface *iface; snd_printdd(KERN_DEBUG "%p %p\n", hw, file); + + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->chip.dev, 0); + usb_autopm_put_interface(iface); + } + iface = usb_ifnum_to_if(us122l->chip.dev, 1); usb_autopm_put_interface(iface); if (us122l->first == file) us122l->first = NULL; @@ -443,6 +476,13 @@ static bool us122l_create_card(struct snd_card *card) int err; struct us122l *us122l = US122L(card); + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->chip.dev, 0, 1); + if (err) { + snd_printk(KERN_ERR "usb_set_interface error \n"); + return false; + } + } err = usb_set_interface(us122l->chip.dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -455,7 +495,10 @@ static bool us122l_create_card(struct snd_card *card) if (!us122l_start(us122l, 44100, 256)) return false; - err = us122l_create_usbmidi(card); + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) + err = us144_create_usbmidi(card); + else + err = us122l_create_usbmidi(card); if (err < 0) { snd_printk(KERN_ERR "us122l_create_usbmidi error %i \n", err); us122l_stop(us122l); @@ -542,6 +585,7 @@ static int us122l_usb_probe(struct usb_interface *intf, return err; } + usb_get_intf(usb_ifnum_to_if(device, 0)); usb_get_dev(device); *cardp = card; return 0; @@ -550,9 +594,16 @@ static int us122l_usb_probe(struct usb_interface *intf, static int snd_us122l_probe(struct usb_interface *intf, const struct usb_device_id *id) { + struct usb_device *device = interface_to_usbdev(intf); struct snd_card *card; int err; + if (device->descriptor.idProduct == USB_ID_US144 + && device->speed == USB_SPEED_HIGH) { + snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); + return -ENOENT; + } + snd_printdd(KERN_DEBUG"%p:%i\n", intf, intf->cur_altsetting->desc.bInterfaceNumber); if (intf->cur_altsetting->desc.bInterfaceNumber != 1) @@ -591,7 +642,8 @@ static void snd_us122l_disconnect(struct usb_interface *intf) snd_usbmidi_disconnect(p); } - usb_put_intf(intf); + usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 0)); + usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 1)); usb_put_dev(us122l->chip.dev); while (atomic_read(&us122l->mmap_count)) @@ -642,6 +694,13 @@ static int snd_us122l_resume(struct usb_interface *intf) mutex_lock(&us122l->mutex); /* needed, doesn't restart without: */ + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->chip.dev, 0, 1); + if (err) { + snd_printk(KERN_ERR "usb_set_interface error \n"); + goto unlock; + } + } err = usb_set_interface(us122l->chip.dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -675,11 +734,11 @@ static struct usb_device_id snd_us122l_usb_id_table[] = { .idVendor = 0x0644, .idProduct = USB_ID_US122L }, -/* { */ /* US-144 maybe works when @USB1.1. Untested. */ -/* .match_flags = USB_DEVICE_ID_MATCH_DEVICE, */ -/* .idVendor = 0x0644, */ -/* .idProduct = USB_ID_US144 */ -/* }, */ + { /* US-144 only works at USB1.1! Disable module ehci-hcd. */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US144 + }, { /* terminator */ } }; -- cgit v1.2.2 From 766df6d98f9c28dfc6f72c23a010819719e4c3e0 Mon Sep 17 00:00:00 2001 From: Barry Song Date: Wed, 23 Sep 2009 11:51:04 -0400 Subject: ASoC: Blackfin I2S: use dai state rather than local counter Since the active field of the dai already tells us the stream activity, the local counter variable is redundant and can be replaced. Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 15 +-------------- 1 file changed, 1 insertion(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 1e9d161c76c4..fe2c35ddcbf4 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -49,7 +49,6 @@ struct bf5xx_i2s_port { u16 rcr1; u16 tcr2; u16 rcr2; - int counter; int configured; }; @@ -133,16 +132,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - - /*this counter is used for counting how many pcm streams are opened*/ - bf5xx_i2s.counter++; - return 0; -} - static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -201,9 +190,8 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); - bf5xx_i2s.counter--; /* No active stream, SPORT is allowed to be configured again. */ - if (!bf5xx_i2s.counter) + if (!dai->active) bf5xx_i2s.configured = 0; } @@ -284,7 +272,6 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, -- cgit v1.2.2 From f34762b64704814838619c1d258bebf19004f5cd Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Fri, 25 Sep 2009 13:30:26 +0100 Subject: ASoC: pxa-ssp increase max_channels to 8 When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 57f201c94ca8..a2b1e8fd5d85 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -760,13 +760,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -780,13 +780,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -801,13 +801,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -822,13 +822,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- cgit v1.2.2 From f0968e3f7a8ea30728d2580d3043a30ea9994ec6 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 27 Sep 2009 23:08:40 +0200 Subject: ALSA: sscape: add supoort for SPEA Media FX/Reveal SC-600 Move code from the OSS sscape driver in order to support old Soundscape OEM models. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 6 ++- sound/isa/sscape.c | 116 +++++++++++++++++++++++++++++++++++++---------------- 2 files changed, 86 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 51a7e3777e17..b90fc164a79c 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -377,10 +377,12 @@ config SND_SSCAPE select SND_WSS_LIB help Say Y here to include support for Ensoniq SoundScape - soundcards. + and Ensoniq OEM soundcards. The PCM audio is supported on SoundScape Classic, Elite, PnP - and VIVO cards. The MIDI support is very experimental. + and VIVO cards. The supported OEM cards are SPEA Media FX and + Reveal SC-600. + The MIDI support is very experimental. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 66187122377c..b11c35f6aefe 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -127,7 +127,8 @@ enum GA_REG { enum card_type { - SSCAPE, + MEDIA_FX, /* Sequoia S-1000 */ + SSCAPE, /* Sequoia S-2000 */ SSCAPE_PNP, SSCAPE_VIVO, }; @@ -784,20 +785,25 @@ static struct snd_kcontrol_new midi_mixer_ctl = { * These IRQs are encoded as bit patterns so that they can be * written to the control registers. */ -static unsigned __devinit get_irq_config(int irq) +static unsigned __devinit get_irq_config(int sscape_type, int irq) { static const int valid_irq[] = { 9, 5, 7, 10 }; + static const int old_irq[] = { 9, 7, 5, 15 }; unsigned cfg; - for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) { - if (irq == valid_irq[cfg]) - return cfg; - } /* for */ + if (sscape_type == MEDIA_FX) { + for (cfg = 0; cfg < ARRAY_SIZE(old_irq); ++cfg) + if (irq == old_irq[cfg]) + return cfg; + } else { + for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) + if (irq == valid_irq[cfg]) + return cfg; + } return INVALID_IRQ; } - /* * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. @@ -842,11 +848,39 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e) goto _done; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + if (s->ic_type == IC_OPUS) + activate_ad1845_unsafe(s->io_base); if (s->type == SSCAPE_VIVO) wss_io += 4; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + + /* wait for WSS codec */ + for (d = 0; d < 500; d++) { + if ((inb(wss_io) & 0x80) == 0) + break; + spin_unlock_irqrestore(&s->lock, flags); + msleep(1); + spin_lock_irqsave(&s->lock, flags); + } + snd_printd(KERN_INFO "init delay = %d ms\n", d); + + if ((inb(wss_io) & 0x80) != 0) + goto _done; + + if (inb(wss_io + 2) == 0xff) + goto _done; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d); + + if ((inb(wss_io) & 0x80) != 0) + s->type = MEDIA_FX; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { if ((inb(wss_io) & 0x80) == 0) @@ -954,9 +988,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, if (sscape->type == SSCAPE_VIVO) port += 4; - if (dma1 == dma2) - dma2 = -1; - err = snd_wss_create(card, port, -1, irq, dma1, dma2, WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); if (!err) { @@ -1051,21 +1082,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; - - /* - * Check that the user didn't pass us garbage data ... - */ - irq_cfg = get_irq_config(irq[dev]); - if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; - } - - mpu_irq_cfg = get_irq_config(mpu_irq[dev]); - if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; - } + const char *name; /* * Grab IO ports that we will need to probe so that we @@ -1109,8 +1126,41 @@ static int __devinit create_sscape(int dev, struct snd_card *card) goto _release_dma; } - printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n", - sscape->io_base, irq[dev], dma[dev]); + switch (sscape->type) { + case MEDIA_FX: + name = "MediaFX/SoundFX"; + break; + case SSCAPE: + name = "Soundscape"; + break; + case SSCAPE_PNP: + name = "Soundscape PnP"; + break; + case SSCAPE_VIVO: + name = "Soundscape VIVO"; + break; + default: + name = "unknown Soundscape"; + break; + } + + printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", + name, sscape->io_base, irq[dev], dma[dev]); + + /* + * Check that the user didn't pass us garbage data ... + */ + irq_cfg = get_irq_config(sscape->type, irq[dev]); + if (irq_cfg == INVALID_IRQ) { + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + return -ENXIO; + } + + mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); + if (mpu_irq_cfg == INVALID_IRQ) { + printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + return -ENXIO; + } if (sscape->type != SSCAPE_VIVO) { /* @@ -1141,8 +1191,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - activate_ad1845_unsafe(sscape->io_base); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1151,12 +1199,12 @@ static int __devinit create_sscape(int dev, struct snd_card *card) * Enable and configure the DMA channels ... */ sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); - dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40); + dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); - sscape_write_unsafe(sscape->io_base, - GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg); + mpu_irq_cfg |= mpu_irq_cfg << 2; + sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); -- cgit v1.2.2 From 87b61902ce3dec23a2d8256b9cfcf4e28786a320 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:05:18 +0200 Subject: sound: oxygen: do not try to restore nonexistent EEPROM On cards where the EEPROM was deliberately omitted, we do not need to try to restore the EEPROM's contents. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9a8936e20744..c9f271419eb8 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -278,7 +278,11 @@ oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) static void oxygen_restore_eeprom(struct oxygen *chip, const struct pci_device_id *id) { - if (oxygen_read_eeprom(chip, 0) != OXYGEN_EEPROM_ID) { + u16 eeprom_id; + + eeprom_id = oxygen_read_eeprom(chip, 0); + if (eeprom_id != OXYGEN_EEPROM_ID && + (eeprom_id != 0xffff || id->subdevice != 0x8788)) { /* * This function gets called only when a known card model has * been detected, i.e., we know there is a valid subsystem -- cgit v1.2.2 From 362bc24d6746bcd49bb4853fc5aa7d4c728b3f9e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:05:58 +0200 Subject: sound: oxygen: fix for PI7C9X110 compatibility If the card is used with a Pericom PI7C9X110 PCI-E/PCI bridge, reconfigure the latter's PCI buffering to fix an unknown problem. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index c9f271419eb8..9c5e6450eebb 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -307,6 +307,28 @@ static void oxygen_restore_eeprom(struct oxygen *chip, } } +static void pci_bridge_magic(void) +{ + struct pci_dev *pci = NULL; + u32 tmp; + + for (;;) { + /* If there is any Pericom PI7C9X110 PCI-E/PCI bridge ... */ + pci = pci_get_device(0x12d8, 0xe110, pci); + if (!pci) + break; + /* + * ... configure its secondary internal arbiter to park to + * the secondary port, instead of to the last master. + */ + if (!pci_read_config_dword(pci, 0x40, &tmp)) { + tmp |= 1; + pci_write_config_dword(pci, 0x40, tmp); + } + /* Why? Try asking C-Media. */ + } +} + static void oxygen_init(struct oxygen *chip) { unsigned int i; @@ -585,6 +607,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, snd_card_set_dev(card, &pci->dev); card->private_free = oxygen_card_free; + pci_bridge_magic(); oxygen_init(chip); chip->model.init(chip); -- cgit v1.2.2 From 65c3ac885ce9852852b895a4a62212f62cb5f2e9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:11:27 +0200 Subject: sound: virtuoso: split virtuoso.c The virtuoso.c file has become rather big. This patch splits it up so that only code for very similar card models is in one file. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/Makefile | 3 +- sound/pci/oxygen/virtuoso.c | 1105 +------------------------------------- sound/pci/oxygen/xonar.h | 50 ++ sound/pci/oxygen/xonar_cs43xx.c | 304 +++++++++++ sound/pci/oxygen/xonar_hdmi.c | 128 +++++ sound/pci/oxygen/xonar_lib.c | 132 +++++ sound/pci/oxygen/xonar_pcm179x.c | 660 +++++++++++++++++++++++ 7 files changed, 1290 insertions(+), 1092 deletions(-) create mode 100644 sound/pci/oxygen/xonar.h create mode 100644 sound/pci/oxygen/xonar_cs43xx.c create mode 100644 sound/pci/oxygen/xonar_hdmi.c create mode 100644 sound/pci/oxygen/xonar_lib.c create mode 100644 sound/pci/oxygen/xonar_pcm179x.c (limited to 'sound') diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 4ba07d42fd1d..389941cf6100 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,7 +1,8 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o snd-hifier-objs := hifier.o snd-oxygen-objs := oxygen.o -snd-virtuoso-objs := virtuoso.o +snd-virtuoso-objs := virtuoso.o xonar_lib.o \ + xonar_pcm179x.o xonar_cs43xx.o xonar_hdmi.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o obj-$(CONFIG_SND_HIFIER) += snd-hifier.o diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6ebcb6bdd712..6accaf9580b2 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -17,145 +17,12 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -/* - * Xonar D2/D2X - * ------------ - * - * CMI8788: - * - * SPI 0 -> 1st PCM1796 (front) - * SPI 1 -> 2nd PCM1796 (surround) - * SPI 2 -> 3rd PCM1796 (center/LFE) - * SPI 4 -> 4th PCM1796 (back) - * - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 5 <- external power present (D2X only) - * GPIO 7 -> ALT - * GPIO 8 -> enable output to speakers - */ - -/* - * Xonar D1/DX - * ----------- - * - * CMI8788: - * - * I²C <-> CS4398 (front) - * <-> CS4362A (surround, center/LFE, back) - * - * GPI 0 <- external power present (DX only) - * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> enable front panel I/O - * GPIO 2 -> M0 of CS5361 - * GPIO 3 -> M1 of CS5361 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) - * - * CS4398: - * - * AD0 <- 1 - * AD1 <- 1 - * - * CS4362A: - * - * AD0 <- 0 - */ - -/* - * Xonar HDAV1.3 (Deluxe) - * ---------------------- - * - * CMI8788: - * - * I²C <-> PCM1796 (front) - * - * GPI 0 <- external power present - * - * GPIO 0 -> enable output to speakers - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) - * - * TXD -> HDMI controller - * RXD <- HDMI controller - * - * PCM1796 front: AD1,0 <- 0,0 - * - * no daughterboard - * ---------------- - * - * GPIO 4 <- 1 - * - * H6 daughterboard - * ---------------- - * - * GPIO 4 <- 0 - * GPIO 5 <- 0 - * - * I²C <-> PCM1796 (surround) - * <-> PCM1796 (center/LFE) - * <-> PCM1796 (back) - * - * PCM1796 surround: AD1,0 <- 0,1 - * PCM1796 center/LFE: AD1,0 <- 1,0 - * PCM1796 back: AD1,0 <- 1,1 - * - * unknown daughterboard - * --------------------- - * - * GPIO 4 <- 0 - * GPIO 5 <- 1 - * - * I²C <-> CS4362A (surround, center/LFE, back) - * - * CS4362A: AD0 <- 0 - */ - -/* - * Xonar Essence ST (Deluxe)/STX - * ----------------------------- - * - * CMI8788: - * - * I²C <-> PCM1792A - * - * GPI 0 <- external power present - * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> route HP to front panel (0) or rear jack (1) - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 7 -> route output to speaker jacks (0) or HP (1) - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) - * - * PCM1792A: - * - * AD0 <- 0 - * - * H6 daughterboard - * ---------------- - * - * GPIO 4 <- 0 - * GPIO 5 <- 0 - */ - #include #include -#include -#include -#include -#include #include #include #include -#include -#include -#include "oxygen.h" -#include "cm9780.h" -#include "pcm1796.h" -#include "cs4398.h" -#include "cs4362a.h" +#include "xonar.h" MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("Asus AVx00 driver"); @@ -173,972 +40,28 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -enum { - MODEL_D2, - MODEL_D2X, - MODEL_D1, - MODEL_DX, - MODEL_HDAV, /* without daughterboard */ - MODEL_HDAV_H6, /* with H6 daughterboard */ - MODEL_ST, - MODEL_ST_H6, - MODEL_STX, -}; - static struct pci_device_id xonar_ids[] __devinitdata = { - { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 }, - { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX }, - { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, - { OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV }, - { OXYGEN_PCI_SUBID(0x1043, 0x8327), .driver_data = MODEL_DX }, - { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, - { OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX }, - { OXYGEN_PCI_SUBID(0x1043, 0x835d), .driver_data = MODEL_ST }, + { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, + { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8314) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8327) }, + { OXYGEN_PCI_SUBID(0x1043, 0x834f) }, + { OXYGEN_PCI_SUBID(0x1043, 0x835c) }, + { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; MODULE_DEVICE_TABLE(pci, xonar_ids); - -#define GPIO_CS53x1_M_MASK 0x000c -#define GPIO_CS53x1_M_SINGLE 0x0000 -#define GPIO_CS53x1_M_DOUBLE 0x0004 -#define GPIO_CS53x1_M_QUAD 0x0008 - -#define GPIO_D2X_EXT_POWER 0x0020 -#define GPIO_D2_ALT 0x0080 -#define GPIO_D2_OUTPUT_ENABLE 0x0100 - -#define GPI_DX_EXT_POWER 0x01 -#define GPIO_DX_OUTPUT_ENABLE 0x0001 -#define GPIO_DX_FRONT_PANEL 0x0002 -#define GPIO_DX_INPUT_ROUTE 0x0100 - -#define GPIO_DB_MASK 0x0030 -#define GPIO_DB_H6 0x0000 -#define GPIO_DB_XX 0x0020 - -#define GPIO_ST_HP_REAR 0x0002 -#define GPIO_ST_HP 0x0080 - -#define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ADx=i, /W=0 */ -#define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ -#define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ - -struct xonar_data { - unsigned int anti_pop_delay; - unsigned int dacs; - u16 output_enable_bit; - u8 ext_power_reg; - u8 ext_power_int_reg; - u8 ext_power_bit; - u8 has_power; - u8 pcm1796_oversampling; - u8 cs4398_fm; - u8 cs4362a_fm; - u8 hdmi_params[5]; -}; - -static void xonar_gpio_changed(struct oxygen *chip); - -static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, - u8 reg, u8 value) -{ - /* maps ALSA channel pair number to SPI output */ - static const u8 codec_map[4] = { - 0, 1, 2, 4 - }; - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | - OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_160 | - (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_CEN_LATCH_CLOCK_HI, - (reg << 8) | value); -} - -static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec, - u8 reg, u8 value) -{ - oxygen_write_i2c(chip, I2C_DEVICE_PCM1796(codec), reg, value); -} - -static void pcm1796_write(struct oxygen *chip, unsigned int codec, - u8 reg, u8 value) -{ - if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) == - OXYGEN_FUNCTION_SPI) - pcm1796_write_spi(chip, codec, reg, value); - else - pcm1796_write_i2c(chip, codec, reg, value); -} - -static void cs4398_write(struct oxygen *chip, u8 reg, u8 value) -{ - oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value); -} - -static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) -{ - oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); -} - -static void hdmi_write_command(struct oxygen *chip, u8 command, - unsigned int count, const u8 *params) -{ - unsigned int i; - u8 checksum; - - oxygen_write_uart(chip, 0xfb); - oxygen_write_uart(chip, 0xef); - oxygen_write_uart(chip, command); - oxygen_write_uart(chip, count); - for (i = 0; i < count; ++i) - oxygen_write_uart(chip, params[i]); - checksum = 0xfb + 0xef + command + count; - for (i = 0; i < count; ++i) - checksum += params[i]; - oxygen_write_uart(chip, checksum); -} - -static void xonar_enable_output(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - msleep(data->anti_pop_delay); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); -} - -static void xonar_common_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - if (data->ext_power_reg) { - oxygen_set_bits8(chip, data->ext_power_int_reg, - data->ext_power_bit); - chip->interrupt_mask |= OXYGEN_INT_GPIO; - chip->model.gpio_changed = xonar_gpio_changed; - data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) - & data->ext_power_bit); - } - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_CS53x1_M_MASK | data->output_enable_bit); - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); - oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - xonar_enable_output(chip); -} - -static void update_pcm1796_volume(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - - for (i = 0; i < data->dacs; ++i) { - pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); - } -} - -static void update_pcm1796_mute(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - u8 value; - - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; - if (chip->dac_mute) - value |= PCM1796_MUTE; - for (i = 0; i < data->dacs; ++i) - pcm1796_write(chip, i, 18, value); -} - -static void pcm1796_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - - for (i = 0; i < data->dacs; ++i) { - pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); - pcm1796_write(chip, i, 20, data->pcm1796_oversampling); - pcm1796_write(chip, i, 21, 0); - } - update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ - update_pcm1796_volume(chip); -} - -static void xonar_d2_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->anti_pop_delay = 300; - data->dacs = 4; - data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; - data->pcm1796_oversampling = PCM1796_OS_64; - - pcm1796_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); - - xonar_common_init(chip); - - snd_component_add(chip->card, "PCM1796"); - snd_component_add(chip->card, "CS5381"); -} - -static void xonar_d2x_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->ext_power_reg = OXYGEN_GPIO_DATA; - data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; - data->ext_power_bit = GPIO_D2X_EXT_POWER; - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER); - - xonar_d2_init(chip); -} - -static void update_cs4362a_volumes(struct oxygen *chip) -{ - u8 mute; - - mute = chip->dac_mute ? CS4362A_MUTE : 0; - cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); - cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); - cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); - cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); - cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); - cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); -} - -static void update_cs43xx_volume(struct oxygen *chip) -{ - cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); - cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); - update_cs4362a_volumes(chip); -} - -static void update_cs43xx_mute(struct oxygen *chip) -{ - u8 reg; - - reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; - if (chip->dac_mute) - reg |= CS4398_MUTE_B | CS4398_MUTE_A; - cs4398_write(chip, 4, reg); - update_cs4362a_volumes(chip); -} - -static void cs43xx_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - /* set CPEN (control port mode) and power down */ - cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); - cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); - /* configure */ - cs4398_write(chip, 2, data->cs4398_fm); - cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); - cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | - CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); - cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); - cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE | - CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); - cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); - cs4362a_write(chip, 0x05, 0); - cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm); - cs4362a_write(chip, 0x0c, data->cs4362a_fm); - update_cs43xx_volume(chip); - update_cs43xx_mute(chip); - /* clear power down */ - cs4398_write(chip, 8, CS4398_CPEN); - cs4362a_write(chip, 0x01, CS4362A_CPEN); -} - -static void xonar_d1_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->anti_pop_delay = 800; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - - cs43xx_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, - GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); - - xonar_common_init(chip); - - snd_component_add(chip->card, "CS4398"); - snd_component_add(chip->card, "CS4362A"); - snd_component_add(chip->card, "CS5361"); -} - -static void xonar_dx_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - - xonar_d1_init(chip); -} - -static void xonar_hdav_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - u8 param; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - - data->anti_pop_delay = 100; - data->dacs = chip->model.private_data == MODEL_HDAV_H6 ? 4 : 1; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - data->pcm1796_oversampling = PCM1796_OS_64; - - pcm1796_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DX_INPUT_ROUTE); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DX_INPUT_ROUTE); - - oxygen_reset_uart(chip); - param = 0; - hdmi_write_command(chip, 0x61, 1, ¶m); - param = 1; - hdmi_write_command(chip, 0x74, 1, ¶m); - data->hdmi_params[1] = IEC958_AES3_CON_FS_48000; - data->hdmi_params[4] = 1; - hdmi_write_command(chip, 0x54, 5, data->hdmi_params); - - xonar_common_init(chip); - - snd_component_add(chip->card, "PCM1796"); - snd_component_add(chip->card, "CS5381"); -} - -static void xonar_st_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - - if (chip->model.private_data == MODEL_ST_H6) - chip->model.dac_channels = 8; - data->anti_pop_delay = 100; - data->dacs = chip->model.private_data == MODEL_ST_H6 ? 4 : 1; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->pcm1796_oversampling = PCM1796_OS_64; - - pcm1796_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, - GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); - - xonar_common_init(chip); - - snd_component_add(chip->card, "PCM1792A"); - snd_component_add(chip->card, "CS5381"); -} - -static void xonar_stx_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - - xonar_st_init(chip); -} - -static void xonar_disable_output(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); -} - -static void xonar_d2_cleanup(struct oxygen *chip) -{ - xonar_disable_output(chip); -} - -static void xonar_d1_cleanup(struct oxygen *chip) -{ - xonar_disable_output(chip); - cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); - oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); -} - -static void xonar_hdav_cleanup(struct oxygen *chip) -{ - u8 param = 0; - - hdmi_write_command(chip, 0x74, 1, ¶m); - xonar_disable_output(chip); -} - -static void xonar_st_cleanup(struct oxygen *chip) -{ - xonar_disable_output(chip); -} - -static void xonar_d2_suspend(struct oxygen *chip) -{ - xonar_d2_cleanup(chip); -} - -static void xonar_d1_suspend(struct oxygen *chip) -{ - xonar_d1_cleanup(chip); -} - -static void xonar_hdav_suspend(struct oxygen *chip) -{ - xonar_hdav_cleanup(chip); - msleep(2); -} - -static void xonar_st_suspend(struct oxygen *chip) -{ - xonar_st_cleanup(chip); -} - -static void xonar_d2_resume(struct oxygen *chip) -{ - pcm1796_init(chip); - xonar_enable_output(chip); -} - -static void xonar_d1_resume(struct oxygen *chip) -{ - oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); - msleep(1); - cs43xx_init(chip); - xonar_enable_output(chip); -} - -static void xonar_hdav_resume(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - u8 param; - - oxygen_reset_uart(chip); - param = 0; - hdmi_write_command(chip, 0x61, 1, ¶m); - param = 1; - hdmi_write_command(chip, 0x74, 1, ¶m); - hdmi_write_command(chip, 0x54, 5, data->hdmi_params); - pcm1796_init(chip); - xonar_enable_output(chip); -} - -static void xonar_st_resume(struct oxygen *chip) -{ - pcm1796_init(chip); - xonar_enable_output(chip); -} - -static void xonar_hdav_pcm_hardware_filter(unsigned int channel, - struct snd_pcm_hardware *hardware) -{ - if (channel == PCM_MULTICH) { - hardware->rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000; - hardware->rate_min = 44100; - } -} - -static void set_pcm1796_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - - data->pcm1796_oversampling = - params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; - for (i = 0; i < data->dacs; ++i) - pcm1796_write(chip, i, 20, data->pcm1796_oversampling); -} - -static void set_cs53x1_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - unsigned int value; - - if (params_rate(params) <= 54000) - value = GPIO_CS53x1_M_SINGLE; - else if (params_rate(params) <= 108000) - value = GPIO_CS53x1_M_DOUBLE; - else - value = GPIO_CS53x1_M_QUAD; - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - value, GPIO_CS53x1_M_MASK); -} - -static void set_cs43xx_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct xonar_data *data = chip->model_data; - - data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - if (params_rate(params) <= 50000) { - data->cs4398_fm |= CS4398_FM_SINGLE; - data->cs4362a_fm |= CS4362A_FM_SINGLE; - } else if (params_rate(params) <= 100000) { - data->cs4398_fm |= CS4398_FM_DOUBLE; - data->cs4362a_fm |= CS4362A_FM_DOUBLE; - } else { - data->cs4398_fm |= CS4398_FM_QUAD; - data->cs4362a_fm |= CS4362A_FM_QUAD; - } - cs4398_write(chip, 2, data->cs4398_fm); - cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm); - cs4362a_write(chip, 0x0c, data->cs4362a_fm); -} - -static void set_hdmi_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct xonar_data *data = chip->model_data; - - data->hdmi_params[0] = 0; /* 1 = non-audio */ - switch (params_rate(params)) { - case 44100: - data->hdmi_params[1] = IEC958_AES3_CON_FS_44100; - break; - case 48000: - data->hdmi_params[1] = IEC958_AES3_CON_FS_48000; - break; - default: /* 96000 */ - data->hdmi_params[1] = IEC958_AES3_CON_FS_96000; - break; - case 192000: - data->hdmi_params[1] = IEC958_AES3_CON_FS_192000; - break; - } - data->hdmi_params[2] = params_channels(params) / 2 - 1; - if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE) - data->hdmi_params[3] = 0; - else - data->hdmi_params[3] = 0xc0; - data->hdmi_params[4] = 1; /* ? */ - hdmi_write_command(chip, 0x54, 5, data->hdmi_params); -} - -static void set_hdav_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - set_pcm1796_params(chip, params); - set_hdmi_params(chip, params); -} - -static void xonar_gpio_changed(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - u8 has_power; - - has_power = !!(oxygen_read8(chip, data->ext_power_reg) - & data->ext_power_bit); - if (has_power != data->has_power) { - data->has_power = has_power; - if (has_power) { - snd_printk(KERN_NOTICE "power restored\n"); - } else { - snd_printk(KERN_CRIT - "Hey! Don't unplug the power cable!\n"); - /* TODO: stop PCMs */ - } - } -} - -static void xonar_hdav_uart_input(struct oxygen *chip) -{ - if (chip->uart_input_count >= 2 && - chip->uart_input[chip->uart_input_count - 2] == 'O' && - chip->uart_input[chip->uart_input_count - 1] == 'K') { - printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n"); - print_hex_dump_bytes("", DUMP_PREFIX_OFFSET, - chip->uart_input, chip->uart_input_count); - chip->uart_input_count = 0; - } -} - -static int gpio_bit_switch_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 bit = ctl->private_value; - - value->value.integer.value[0] = - !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit); - return 0; -} - -static int gpio_bit_switch_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 bit = ctl->private_value; - u16 old_bits, new_bits; - int changed; - - spin_lock_irq(&chip->reg_lock); - old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); - if (value->value.integer.value[0]) - new_bits = old_bits | bit; - else - new_bits = old_bits & ~bit; - changed = new_bits != old_bits; - if (changed) - oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits); - spin_unlock_irq(&chip->reg_lock); - return changed; -} - -static const struct snd_kcontrol_new alt_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Loopback Switch", - .info = snd_ctl_boolean_mono_info, - .get = gpio_bit_switch_get, - .put = gpio_bit_switch_put, - .private_value = GPIO_D2_ALT, -}; - -static const struct snd_kcontrol_new front_panel_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Front Panel Switch", - .info = snd_ctl_boolean_mono_info, - .get = gpio_bit_switch_get, - .put = gpio_bit_switch_put, - .private_value = GPIO_DX_FRONT_PANEL, -}; - -static int st_output_switch_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[3] = { - "Speakers", "Headphones", "FP Headphones" - }; - - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; -} - -static int st_output_switch_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 gpio; - - gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA); - if (!(gpio & GPIO_ST_HP)) - value->value.enumerated.item[0] = 0; - else if (gpio & GPIO_ST_HP_REAR) - value->value.enumerated.item[0] = 1; - else - value->value.enumerated.item[0] = 2; - return 0; -} - - -static int st_output_switch_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 gpio_old, gpio; - - mutex_lock(&chip->mutex); - gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA); - gpio = gpio_old; - switch (value->value.enumerated.item[0]) { - case 0: - gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR); - break; - case 1: - gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR; - break; - case 2: - gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR; - break; - } - oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); - mutex_unlock(&chip->mutex); - return gpio != gpio_old; -} - -static const struct snd_kcontrol_new st_output_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Output", - .info = st_output_switch_info, - .get = st_output_switch_get, - .put = st_output_switch_put, -}; - -static void xonar_line_mic_ac97_switch(struct oxygen *chip, - unsigned int reg, unsigned int mute) -{ - if (reg == AC97_LINE) { - spin_lock_irq(&chip->reg_lock); - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - mute ? GPIO_DX_INPUT_ROUTE : 0, - GPIO_DX_INPUT_ROUTE); - spin_unlock_irq(&chip->reg_lock); - } -} - -static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0); -static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); - -static int xonar_d2_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - /* CD in is actually connected to the video in pin */ - template->private_value ^= AC97_CD ^ AC97_VIDEO; - return 0; -} - -static int xonar_d1_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - -static int xonar_st_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - if (!strcmp(template->name, "Stereo Upmixing")) - return 1; /* stereo only - we don't need upmixing */ - return 0; -} - -static int xonar_d2_mixer_init(struct oxygen *chip) -{ - return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); -} - -static int xonar_d1_mixer_init(struct oxygen *chip) -{ - return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); -} - -static int xonar_st_mixer_init(struct oxygen *chip) -{ - return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip)); -} - -static const struct oxygen_model model_xonar_d2 = { - .longname = "Asus Virtuoso 200", - .chip = "AV200", - .init = xonar_d2_init, - .control_filter = xonar_d2_control_filter, - .mixer_init = xonar_d2_mixer_init, - .cleanup = xonar_d2_cleanup, - .suspend = xonar_d2_suspend, - .resume = xonar_d2_resume, - .set_dac_params = set_pcm1796_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_pcm1796_volume, - .update_dac_mute = update_pcm1796_mute, - .dac_tlv = pcm1796_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF | - MIDI_OUTPUT | - MIDI_INPUT, - .dac_channels = 8, - .dac_volume_min = 255 - 2*60, - .dac_volume_max = 255, - .misc_flags = OXYGEN_MISC_MIDI, - .function_flags = OXYGEN_FUNCTION_SPI | - OXYGEN_FUNCTION_ENABLE_SPI_4_5, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static const struct oxygen_model model_xonar_d1 = { - .longname = "Asus Virtuoso 100", - .chip = "AV200", - .init = xonar_d1_init, - .control_filter = xonar_d1_control_filter, - .mixer_init = xonar_d1_mixer_init, - .cleanup = xonar_d1_cleanup, - .suspend = xonar_d1_suspend, - .resume = xonar_d1_resume, - .set_dac_params = set_cs43xx_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_cs43xx_volume, - .update_dac_mute = update_cs43xx_mute, - .ac97_switch = xonar_line_mic_ac97_switch, - .dac_tlv = cs4362a_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, - .dac_channels = 8, - .dac_volume_min = 127 - 60, - .dac_volume_max = 127, - .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static const struct oxygen_model model_xonar_hdav = { - .longname = "Asus Virtuoso 200", - .chip = "AV200", - .init = xonar_hdav_init, - .cleanup = xonar_hdav_cleanup, - .suspend = xonar_hdav_suspend, - .resume = xonar_hdav_resume, - .pcm_hardware_filter = xonar_hdav_pcm_hardware_filter, - .set_dac_params = set_hdav_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_pcm1796_volume, - .update_dac_mute = update_pcm1796_mute, - .uart_input = xonar_hdav_uart_input, - .ac97_switch = xonar_line_mic_ac97_switch, - .dac_tlv = pcm1796_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF, - .dac_channels = 8, - .dac_volume_min = 255 - 2*60, - .dac_volume_max = 255, - .misc_flags = OXYGEN_MISC_MIDI, - .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static const struct oxygen_model model_xonar_st = { - .longname = "Asus Virtuoso 100", - .chip = "AV200", - .init = xonar_st_init, - .control_filter = xonar_st_control_filter, - .mixer_init = xonar_st_mixer_init, - .cleanup = xonar_st_cleanup, - .suspend = xonar_st_suspend, - .resume = xonar_st_resume, - .set_dac_params = set_pcm1796_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_pcm1796_volume, - .update_dac_mute = update_pcm1796_mute, - .ac97_switch = xonar_line_mic_ac97_switch, - .dac_tlv = pcm1796_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, - .dac_channels = 2, - .dac_volume_min = 255 - 2*60, - .dac_volume_max = 255, - .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - static int __devinit get_xonar_model(struct oxygen *chip, const struct pci_device_id *id) { - static const struct oxygen_model *const models[] = { - [MODEL_D1] = &model_xonar_d1, - [MODEL_DX] = &model_xonar_d1, - [MODEL_D2] = &model_xonar_d2, - [MODEL_D2X] = &model_xonar_d2, - [MODEL_HDAV] = &model_xonar_hdav, - [MODEL_ST] = &model_xonar_st, - [MODEL_STX] = &model_xonar_st, - }; - static const char *const names[] = { - [MODEL_D1] = "Xonar D1", - [MODEL_DX] = "Xonar DX", - [MODEL_D2] = "Xonar D2", - [MODEL_D2X] = "Xonar D2X", - [MODEL_HDAV] = "Xonar HDAV1.3", - [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", - [MODEL_ST] = "Xonar Essence ST", - [MODEL_ST_H6] = "Xonar Essence ST+H6", - [MODEL_STX] = "Xonar Essence STX", - }; - unsigned int model = id->driver_data; - - if (model >= ARRAY_SIZE(models) || !models[model]) - return -EINVAL; - chip->model = *models[model]; - - switch (model) { - case MODEL_D2X: - chip->model.init = xonar_d2x_init; - break; - case MODEL_DX: - chip->model.init = xonar_dx_init; - break; - case MODEL_HDAV: - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); - switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { - case GPIO_DB_H6: - model = MODEL_HDAV_H6; - break; - case GPIO_DB_XX: - snd_printk(KERN_ERR "unknown daughterboard\n"); - return -ENODEV; - } - break; - case MODEL_ST: - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); - switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { - case GPIO_DB_H6: - model = MODEL_ST_H6; - break; - } - break; - case MODEL_STX: - chip->model.init = xonar_stx_init; - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); - break; - } - - chip->model.shortname = names[model]; - chip->model.private_data = model; - return 0; + if (get_xonar_pcm179x_model(chip, id) >= 0) + return 0; + if (get_xonar_cs43xx_model(chip, id) >= 0) + return 0; + return -EINVAL; } static int __devinit xonar_probe(struct pci_dev *pci, diff --git a/sound/pci/oxygen/xonar.h b/sound/pci/oxygen/xonar.h new file mode 100644 index 000000000000..89b3ed814d64 --- /dev/null +++ b/sound/pci/oxygen/xonar.h @@ -0,0 +1,50 @@ +#ifndef XONAR_H_INCLUDED +#define XONAR_H_INCLUDED + +#include "oxygen.h" + +struct xonar_generic { + unsigned int anti_pop_delay; + u16 output_enable_bit; + u8 ext_power_reg; + u8 ext_power_int_reg; + u8 ext_power_bit; + u8 has_power; +}; + +struct xonar_hdmi { + u8 params[5]; +}; + +/* generic helper functions */ + +void xonar_enable_output(struct oxygen *chip); +void xonar_disable_output(struct oxygen *chip); +void xonar_init_ext_power(struct oxygen *chip); +void xonar_init_cs53x1(struct oxygen *chip); +void xonar_set_cs53x1_params(struct oxygen *chip, + struct snd_pcm_hw_params *params); +int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value); +int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value); + +/* model-specific card drivers */ + +int get_xonar_pcm179x_model(struct oxygen *chip, + const struct pci_device_id *id); +int get_xonar_cs43xx_model(struct oxygen *chip, + const struct pci_device_id *id); + +/* HDMI helper functions */ + +void xonar_hdmi_init(struct oxygen *chip, struct xonar_hdmi *data); +void xonar_hdmi_cleanup(struct oxygen *chip); +void xonar_hdmi_resume(struct oxygen *chip, struct xonar_hdmi *hdmi); +void xonar_hdmi_pcm_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware); +void xonar_set_hdmi_params(struct oxygen *chip, struct xonar_hdmi *hdmi, + struct snd_pcm_hw_params *params); +void xonar_hdmi_uart_input(struct oxygen *chip); + +#endif diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c new file mode 100644 index 000000000000..8fb5797577dd --- /dev/null +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -0,0 +1,304 @@ +/* + * card driver for models with CS4398/CS4362A DACs (Xonar D1/DX) + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see . + */ + +/* + * Xonar D1/DX + * ----------- + * + * CMI8788: + * + * I²C <-> CS4398 (front) + * <-> CS4362A (surround, center/LFE, back) + * + * GPI 0 <- external power present (DX only) + * + * GPIO 0 -> enable output to speakers + * GPIO 1 -> enable front panel I/O + * GPIO 2 -> M0 of CS5361 + * GPIO 3 -> M1 of CS5361 + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * CS4398: + * + * AD0 <- 1 + * AD1 <- 1 + * + * CS4362A: + * + * AD0 <- 0 + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "xonar.h" +#include "cs4398.h" +#include "cs4362a.h" + +#define GPI_EXT_POWER 0x01 +#define GPIO_D1_OUTPUT_ENABLE 0x0001 +#define GPIO_D1_FRONT_PANEL 0x0002 +#define GPIO_D1_INPUT_ROUTE 0x0100 + +#define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ +#define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ + +struct xonar_cs43xx { + struct xonar_generic generic; + u8 cs4398_fm; + u8 cs4362a_fm; +}; + +static void cs4398_write(struct oxygen *chip, u8 reg, u8 value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value); +} + +static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); +} + +static void update_cs4362a_volumes(struct oxygen *chip) +{ + u8 mute; + + mute = chip->dac_mute ? CS4362A_MUTE : 0; + cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); + cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); + cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); + cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); + cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); + cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); +} + +static void update_cs43xx_volume(struct oxygen *chip) +{ + cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); + cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_mute(struct oxygen *chip) +{ + u8 reg; + + reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; + if (chip->dac_mute) + reg |= CS4398_MUTE_B | CS4398_MUTE_A; + cs4398_write(chip, 4, reg); + update_cs4362a_volumes(chip); +} + +static void cs43xx_init(struct oxygen *chip) +{ + struct xonar_cs43xx *data = chip->model_data; + + /* set CPEN (control port mode) and power down */ + cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); + cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); + /* configure */ + cs4398_write(chip, 2, data->cs4398_fm); + cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); + cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | + CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); + cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); + cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE | + CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); + cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); + cs4362a_write(chip, 0x05, 0); + cs4362a_write(chip, 0x06, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); + update_cs43xx_volume(chip); + update_cs43xx_mute(chip); + /* clear power down */ + cs4398_write(chip, 8, CS4398_CPEN); + cs4362a_write(chip, 0x01, CS4362A_CPEN); +} + +static void xonar_d1_init(struct oxygen *chip) +{ + struct xonar_cs43xx *data = chip->model_data; + + data->generic.anti_pop_delay = 800; + data->generic.output_enable_bit = GPIO_D1_OUTPUT_ENABLE; + data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + cs43xx_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); + + xonar_init_cs53x1(chip); + xonar_enable_output(chip); + + snd_component_add(chip->card, "CS4398"); + snd_component_add(chip->card, "CS4362A"); + snd_component_add(chip->card, "CS5361"); +} + +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_cs43xx *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_d1_init(chip); +} + +static void xonar_d1_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); + cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); + oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); +} + +static void xonar_d1_suspend(struct oxygen *chip) +{ + xonar_d1_cleanup(chip); +} + +static void xonar_d1_resume(struct oxygen *chip) +{ + oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); + msleep(1); + cs43xx_init(chip); + xonar_enable_output(chip); +} + +static void set_cs43xx_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_cs43xx *data = chip->model_data; + + data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + if (params_rate(params) <= 50000) { + data->cs4398_fm |= CS4398_FM_SINGLE; + data->cs4362a_fm |= CS4362A_FM_SINGLE; + } else if (params_rate(params) <= 100000) { + data->cs4398_fm |= CS4398_FM_DOUBLE; + data->cs4362a_fm |= CS4362A_FM_DOUBLE; + } else { + data->cs4398_fm |= CS4398_FM_QUAD; + data->cs4362a_fm |= CS4362A_FM_QUAD; + } + cs4398_write(chip, 2, data->cs4398_fm); + cs4362a_write(chip, 0x06, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); +} + +static const struct snd_kcontrol_new front_panel_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Front Panel Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_D1_FRONT_PANEL, +}; + +static void xonar_d1_line_mic_ac97_switch(struct oxygen *chip, + unsigned int reg, unsigned int mute) +{ + if (reg == AC97_LINE) { + spin_lock_irq(&chip->reg_lock); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + mute ? GPIO_D1_INPUT_ROUTE : 0, + GPIO_D1_INPUT_ROUTE); + spin_unlock_irq(&chip->reg_lock); + } +} + +static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); + +static int xonar_d1_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + return 0; +} + +static int xonar_d1_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); +} + +static const struct oxygen_model model_xonar_d1 = { + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .init = xonar_d1_init, + .control_filter = xonar_d1_control_filter, + .mixer_init = xonar_d1_mixer_init, + .cleanup = xonar_d1_cleanup, + .suspend = xonar_d1_suspend, + .resume = xonar_d1_resume, + .set_dac_params = set_cs43xx_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_cs43xx_volume, + .update_dac_mute = update_cs43xx_mute, + .ac97_switch = xonar_d1_line_mic_ac97_switch, + .dac_tlv = cs4362a_db_scale, + .model_data_size = sizeof(struct xonar_cs43xx), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 8, + .dac_volume_min = 127 - 60, + .dac_volume_max = 127, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +int __devinit get_xonar_cs43xx_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + switch (id->subdevice) { + case 0x834f: + chip->model = model_xonar_d1; + chip->model.shortname = "Xonar D1"; + break; + case 0x8275: + case 0x8327: + chip->model = model_xonar_d1; + chip->model.shortname = "Xonar DX"; + chip->model.init = xonar_dx_init; + break; + default: + return -EINVAL; + } + return 0; +} diff --git a/sound/pci/oxygen/xonar_hdmi.c b/sound/pci/oxygen/xonar_hdmi.c new file mode 100644 index 000000000000..b12db1f1cea9 --- /dev/null +++ b/sound/pci/oxygen/xonar_hdmi.c @@ -0,0 +1,128 @@ +/* + * helper functions for HDMI models (Xonar HDAV1.3) + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see . + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "xonar.h" + +static void hdmi_write_command(struct oxygen *chip, u8 command, + unsigned int count, const u8 *params) +{ + unsigned int i; + u8 checksum; + + oxygen_write_uart(chip, 0xfb); + oxygen_write_uart(chip, 0xef); + oxygen_write_uart(chip, command); + oxygen_write_uart(chip, count); + for (i = 0; i < count; ++i) + oxygen_write_uart(chip, params[i]); + checksum = 0xfb + 0xef + command + count; + for (i = 0; i < count; ++i) + checksum += params[i]; + oxygen_write_uart(chip, checksum); +} + +static void xonar_hdmi_init_commands(struct oxygen *chip, + struct xonar_hdmi *hdmi) +{ + u8 param; + + oxygen_reset_uart(chip); + param = 0; + hdmi_write_command(chip, 0x61, 1, ¶m); + param = 1; + hdmi_write_command(chip, 0x74, 1, ¶m); + hdmi_write_command(chip, 0x54, 5, hdmi->params); +} + +void xonar_hdmi_init(struct oxygen *chip, struct xonar_hdmi *hdmi) +{ + hdmi->params[1] = IEC958_AES3_CON_FS_48000; + hdmi->params[4] = 1; + xonar_hdmi_init_commands(chip, hdmi); +} + +void xonar_hdmi_cleanup(struct oxygen *chip) +{ + u8 param = 0; + + hdmi_write_command(chip, 0x74, 1, ¶m); +} + +void xonar_hdmi_resume(struct oxygen *chip, struct xonar_hdmi *hdmi) +{ + xonar_hdmi_init_commands(chip, hdmi); +} + +void xonar_hdmi_pcm_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + if (channel == PCM_MULTICH) { + hardware->rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000; + hardware->rate_min = 44100; + } +} + +void xonar_set_hdmi_params(struct oxygen *chip, struct xonar_hdmi *hdmi, + struct snd_pcm_hw_params *params) +{ + hdmi->params[0] = 0; /* 1 = non-audio */ + switch (params_rate(params)) { + case 44100: + hdmi->params[1] = IEC958_AES3_CON_FS_44100; + break; + case 48000: + hdmi->params[1] = IEC958_AES3_CON_FS_48000; + break; + default: /* 96000 */ + hdmi->params[1] = IEC958_AES3_CON_FS_96000; + break; + case 192000: + hdmi->params[1] = IEC958_AES3_CON_FS_192000; + break; + } + hdmi->params[2] = params_channels(params) / 2 - 1; + if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE) + hdmi->params[3] = 0; + else + hdmi->params[3] = 0xc0; + hdmi->params[4] = 1; /* ? */ + hdmi_write_command(chip, 0x54, 5, hdmi->params); +} + +void xonar_hdmi_uart_input(struct oxygen *chip) +{ + if (chip->uart_input_count >= 2 && + chip->uart_input[chip->uart_input_count - 2] == 'O' && + chip->uart_input[chip->uart_input_count - 1] == 'K') { + printk(KERN_DEBUG "message from HDMI chip received:\n"); + print_hex_dump_bytes("", DUMP_PREFIX_OFFSET, + chip->uart_input, chip->uart_input_count); + chip->uart_input_count = 0; + } +} diff --git a/sound/pci/oxygen/xonar_lib.c b/sound/pci/oxygen/xonar_lib.c new file mode 100644 index 000000000000..b3ff71316653 --- /dev/null +++ b/sound/pci/oxygen/xonar_lib.c @@ -0,0 +1,132 @@ +/* + * helper functions for Asus Xonar cards + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see . + */ + +#include +#include +#include +#include +#include +#include "xonar.h" + + +#define GPIO_CS53x1_M_MASK 0x000c +#define GPIO_CS53x1_M_SINGLE 0x0000 +#define GPIO_CS53x1_M_DOUBLE 0x0004 +#define GPIO_CS53x1_M_QUAD 0x0008 + + +void xonar_enable_output(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit); + msleep(data->anti_pop_delay); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + +void xonar_disable_output(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + +static void xonar_ext_power_gpio_changed(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + u8 has_power; + + has_power = !!(oxygen_read8(chip, data->ext_power_reg) + & data->ext_power_bit); + if (has_power != data->has_power) { + data->has_power = has_power; + if (has_power) { + snd_printk(KERN_NOTICE "power restored\n"); + } else { + snd_printk(KERN_CRIT + "Hey! Don't unplug the power cable!\n"); + /* TODO: stop PCMs */ + } + } +} + +void xonar_init_ext_power(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + + oxygen_set_bits8(chip, data->ext_power_int_reg, + data->ext_power_bit); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + chip->model.gpio_changed = xonar_ext_power_gpio_changed; + data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) + & data->ext_power_bit); +} + +void xonar_init_cs53x1(struct oxygen *chip) +{ + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); +} + +void xonar_set_cs53x1_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + unsigned int value; + + if (params_rate(params) <= 54000) + value = GPIO_CS53x1_M_SINGLE; + else if (params_rate(params) <= 108000) + value = GPIO_CS53x1_M_DOUBLE; + else + value = GPIO_CS53x1_M_QUAD; + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + value, GPIO_CS53x1_M_MASK); +} + +int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 bit = ctl->private_value; + + value->value.integer.value[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit); + return 0; +} + +int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 bit = ctl->private_value; + u16 old_bits, new_bits; + int changed; + + spin_lock_irq(&chip->reg_lock); + old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (value->value.integer.value[0]) + new_bits = old_bits | bit; + else + new_bits = old_bits & ~bit; + changed = new_bits != old_bits; + if (changed) + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits); + spin_unlock_irq(&chip->reg_lock); + return changed; +} diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c new file mode 100644 index 000000000000..eb5f015fcd23 --- /dev/null +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -0,0 +1,660 @@ +/* + * card driver for models with PCM1796 DACs (Xonar D2/D2X/HDAV1.3/ST/STX) + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see . + */ + +/* + * Xonar D2/D2X + * ------------ + * + * CMI8788: + * + * SPI 0 -> 1st PCM1796 (front) + * SPI 1 -> 2nd PCM1796 (surround) + * SPI 2 -> 3rd PCM1796 (center/LFE) + * SPI 4 -> 4th PCM1796 (back) + * + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 5 <- external power present (D2X only) + * GPIO 7 -> ALT + * GPIO 8 -> enable output to speakers + */ + +/* + * Xonar HDAV1.3 (Deluxe) + * ---------------------- + * + * CMI8788: + * + * I²C <-> PCM1796 (front) + * + * GPI 0 <- external power present + * + * GPIO 0 -> enable output to speakers + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * TXD -> HDMI controller + * RXD <- HDMI controller + * + * PCM1796 front: AD1,0 <- 0,0 + * + * no daughterboard + * ---------------- + * + * GPIO 4 <- 1 + * + * H6 daughterboard + * ---------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 0 + * + * I²C <-> PCM1796 (surround) + * <-> PCM1796 (center/LFE) + * <-> PCM1796 (back) + * + * PCM1796 surround: AD1,0 <- 0,1 + * PCM1796 center/LFE: AD1,0 <- 1,0 + * PCM1796 back: AD1,0 <- 1,1 + * + * unknown daughterboard + * --------------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 1 + * + * I²C <-> CS4362A (surround, center/LFE, back) + * + * CS4362A: AD0 <- 0 + */ + +/* + * Xonar Essence ST (Deluxe)/STX + * ----------------------------- + * + * CMI8788: + * + * I²C <-> PCM1792A + * + * GPI 0 <- external power present (STX only) + * + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 7 -> route output to speaker jacks (0) or HP (1) + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * PCM1792A: + * + * AD1,0 <- 0,0 + * + * H6 daughterboard + * ---------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 0 + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "xonar.h" +#include "cm9780.h" +#include "pcm1796.h" + + +#define GPIO_D2X_EXT_POWER 0x0020 +#define GPIO_D2_ALT 0x0080 +#define GPIO_D2_OUTPUT_ENABLE 0x0100 + +#define GPI_EXT_POWER 0x01 +#define GPIO_INPUT_ROUTE 0x0100 + +#define GPIO_HDAV_OUTPUT_ENABLE 0x0001 + +#define GPIO_DB_MASK 0x0030 +#define GPIO_DB_H6 0x0000 + +#define GPIO_ST_OUTPUT_ENABLE 0x0001 +#define GPIO_ST_HP_REAR 0x0002 +#define GPIO_ST_HP 0x0080 + +#define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ii, /W=0 */ + + +struct xonar_pcm179x { + struct xonar_generic generic; + unsigned int dacs; + u8 oversampling; +}; + +struct xonar_hdav { + struct xonar_pcm179x pcm179x; + struct xonar_hdmi hdmi; +}; + + +static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + /* maps ALSA channel pair number to SPI output */ + static const u8 codec_map[4] = { + 0, 1, 2, 4 + }; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI, + (reg << 8) | value); +} + +static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_PCM1796(codec), reg, value); +} + +static void pcm1796_write(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) == + OXYGEN_FUNCTION_SPI) + pcm1796_write_spi(chip, codec, reg, value); + else + pcm1796_write_i2c(chip, codec, reg, value); +} + +static void update_pcm1796_volume(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + + for (i = 0; i < data->dacs; ++i) { + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); + } +} + +static void update_pcm1796_mute(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + u8 value; + + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + if (chip->dac_mute) + value |= PCM1796_MUTE; + for (i = 0; i < data->dacs; ++i) + pcm1796_write(chip, i, 18, value); +} + +static void pcm1796_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + + for (i = 0; i < data->dacs; ++i) { + pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); + pcm1796_write(chip, i, 20, data->oversampling); + pcm1796_write(chip, i, 21, 0); + } + update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ + update_pcm1796_volume(chip); +} + +static void xonar_d2_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.anti_pop_delay = 300; + data->generic.output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + data->dacs = 4; + data->oversampling = PCM1796_OS_64; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); + + oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); + + xonar_init_cs53x1(chip); + xonar_enable_output(chip); + + snd_component_add(chip->card, "PCM1796"); + snd_component_add(chip->card, "CS5381"); +} + +static void xonar_d2x_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPIO_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; + data->generic.ext_power_bit = GPIO_D2X_EXT_POWER; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER); + xonar_init_ext_power(chip); + xonar_d2_init(chip); +} + +static void xonar_hdav_init(struct oxygen *chip) +{ + struct xonar_hdav *data = chip->model_data; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + data->pcm179x.generic.anti_pop_delay = 100; + data->pcm179x.generic.output_enable_bit = GPIO_HDAV_OUTPUT_ENABLE; + data->pcm179x.generic.ext_power_reg = OXYGEN_GPI_DATA; + data->pcm179x.generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->pcm179x.generic.ext_power_bit = GPI_EXT_POWER; + data->pcm179x.dacs = chip->model.private_data ? 4 : 1; + data->pcm179x.oversampling = PCM1796_OS_64; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_INPUT_ROUTE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_INPUT_ROUTE); + + xonar_init_cs53x1(chip); + xonar_init_ext_power(chip); + xonar_hdmi_init(chip, &data->hdmi); + xonar_enable_output(chip); + + snd_component_add(chip->card, "PCM1796"); + snd_component_add(chip->card, "CS5381"); +} + +static void xonar_st_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + data->generic.anti_pop_delay = 100; + data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; + data->dacs = chip->model.private_data ? 4 : 1; + data->oversampling = PCM1796_OS_64; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + + xonar_init_cs53x1(chip); + xonar_enable_output(chip); + + snd_component_add(chip->card, "PCM1792A"); + snd_component_add(chip->card, "CS5381"); +} + +static void xonar_stx_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init(chip); +} + +static void xonar_d2_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + +static void xonar_hdav_cleanup(struct oxygen *chip) +{ + xonar_hdmi_cleanup(chip); + xonar_disable_output(chip); + msleep(2); +} + +static void xonar_st_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + +static void xonar_d2_suspend(struct oxygen *chip) +{ + xonar_d2_cleanup(chip); +} + +static void xonar_hdav_suspend(struct oxygen *chip) +{ + xonar_hdav_cleanup(chip); +} + +static void xonar_st_suspend(struct oxygen *chip) +{ + xonar_st_cleanup(chip); +} + +static void xonar_d2_resume(struct oxygen *chip) +{ + pcm1796_init(chip); + xonar_enable_output(chip); +} + +static void xonar_hdav_resume(struct oxygen *chip) +{ + struct xonar_hdav *data = chip->model_data; + + pcm1796_init(chip); + xonar_hdmi_resume(chip, &data->hdmi); + xonar_enable_output(chip); +} + +static void xonar_st_resume(struct oxygen *chip) +{ + pcm1796_init(chip); + xonar_enable_output(chip); +} + +static void set_pcm1796_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + + data->oversampling = + params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; + for (i = 0; i < data->dacs; ++i) + pcm1796_write(chip, i, 20, data->oversampling); +} + +static void set_hdav_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_hdav *data = chip->model_data; + + set_pcm1796_params(chip, params); + xonar_set_hdmi_params(chip, &data->hdmi, params); +} + +static const struct snd_kcontrol_new alt_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Loopback Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_D2_ALT, +}; + +static int st_output_switch_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Speakers", "Headphones", "FP Headphones" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int st_output_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio; + + gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (!(gpio & GPIO_ST_HP)) + value->value.enumerated.item[0] = 0; + else if (gpio & GPIO_ST_HP_REAR) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + + +static int st_output_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio_old, gpio; + + mutex_lock(&chip->mutex); + gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA); + gpio = gpio_old; + switch (value->value.enumerated.item[0]) { + case 0: + gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR); + break; + case 1: + gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR; + break; + case 2: + gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR; + break; + } + oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); + mutex_unlock(&chip->mutex); + return gpio != gpio_old; +} + +static const struct snd_kcontrol_new st_output_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output", + .info = st_output_switch_info, + .get = st_output_switch_get, + .put = st_output_switch_put, +}; + +static void xonar_line_mic_ac97_switch(struct oxygen *chip, + unsigned int reg, unsigned int mute) +{ + if (reg == AC97_LINE) { + spin_lock_irq(&chip->reg_lock); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + mute ? GPIO_INPUT_ROUTE : 0, + GPIO_INPUT_ROUTE); + spin_unlock_irq(&chip->reg_lock); + } +} + +static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0); + +static int xonar_d2_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + /* CD in is actually connected to the video in pin */ + template->private_value ^= AC97_CD ^ AC97_VIDEO; + return 0; +} + +static int xonar_st_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + if (!strcmp(template->name, "Stereo Upmixing")) + return 1; /* stereo only - we don't need upmixing */ + return 0; +} + +static int xonar_d2_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); +} + +static int xonar_st_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip)); +} + +static const struct oxygen_model model_xonar_d2 = { + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_d2_init, + .control_filter = xonar_d2_control_filter, + .mixer_init = xonar_d2_mixer_init, + .cleanup = xonar_d2_cleanup, + .suspend = xonar_d2_suspend, + .resume = xonar_d2_resume, + .set_dac_params = set_pcm1796_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_pcm179x), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF | + MIDI_OUTPUT | + MIDI_INPUT, + .dac_channels = 8, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .misc_flags = OXYGEN_MISC_MIDI, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +static const struct oxygen_model model_xonar_hdav = { + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_hdav_init, + .cleanup = xonar_hdav_cleanup, + .suspend = xonar_hdav_suspend, + .resume = xonar_hdav_resume, + .pcm_hardware_filter = xonar_hdmi_pcm_hardware_filter, + .set_dac_params = set_hdav_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .uart_input = xonar_hdmi_uart_input, + .ac97_switch = xonar_line_mic_ac97_switch, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_hdav), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF, + .dac_channels = 8, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .misc_flags = OXYGEN_MISC_MIDI, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +static const struct oxygen_model model_xonar_st = { + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .init = xonar_st_init, + .control_filter = xonar_st_control_filter, + .mixer_init = xonar_st_mixer_init, + .cleanup = xonar_st_cleanup, + .suspend = xonar_st_suspend, + .resume = xonar_st_resume, + .set_dac_params = set_pcm1796_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .ac97_switch = xonar_line_mic_ac97_switch, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_pcm179x), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 2, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +int __devinit get_xonar_pcm179x_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + switch (id->subdevice) { + case 0x8269: + chip->model = model_xonar_d2; + chip->model.shortname = "Xonar D2"; + break; + case 0x82b7: + chip->model = model_xonar_d2; + chip->model.shortname = "Xonar D2X"; + chip->model.init = xonar_d2x_init; + break; + case 0x8314: + chip->model = model_xonar_hdav; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { + default: + chip->model.shortname = "Xonar HDAV1.3"; + break; + case GPIO_DB_H6: + chip->model.shortname = "Xonar HDAV1.3+H6"; + chip->model.private_data = 1; + break; + } + break; + case 0x835d: + chip->model = model_xonar_st; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { + default: + chip->model.shortname = "Xonar ST"; + break; + case GPIO_DB_H6: + chip->model.shortname = "Xonar ST+H6"; + chip->model.dac_channels = 8; + chip->model.private_data = 1; + break; + } + break; + case 0x835c: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar STX"; + chip->model.init = xonar_stx_init; + break; + default: + return -EINVAL; + } + return 0; +} -- cgit v1.2.2 From 268304f4c4f0b8677d67400f04ad4e0271ec3742 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:15:01 +0200 Subject: sound: virtuoso: fix Xonar Essence ST support The Essence ST uses the CS2000 chip to generate the DAC master clock, so we better initialize and program it appropriately. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/cs2000.h | 83 ++++++++++++++++++++++++++++ sound/pci/oxygen/xonar_pcm179x.c | 113 ++++++++++++++++++++++++++++++++++++--- 2 files changed, 190 insertions(+), 6 deletions(-) create mode 100644 sound/pci/oxygen/cs2000.h (limited to 'sound') diff --git a/sound/pci/oxygen/cs2000.h b/sound/pci/oxygen/cs2000.h new file mode 100644 index 000000000000..c3501bdb5edc --- /dev/null +++ b/sound/pci/oxygen/cs2000.h @@ -0,0 +1,83 @@ +#ifndef CS2000_H_INCLUDED +#define CS2000_H_INCLUDED + +#define CS2000_DEV_ID 0x01 +#define CS2000_DEV_CTRL 0x02 +#define CS2000_DEV_CFG_1 0x03 +#define CS2000_DEV_CFG_2 0x04 +#define CS2000_GLOBAL_CFG 0x05 +#define CS2000_RATIO_0 0x06 /* 32 bits, big endian */ +#define CS2000_RATIO_1 0x0a +#define CS2000_RATIO_2 0x0e +#define CS2000_RATIO_3 0x12 +#define CS2000_FUN_CFG_1 0x16 +#define CS2000_FUN_CFG_2 0x17 +#define CS2000_FUN_CFG_3 0x1e + +/* DEV_ID */ +#define CS2000_DEVICE_MASK 0xf8 +#define CS2000_REVISION_MASK 0x07 + +/* DEV_CTRL */ +#define CS2000_UNLOCK 0x80 +#define CS2000_AUX_OUT_DIS 0x02 +#define CS2000_CLK_OUT_DIS 0x01 + +/* DEV_CFG_1 */ +#define CS2000_R_MOD_SEL_MASK 0xe0 +#define CS2000_R_MOD_SEL_1 0x00 +#define CS2000_R_MOD_SEL_2 0x20 +#define CS2000_R_MOD_SEL_4 0x40 +#define CS2000_R_MOD_SEL_8 0x60 +#define CS2000_R_MOD_SEL_1_2 0x80 +#define CS2000_R_MOD_SEL_1_4 0xa0 +#define CS2000_R_MOD_SEL_1_8 0xc0 +#define CS2000_R_MOD_SEL_1_16 0xe0 +#define CS2000_R_SEL_MASK 0x18 +#define CS2000_R_SEL_SHIFT 3 +#define CS2000_AUX_OUT_SRC_MASK 0x06 +#define CS2000_AUX_OUT_SRC_REF_CLK 0x00 +#define CS2000_AUX_OUT_SRC_CLK_IN 0x02 +#define CS2000_AUX_OUT_SRC_CLK_OUT 0x04 +#define CS2000_AUX_OUT_SRC_PLL_LOCK 0x06 +#define CS2000_EN_DEV_CFG_1 0x01 + +/* DEV_CFG_2 */ +#define CS2000_LOCK_CLK_MASK 0x06 +#define CS2000_LOCK_CLK_SHIFT 1 +#define CS2000_FRAC_N_SRC_MASK 0x01 +#define CS2000_FRAC_N_SRC_STATIC 0x00 +#define CS2000_FRAC_N_SRC_DYNAMIC 0x01 + +/* GLOBAL_CFG */ +#define CS2000_FREEZE 0x08 +#define CS2000_EN_DEV_CFG_2 0x01 + +/* FUN_CFG_1 */ +#define CS2000_CLK_SKIP_EN 0x80 +#define CS2000_AUX_LOCK_CFG_MASK 0x40 +#define CS2000_AUX_LOCK_CFG_PP_HIGH 0x00 +#define CS2000_AUX_LOCK_CFG_OD_LOW 0x40 +#define CS2000_REF_CLK_DIV_MASK 0x18 +#define CS2000_REF_CLK_DIV_4 0x00 +#define CS2000_REF_CLK_DIV_2 0x08 +#define CS2000_REF_CLK_DIV_1 0x10 + +/* FUN_CFG_2 */ +#define CS2000_CLK_OUT_UNL 0x10 +#define CS2000_L_F_RATIO_CFG_MASK 0x08 +#define CS2000_L_F_RATIO_CFG_20_12 0x00 +#define CS2000_L_F_RATIO_CFG_12_20 0x08 + +/* FUN_CFG_3 */ +#define CS2000_CLK_IN_BW_MASK 0x70 +#define CS2000_CLK_IN_BW_1 0x00 +#define CS2000_CLK_IN_BW_2 0x10 +#define CS2000_CLK_IN_BW_4 0x20 +#define CS2000_CLK_IN_BW_8 0x30 +#define CS2000_CLK_IN_BW_16 0x40 +#define CS2000_CLK_IN_BW_32 0x50 +#define CS2000_CLK_IN_BW_64 0x60 +#define CS2000_CLK_IN_BW_128 0x70 + +#endif diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index eb5f015fcd23..522efde0d52e 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -91,6 +91,9 @@ * CMI8788: * * I²C <-> PCM1792A + * <-> CS2000 (ST only) + * + * ADC1 MCLK -> REF_CLK of CS2000 (ST only) * * GPI 0 <- external power present (STX only) * @@ -124,6 +127,7 @@ #include "xonar.h" #include "cm9780.h" #include "pcm1796.h" +#include "cs2000.h" #define GPIO_D2X_EXT_POWER 0x0020 @@ -143,12 +147,14 @@ #define GPIO_ST_HP 0x0080 #define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ii, /W=0 */ +#define I2C_DEVICE_CS2000 0x9c /* 100111, 0, /W=0 */ struct xonar_pcm179x { struct xonar_generic generic; unsigned int dacs; u8 oversampling; + u8 cs2000_fun_cfg_1; }; struct xonar_hdav { @@ -188,6 +194,11 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec, pcm1796_write_i2c(chip, codec, reg, value); } +static void cs2000_write(struct oxygen *chip, u8 reg, u8 value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_CS2000, reg, value); +} + static void update_pcm1796_volume(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; @@ -292,14 +303,17 @@ static void xonar_hdav_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } -static void xonar_st_init(struct oxygen *chip) +static void xonar_st_init_i2c(struct oxygen *chip) { - struct xonar_pcm179x *data = chip->model_data; - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | OXYGEN_2WIRE_INTERRUPT_MASK | OXYGEN_2WIRE_SPEED_FAST); +} + +static void xonar_st_init_common(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; data->generic.anti_pop_delay = 100; data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; @@ -320,15 +334,57 @@ static void xonar_st_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } +static void cs2000_registers_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + cs2000_write(chip, CS2000_GLOBAL_CFG, CS2000_FREEZE); + cs2000_write(chip, CS2000_DEV_CTRL, 0); + cs2000_write(chip, CS2000_DEV_CFG_1, + CS2000_R_MOD_SEL_1 | + (0 << CS2000_R_SEL_SHIFT) | + CS2000_AUX_OUT_SRC_REF_CLK | + CS2000_EN_DEV_CFG_1); + cs2000_write(chip, CS2000_DEV_CFG_2, + (0 << CS2000_LOCK_CLK_SHIFT) | + CS2000_FRAC_N_SRC_STATIC); + cs2000_write(chip, CS2000_RATIO_0 + 0, 0x00); /* 1.0 */ + cs2000_write(chip, CS2000_RATIO_0 + 1, 0x10); + cs2000_write(chip, CS2000_RATIO_0 + 2, 0x00); + cs2000_write(chip, CS2000_RATIO_0 + 3, 0x00); + cs2000_write(chip, CS2000_FUN_CFG_1, data->cs2000_fun_cfg_1); + cs2000_write(chip, CS2000_FUN_CFG_2, 0); + cs2000_write(chip, CS2000_GLOBAL_CFG, CS2000_EN_DEV_CFG_2); +} + +static void xonar_st_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; + + oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, + OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | + OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + + xonar_st_init_i2c(chip); + cs2000_registers_init(chip); + xonar_st_init_common(chip); + + snd_component_add(chip->card, "CS2000"); +} + static void xonar_stx_init(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; + xonar_st_init_i2c(chip); data->generic.ext_power_reg = OXYGEN_GPI_DATA; data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->generic.ext_power_bit = GPI_EXT_POWER; xonar_init_ext_power(chip); - xonar_st_init(chip); + xonar_st_init_common(chip); } static void xonar_d2_cleanup(struct oxygen *chip) @@ -378,12 +434,18 @@ static void xonar_hdav_resume(struct oxygen *chip) xonar_enable_output(chip); } -static void xonar_st_resume(struct oxygen *chip) +static void xonar_stx_resume(struct oxygen *chip) { pcm1796_init(chip); xonar_enable_output(chip); } +static void xonar_st_resume(struct oxygen *chip) +{ + cs2000_registers_init(chip); + xonar_stx_resume(chip); +} + static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -396,6 +458,43 @@ static void set_pcm1796_params(struct oxygen *chip, pcm1796_write(chip, i, 20, data->oversampling); } +static void set_cs2000_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + /* XXX Why is the I2S A MCLK half the actual I2S multich MCLK? */ + static const u8 rate_mclks[] = { + [OXYGEN_RATE_32000] = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_128, + [OXYGEN_RATE_44100] = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_128, + [OXYGEN_RATE_48000] = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_128, + [OXYGEN_RATE_64000] = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256, + [OXYGEN_RATE_88200] = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256, + [OXYGEN_RATE_96000] = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256, + [OXYGEN_RATE_176400] = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256, + [OXYGEN_RATE_192000] = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256, + }; + struct xonar_pcm179x *data = chip->model_data; + unsigned int rate_index; + u8 rate_mclk; + + rate_index = oxygen_read16(chip, OXYGEN_I2S_MULTICH_FORMAT) + & OXYGEN_I2S_RATE_MASK; + rate_mclk = rate_mclks[rate_index]; + oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, + OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); + if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_128) + data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; + else + data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_2; + cs2000_write(chip, CS2000_FUN_CFG_1, data->cs2000_fun_cfg_1); +} + +static void set_st_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + set_cs2000_params(chip, params); + set_pcm1796_params(chip, params); +} + static void set_hdav_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -590,7 +689,7 @@ static const struct oxygen_model model_xonar_st = { .cleanup = xonar_st_cleanup, .suspend = xonar_st_suspend, .resume = xonar_st_resume, - .set_dac_params = set_pcm1796_params, + .set_dac_params = set_st_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, @@ -652,6 +751,8 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, chip->model = model_xonar_st; chip->model.shortname = "Xonar STX"; chip->model.init = xonar_stx_init; + chip->model.resume = xonar_stx_resume; + chip->model.set_dac_params = set_pcm1796_params; break; default: return -EINVAL; -- cgit v1.2.2 From 75919d7c057be888c7cd7b192fad02182260b04a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:15:49 +0200 Subject: sound: oxygen: better defaults for upmixing control On card models with two-channel outputs, the base driver can automatically disable the upmixing control so that the model drivers do not need to do this. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 8 -------- sound/pci/oxygen/oxygen_mixer.c | 3 +++ sound/pci/oxygen/xonar_pcm179x.c | 2 -- 3 files changed, 3 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 84ef13183419..9026a143a5ec 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -141,19 +141,11 @@ static void set_cs5340_params(struct oxygen *chip, static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); -static int hifier_control_filter(struct snd_kcontrol_new *template) -{ - if (!strcmp(template->name, "Stereo Upmixing")) - return 1; /* stereo only - we don't need upmixing */ - return 0; -} - static const struct oxygen_model model_hifier = { .shortname = "C-Media CMI8787", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", .init = hifier_init, - .control_filter = hifier_control_filter, .cleanup = hifier_cleanup, .resume = hifier_resume, .set_dac_params = set_ak4396_params, diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 5401c547c4e3..e8e911a86c8e 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -954,6 +954,9 @@ static int add_controls(struct oxygen *chip, if (err == 1) continue; } + if (!strcmp(template.name, "Stereo Upmixing") && + chip->model.dac_channels == 2) + continue; if (!strcmp(template.name, "Master Playback Volume") && chip->model.dac_tlv) { template.tlv.p = chip->model.dac_tlv; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 522efde0d52e..07aaa893d323 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -605,8 +605,6 @@ static int xonar_st_control_filter(struct snd_kcontrol_new *template) { if (!strncmp(template->name, "CD Capture ", 11)) return 1; /* no CD input */ - if (!strcmp(template->name, "Stereo Upmixing")) - return 1; /* stereo only - we don't need upmixing */ return 0; } -- cgit v1.2.2 From 3d8bb454c4fbe18cea1adfd4183a4a9ef5f0ef04 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:16:41 +0200 Subject: sound: oxygen: add stereo upmixing to center/LFE channels Add the possibility to route a mix of the two channels of stereo data to the center and LFE outputs. This is implemented only for models where the DACs support this, i.e., for the Xonar D1 and DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 1 + sound/pci/oxygen/oxygen_mixer.c | 33 ++++++++++++++++++++++++--------- sound/pci/oxygen/oxygen_pcm.c | 6 ++++-- sound/pci/oxygen/xonar_cs43xx.c | 39 +++++++++++++++++++++++++++++---------- 4 files changed, 58 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index bd615dbffadb..2ac3b3c8253f 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -84,6 +84,7 @@ struct oxygen_model { struct snd_pcm_hw_params *params); void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); + void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed); void (*gpio_changed)(struct oxygen *chip); void (*uart_input)(struct oxygen *chip); void (*ac97_switch)(struct oxygen *chip, diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index e8e911a86c8e..5dfb5fb73381 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -99,11 +99,15 @@ static int dac_mute_put(struct snd_kcontrol *ctl, static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { - static const char *const names[3] = { - "Front", "Front+Surround", "Front+Surround+Back" + static const char *const names[5] = { + "Front", + "Front+Surround", + "Front+Surround+Back", + "Front+Surround+Center/LFE", + "Front+Surround+Center/LFE+Back", }; struct oxygen *chip = ctl->private_data; - unsigned int count = 2 + (chip->model.dac_channels == 8); + unsigned int count = chip->model.update_center_lfe_mix ? 5 : 3; info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; info->count = 1; @@ -127,7 +131,7 @@ static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) void oxygen_update_dac_routing(struct oxygen *chip) { /* DAC 0: front, DAC 1: surround, DAC 2: center/LFE, DAC 3: back */ - static const unsigned int reg_values[3] = { + static const unsigned int reg_values[5] = { /* stereo -> front */ (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | @@ -143,6 +147,16 @@ void oxygen_update_dac_routing(struct oxygen *chip) (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), + /* stereo -> front+surround+center/LFE */ + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), + /* stereo -> front+surround+center/LFE+back */ + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), }; u8 channels; unsigned int reg_value; @@ -167,22 +181,23 @@ void oxygen_update_dac_routing(struct oxygen *chip) OXYGEN_PLAY_DAC1_SOURCE_MASK | OXYGEN_PLAY_DAC2_SOURCE_MASK | OXYGEN_PLAY_DAC3_SOURCE_MASK); + if (chip->model.update_center_lfe_mix) + chip->model.update_center_lfe_mix(chip, chip->dac_routing > 2); } static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; - unsigned int count = 2 + (chip->model.dac_channels == 8); + unsigned int count = chip->model.update_center_lfe_mix ? 5 : 3; int changed; + if (value->value.enumerated.item[0] >= count) + return -EINVAL; mutex_lock(&chip->mutex); changed = value->value.enumerated.item[0] != chip->dac_routing; if (changed) { - chip->dac_routing = min(value->value.enumerated.item[0], - count - 1); - spin_lock_irq(&chip->reg_lock); + chip->dac_routing = value->value.enumerated.item[0]; oxygen_update_dac_routing(chip); - spin_unlock_irq(&chip->reg_lock); } mutex_unlock(&chip->mutex); return changed; diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index ef2345d82b86..1e98333366df 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -435,6 +435,7 @@ static int oxygen_spdif_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; + mutex_lock(&chip->mutex); spin_lock_irq(&chip->reg_lock); oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_OUT_ENABLE); @@ -446,6 +447,7 @@ static int oxygen_spdif_hw_params(struct snd_pcm_substream *substream, OXYGEN_SPDIF_OUT_RATE_MASK); oxygen_update_spdif_source(chip); spin_unlock_irq(&chip->reg_lock); + mutex_unlock(&chip->mutex); return 0; } @@ -459,6 +461,7 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; + mutex_lock(&chip->mutex); spin_lock_irq(&chip->reg_lock); oxygen_write8_masked(chip, OXYGEN_PLAY_CHANNELS, oxygen_play_channels(hw_params), @@ -475,12 +478,11 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_update_dac_routing(chip); oxygen_update_spdif_source(chip); spin_unlock_irq(&chip->reg_lock); - mutex_lock(&chip->mutex); chip->model.set_dac_params(chip, hw_params); + oxygen_update_dac_routing(chip); mutex_unlock(&chip->mutex); return 0; } diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 8fb5797577dd..0fa05ed6681d 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -67,6 +67,7 @@ struct xonar_cs43xx { struct xonar_generic generic; u8 cs4398_fm; u8 cs4362a_fm; + u8 cs4362a_fm_c; }; static void cs4398_write(struct oxygen *chip, u8 reg, u8 value) @@ -128,7 +129,7 @@ static void cs43xx_init(struct oxygen *chip) cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); cs4362a_write(chip, 0x05, 0); cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm_c); cs4362a_write(chip, 0x0c, data->cs4362a_fm); update_cs43xx_volume(chip); update_cs43xx_mute(chip); @@ -146,6 +147,7 @@ static void xonar_d1_init(struct oxygen *chip) data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; data->cs4362a_fm = CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + data->cs4362a_fm_c = data->cs4362a_fm; oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | @@ -202,25 +204,41 @@ static void set_cs43xx_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { struct xonar_cs43xx *data = chip->model_data; + u8 cs4398_fm, cs4362a_fm; - data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; if (params_rate(params) <= 50000) { - data->cs4398_fm |= CS4398_FM_SINGLE; - data->cs4362a_fm |= CS4362A_FM_SINGLE; + cs4398_fm = CS4398_FM_SINGLE; + cs4362a_fm = CS4362A_FM_SINGLE; } else if (params_rate(params) <= 100000) { - data->cs4398_fm |= CS4398_FM_DOUBLE; - data->cs4362a_fm |= CS4362A_FM_DOUBLE; + cs4398_fm = CS4398_FM_DOUBLE; + cs4362a_fm = CS4362A_FM_DOUBLE; } else { - data->cs4398_fm |= CS4398_FM_QUAD; - data->cs4362a_fm |= CS4362A_FM_QUAD; + cs4398_fm = CS4398_FM_QUAD; + cs4362a_fm = CS4362A_FM_QUAD; } + data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST | cs4398_fm; + data->cs4362a_fm = + (data->cs4362a_fm & ~CS4362A_FM_MASK) | cs4362a_fm; + data->cs4362a_fm_c = + (data->cs4362a_fm_c & ~CS4362A_FM_MASK) | cs4362a_fm; cs4398_write(chip, 2, data->cs4398_fm); cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm_c); cs4362a_write(chip, 0x0c, data->cs4362a_fm); } +static void update_cs43xx_center_lfe_mix(struct oxygen *chip, bool mixed) +{ + struct xonar_cs43xx *data = chip->model_data; + + data->cs4362a_fm_c &= ~CS4362A_ATAPI_MASK; + if (mixed) + data->cs4362a_fm_c |= CS4362A_ATAPI_B_LR | CS4362A_ATAPI_A_LR; + else + data->cs4362a_fm_c |= CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + cs4362a_write(chip, 0x09, data->cs4362a_fm_c); +} + static const struct snd_kcontrol_new front_panel_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Front Panel Switch", @@ -269,6 +287,7 @@ static const struct oxygen_model model_xonar_d1 = { .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, .update_dac_mute = update_cs43xx_mute, + .update_center_lfe_mix = update_cs43xx_center_lfe_mix, .ac97_switch = xonar_d1_line_mic_ac97_switch, .dac_tlv = cs4362a_db_scale, .model_data_size = sizeof(struct xonar_cs43xx), -- cgit v1.2.2 From dc0adf48daa81b05765d3c5ebab76321f77e9d21 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:17:36 +0200 Subject: sound: oxygen: more hardware documentation Add some comments describing the hardware pin routing. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 6 ++++++ sound/pci/oxygen/oxygen.c | 6 ++++++ sound/pci/oxygen/xonar_cs43xx.c | 4 ++++ sound/pci/oxygen/xonar_pcm179x.c | 17 +++++++++++++++++ 4 files changed, 33 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 9026a143a5ec..19e9e0123304 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -17,6 +17,12 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +/* + * CMI8788: + * + * SPI 0 -> AK4396 + */ + #include #include #include diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 72db4c39007f..53dff7193f31 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -18,6 +18,8 @@ */ /* + * CMI8788: + * * SPI 0 -> 1st AK4396 (front) * SPI 1 -> 2nd AK4396 (surround) * SPI 2 -> 3rd AK4396 (center/LFE) @@ -27,6 +29,10 @@ * GPIO 0 -> DFS0 of AK5385 * GPIO 1 -> DFS1 of AK5385 * GPIO 8 -> enable headphone amplifier on HT-Omega models + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to ADC input */ #include diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 0fa05ed6681d..a8ec4e8271a4 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -41,6 +41,10 @@ * CS4362A: * * AD0 <- 0 + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5361 input */ #include diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 07aaa893d323..97574dbec2b6 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -32,6 +32,10 @@ * GPIO 5 <- external power present (D2X only) * GPIO 7 -> ALT * GPIO 8 -> enable output to speakers + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input */ /* @@ -54,6 +58,10 @@ * * PCM1796 front: AD1,0 <- 0,0 * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * * no daughterboard * ---------------- * @@ -107,6 +115,15 @@ * PCM1792A: * * AD1,0 <- 0,0 + * SCK <- CLK_OUT of CS2000 (ST only) + * + * CS2000: + * + * AD0 <- 0 + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input * * H6 daughterboard * ---------------- -- cgit v1.2.2 From 6f0de3ce068e48b033b5e4d0822b47218e9d206c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:18:45 +0200 Subject: sound: oxygen: cache codec registers Keep a cache of codec registers to avoid unnecessary writes. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 46 ++++++++----- sound/pci/oxygen/oxygen.c | 107 ++++++++++++++++-------------- sound/pci/oxygen/xonar_cs43xx.c | 140 ++++++++++++++++++++++++--------------- sound/pci/oxygen/xonar_pcm179x.c | 109 ++++++++++++++++++++---------- 4 files changed, 250 insertions(+), 152 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 19e9e0123304..2079c100aabc 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -57,23 +57,28 @@ static struct pci_device_id hifier_ids[] __devinitdata = { MODULE_DEVICE_TABLE(pci, hifier_ids); struct hifier_data { - u8 ak4396_ctl2; + u8 ak4396_regs[5]; }; static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) { + struct hifier_data *data = chip->model_data; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (0 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); + data->ak4396_regs[reg] = value; } -static void update_ak4396_volume(struct oxygen *chip) +static void ak4396_write_cached(struct oxygen *chip, u8 reg, u8 value) { - ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); + struct hifier_data *data = chip->model_data; + + if (value != data->ak4396_regs[reg]) + ak4396_write(chip, reg, value); } static void hifier_registers_init(struct oxygen *chip) @@ -81,16 +86,19 @@ static void hifier_registers_init(struct oxygen *chip) struct hifier_data *data = chip->model_data; ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); + ak4396_write(chip, AK4396_CONTROL_2, + data->ak4396_regs[AK4396_CONTROL_2]); ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - update_ak4396_volume(chip); + ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); } static void hifier_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + data->ak4396_regs[AK4396_CONTROL_2] = + AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; hifier_registers_init(chip); snd_component_add(chip->card, "AK4396"); @@ -112,20 +120,29 @@ static void set_ak4396_params(struct oxygen *chip, struct hifier_data *data = chip->model_data; u8 value; - value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; + value = data->ak4396_regs[AK4396_CONTROL_2] & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; else if (params_rate(params) <= 108000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; - data->ak4396_ctl2 = value; msleep(1); /* wait for the new MCLK to become stable */ - ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB); - ak4396_write(chip, AK4396_CONTROL_2, value); - ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); + if (value != data->ak4396_regs[AK4396_CONTROL_2]) { + ak4396_write(chip, AK4396_CONTROL_1, + AK4396_DIF_24_MSB); + ak4396_write(chip, AK4396_CONTROL_2, value); + ak4396_write(chip, AK4396_CONTROL_1, + AK4396_DIF_24_MSB | AK4396_RSTN); + } +} + +static void update_ak4396_volume(struct oxygen *chip) +{ + ak4396_write_cached(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write_cached(chip, AK4396_RCH_ATT, chip->dac_volume[1]); } static void update_ak4396_mute(struct oxygen *chip) @@ -133,11 +150,10 @@ static void update_ak4396_mute(struct oxygen *chip) struct hifier_data *data = chip->model_data; u8 value; - value = data->ak4396_ctl2 & ~AK4396_SMUTE; + value = data->ak4396_regs[AK4396_CONTROL_2] & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; - data->ak4396_ctl2 = value; - ak4396_write(chip, AK4396_CONTROL_2, value); + ak4396_write_cached(chip, AK4396_CONTROL_2, value); } static void set_cs5340_params(struct oxygen *chip, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 53dff7193f31..c986c5ebf65b 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -97,8 +97,8 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_CLARO_HP 0x0100 struct generic_data { - u8 ak4396_ctl2; - u16 saved_wm8785_registers[2]; + u8 ak4396_regs[4][5]; + u16 wm8785_regs[1]; }; static void ak4396_write(struct oxygen *chip, unsigned int codec, @@ -108,12 +108,24 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, static const u8 codec_spi_map[4] = { 0, 1, 2, 4 }; + struct generic_data *data = chip->model_data; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (codec_spi_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); + data->ak4396_regs[codec][reg] = value; +} + +static void ak4396_write_cached(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + struct generic_data *data = chip->model_data; + + if (value != data->ak4396_regs[codec][reg]) + ak4396_write(chip, codec, reg, value); } static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) @@ -126,20 +138,8 @@ static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) (3 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); - if (reg < ARRAY_SIZE(data->saved_wm8785_registers)) - data->saved_wm8785_registers[reg] = value; -} - -static void update_ak4396_volume(struct oxygen *chip) -{ - unsigned int i; - - for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_LCH_ATT, chip->dac_volume[i * 2]); - ak4396_write(chip, i, - AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); - } + if (reg < ARRAY_SIZE(data->wm8785_regs)) + data->wm8785_regs[reg] = value; } static void ak4396_registers_init(struct oxygen *chip) @@ -148,21 +148,25 @@ static void ak4396_registers_init(struct oxygen *chip) unsigned int i; for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, i, - AK4396_CONTROL_2, data->ak4396_ctl2); - ak4396_write(chip, i, - AK4396_CONTROL_3, AK4396_PCM); + ak4396_write(chip, i, AK4396_CONTROL_1, + AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write(chip, i, AK4396_CONTROL_2, + data->ak4396_regs[0][AK4396_CONTROL_2]); + ak4396_write(chip, i, AK4396_CONTROL_3, + AK4396_PCM); + ak4396_write(chip, i, AK4396_LCH_ATT, + chip->dac_volume[i * 2]); + ak4396_write(chip, i, AK4396_RCH_ATT, + chip->dac_volume[i * 2 + 1]); } - update_ak4396_volume(chip); } static void ak4396_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + data->ak4396_regs[0][AK4396_CONTROL_2] = + AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_registers_init(chip); snd_component_add(chip->card, "AK4396"); } @@ -179,17 +183,15 @@ static void wm8785_registers_init(struct oxygen *chip) struct generic_data *data = chip->model_data; wm8785_write(chip, WM8785_R7, 0); - wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); - wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); + wm8785_write(chip, WM8785_R0, data->wm8785_regs[0]); } static void wm8785_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; - data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | - WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; - data->saved_wm8785_registers[1] = WM8785_WL_24; + data->wm8785_regs[0] = + WM8785_MCR_SLAVE | WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; wm8785_registers_init(chip); snd_component_add(chip->card, "WM8785"); } @@ -270,24 +272,36 @@ static void set_ak4396_params(struct oxygen *chip, unsigned int i; u8 value; - value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; + value = data->ak4396_regs[0][AK4396_CONTROL_2] & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; else if (params_rate(params) <= 108000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; - data->ak4396_ctl2 = value; msleep(1); /* wait for the new MCLK to become stable */ + if (value != data->ak4396_regs[0][AK4396_CONTROL_2]) { + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, AK4396_CONTROL_1, + AK4396_DIF_24_MSB); + ak4396_write(chip, i, AK4396_CONTROL_2, value); + ak4396_write(chip, i, AK4396_CONTROL_1, + AK4396_DIF_24_MSB | AK4396_RSTN); + } + } +} + +static void update_ak4396_volume(struct oxygen *chip) +{ + unsigned int i; + for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_CONTROL_1, AK4396_DIF_24_MSB); - ak4396_write(chip, i, - AK4396_CONTROL_2, value); - ak4396_write(chip, i, - AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write_cached(chip, i, AK4396_LCH_ATT, + chip->dac_volume[i * 2]); + ak4396_write_cached(chip, i, AK4396_RCH_ATT, + chip->dac_volume[i * 2 + 1]); } } @@ -297,21 +311,19 @@ static void update_ak4396_mute(struct oxygen *chip) unsigned int i; u8 value; - value = data->ak4396_ctl2 & ~AK4396_SMUTE; + value = data->ak4396_regs[0][AK4396_CONTROL_2] & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; - data->ak4396_ctl2 = value; for (i = 0; i < 4; ++i) - ak4396_write(chip, i, AK4396_CONTROL_2, value); + ak4396_write_cached(chip, i, AK4396_CONTROL_2, value); } static void set_wm8785_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { + struct generic_data *data = chip->model_data; unsigned int value; - wm8785_write(chip, WM8785_R7, 0); - value = WM8785_MCR_SLAVE | WM8785_FORMAT_LJUST; if (params_rate(params) <= 48000) value |= WM8785_OSR_SINGLE; @@ -319,13 +331,10 @@ static void set_wm8785_params(struct oxygen *chip, value |= WM8785_OSR_DOUBLE; else value |= WM8785_OSR_QUAD; - wm8785_write(chip, WM8785_R0, value); - - if (snd_pcm_format_width(params_format(params)) <= 16) - value = WM8785_WL_16; - else - value = WM8785_WL_24; - wm8785_write(chip, WM8785_R1, value); + if (value != data->wm8785_regs[0]) { + wm8785_write(chip, WM8785_R7, 0); + wm8785_write(chip, WM8785_R0, value); + } } static void set_ak5385_params(struct oxygen *chip, diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index a8ec4e8271a4..330c5e755917 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -69,62 +69,58 @@ struct xonar_cs43xx { struct xonar_generic generic; - u8 cs4398_fm; - u8 cs4362a_fm; - u8 cs4362a_fm_c; + u8 cs4398_regs[7]; + u8 cs4362a_regs[15]; }; static void cs4398_write(struct oxygen *chip, u8 reg, u8 value) { - oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value); -} + struct xonar_cs43xx *data = chip->model_data; -static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) -{ - oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); + oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value); + if (reg < ARRAY_SIZE(data->cs4398_regs)) + data->cs4398_regs[reg] = value; } -static void update_cs4362a_volumes(struct oxygen *chip) +static void cs4398_write_cached(struct oxygen *chip, u8 reg, u8 value) { - u8 mute; + struct xonar_cs43xx *data = chip->model_data; - mute = chip->dac_mute ? CS4362A_MUTE : 0; - cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); - cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); - cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); - cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); - cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); - cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); + if (value != data->cs4398_regs[reg]) + cs4398_write(chip, reg, value); } -static void update_cs43xx_volume(struct oxygen *chip) +static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) { - cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); - cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); - update_cs4362a_volumes(chip); + struct xonar_cs43xx *data = chip->model_data; + + oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); + if (reg < ARRAY_SIZE(data->cs4362a_regs)) + data->cs4362a_regs[reg] = value; } -static void update_cs43xx_mute(struct oxygen *chip) +static void cs4362a_write_cached(struct oxygen *chip, u8 reg, u8 value) { - u8 reg; + struct xonar_cs43xx *data = chip->model_data; - reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; - if (chip->dac_mute) - reg |= CS4398_MUTE_B | CS4398_MUTE_A; - cs4398_write(chip, 4, reg); - update_cs4362a_volumes(chip); + if (value != data->cs4362a_regs[reg]) + cs4362a_write(chip, reg, value); } -static void cs43xx_init(struct oxygen *chip) +static void cs43xx_registers_init(struct oxygen *chip) { struct xonar_cs43xx *data = chip->model_data; + unsigned int i; /* set CPEN (control port mode) and power down */ cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); /* configure */ - cs4398_write(chip, 2, data->cs4398_fm); + cs4398_write(chip, 2, data->cs4398_regs[2]); cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); + cs4398_write(chip, 4, data->cs4398_regs[4]); + cs4398_write(chip, 5, data->cs4398_regs[5]); + cs4398_write(chip, 6, data->cs4398_regs[6]); cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); @@ -132,11 +128,8 @@ static void cs43xx_init(struct oxygen *chip) CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); cs4362a_write(chip, 0x05, 0); - cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm_c); - cs4362a_write(chip, 0x0c, data->cs4362a_fm); - update_cs43xx_volume(chip); - update_cs43xx_mute(chip); + for (i = 6; i <= 14; ++i) + cs4362a_write(chip, i, data->cs4362a_regs[i]); /* clear power down */ cs4398_write(chip, 8, CS4398_CPEN); cs4362a_write(chip, 0x01, CS4362A_CPEN); @@ -148,17 +141,29 @@ static void xonar_d1_init(struct oxygen *chip) data->generic.anti_pop_delay = 800; data->generic.output_enable_bit = GPIO_D1_OUTPUT_ENABLE; - data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_FM_SINGLE | + data->cs4398_regs[2] = + CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4398_regs[4] = CS4398_MUTEP_LOW | + CS4398_MUTE_B | CS4398_MUTE_A | CS4398_PAMUTE; + data->cs4398_regs[5] = 60 * 2; + data->cs4398_regs[6] = 60 * 2; + data->cs4362a_regs[6] = CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - data->cs4362a_fm_c = data->cs4362a_fm; + data->cs4362a_regs[7] = 60 | CS4362A_MUTE; + data->cs4362a_regs[8] = 60 | CS4362A_MUTE; + data->cs4362a_regs[9] = data->cs4362a_regs[6]; + data->cs4362a_regs[10] = 60 | CS4362A_MUTE; + data->cs4362a_regs[11] = 60 | CS4362A_MUTE; + data->cs4362a_regs[12] = data->cs4362a_regs[6]; + data->cs4362a_regs[13] = 60 | CS4362A_MUTE; + data->cs4362a_regs[14] = 60 | CS4362A_MUTE; oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | OXYGEN_2WIRE_INTERRUPT_MASK | OXYGEN_2WIRE_SPEED_FAST); - cs43xx_init(chip); + cs43xx_registers_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); @@ -200,7 +205,7 @@ static void xonar_d1_resume(struct oxygen *chip) { oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); msleep(1); - cs43xx_init(chip); + cs43xx_registers_init(chip); xonar_enable_output(chip); } @@ -220,27 +225,56 @@ static void set_cs43xx_params(struct oxygen *chip, cs4398_fm = CS4398_FM_QUAD; cs4362a_fm = CS4362A_FM_QUAD; } - data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST | cs4398_fm; - data->cs4362a_fm = - (data->cs4362a_fm & ~CS4362A_FM_MASK) | cs4362a_fm; - data->cs4362a_fm_c = - (data->cs4362a_fm_c & ~CS4362A_FM_MASK) | cs4362a_fm; - cs4398_write(chip, 2, data->cs4398_fm); - cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm_c); - cs4362a_write(chip, 0x0c, data->cs4362a_fm); + cs4398_fm |= CS4398_DEM_NONE | CS4398_DIF_LJUST; + cs4398_write_cached(chip, 2, cs4398_fm); + cs4362a_fm |= data->cs4362a_regs[6] & ~CS4362A_FM_MASK; + cs4362a_write_cached(chip, 6, cs4362a_fm); + cs4362a_write_cached(chip, 12, cs4362a_fm); + cs4362a_fm &= CS4362A_FM_MASK; + cs4362a_fm |= data->cs4362a_regs[9] & ~CS4362A_FM_MASK; + cs4362a_write_cached(chip, 9, cs4362a_fm); +} + +static void update_cs4362a_volumes(struct oxygen *chip) +{ + unsigned int i; + u8 mute; + + mute = chip->dac_mute ? CS4362A_MUTE : 0; + for (i = 0; i < 6; ++i) + cs4362a_write_cached(chip, 7 + i + i / 2, + (127 - chip->dac_volume[2 + i]) | mute); +} + +static void update_cs43xx_volume(struct oxygen *chip) +{ + cs4398_write_cached(chip, 5, (127 - chip->dac_volume[0]) * 2); + cs4398_write_cached(chip, 6, (127 - chip->dac_volume[1]) * 2); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_mute(struct oxygen *chip) +{ + u8 reg; + + reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; + if (chip->dac_mute) + reg |= CS4398_MUTE_B | CS4398_MUTE_A; + cs4398_write_cached(chip, 4, reg); + update_cs4362a_volumes(chip); } static void update_cs43xx_center_lfe_mix(struct oxygen *chip, bool mixed) { struct xonar_cs43xx *data = chip->model_data; + u8 reg; - data->cs4362a_fm_c &= ~CS4362A_ATAPI_MASK; + reg = data->cs4362a_regs[9] & ~CS4362A_ATAPI_MASK; if (mixed) - data->cs4362a_fm_c |= CS4362A_ATAPI_B_LR | CS4362A_ATAPI_A_LR; + reg |= CS4362A_ATAPI_B_LR | CS4362A_ATAPI_A_LR; else - data->cs4362a_fm_c |= CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - cs4362a_write(chip, 0x09, data->cs4362a_fm_c); + reg |= CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + cs4362a_write_cached(chip, 9, reg); } static const struct snd_kcontrol_new front_panel_switch = { diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 97574dbec2b6..e17ee5e8e510 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -166,11 +166,13 @@ #define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ii, /W=0 */ #define I2C_DEVICE_CS2000 0x9c /* 100111, 0, /W=0 */ +#define PCM1796_REG_BASE 16 + struct xonar_pcm179x { struct xonar_generic generic; unsigned int dacs; - u8 oversampling; + u8 pcm1796_regs[4][5]; u8 cs2000_fun_cfg_1; }; @@ -204,54 +206,71 @@ static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec, static void pcm1796_write(struct oxygen *chip, unsigned int codec, u8 reg, u8 value) { + struct xonar_pcm179x *data = chip->model_data; + if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) == OXYGEN_FUNCTION_SPI) pcm1796_write_spi(chip, codec, reg, value); else pcm1796_write_i2c(chip, codec, reg, value); + if ((unsigned int)(reg - PCM1796_REG_BASE) + < ARRAY_SIZE(data->pcm1796_regs[codec])) + data->pcm1796_regs[codec][reg - PCM1796_REG_BASE] = value; } -static void cs2000_write(struct oxygen *chip, u8 reg, u8 value) +static void pcm1796_write_cached(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) { - oxygen_write_i2c(chip, I2C_DEVICE_CS2000, reg, value); + struct xonar_pcm179x *data = chip->model_data; + + if (value != data->pcm1796_regs[codec][reg - PCM1796_REG_BASE]) + pcm1796_write(chip, codec, reg, value); } -static void update_pcm1796_volume(struct oxygen *chip) +static void cs2000_write(struct oxygen *chip, u8 reg, u8 value) { struct xonar_pcm179x *data = chip->model_data; - unsigned int i; - for (i = 0; i < data->dacs; ++i) { - pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); - } + oxygen_write_i2c(chip, I2C_DEVICE_CS2000, reg, value); + if (reg == CS2000_FUN_CFG_1) + data->cs2000_fun_cfg_1 = value; } -static void update_pcm1796_mute(struct oxygen *chip) +static void cs2000_write_cached(struct oxygen *chip, u8 reg, u8 value) { struct xonar_pcm179x *data = chip->model_data; - unsigned int i; - u8 value; - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; - if (chip->dac_mute) - value |= PCM1796_MUTE; - for (i = 0; i < data->dacs; ++i) - pcm1796_write(chip, i, 18, value); + if (reg != CS2000_FUN_CFG_1 || + value != data->cs2000_fun_cfg_1) + cs2000_write(chip, reg, value); } -static void pcm1796_init(struct oxygen *chip) +static void pcm1796_registers_init(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; unsigned int i; for (i = 0; i < data->dacs; ++i) { + /* set ATLD before ATL/ATR */ + pcm1796_write(chip, i, 18, + data->pcm1796_regs[0][18 - PCM1796_REG_BASE]); + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); - pcm1796_write(chip, i, 20, data->oversampling); + pcm1796_write(chip, i, 20, + data->pcm1796_regs[0][20 - PCM1796_REG_BASE]); pcm1796_write(chip, i, 21, 0); } - update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ - update_pcm1796_volume(chip); +} + +static void pcm1796_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = PCM1796_MUTE | + PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = PCM1796_OS_64; + pcm1796_registers_init(chip); } static void xonar_d2_init(struct oxygen *chip) @@ -261,7 +280,6 @@ static void xonar_d2_init(struct oxygen *chip) data->generic.anti_pop_delay = 300; data->generic.output_enable_bit = GPIO_D2_OUTPUT_ENABLE; data->dacs = 4; - data->oversampling = PCM1796_OS_64; pcm1796_init(chip); @@ -304,7 +322,6 @@ static void xonar_hdav_init(struct oxygen *chip) data->pcm179x.generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->pcm179x.generic.ext_power_bit = GPI_EXT_POWER; data->pcm179x.dacs = chip->model.private_data ? 4 : 1; - data->pcm179x.oversampling = PCM1796_OS_64; pcm1796_init(chip); @@ -335,7 +352,6 @@ static void xonar_st_init_common(struct oxygen *chip) data->generic.anti_pop_delay = 100; data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; data->dacs = chip->model.private_data ? 4 : 1; - data->oversampling = PCM1796_OS_64; pcm1796_init(chip); @@ -438,7 +454,7 @@ static void xonar_st_suspend(struct oxygen *chip) static void xonar_d2_resume(struct oxygen *chip) { - pcm1796_init(chip); + pcm1796_registers_init(chip); xonar_enable_output(chip); } @@ -446,14 +462,14 @@ static void xonar_hdav_resume(struct oxygen *chip) { struct xonar_hdav *data = chip->model_data; - pcm1796_init(chip); + pcm1796_registers_init(chip); xonar_hdmi_resume(chip, &data->hdmi); xonar_enable_output(chip); } static void xonar_stx_resume(struct oxygen *chip) { - pcm1796_init(chip); + pcm1796_registers_init(chip); xonar_enable_output(chip); } @@ -468,11 +484,35 @@ static void set_pcm1796_params(struct oxygen *chip, { struct xonar_pcm179x *data = chip->model_data; unsigned int i; + u8 reg; + + reg = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; + for (i = 0; i < data->dacs; ++i) + pcm1796_write_cached(chip, i, 20, reg); +} + +static void update_pcm1796_volume(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + + for (i = 0; i < data->dacs; ++i) { + pcm1796_write_cached(chip, i, 16, chip->dac_volume[i * 2]); + pcm1796_write_cached(chip, i, 17, chip->dac_volume[i * 2 + 1]); + } +} + +static void update_pcm1796_mute(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + u8 value; - data->oversampling = - params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + if (chip->dac_mute) + value |= PCM1796_MUTE; for (i = 0; i < data->dacs; ++i) - pcm1796_write(chip, i, 20, data->oversampling); + pcm1796_write_cached(chip, i, 18, value); } static void set_cs2000_params(struct oxygen *chip, @@ -489,9 +529,8 @@ static void set_cs2000_params(struct oxygen *chip, [OXYGEN_RATE_176400] = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256, [OXYGEN_RATE_192000] = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256, }; - struct xonar_pcm179x *data = chip->model_data; unsigned int rate_index; - u8 rate_mclk; + u8 rate_mclk, reg; rate_index = oxygen_read16(chip, OXYGEN_I2S_MULTICH_FORMAT) & OXYGEN_I2S_RATE_MASK; @@ -499,10 +538,10 @@ static void set_cs2000_params(struct oxygen *chip, oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_128) - data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; + reg = CS2000_REF_CLK_DIV_1; else - data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_2; - cs2000_write(chip, CS2000_FUN_CFG_1, data->cs2000_fun_cfg_1); + reg = CS2000_REF_CLK_DIV_2; + cs2000_write_cached(chip, CS2000_FUN_CFG_1, reg); } static void set_st_params(struct oxygen *chip, -- cgit v1.2.2 From a361e247b4e36c567b44fef354ab595458422d44 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:19:19 +0200 Subject: sound: virtuoso: add headphone impedance control Add a mixer control to adjust the headphone amplifier output for headphones with different impedances. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_pcm179x.c | 110 +++++++++++++++++++++++++++++++++++---- 1 file changed, 99 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index e17ee5e8e510..cf94e4432a3f 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -173,6 +173,8 @@ struct xonar_pcm179x { struct xonar_generic generic; unsigned int dacs; u8 pcm1796_regs[4][5]; + bool hp_active; + s8 hp_gain_offset; u8 cs2000_fun_cfg_1; }; @@ -249,13 +251,17 @@ static void pcm1796_registers_init(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; unsigned int i; + s8 gain_offset; + gain_offset = data->hp_active ? data->hp_gain_offset : 0; for (i = 0; i < data->dacs; ++i) { /* set ATLD before ATL/ATR */ pcm1796_write(chip, i, 18, data->pcm1796_regs[0][18 - PCM1796_REG_BASE]); - pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2] + + gain_offset); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1] + + gain_offset); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); pcm1796_write(chip, i, 20, data->pcm1796_regs[0][20 - PCM1796_REG_BASE]); @@ -352,6 +358,7 @@ static void xonar_st_init_common(struct oxygen *chip) data->generic.anti_pop_delay = 100; data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; data->dacs = chip->model.private_data ? 4 : 1; + data->hp_gain_offset = 2*-18; pcm1796_init(chip); @@ -495,10 +502,14 @@ static void update_pcm1796_volume(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; unsigned int i; + s8 gain_offset; + gain_offset = data->hp_active ? data->hp_gain_offset : 0; for (i = 0; i < data->dacs; ++i) { - pcm1796_write_cached(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write_cached(chip, i, 17, chip->dac_volume[i * 2 + 1]); + pcm1796_write_cached(chip, i, 16, chip->dac_volume[i * 2] + + gain_offset); + pcm1796_write_cached(chip, i, 17, chip->dac_volume[i * 2 + 1] + + gain_offset); } } @@ -606,6 +617,7 @@ static int st_output_switch_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; u16 gpio_old, gpio; mutex_lock(&chip->mutex); @@ -623,16 +635,83 @@ static int st_output_switch_put(struct snd_kcontrol *ctl, break; } oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); + data->hp_active = gpio & GPIO_ST_HP; + update_pcm1796_volume(chip); mutex_unlock(&chip->mutex); return gpio != gpio_old; } -static const struct snd_kcontrol_new st_output_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Output", - .info = st_output_switch_info, - .get = st_output_switch_get, - .put = st_output_switch_put, +static int st_hp_volume_offset_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "< 64 ohms", "64-300 ohms", "300-600 ohms" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item > 2) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int st_hp_volume_offset_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + mutex_lock(&chip->mutex); + if (data->hp_gain_offset < 2*-6) + value->value.enumerated.item[0] = 0; + else if (data->hp_gain_offset < 0) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + mutex_unlock(&chip->mutex); + return 0; +} + + +static int st_hp_volume_offset_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + static const s8 offsets[] = { 2*-18, 2*-6, 0 }; + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + s8 offset; + int changed; + + if (value->value.enumerated.item[0] > 2) + return -EINVAL; + offset = offsets[value->value.enumerated.item[0]]; + mutex_lock(&chip->mutex); + changed = offset != data->hp_gain_offset; + if (changed) { + data->hp_gain_offset = offset; + update_pcm1796_volume(chip); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new st_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output", + .info = st_output_switch_info, + .get = st_output_switch_get, + .put = st_output_switch_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphones Impedance Playback Enum", + .info = st_hp_volume_offset_info, + .get = st_hp_volume_offset_get, + .put = st_hp_volume_offset_put, + }, }; static void xonar_line_mic_ac97_switch(struct oxygen *chip, @@ -671,7 +750,16 @@ static int xonar_d2_mixer_init(struct oxygen *chip) static int xonar_st_mixer_init(struct oxygen *chip) { - return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip)); + unsigned int i; + int err; + + for (i = 0; i < ARRAY_SIZE(st_controls); ++i) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&st_controls[i], chip)); + if (err < 0) + return err; + } + return 0; } static const struct oxygen_model model_xonar_d2 = { -- cgit v1.2.2 From 76ffe1e3fb2f65e98d7ed001c5a2b6f334655364 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:20:11 +0200 Subject: sound: oxygen: allow custom MCLK rates Add a callback that allows model drivers to modify the default I2S MCLK rate. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 1 + sound/pci/oxygen/oxygen.c | 1 + sound/pci/oxygen/oxygen.h | 4 ++++ sound/pci/oxygen/oxygen_pcm.c | 13 +++++++++---- sound/pci/oxygen/xonar_cs43xx.c | 1 + sound/pci/oxygen/xonar_pcm179x.c | 3 +++ 6 files changed, 19 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 2079c100aabc..e3c229b63311 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -170,6 +170,7 @@ static const struct oxygen_model model_hifier = { .init = hifier_init, .cleanup = hifier_cleanup, .resume = hifier_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_ak4396_params, .set_adc_params = set_cs5340_params, .update_dac_volume = update_ak4396_volume, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index c986c5ebf65b..d12fd9efe94e 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -361,6 +361,7 @@ static const struct oxygen_model model_generic = { .init = generic_init, .cleanup = generic_cleanup, .resume = generic_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 2ac3b3c8253f..6147216af744 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -78,6 +78,8 @@ struct oxygen_model { void (*resume)(struct oxygen *chip); void (*pcm_hardware_filter)(unsigned int channel, struct snd_pcm_hardware *hardware); + unsigned int (*get_i2s_mclk)(struct oxygen *chip, unsigned int channel, + struct snd_pcm_hw_params *hw_params); void (*set_dac_params)(struct oxygen *chip, struct snd_pcm_hw_params *params); void (*set_adc_params)(struct oxygen *chip, @@ -163,6 +165,8 @@ void oxygen_update_spdif_source(struct oxygen *chip); /* oxygen_pcm.c */ int oxygen_pcm_init(struct oxygen *chip); +unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, unsigned int channel, + struct snd_pcm_hw_params *hw_params); /* oxygen_io.c */ diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 1e98333366df..9dff6954c397 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -271,13 +271,16 @@ static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params) } } -static unsigned int oxygen_i2s_mclk(struct snd_pcm_hw_params *hw_params) +unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, + unsigned int channel, + struct snd_pcm_hw_params *hw_params) { if (params_rate(hw_params) <= 96000) return OXYGEN_I2S_MCLK_256; else return OXYGEN_I2S_MCLK_128; } +EXPORT_SYMBOL(oxygen_default_i2s_mclk); static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params) { @@ -354,7 +357,7 @@ static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, OXYGEN_REC_FORMAT_A_MASK); oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, oxygen_rate(hw_params) | - oxygen_i2s_mclk(hw_params) | + chip->model.get_i2s_mclk(chip, PCM_A, hw_params) | chip->model.adc_i2s_format | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | @@ -390,7 +393,8 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, if (!is_ac97) oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, oxygen_rate(hw_params) | - oxygen_i2s_mclk(hw_params) | + chip->model.get_i2s_mclk(chip, PCM_B, + hw_params) | chip->model.adc_i2s_format | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | @@ -472,7 +476,8 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, oxygen_rate(hw_params) | chip->model.dac_i2s_format | - oxygen_i2s_mclk(hw_params) | + chip->model.get_i2s_mclk(chip, PCM_MULTICH, + hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 330c5e755917..a83f827feb34 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -321,6 +321,7 @@ static const struct oxygen_model model_xonar_d1 = { .cleanup = xonar_d1_cleanup, .suspend = xonar_d1_suspend, .resume = xonar_d1_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_cs43xx_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index cf94e4432a3f..35b3fb4071fb 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -771,6 +771,7 @@ static const struct oxygen_model model_xonar_d2 = { .cleanup = xonar_d2_cleanup, .suspend = xonar_d2_suspend, .resume = xonar_d2_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_pcm1796_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -801,6 +802,7 @@ static const struct oxygen_model model_xonar_hdav = { .suspend = xonar_hdav_suspend, .resume = xonar_hdav_resume, .pcm_hardware_filter = xonar_hdmi_pcm_hardware_filter, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_hdav_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -831,6 +833,7 @@ static const struct oxygen_model model_xonar_st = { .cleanup = xonar_st_cleanup, .suspend = xonar_st_suspend, .resume = xonar_st_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_st_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, -- cgit v1.2.2 From 973dca93a3d46cca7e4743300f8a510b779906af Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:20:47 +0200 Subject: sound: virtuoso: add PCM1796 oversampling control Add a control to increase the oversampling factor to 128x on cards with PCM1796 or PCM1792A DACs. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_pcm179x.c | 182 +++++++++++++++++++++++++++++++++------ 1 file changed, 157 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 35b3fb4071fb..7f153fb1848d 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -173,8 +173,11 @@ struct xonar_pcm179x { struct xonar_generic generic; unsigned int dacs; u8 pcm1796_regs[4][5]; + unsigned int current_rate; + bool os_128; bool hp_active; s8 hp_gain_offset; + bool has_cs2000; u8 cs2000_fun_cfg_1; }; @@ -277,6 +280,7 @@ static void pcm1796_init(struct oxygen *chip) PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = PCM1796_OS_64; pcm1796_registers_init(chip); + data->current_rate = 48000; } static void xonar_d2_init(struct oxygen *chip) @@ -401,6 +405,7 @@ static void xonar_st_init(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; + data->has_cs2000 = 1; data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, @@ -486,18 +491,57 @@ static void xonar_st_resume(struct oxygen *chip) xonar_stx_resume(chip); } -static void set_pcm1796_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) +static unsigned int mclk_from_rate(struct oxygen *chip, unsigned int rate) +{ + struct xonar_pcm179x *data = chip->model_data; + + if (rate <= 32000) + return OXYGEN_I2S_MCLK_512; + else if (rate <= 48000 && data->os_128) + return OXYGEN_I2S_MCLK_512; + else if (rate <= 96000) + return OXYGEN_I2S_MCLK_256; + else + return OXYGEN_I2S_MCLK_128; +} + +static unsigned int get_pcm1796_i2s_mclk(struct oxygen *chip, + unsigned int channel, + struct snd_pcm_hw_params *params) +{ + if (channel == PCM_MULTICH) + return mclk_from_rate(chip, params_rate(params)); + else + return oxygen_default_i2s_mclk(chip, channel, params); +} + +static void update_pcm1796_oversampling(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; unsigned int i; u8 reg; - reg = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; + if (data->current_rate <= 32000) + reg = PCM1796_OS_128; + else if (data->current_rate <= 48000 && data->os_128) + reg = PCM1796_OS_128; + else if (data->current_rate <= 96000 || data->os_128) + reg = PCM1796_OS_64; + else + reg = PCM1796_OS_32; for (i = 0; i < data->dacs; ++i) pcm1796_write_cached(chip, i, 20, reg); } +static void set_pcm1796_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->current_rate = params_rate(params); + update_pcm1796_oversampling(chip); +} + static void update_pcm1796_volume(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; @@ -526,26 +570,44 @@ static void update_pcm1796_mute(struct oxygen *chip) pcm1796_write_cached(chip, i, 18, value); } -static void set_cs2000_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) +static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) { - /* XXX Why is the I2S A MCLK half the actual I2S multich MCLK? */ - static const u8 rate_mclks[] = { - [OXYGEN_RATE_32000] = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_128, - [OXYGEN_RATE_44100] = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_128, - [OXYGEN_RATE_48000] = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_128, - [OXYGEN_RATE_64000] = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256, - [OXYGEN_RATE_88200] = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256, - [OXYGEN_RATE_96000] = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256, - [OXYGEN_RATE_176400] = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256, - [OXYGEN_RATE_192000] = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256, - }; - unsigned int rate_index; + struct xonar_pcm179x *data = chip->model_data; u8 rate_mclk, reg; - rate_index = oxygen_read16(chip, OXYGEN_I2S_MULTICH_FORMAT) - & OXYGEN_I2S_RATE_MASK; - rate_mclk = rate_mclks[rate_index]; + switch (rate) { + /* XXX Why is the I2S A MCLK half the actual I2S MCLK? */ + case 32000: + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + break; + case 44100: + if (data->os_128) + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + else + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_128; + break; + default: /* 48000 */ + if (data->os_128) + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + else + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_128; + break; + case 64000: + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + break; + case 88200: + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + break; + case 96000: + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + break; + case 176400: + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + break; + case 192000: + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + break; + } oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_128) @@ -558,7 +620,7 @@ static void set_cs2000_params(struct oxygen *chip, static void set_st_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { - set_cs2000_params(chip, params); + update_cs2000_rate(chip, params_rate(params)); set_pcm1796_params(chip, params); } @@ -580,6 +642,59 @@ static const struct snd_kcontrol_new alt_switch = { .private_value = GPIO_D2_ALT, }; +static int os_128_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "64x", "128x" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int os_128_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + value->value.enumerated.item[0] = data->os_128; + return 0; +} + +static int os_128_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + int changed; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != data->os_128; + if (changed) { + data->os_128 = value->value.enumerated.item[0]; + if (data->has_cs2000) + update_cs2000_rate(chip, data->current_rate); + oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, + mclk_from_rate(chip, data->current_rate), + OXYGEN_I2S_MCLK_MASK); + update_pcm1796_oversampling(chip); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new os_128_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Oversampling Playback Enum", + .info = os_128_info, + .get = os_128_get, + .put = os_128_put, +}; + static int st_output_switch_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -745,7 +860,20 @@ static int xonar_st_control_filter(struct snd_kcontrol_new *template) static int xonar_d2_mixer_init(struct oxygen *chip) { - return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); + if (err < 0) + return err; + return 0; +} + +static int xonar_hdav_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); } static int xonar_st_mixer_init(struct oxygen *chip) @@ -759,6 +887,9 @@ static int xonar_st_mixer_init(struct oxygen *chip) if (err < 0) return err; } + err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); + if (err < 0) + return err; return 0; } @@ -771,7 +902,7 @@ static const struct oxygen_model model_xonar_d2 = { .cleanup = xonar_d2_cleanup, .suspend = xonar_d2_suspend, .resume = xonar_d2_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, + .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_pcm1796_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -798,11 +929,12 @@ static const struct oxygen_model model_xonar_hdav = { .longname = "Asus Virtuoso 200", .chip = "AV200", .init = xonar_hdav_init, + .mixer_init = xonar_hdav_mixer_init, .cleanup = xonar_hdav_cleanup, .suspend = xonar_hdav_suspend, .resume = xonar_hdav_resume, .pcm_hardware_filter = xonar_hdmi_pcm_hardware_filter, - .get_i2s_mclk = oxygen_default_i2s_mclk, + .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_hdav_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -833,7 +965,7 @@ static const struct oxygen_model model_xonar_st = { .cleanup = xonar_st_cleanup, .suspend = xonar_st_suspend, .resume = xonar_st_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, + .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_st_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, -- cgit v1.2.2 From 4852ad02476ab2bbc874f6f8fda9e677e0f09c87 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:21:21 +0200 Subject: sound: oxygen: add digital filter control Add a control to select between sharp and slow roll-of filter responses of the DACs. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 65 ++++++++++++++++++++++++++++++ sound/pci/oxygen/xonar_cs43xx.c | 82 +++++++++++++++++++++++++++++++++++--- sound/pci/oxygen/xonar_pcm179x.c | 85 ++++++++++++++++++++++++++++++++++++++-- 3 files changed, 223 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index d12fd9efe94e..3ad9eb00aebd 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -352,6 +352,70 @@ static void set_ak5385_params(struct oxygen *chip, value, GPIO_AK5385_DFS_MASK); } +static int rolloff_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "Sharp Roll-off", "Slow Roll-off" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int rolloff_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->ak4396_regs[0][AK4396_CONTROL_2] & AK4396_SLOW) != 0; + return 0; +} + +static int rolloff_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + unsigned int i; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->ak4396_regs[0][AK4396_CONTROL_2]; + if (value->value.enumerated.item[0]) + reg |= AK4396_SLOW; + else + reg &= ~AK4396_SLOW; + changed = reg != data->ak4396_regs[0][AK4396_CONTROL_2]; + if (changed) { + for (i = 0; i < 4; ++i) + ak4396_write(chip, i, AK4396_CONTROL_2, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new rolloff_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Filter Playback Enum", + .info = rolloff_info, + .get = rolloff_get, + .put = rolloff_put, +}; + +static int generic_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); +} + static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static const struct oxygen_model model_generic = { @@ -359,6 +423,7 @@ static const struct oxygen_model model_generic = { .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", .init = generic_init, + .mixer_init = generic_mixer_init, .cleanup = generic_cleanup, .resume = generic_resume, .get_i2s_mclk = oxygen_default_i2s_mclk, diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index a83f827feb34..16c226bfcd2b 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -69,7 +69,7 @@ struct xonar_cs43xx { struct xonar_generic generic; - u8 cs4398_regs[7]; + u8 cs4398_regs[8]; u8 cs4362a_regs[15]; }; @@ -121,12 +121,11 @@ static void cs43xx_registers_init(struct oxygen *chip) cs4398_write(chip, 4, data->cs4398_regs[4]); cs4398_write(chip, 5, data->cs4398_regs[5]); cs4398_write(chip, 6, data->cs4398_regs[6]); - cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | - CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); + cs4398_write(chip, 7, data->cs4398_regs[7]); cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE | CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); - cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); + cs4362a_write(chip, 0x04, data->cs4362a_regs[0x04]); cs4362a_write(chip, 0x05, 0); for (i = 6; i <= 14; ++i) cs4362a_write(chip, i, data->cs4362a_regs[i]); @@ -147,6 +146,9 @@ static void xonar_d1_init(struct oxygen *chip) CS4398_MUTE_B | CS4398_MUTE_A | CS4398_PAMUTE; data->cs4398_regs[5] = 60 * 2; data->cs4398_regs[6] = 60 * 2; + data->cs4398_regs[7] = CS4398_RMP_DN | CS4398_RMP_UP | + CS4398_ZERO_CROSS | CS4398_SOFT_RAMP; + data->cs4362a_regs[4] = CS4362A_RMP_DN | CS4362A_DEM_NONE; data->cs4362a_regs[6] = CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; data->cs4362a_regs[7] = 60 | CS4362A_MUTE; @@ -286,6 +288,68 @@ static const struct snd_kcontrol_new front_panel_switch = { .private_value = GPIO_D1_FRONT_PANEL, }; +static int rolloff_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "Fast Roll-off", "Slow Roll-off" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int rolloff_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_cs43xx *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->cs4398_regs[7] & CS4398_FILT_SEL) != 0; + return 0; +} + +static int rolloff_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_cs43xx *data = chip->model_data; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->cs4398_regs[7]; + if (value->value.enumerated.item[0]) + reg |= CS4398_FILT_SEL; + else + reg &= ~CS4398_FILT_SEL; + changed = reg != data->cs4398_regs[7]; + if (changed) { + cs4398_write(chip, 7, reg); + if (reg & CS4398_FILT_SEL) + reg = data->cs4362a_regs[0x04] | CS4362A_FILT_SEL; + else + reg = data->cs4362a_regs[0x04] & ~CS4362A_FILT_SEL; + cs4362a_write(chip, 0x04, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new rolloff_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Filter Playback Enum", + .info = rolloff_info, + .get = rolloff_get, + .put = rolloff_put, +}; + static void xonar_d1_line_mic_ac97_switch(struct oxygen *chip, unsigned int reg, unsigned int mute) { @@ -309,7 +373,15 @@ static int xonar_d1_control_filter(struct snd_kcontrol_new *template) static int xonar_d1_mixer_init(struct oxygen *chip) { - return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); + if (err < 0) + return err; + return 0; } static const struct oxygen_model model_xonar_d1 = { diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 7f153fb1848d..ba18fb546b4f 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -265,7 +265,8 @@ static void pcm1796_registers_init(struct oxygen *chip) + gain_offset); pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1] + gain_offset); - pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); + pcm1796_write(chip, i, 19, + data->pcm1796_regs[0][19 - PCM1796_REG_BASE]); pcm1796_write(chip, i, 20, data->pcm1796_regs[0][20 - PCM1796_REG_BASE]); pcm1796_write(chip, i, 21, 0); @@ -278,6 +279,8 @@ static void pcm1796_init(struct oxygen *chip) data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = PCM1796_MUTE | PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = + PCM1796_FLT_SHARP | PCM1796_ATS_1; data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = PCM1796_OS_64; pcm1796_registers_init(chip); data->current_rate = 48000; @@ -642,6 +645,67 @@ static const struct snd_kcontrol_new alt_switch = { .private_value = GPIO_D2_ALT, }; +static int rolloff_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "Sharp Roll-off", "Slow Roll-off" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int rolloff_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->pcm1796_regs[0][19 - PCM1796_REG_BASE] & + PCM1796_FLT_MASK) != PCM1796_FLT_SHARP; + return 0; +} + +static int rolloff_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->pcm1796_regs[0][19 - PCM1796_REG_BASE]; + reg &= ~PCM1796_FLT_MASK; + if (!value->value.enumerated.item[0]) + reg |= PCM1796_FLT_SHARP; + else + reg |= PCM1796_FLT_SLOW; + changed = reg != data->pcm1796_regs[0][19 - PCM1796_REG_BASE]; + if (changed) { + for (i = 0; i < data->dacs; ++i) + pcm1796_write(chip, i, 19, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new rolloff_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Filter Playback Enum", + .info = rolloff_info, + .get = rolloff_get, + .put = rolloff_put, +}; + static int os_128_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { static const char *const names[2] = { "64x", "128x" }; @@ -858,6 +922,19 @@ static int xonar_st_control_filter(struct snd_kcontrol_new *template) return 0; } +static int add_pcm1796_controls(struct oxygen *chip) +{ + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); + if (err < 0) + return err; + return 0; +} + static int xonar_d2_mixer_init(struct oxygen *chip) { int err; @@ -865,7 +942,7 @@ static int xonar_d2_mixer_init(struct oxygen *chip) err = snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); if (err < 0) return err; - err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); + err = add_pcm1796_controls(chip); if (err < 0) return err; return 0; @@ -873,7 +950,7 @@ static int xonar_d2_mixer_init(struct oxygen *chip) static int xonar_hdav_mixer_init(struct oxygen *chip) { - return snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); + return add_pcm1796_controls(chip); } static int xonar_st_mixer_init(struct oxygen *chip) @@ -887,7 +964,7 @@ static int xonar_st_mixer_init(struct oxygen *chip) if (err < 0) return err; } - err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); + err = add_pcm1796_controls(chip); if (err < 0) return err; return 0; -- cgit v1.2.2 From 1ff048869eb8e8408856e23b3dc6af094491f837 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:21:51 +0200 Subject: sound: oxygen: add high-pass filter control Add a control that allows disabling the high-pass filter of the WM8785 ADC. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 73 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 71 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 3ad9eb00aebd..acbedebcffd9 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -98,7 +98,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); struct generic_data { u8 ak4396_regs[4][5]; - u16 wm8785_regs[1]; + u16 wm8785_regs[3]; }; static void ak4396_write(struct oxygen *chip, unsigned int codec, @@ -184,6 +184,7 @@ static void wm8785_registers_init(struct oxygen *chip) wm8785_write(chip, WM8785_R7, 0); wm8785_write(chip, WM8785_R0, data->wm8785_regs[0]); + wm8785_write(chip, WM8785_R2, data->wm8785_regs[2]); } static void wm8785_init(struct oxygen *chip) @@ -192,6 +193,7 @@ static void wm8785_init(struct oxygen *chip) data->wm8785_regs[0] = WM8785_MCR_SLAVE | WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; + data->wm8785_regs[2] = WM8785_HPFR | WM8785_HPFL; wm8785_registers_init(chip); snd_component_add(chip->card, "WM8785"); } @@ -334,6 +336,7 @@ static void set_wm8785_params(struct oxygen *chip, if (value != data->wm8785_regs[0]) { wm8785_write(chip, WM8785_R7, 0); wm8785_write(chip, WM8785_R0, value); + wm8785_write(chip, WM8785_R2, data->wm8785_regs[2]); } } @@ -411,11 +414,75 @@ static const struct snd_kcontrol_new rolloff_control = { .put = rolloff_put, }; +static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "None", "High-pass Filter" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->wm8785_regs[WM8785_R2] & WM8785_HPFR) != 0; + return 0; +} + +static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + unsigned int reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8785_regs[WM8785_R2] & ~(WM8785_HPFR | WM8785_HPFL); + if (value->value.enumerated.item[0]) + reg |= WM8785_HPFR | WM8785_HPFL; + changed = reg != data->wm8785_regs[WM8785_R2]; + if (changed) + wm8785_write(chip, WM8785_R2, reg); + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new hpf_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, +}; + static int generic_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); } +static int generic_wm8785_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&hpf_control, chip)); + if (err < 0) + return err; + return 0; +} + static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static const struct oxygen_model model_generic = { @@ -423,7 +490,7 @@ static const struct oxygen_model model_generic = { .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", .init = generic_init, - .mixer_init = generic_mixer_init, + .mixer_init = generic_wm8785_mixer_init, .cleanup = generic_cleanup, .resume = generic_resume, .get_i2s_mclk = oxygen_default_i2s_mclk, @@ -455,6 +522,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, switch (id->driver_data) { case MODEL_MERIDIAN: chip->model.init = meridian_init; + chip->model.mixer_init = generic_mixer_init; chip->model.resume = meridian_resume; chip->model.set_adc_params = set_ak5385_params; chip->model.device_config = PLAYBACK_0_TO_I2S | @@ -470,6 +538,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, break; case MODEL_CLARO_HALO: chip->model.init = claro_halo_init; + chip->model.mixer_init = generic_mixer_init; chip->model.cleanup = claro_cleanup; chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; -- cgit v1.2.2 From 62428f7b8c873d43be8201e66392c3aad82fec93 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Sep 2009 11:22:18 +0200 Subject: sound: oxygen: fix input monitor control names Insert "Playback" into the input monitor control names to prevent alsa-lib from treating these controls as global controls. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 5dfb5fb73381..f375b8a27862 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -805,7 +805,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", + .name = "Analog Input Monitor Playback Switch", .info = snd_ctl_boolean_mono_info, .get = monitor_get, .put = monitor_put, @@ -813,7 +813,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", + .name = "Analog Input Monitor Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = monitor_volume_info, @@ -830,7 +830,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", + .name = "Analog Input Monitor Playback Switch", .info = snd_ctl_boolean_mono_info, .get = monitor_get, .put = monitor_put, @@ -838,7 +838,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", + .name = "Analog Input Monitor Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = monitor_volume_info, @@ -855,7 +855,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", + .name = "Analog Input Monitor Playback Switch", .index = 1, .info = snd_ctl_boolean_mono_info, .get = monitor_get, @@ -864,7 +864,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", + .name = "Analog Input Monitor Playback Volume", .index = 1, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, @@ -882,7 +882,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Monitor Switch", + .name = "Digital Input Monitor Playback Switch", .info = snd_ctl_boolean_mono_info, .get = monitor_get, .put = monitor_put, @@ -890,7 +890,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Monitor Volume", + .name = "Digital Input Monitor Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = monitor_volume_info, -- cgit v1.2.2 From 71623855e20c3febebb5fa60528cde2592678bd5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Sep 2009 13:14:04 +0200 Subject: ALSA: hda - Enable MSI as default Since the recent kernel can handle MSI properly on non-Intel platforms, let's enable MSI as default. If any borken device is found, we can add the quirk entry to the list, which is currently empty. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c9ad182e1b4b..d0effa3563e2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -60,7 +60,7 @@ static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; static int single_cmd; -static int enable_msi; +static int enable_msi = -1; #ifdef CONFIG_SND_HDA_PATCH_LOADER static char *patch[SNDRV_CARDS]; #endif @@ -2300,11 +2300,9 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) } /* - * white-list for enable_msi + * white/black-list for enable_msi */ -static struct snd_pci_quirk msi_white_list[] __devinitdata = { - SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1), - SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1), +static struct snd_pci_quirk msi_black_list[] __devinitdata = { {} }; @@ -2312,10 +2310,12 @@ static void __devinit check_msi(struct azx *chip) { const struct snd_pci_quirk *q; - chip->msi = enable_msi; - if (chip->msi) + if (enable_msi >= 0) { + chip->msi = !!enable_msi; return; - q = snd_pci_quirk_lookup(chip->pci, msi_white_list); + } + chip->msi = 1; /* enable MSI as default */ + q = snd_pci_quirk_lookup(chip->pci, msi_black_list); if (q) { printk(KERN_INFO "hda_intel: msi for device %04x:%04x set to %d\n", -- cgit v1.2.2 From 4fa9c1a5953441e06dbde7b6a655cbf6618e61dd Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Wed, 30 Sep 2009 17:32:27 -0400 Subject: ASoC: DaVinci: McASP FIFO related updates The DMA params for McASP with FIFO has been updated so that it works for various FIFO levels. A member- 'fifo_level' has been added to the DMA params data structure. The fifo_level can be adjusted by the tx[rx]_numevt platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This implementation has been tested for numevt values 1, 2, 4, 8. Signed-off-by: Chaithrika U S Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 2 ++ sound/soc/davinci/davinci-mcasp.c | 17 +++++++---------- sound/soc/davinci/davinci-pcm.c | 21 ++++++++++++++++++--- sound/soc/davinci/davinci-pcm.h | 1 + 4 files changed, 28 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae707048021..2ab809359c08 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -397,6 +397,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, } dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = 0; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a4c978..50ad0519a8fa 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -714,16 +714,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; int word_length; - u8 numevt; + u8 fifo_level; davinci_hw_common_param(dev, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - numevt = dev->txnumevt; + fifo_level = dev->txnumevt; else - numevt = dev->rxnumevt; - - if (!numevt) - numevt = 1; + fifo_level = dev->rxnumevt; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -751,12 +748,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (dev->version == MCASP_VERSION_2) { - dma_params->data_type *= numevt; - dma_params->acnt = 4 * numevt; - } else + if (dev->version == MCASP_VERSION_2 && !fifo_level) + dma_params->acnt = 4; + else dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = fifo_level; davinci_config_channel_size(dev, word_length); return 0; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 359e99ec7244..1152d8ba8970 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -66,38 +66,53 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t dma_pos; dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; + unsigned short src_cidx, dst_cidx; unsigned int data_type; unsigned short acnt; unsigned int count; + unsigned int fifo_level; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; + fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; + if (fifo_level) + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 0; + src_cidx = data_type * fifo_level; + dst_cidx = 0; } else { src = prtd->params->dma_addr; dst = dma_pos; src_bidx = 0; dst_bidx = data_type; + src_cidx = 0; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); - edma_set_src_index(lch, src_bidx, 0); - edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + + edma_set_src_index(lch, src_bidx, src_cidx); + edma_set_dest_index(lch, dst_bidx, dst_cidx); + + if (!fifo_level) + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + else + edma_set_transfer_params(lch, acnt, fifo_level, count, + fifo_level, ABSYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 8746606efc89..c8b0d2baf05a 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -23,6 +23,7 @@ struct davinci_pcm_dma_params { enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; + unsigned int fifo_level; }; -- cgit v1.2.2 From c36b2fc73a6c0e7b185b17d594b38398ce1f7fff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Sep 2009 14:31:38 +0100 Subject: ASoC: Clean up WM8974 PLL configuration Don't use a static for WM8974 PLL factors - we don't support more than one device so it won't happen but no sense in leaving the race condition hanging around. Also, pre_div is a single bit and it's a bit simpler if we move the handling of the factor of 4 in the output into the coefficient setup. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 5104c8aa34f6..f30f86b3bda0 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -330,36 +330,38 @@ static int wm8974_add_widgets(struct snd_soc_codec *codec) } struct pll_ { - unsigned int pre_div:4; /* prescale - 1 */ + unsigned int pre_div:1; unsigned int n:4; unsigned int k; }; -static struct pll_ pll_div; - /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) -static void pll_factors(unsigned int target, unsigned int source) +static void pll_factors(struct pll_ *pll_div, + unsigned int target, unsigned int source) { unsigned long long Kpart; unsigned int K, Ndiv, Nmod; + /* There is a fixed divide by 4 in the output path */ + target *= 4; + Ndiv = target / source; if (Ndiv < 6) { - source >>= 1; - pll_div.pre_div = 1; + source /= 2; + pll_div->pre_div = 1; Ndiv = target / source; } else - pll_div.pre_div = 0; + pll_div->pre_div = 0; if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING "WM8974 N value %u outwith recommended range!\n", Ndiv); - pll_div.n = Ndiv; + pll_div->n = Ndiv; Nmod = target % source; Kpart = FIXED_PLL_SIZE * (long long)Nmod; @@ -374,13 +376,14 @@ static void pll_factors(unsigned int target, unsigned int source) /* Move down to proper range now rounding is done */ K /= 10; - pll_div.k = K; + pll_div->k = K; } static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; + struct pll_ pll_div; u16 reg; if (freq_in == 0 || freq_out == 0) { @@ -394,7 +397,7 @@ static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } - pll_factors(freq_out*4, freq_in); + pll_factors(&pll_div, freq_out, freq_in); wm8974_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n); wm8974_write(codec, WM8974_PLLK1, pll_div.k >> 18); -- cgit v1.2.2 From aa983d9d63c38f596fb87754205da9b7a8d2f6fd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Sep 2009 14:16:11 +0100 Subject: ASoC: Factor out analogue platform data from WM8993 This is also shared with newer CODECs. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 36 +++++++++--------------------------- sound/soc/codecs/wm_hubs.c | 35 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_hubs.h | 5 +++++ 3 files changed, 49 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 6b32a2852603..dac397712147 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1572,33 +1572,15 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, /* Use automatic clock configuration */ snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0); - if (!wm8993->pdata.lineout1_diff) - snd_soc_update_bits(codec, WM8993_LINE_MIXER1, - WM8993_LINEOUT1_MODE, - WM8993_LINEOUT1_MODE); - if (!wm8993->pdata.lineout2_diff) - snd_soc_update_bits(codec, WM8993_LINE_MIXER2, - WM8993_LINEOUT2_MODE, - WM8993_LINEOUT2_MODE); - - if (wm8993->pdata.lineout1fb) - snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, - WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); - - if (wm8993->pdata.lineout2fb) - snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, - WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); - - /* Apply the microphone bias/detection configuration - the - * platform data is directly applicable to the register. */ - snd_soc_update_bits(codec, WM8993_MICBIAS, - WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | - WM8993_MICB1_LVL | WM8993_MICB2_LVL, - wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT | - wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT | - wm8993->pdata.micbias1_lvl | - wm8993->pdata.micbias1_lvl << 1); - + wm_hubs_handle_analogue_pdata(codec, wm8993->pdata.lineout1_diff, + wm8993->pdata.lineout2_diff, + wm8993->pdata.lineout1fb, + wm8993->pdata.lineout2fb, + wm8993->pdata.jd_scthr, + wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_lvl, + wm8993->pdata.micbias2_lvl); + ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e542027eea89..810a563d0ebf 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -738,6 +738,41 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); +int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, + int lineout1_diff, int lineout2_diff, + int lineout1fb, int lineout2fb, + int jd_scthr, int jd_thr, int micbias1_lvl, + int micbias2_lvl) +{ + if (!lineout1_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER1, + WM8993_LINEOUT1_MODE, + WM8993_LINEOUT1_MODE); + if (!lineout2_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER2, + WM8993_LINEOUT2_MODE, + WM8993_LINEOUT2_MODE); + + if (lineout1fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); + + if (lineout2fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + + snd_soc_update_bits(codec, WM8993_MICBIAS, + WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | + WM8993_MICB1_LVL | WM8993_MICB2_LVL, + jd_scthr << WM8993_JD_SCTHR_SHIFT | + jd_thr << WM8993_JD_THR_SHIFT | + micbias1_lvl | + micbias2_lvl << WM8993_MICB2_LVL_SHIFT); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata); + MODULE_DESCRIPTION("Shared support for Wolfson hubs products"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index ec09cb6a2939..36d3fba1de8b 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -20,5 +20,10 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int); +extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, + int lineout1_diff, int lineout2_diff, + int lineout1fb, int lineout2fb, + int jd_scthr, int jd_thr, + int micbias1_lvl, int micbias2_lvl); #endif -- cgit v1.2.2 From bb26276744a80d066681836f4d49c70010b129d6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Oct 2009 07:39:45 +0200 Subject: ASoC: Fix build errors of wm8711.c with SPI Fix a couple of typos and a missing header file inclusion to build wm8711.c properly with CONFIG_SPI_MASTER. Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8711.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index ae083eb92fb7..90ec8c58e2f4 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -632,9 +633,9 @@ static int __init wm8711_modinit(void) } #endif #if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&wm8731_spi_driver); + ret = spi_register_driver(&wm8711_spi_driver); if (ret != 0) { - printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n", + printk(KERN_ERR "Failed to register WM8711 SPI driver: %d\n", ret); } #endif @@ -648,7 +649,7 @@ static void __exit wm8711_exit(void) i2c_del_driver(&wm8711_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8731_spi_driver); + spi_unregister_driver(&wm8711_spi_driver); #endif } module_exit(wm8711_exit); -- cgit v1.2.2 From acd47100914b2896d0699febefd077f85c4dd272 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 1 Oct 2009 00:10:34 +0200 Subject: ALSA: sscape: convert to firmware loader framework The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 8 +- sound/isa/sscape.c | 328 +++++++++++++++++------------------------------------ 2 files changed, 112 insertions(+), 224 deletions(-) (limited to 'sound') diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b90fc164a79c..02fe81ca88fd 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -372,9 +372,9 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape driver" - select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB + select FW_LOADER help Say Y here to include support for Ensoniq SoundScape and Ensoniq OEM soundcards. @@ -382,7 +382,11 @@ config SND_SSCAPE The PCM audio is supported on SoundScape Classic, Elite, PnP and VIVO cards. The supported OEM cards are SPEA Media FX and Reveal SC-600. - The MIDI support is very experimental. + The MIDI support is very experimental and requires binary + firmware files called "scope.cod" and "sndscape.co?" where the + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found + in DOS or Windows driver packages. One has to put the firmware + files into the /lib/firmware directory. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b11c35f6aefe..1ce465cc66a8 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1,5 +1,5 @@ /* - * Low-level ALSA driver for the ENSONIQ SoundScape PnP + * Low-level ALSA driver for the ENSONIQ SoundScape * Copyright (c) by Chris Rankin * * This driver was written in part using information obtained from @@ -25,22 +25,26 @@ #include #include #include +#include #include #include #include #include #include -#include #include #include #include -#include - MODULE_AUTHOR("Chris Rankin"); -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); MODULE_LICENSE("GPL"); +MODULE_FIRMWARE("sndscape.co0"); +MODULE_FIRMWARE("sndscape.co1"); +MODULE_FIRMWARE("sndscape.co2"); +MODULE_FIRMWARE("sndscape.co3"); +MODULE_FIRMWARE("sndscape.co4"); +MODULE_FIRMWARE("scope.cod"); static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; @@ -142,14 +146,12 @@ struct soundscape { struct resource *wss_res; struct snd_wss *chip; struct snd_mpu401 *mpu; - struct snd_hwdep *hw; /* * The MIDI device won't work until we've loaded * its firmware via a hardware-dependent device IOCTL */ spinlock_t fwlock; - int hw_in_use; unsigned long midi_usage; unsigned char midi_vol; }; @@ -167,12 +169,6 @@ static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) return (struct soundscape *) (mpu->private_data); } -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw) -{ - return (struct soundscape *) (hw->private_data); -} - - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -393,12 +389,12 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); - if ((x & 0xfe) == 0xfe) + if (x == 0xfe || x == 0xff) return 1; msleep(10); @@ -420,10 +416,10 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; @@ -438,14 +434,14 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) * Upload a byte-stream into the SoundScape using DMA channel A. */ static int upload_dma_data(struct soundscape *s, - const unsigned char __user *data, + const unsigned char *data, size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; spin_lock_irqsave(&s->lock, flags); @@ -458,7 +454,6 @@ static int upload_dma_data(struct soundscape *s, /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMACFG_REG, 0x50); sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); @@ -468,35 +463,17 @@ static int upload_dma_data(struct soundscape *s, sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); /* - * Upload the user's data (firmware?) to the SoundScape + * Upload the firmware to the SoundScape * board through the DMA channel ... */ while (size != 0) { unsigned long len; - /* - * Apparently, copying to/from userspace can sleep. - * We are therefore forbidden from holding any - * spinlocks while we copy ... - */ - spin_unlock_irqrestore(&s->lock, flags); - - /* - * Remember that the data that we want to DMA - * comes from USERSPACE. We have already verified - * the userspace pointer ... - */ len = min(size, dma.bytes); - len -= __copy_from_user(dma.area, data, len); + memcpy(dma.area, data, len); data += len; size -= len; - /* - * Grab that spinlock again, now that we've - * finished copying! - */ - spin_lock_irqsave(&s->lock, flags); - snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { @@ -512,6 +489,7 @@ static int upload_dma_data(struct soundscape *s, } /* while */ set_host_mode_unsafe(s->io_base); + outb(0x0, s->io_base); /* * Boot the board ... (I think) @@ -537,7 +515,7 @@ _release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); free_dmabuf(&dma); return ret; @@ -547,162 +525,69 @@ _release_dma: * Upload the bootblock(?) into the SoundScape. The only * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. - * - * NOTE: The boot-block data resides in USER-SPACE!!! - * However, we have already verified its memory - * addresses by the time we get here. */ -static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb) +static int sscape_upload_bootblock(struct snd_card *card) { + struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; + const struct firmware *init_fw = NULL; int data = 0; int ret; - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); - - spin_lock_irqsave(&sscape->lock, flags); - if (ret == 0) { - data = host_read_ctrl_unsafe(sscape->io_base, 100); - } - set_midi_mode_unsafe(sscape->io_base); - spin_unlock_irqrestore(&sscape->lock, flags); - - if (ret == 0) { - if (data < 0) { - snd_printk(KERN_ERR "sscape: timeout reading firmware version\n"); - ret = -EAGAIN; - } - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { - ret = -EFAULT; - } + ret = request_firmware(&init_fw, "scope.cod", card->dev); + if (ret < 0) { + snd_printk(KERN_ERR "Error loading scope.cod"); + return ret; } + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); - return ret; -} + release_firmware(init_fw); -/* - * Upload the microcode into the SoundScape. The - * microcode is 64K of data, and if we try to copy - * it into a local variable then we will SMASH THE - * KERNEL'S STACK! We therefore leave it in USER - * SPACE, and save ourselves from copying it at all. - */ -static int sscape_upload_microcode(struct soundscape *sscape, - const struct sscape_microcode __user *mc) -{ - unsigned long flags; - char __user *code; - int err; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now. - * - * NOTE: This buffer is 64K long! That's WAY too big to - * copy into a stack-temporary anyway. - */ - if ( get_user(code, &mc->code) || - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) - return -EFAULT; + spin_lock_irqsave(&sscape->lock, flags); + if (ret == 0) + data = host_read_ctrl_unsafe(sscape->io_base, 100); - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) { - snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n"); - } + if (data & 0x10) + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); - spin_lock_irqsave(&sscape->lock, flags); - set_midi_mode_unsafe(sscape->io_base); spin_unlock_irqrestore(&sscape->lock, flags); - initialise_mpu401(sscape->mpu); + data &= 0xf; + if (ret == 0 && data > 7) { + snd_printk(KERN_ERR "timeout reading firmware version\n"); + ret = -EAGAIN; + } - return err; + return (ret == 0) ? data : ret; } /* - * Hardware-specific device functions, to implement special - * IOCTLs for the SoundScape card. This is how we upload - * the microcode into the card, for example, and so we - * must ensure that no two processes can open this device - * simultaneously, and that we can't open it at all if - * someone is using the MIDI device. + * Upload the microcode into the SoundScape. */ -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) +static int sscape_upload_microcode(struct snd_card *card, int version) { - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; + struct soundscape *sscape = get_card_soundscape(card); + const struct firmware *init_fw = NULL; + char name[14]; int err; - spin_lock_irqsave(&sscape->fwlock, flags); + snprintf(name, sizeof(name), "sndscape.co%d", version); - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { - err = -EBUSY; - } else { - sscape->hw_in_use = 1; - err = 0; + err = request_firmware(&init_fw, name, card->dev); + if (err < 0) { + snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + return err; } + err = upload_dma_data(sscape, init_fw->data, init_fw->size); + if (err == 0) + snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + init_fw->size >> 10); - spin_unlock_irqrestore(&sscape->fwlock, flags); - return err; -} - -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) -{ - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - sscape->hw_in_use = 0; - spin_unlock_irqrestore(&sscape->fwlock, flags); - return 0; -} - -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct soundscape *sscape = get_hwdep_soundscape(hw); - int err = -EBUSY; - - switch (cmd) { - case SND_SSCAPE_LOAD_BOOTB: - { - register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now ... - */ - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) - return -EFAULT; - - /* - * Now check that we can write the firmware version number too... - */ - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) - return -EFAULT; - - err = sscape_upload_bootblock(sscape, bb); - } - break; - - case SND_SSCAPE_LOAD_MCODE: - { - register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg; - - err = sscape_upload_microcode(sscape, mc); - } - break; - - default: - err = -EINVAL; - break; - } /* switch */ + release_firmware(init_fw); return err; } - /* * Mixer control for the SoundScape's MIDI device. */ @@ -920,7 +805,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) spin_lock_irqsave(&sscape->fwlock, flags); - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { + if (sscape->midi_usage == ULONG_MAX) { err = -EBUSY; } else { ++(sscape->midi_usage); @@ -1053,13 +938,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, } } - strcpy(card->driver, "SoundScape"); - strcpy(card->shortname, pcm->name); - snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", - pcm->name, chip->port, chip->irq, - chip->dma1, chip->dma2); - sscape->chip = chip; } @@ -1162,29 +1040,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) return -ENXIO; } - if (sscape->type != SSCAPE_VIVO) { - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "firmware device\n"); - goto _release_dma; - } - strlcpy(sscape->hw->name, "SoundScape M68K", - sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; - } - /* * Tell the on-board devices where their resources are (I think - * I can't be sure without a datasheet ... So many magic values!) @@ -1222,28 +1077,56 @@ static int __devinit create_sscape(int dev, struct snd_card *card) wss_port[dev], irq[dev]); goto _release_dma; } + strcpy(card->driver, "SoundScape"); + strcpy(card->shortname, name); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + name, sscape->chip->port, sscape->chip->irq, + sscape->chip->dma1, sscape->chip->dma2); + #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); - goto _release_dma; - } + err = sscape_upload_bootblock(card); + if (err >= 0) + err = sscape_upload_microcode(card, err); - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + if (err == 0) { + err = create_mpu401(card, MIDI_DEVNUM, port[dev], + mpu_irq[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%lx\n", + port[dev]); + goto _release_dma; + } + + /* + * Enable the master IRQ ... + */ + sscape_write(sscape, GA_INTENA_REG, 0x80); + + /* + * Initialize mixer + */ + spin_lock_irqsave(&sscape->lock, flags); + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_XXX_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_EXTMIDI, 100); + host_write_ctrl_unsafe(sscape->io_base, + 0, 100); + host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100); + + set_midi_mode_unsafe(sscape->io_base); + spin_unlock_irqrestore(&sscape->lock, flags); + } } /* @@ -1301,11 +1184,12 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + snd_card_set_dev(card, pdev); + ret = create_sscape(dev, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, pdev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; @@ -1426,12 +1310,12 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + snd_card_set_dev(card, &pcard->card->dev); ret = create_sscape(idx, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, &pcard->card->dev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; -- cgit v1.2.2 From 140318aaa924ce9664ff59366993228cf1547f1d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Oct 2009 08:40:32 +0200 Subject: ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c Fix the missing argument of snd_soc_dai_set_pll() in neo1973_*.c, which was forgotten in the commit 85488037bb. Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 6 +++--- sound/soc/s3c24xx/neo1973_wm8753.c | 6 +++--- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 6ddd1b3b16b3..26409a9cef9e 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -133,7 +133,7 @@ static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); } /* @@ -183,7 +183,7 @@ static int neo1973_gta02_voice_hw_params( return ret; /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -197,7 +197,7 @@ static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } static struct snd_soc_ops neo1973_gta02_voice_ops = { diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 16009eba9cba..c9b794843a70 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -153,7 +153,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); } /* @@ -203,7 +203,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, return ret; /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -219,7 +219,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { -- cgit v1.2.2 From 88439ac793934a47f47ad285656b63d09f5937c8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 1 Oct 2009 10:32:47 +0300 Subject: ASoC: add support for multiple cards/codecs in debugfs In order to support multiple codecs on the same system in the debugfs the directory hierarchy need to be changed by adding directory per codec under the asoc direcorty: debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg /dapm_pop_time /dapm/{widgets} With the original implementation only the debugfs files are only created for the first codec, other codecs loaded later would fail to create the debugfs files (since they are already exist). Furthermore in this situation any of the codecs has been removed, would cause the debugfs entries to disappear, regardless if the codec, which created them are still loaded (the one which loaded first). Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f5b356f8acfb..e4ab36daf3f7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1254,21 +1254,35 @@ static const struct file_operations codec_reg_fops = { static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { + char codec_root[128]; + + snprintf(codec_root, sizeof(codec_root), + "%s-%s", dev_name(codec->socdev->dev), codec->name); + + codec->debugfs_codec_root = debugfs_create_dir(codec_root, + debugfs_root); + if (!codec->debugfs_codec_root) { + printk(KERN_WARNING + "ASoC: Failed to create codec debugfs directory\n"); + return; + } + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - debugfs_root, codec, - &codec_reg_fops); + codec->debugfs_codec_root, + codec, &codec_reg_fops); if (!codec->debugfs_reg) printk(KERN_WARNING "ASoC: Failed to create codec register debugfs file\n"); codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, - debugfs_root, + codec->debugfs_codec_root, &codec->pop_time); if (!codec->debugfs_pop_time) printk(KERN_WARNING "Failed to create pop time debugfs file\n"); - codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root); + codec->debugfs_dapm = debugfs_create_dir("dapm", + codec->debugfs_codec_root); if (!codec->debugfs_dapm) printk(KERN_WARNING "Failed to create DAPM debugfs directory\n"); @@ -1278,9 +1292,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) { - debugfs_remove_recursive(codec->debugfs_dapm); - debugfs_remove(codec->debugfs_pop_time); - debugfs_remove(codec->debugfs_reg); + debugfs_remove_recursive(codec->debugfs_codec_root); } #else -- cgit v1.2.2 From 0afe5f891501609f31146798fb41784f4adad27c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 2 Oct 2009 09:20:00 +0200 Subject: ALSA: hda - Clean up name string creation in patch_realtek.c Use a common helper to create playback controls. This gives less chance of typos. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 137 ++++++++++++++++++------------------------ 1 file changed, 57 insertions(+), 80 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7810d3dcad83..a751858811e1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4309,6 +4309,20 @@ static int add_control(struct alc_spec *spec, int type, const char *name, return 0; } +static int add_control_with_pfx(struct alc_spec *spec, int type, + const char *pfx, const char *dir, + const char *sfx, unsigned long val) +{ + char name[32]; + snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); + return add_control(spec, type, name, val); +} + +#define add_pb_vol_ctrl(spec, type, pfx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Volume", val) +#define add_pb_sw_ctrl(spec, type, pfx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Switch", val) + #define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17) #define alc880_fixed_pin_idx(nid) ((nid) - 0x14) #define alc880_is_multi_pin(nid) ((nid) >= 0x18) @@ -4362,7 +4376,6 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - char name[32]; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; @@ -4375,26 +4388,26 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i])); if (i == 2) { /* Center/LFE */ - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Center Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "Center", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "LFE Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "LFE", HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "Center Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "Center", HDA_COMPOSE_AMP_VAL(nid, 1, 2, HDA_INPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "LFE Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "LFE", HDA_COMPOSE_AMP_VAL(nid, 2, 2, HDA_INPUT)); if (err < 0) @@ -4406,14 +4419,12 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, pfx = "Speaker"; else pfx = chname[i]; - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) @@ -4429,7 +4440,6 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, { hda_nid_t nid; int err; - char name[32]; if (!pin) return 0; @@ -4443,21 +4453,18 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, spec->multiout.extra_out_nid[0] = nid; /* control HP volume/switch on the output mixer amp */ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin)); - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) return err; } else if (alc880_is_multi_pin(pin)) { /* set manual connection */ /* we have only a switch on HP-out PIN */ - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -4470,16 +4477,13 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname, int idx, hda_nid_t mix_nid) { - char name[32]; int err; - sprintf(name, "%s Playback Volume", ctlname); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", ctlname); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -5972,7 +5976,6 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, { hda_nid_t nid_vol; unsigned long vol_val, sw_val; - char name[32]; int err; if (nid >= 0x0f && nid < 0x11) { @@ -5992,14 +5995,12 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, if (!(*vol_bits & (1 << nid_vol))) { /* first control for the volume widget */ - snprintf(name, sizeof(name), "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, vol_val); if (err < 0) return err; *vol_bits |= (1 << nid_vol); } - snprintf(name, sizeof(name), "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val); + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, sw_val); if (err < 0) return err; return 1; @@ -10936,7 +10937,6 @@ static int alc262_check_volbit(hda_nid_t nid) static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid, const char *pfx, int *vbits) { - char name[32]; unsigned long val; int vbit; @@ -10946,28 +10946,25 @@ static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid, if (*vbits & vbit) /* a volume control for this mixer already there */ return 0; *vbits |= vbit; - snprintf(name, sizeof(name), "%s Playback Volume", pfx); if (vbit == 2) val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT); else val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT); - return add_control(spec, ALC_CTL_WIDGET_VOL, name, val); + return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, val); } static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid, const char *pfx) { - char name[32]; unsigned long val; if (!nid) return 0; - snprintf(name, sizeof(name), "%s Playback Switch", pfx); if (nid == 0x16) val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT); else val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); - return add_control(spec, ALC_CTL_WIDGET_MUTE, name, val); + return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val); } /* add playback controls from the parsed DAC table */ @@ -12305,11 +12302,9 @@ static struct snd_kcontrol_new alc268_test_mixer[] = { static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, const char *ctlname, int idx) { - char name[32]; hda_nid_t dac; int err; - sprintf(name, "%s Playback Volume", ctlname); switch (nid) { case 0x14: case 0x16: @@ -12323,7 +12318,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, } if (spec->multiout.dac_nids[0] != dac && spec->multiout.dac_nids[1] != dac) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, HDA_COMPOSE_AMP_VAL(dac, 3, idx, HDA_OUTPUT)); if (err < 0) @@ -12331,12 +12326,11 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; } - sprintf(name, "%s Playback Switch", ctlname); if (nid != 0x16) - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); else /* mono */ - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT)); if (err < 0) return err; @@ -12366,8 +12360,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->speaker_pins[0]; if (nid == 0x1d) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Speaker Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, "Speaker", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) return err; @@ -12385,8 +12378,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->line_out_pins[1] | cfg->line_out_pins[2]; if (nid == 0x16) { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Mono Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, "Mono", HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -14235,9 +14227,7 @@ static int alc861_auto_fill_dac_nids(struct hda_codec *codec, static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx, hda_nid_t nid, unsigned int chs) { - char name[32]; - snprintf(name, sizeof(name), "%s Playback Switch", pfx); - return add_control(codec->spec, ALC_CTL_WIDGET_MUTE, name, + return add_pb_sw_ctrl(codec->spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); } @@ -15360,7 +15350,6 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - char name[32]; static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"}; hda_nid_t nid_v, nid_s; int i, err; @@ -15377,26 +15366,26 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, if (i == 2) { /* Center/LFE */ - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Center Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "Center", HDA_COMPOSE_AMP_VAL(nid_v, 1, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "LFE Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "LFE", HDA_COMPOSE_AMP_VAL(nid_v, 2, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "Center Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "Center", HDA_COMPOSE_AMP_VAL(nid_s, 1, 2, HDA_INPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "LFE Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "LFE", HDA_COMPOSE_AMP_VAL(nid_s, 2, 2, HDA_INPUT)); if (err < 0) @@ -15411,8 +15400,7 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, pfx = "PCM"; } else pfx = chname[i]; - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) @@ -15420,8 +15408,7 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, if (cfg->line_outs == 1 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) pfx = "Speaker"; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) @@ -15439,7 +15426,6 @@ static int alc861vd_auto_create_extra_out(struct alc_spec *spec, { hda_nid_t nid_v, nid_s; int err; - char name[32]; if (!pin) return 0; @@ -15457,21 +15443,18 @@ static int alc861vd_auto_create_extra_out(struct alc_spec *spec, nid_s = alc861vd_idx_to_mixer_switch( alc880_fixed_pin_idx(pin)); - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) return err; } else if (alc880_is_multi_pin(pin)) { /* set manual connection */ /* we have only a switch on HP-out PIN */ - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -17213,21 +17196,17 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec, return 0; } -static int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx, +static inline int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx, hda_nid_t nid, unsigned int chs) { - char name[32]; - sprintf(name, "%s Playback Volume", pfx); - return add_control(spec, ALC_CTL_WIDGET_VOL, name, + return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); } -static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx, +static inline int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx, hda_nid_t nid, unsigned int chs) { - char name[32]; - sprintf(name, "%s Playback Switch", pfx); - return add_control(spec, ALC_CTL_WIDGET_MUTE, name, + return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT)); } @@ -17305,13 +17284,11 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, return 0; nid = alc662_look_for_dac(codec, pin); if (!nid) { - char name[32]; /* the corresponding DAC is already occupied */ if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)) return 0; /* no way */ /* create a switch only */ - sprintf(name, "%s Playback Switch", pfx); - return add_control(spec, ALC_CTL_WIDGET_MUTE, name, + return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); } -- cgit v1.2.2 From ce3e3737a3361e0c7030f8598eec36bb82050de6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 2 Oct 2009 09:17:37 +0300 Subject: ASoC: Improve the debugfs hierarchy Change the way the debugfs entries are created: If the codec->dev is valid, than use: debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/ if the codec->dev is NULL: debugfs/asoc/{codec->name}/ as root for the debugfs entries. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e4ab36daf3f7..1dec9d21c55e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1256,8 +1256,12 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { char codec_root[128]; - snprintf(codec_root, sizeof(codec_root), - "%s-%s", dev_name(codec->socdev->dev), codec->name); + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); codec->debugfs_codec_root = debugfs_create_dir(codec_root, debugfs_root); -- cgit v1.2.2 From bcde1f8a80d1bdfd43fb498996dfa89666fd7fe3 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 2 Oct 2009 18:41:29 +0200 Subject: ALSA: sscape: remove MIDI instances counting with limit ULONG_MAX There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 101 +++++++++++++++-------------------------------------- 1 file changed, 29 insertions(+), 72 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465cc66a8..c739374af20e 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -147,12 +147,6 @@ struct soundscape { struct snd_wss *chip; struct snd_mpu401 *mpu; - /* - * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL - */ - spinlock_t fwlock; - unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +158,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +168,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR "sscape: Failed to allocate " + "%lu bytes for DMA\n", + size); return NULL; } } @@ -482,7 +473,8 @@ static int upload_dma_data(struct soundscape *s, */ spin_unlock_irqrestore(&s->lock, flags); - snd_printk(KERN_ERR "sscape: DMA upload has timed out\n"); + snd_printk(KERN_ERR + "sscape: DMA upload has timed out\n"); ret = -EAGAIN; goto _release_dma; } @@ -504,10 +496,12 @@ static int upload_dma_data(struct soundscape *s, */ ret = 0; if (!obp_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); + snd_printk(KERN_ERR "sscape: No response " + "from on-board processor after upload\n"); ret = -EAGAIN; } else if (!host_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); + snd_printk(KERN_ERR + "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -536,7 +530,7 @@ static int sscape_upload_bootblock(struct snd_card *card) ret = request_firmware(&init_fw, "scope.cod", card->dev); if (ret < 0) { - snd_printk(KERN_ERR "Error loading scope.cod"); + snd_printk(KERN_ERR "sscape: Error loading scope.cod"); return ret; } ret = upload_dma_data(sscape, init_fw->data, init_fw->size); @@ -554,7 +548,8 @@ static int sscape_upload_bootblock(struct snd_card *card) data &= 0xf; if (ret == 0 && data > 7) { - snd_printk(KERN_ERR "timeout reading firmware version\n"); + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); ret = -EAGAIN; } @@ -575,12 +570,13 @@ static int sscape_upload_microcode(struct snd_card *card, int version) err = request_firmware(&init_fw, name, card->dev); if (err < 0) { - snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d", + version); return err; } err = upload_dma_data(sscape, init_fw->data, init_fw->size); if (err == 0) - snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n", init_fw->size >> 10); release_firmware(init_fw); @@ -750,7 +746,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); if ((inb(wss_io) & 0x80) != 0) goto _done; @@ -774,7 +769,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); /* * SoundScape successfully detected! @@ -794,38 +788,13 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ static int mpu401_open(struct snd_mpu401 * mpu) { - int err; - if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); - err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); + snd_printk(KERN_ERR "sscape: MIDI disabled, " + "please load firmware\n"); + return -ENODEV; } - return err; -} - -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); + return 0; } /* @@ -845,8 +814,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -993,13 +960,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1036,7 +1003,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,8 +1040,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + snd_printk(KERN_ERR + "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); goto _release_dma; } strcpy(card->driver, "SoundScape"); @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "sscape: Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "sscape: device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- cgit v1.2.2 From 1cb0fdebae08f6daaac81197d8dde1746e0a1d96 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 5 Oct 2009 18:18:57 +0200 Subject: ALSA: sscape: force AD1848 codec mode on old Soundscape Old Soundscape cards (pre PnP) work only with AD1848 codecs. If the CS4231 codec is installed it must be used in AD1848 compatible mode. Also, add gameport support and remove an unused mpu field. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index c739374af20e..279be505b72e 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -54,6 +54,7 @@ static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS] __devinitdata; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -79,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -145,7 +149,6 @@ struct soundscape { struct resource *io_res; struct resource *wss_res; struct snd_wss *chip; - struct snd_mpu401 *mpu; unsigned char midi_vol; }; @@ -815,7 +818,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; - sscape->mpu = mpu; initialise_mpu401(mpu); } @@ -836,12 +838,30 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, register struct soundscape *sscape = get_card_soundscape(card); struct snd_wss *chip; int err; + int codec_type = WSS_HW_DETECT; + + switch (sscape->type) { + case MEDIA_FX: + case SSCAPE: + /* + * There are some freak examples of early Soundscape cards + * with CS4231 instead of AD1848/CS4248. Unfortunately, the + * CS4231 works only in CS4248 compatibility mode on + * these cards so force it. + */ + if (sscape->ic_type != IC_OPUS) + codec_type = WSS_HW_AD1848; + break; - if (sscape->type == SSCAPE_VIVO) + case SSCAPE_VIVO: port += 4; + break; + default: + break; + } err = snd_wss_create(card, port, -1, irq, dma1, dma2, - WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); + codec_type, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -927,6 +947,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; + int val; const char *name; /* @@ -1026,6 +1047,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; + if (joystick[dev]) + val |= 8; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT -- cgit v1.2.2 From ed76f652d5329d9dff0ea7f3953b1357ed7f8e6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Oct 2009 18:27:28 +0200 Subject: ALSA: sscape - Remove invalid __devinitdata to module parameters Module parameters shouldn't be marked as __devinitdata since they can be referred via sysfs even after probing. Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be505b72e..579a59b9e470 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -46,15 +46,15 @@ MODULE_FIRMWARE("sndscape.co3"); MODULE_FIRMWARE("sndscape.co4"); MODULE_FIRMWARE("scope.cod"); -static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; -static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; -static long port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; -static long wss_port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; -static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; -static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; -static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; -static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; -static bool joystick[SNDRV_CARDS] __devinitdata; +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); -- cgit v1.2.2 From 1642e3d42a062221e4df18df260d4703d18ca519 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Oct 2009 16:24:26 +0100 Subject: ASoC: Simplify code for DAPM widget updates We don't need to check for an event callback since we also check for an appropriate event flag when applying mux status changes. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 52 ++++++++++++++++++++++++++-------------------------- 1 file changed, 26 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8eaf1b6e7ef2..613764638c7d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1786,19 +1786,19 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); - if (widget->event) { - if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret < 0) - goto out; - } - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); - if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_POST_REG); - } else - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret < 0) + goto out; + } + + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_POST_REG) + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); out: mutex_unlock(&widget->codec->mutex); @@ -1883,19 +1883,19 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); - if (widget->event) { - if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret < 0) - goto out; - } - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); - if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_POST_REG); - } else - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret < 0) + goto out; + } + + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_POST_REG) + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); out: mutex_unlock(&widget->codec->mutex); -- cgit v1.2.2 From 3a65577d2199a7b33c85fd32838020c39da200f3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Oct 2009 17:23:30 +0100 Subject: ASoC: Push DAPM enumeration register change test out Don't assume that enumerations are backed by registers when updating mux power. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 613764638c7d..311467b95afb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1202,8 +1202,8 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mask, - int mux, int val, struct soc_enum *e) + struct snd_kcontrol *kcontrol, int change, + int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -1212,7 +1212,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, widget->id != snd_soc_dapm_value_mux) return -ENODEV; - if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) + if (!change) return 0; /* find dapm widget path assoc with kcontrol */ @@ -1765,7 +1765,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux; + unsigned int val, mux, change; unsigned int mask, bitmask; int ret = 0; @@ -1785,7 +1785,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + dapm_mux_update_power(widget, kcontrol, change, mux, e); if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, @@ -1864,7 +1865,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux; + unsigned int val, mux, change; unsigned int mask; int ret = 0; @@ -1882,7 +1883,8 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + dapm_mux_update_power(widget, kcontrol, change, mux, e); if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- cgit v1.2.2 From d2b247a8be57647d1745535acd58169fbcbe431a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Oct 2009 15:21:04 +0100 Subject: ASoC: Add virtual enumeration support for DAPM muxes Sometimes it is desirable to have a mux which does not reflect any direct register configuration but which will instead only have an effect implicitly (for example, as a result of changing which parts of the device are powered up). Provide a virtual mux for this purpose. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 311467b95afb..d2af872e4771 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1807,6 +1807,54 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); +/** + * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = widget->value; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); + +/** + * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = + (struct soc_enum *)kcontrol->private_value; + int change; + int ret = 0; + + if (ucontrol->value.enumerated.item[0] >= e->max) + return -EINVAL; + + mutex_lock(&widget->codec->mutex); + + change = widget->value != ucontrol->value.enumerated.item[0]; + widget->value = ucontrol->value.enumerated.item[0]; + dapm_mux_update_power(widget, kcontrol, change, widget->value, e); + + mutex_unlock(&widget->codec->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); + /** * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get * callback -- cgit v1.2.2 From 69d2c2ae1dffac5fcd6130e459f250ae035b678f Mon Sep 17 00:00:00 2001 From: Nicolas Ferre Date: Thu, 8 Oct 2009 18:19:49 +0200 Subject: ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file to extend the machine ID checking. Signed-off-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 885ba012557e..e028744c32ce 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static int __init at91sam9g20ek_init(void) struct clk *pllb; int ret; - if (!machine_is_at91sam9g20ek()) + if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) return -ENODEV; /* -- cgit v1.2.2 From 493b67efffc462703d583389aca96f850c18d3b3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 9 Oct 2009 15:55:41 +0300 Subject: ASoC: TPA6130A2 amplifier driver Driver for Texas Instruments TPA6130A2 stereo headphone amplifier. The driver provides playback gain control and also pre-defined DAPM_HP widgets and DAPM routings for power management. The DAPM_HP widget names are: "TPA6130A2 Headphone Left" "TPA6130A2 Headphone Right" From soc machine drivers to use with the tpa6130a2 amplifier, the tpa6130a2_add_controls has to be called, which adds the alsa controls and the DAPM routing needed for the tpa6130a2. After that the machine driver can connect the codec's output with 'TPA6130A2 Left' and 'TPA6130A2 Right': {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"}, {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"}, Internally the left and right channels are powered separately. When none of the channels are needed the amplifier is powered down: hard power: valid GPIO number is passed within platform data soft power: Using the software shutdown of the amplifier Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tpa6130a2.c | 463 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tpa6130a2.h | 62 ++++++ 4 files changed, 531 insertions(+) create mode 100644 sound/soc/codecs/tpa6130a2.c create mode 100644 sound/soc/codecs/tpa6130a2.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3c46f34928ec..fab01c991828 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TPA6130A2 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C @@ -228,3 +229,6 @@ config SND_SOC_WM9713 # Amp config SND_SOC_MAX9877 tristate + +config SND_SOC_TPA6130A2 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fc1c458cbe2f..2f14391b96f9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o +snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o @@ -101,3 +102,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o +obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c new file mode 100644 index 000000000000..1b77c959e2dc --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.c @@ -0,0 +1,463 @@ +/* + * ALSA SoC Texas Instruments TPA6130A2 headset stereo amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tpa6130a2.h" + +struct i2c_client *tpa6130a2_client; + +/* This struct is used to save the context */ +struct tpa6130a2_data { + struct mutex mutex; + unsigned char regs[TPA6130A2_CACHEREGNUM]; + int power_gpio; + unsigned char power_state; +}; + +static int tpa6130a2_i2c_read(int reg) +{ + struct tpa6130a2_data *data; + int val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + /* If powered off, return the cached value */ + if (data->power_state) { + val = i2c_smbus_read_byte_data(tpa6130a2_client, reg); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Read failed\n"); + else + data->regs[reg] = val; + } else { + val = data->regs[reg]; + } + + return val; +} + +static int tpa6130a2_i2c_write(int reg, u8 value) +{ + struct tpa6130a2_data *data; + int val = 0; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (data->power_state) { + val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Write failed\n"); + } + + /* Either powered on or off, we save the context */ + data->regs[reg] = value; + + return val; +} + +static u8 tpa6130a2_read(int reg) +{ + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + return data->regs[reg]; +} + +static void tpa6130a2_initialize(void) +{ + struct tpa6130a2_data *data; + int i; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + for (i = 1; i < TPA6130A2_REG_VERSION; i++) + tpa6130a2_i2c_write(i, data->regs[i]); +} + +void tpa6130a2_power(int power) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + if (power) { + /* Power on */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 1); + data->power_state = 1; + tpa6130a2_initialize(); + } + /* Clear SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } else { + /* set SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + /* Power off */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 0); + data->power_state = 0; + } + } + mutex_unlock(&data->mutex); +} + +static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + + ucontrol->value.integer.value[0] = + (tpa6130a2_read(reg) >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; + + mutex_unlock(&data->mutex); + return 0; +} + +static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + unsigned int val_reg; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (invert) + val = mask - val; + + mutex_lock(&data->mutex); + + val_reg = tpa6130a2_read(reg); + if (((val_reg >> shift) & mask) == val) { + mutex_unlock(&data->mutex); + return 0; + } + + val_reg &= ~(mask << shift); + val_reg |= val << shift; + tpa6130a2_i2c_write(reg, val_reg); + + mutex_unlock(&data->mutex); + + return 1; +} + +/* + * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going + * down in gain. + */ +static const unsigned int tpa6130_tlv[] = { + TLV_DB_RANGE_HEAD(10), + 0, 1, TLV_DB_SCALE_ITEM(-5950, 600, 0), + 2, 3, TLV_DB_SCALE_ITEM(-5000, 250, 0), + 4, 5, TLV_DB_SCALE_ITEM(-4550, 160, 0), + 6, 7, TLV_DB_SCALE_ITEM(-4140, 190, 0), + 8, 9, TLV_DB_SCALE_ITEM(-3650, 120, 0), + 10, 11, TLV_DB_SCALE_ITEM(-3330, 160, 0), + 12, 13, TLV_DB_SCALE_ITEM(-3040, 180, 0), + 14, 20, TLV_DB_SCALE_ITEM(-2710, 110, 0), + 21, 37, TLV_DB_SCALE_ITEM(-1960, 74, 0), + 38, 63, TLV_DB_SCALE_ITEM(-720, 45, 0), +}; + +static const struct snd_kcontrol_new tpa6130a2_controls[] = { + SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, + tpa6130a2_get_reg, tpa6130a2_set_reg, + tpa6130_tlv), +}; + +/* + * Enable or disable channel (left or right) + * The bit number for mute and amplifier are the same per channel: + * bit 6: Right channel + * bit 7: Left channel + * in both registers. + */ +static void tpa6130a2_channel_enable(u8 channel, int enable) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (enable) { + /* Enable channel */ + /* Enable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + + /* Unmute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + } else { + /* Disable channel */ + /* Mute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + + /* Disable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } +} + +static int tpa6130a2_left_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 0); + break; + } + return 0; +} + +static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 0); + break; + } + return 0; +} + +static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_power(1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_power(0); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { + SND_SOC_DAPM_PGA_E("TPA6130A2 Left", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_left_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("TPA6130A2 Right", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_right_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("TPA6130A2 Enable", SND_SOC_NOPM, + 0, 0, tpa6130a2_supply_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + /* Outputs */ + SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL), + SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Left"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Right"}, + + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Enable"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Enable"}, +}; + +int tpa6130a2_add_controls(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, + ARRAY_SIZE(tpa6130a2_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return snd_soc_add_controls(codec, tpa6130a2_controls, + ARRAY_SIZE(tpa6130a2_controls)); + +} +EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); + +static int tpa6130a2_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tpa6130a2_data *data; + struct tpa6130a2_platform_data *pdata; + int ret; + + dev = &client->dev; + + if (client->dev.platform_data == NULL) { + dev_err(dev, "Platform data not set\n"); + dump_stack(); + return -ENODEV; + } + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (data == NULL) { + dev_err(dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + tpa6130a2_client = client; + + i2c_set_clientdata(tpa6130a2_client, data); + + pdata = (struct tpa6130a2_platform_data *)client->dev.platform_data; + data->power_gpio = pdata->power_gpio; + + mutex_init(&data->mutex); + + /* Set default register values */ + data->regs[TPA6130A2_REG_CONTROL] = TPA6130A2_SWS; + data->regs[TPA6130A2_REG_VOL_MUTE] = TPA6130A2_MUTE_R | + TPA6130A2_MUTE_L; + + if (data->power_gpio >= 0) { + ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + if (ret < 0) { + dev_err(dev, "Failed to request power GPIO (%d)\n", + data->power_gpio); + goto fail; + } + gpio_direction_output(data->power_gpio, 0); + } else { + data->power_state = 1; + tpa6130a2_initialize(); + } + + tpa6130a2_power(1); + + /* Read version */ + ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) & + TPA6130A2_VERSION_MASK; + if ((ret != 1) && (ret != 2)) + dev_warn(dev, "UNTESTED version detected (%d)\n", ret); + + /* Disable the chip */ + tpa6130a2_power(0); + + return 0; +fail: + kfree(data); + i2c_set_clientdata(tpa6130a2_client, NULL); + tpa6130a2_client = 0; + + return ret; +} + +static int tpa6130a2_remove(struct i2c_client *client) +{ + struct tpa6130a2_data *data = i2c_get_clientdata(client); + + tpa6130a2_power(0); + + if (data->power_gpio >= 0) + gpio_free(data->power_gpio); + kfree(data); + tpa6130a2_client = 0; + + return 0; +} + +static const struct i2c_device_id tpa6130a2_id[] = { + { "tpa6130a2", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); + +static struct i2c_driver tpa6130a2_i2c_driver = { + .driver = { + .name = "tpa6130a2", + .owner = THIS_MODULE, + }, + .probe = tpa6130a2_probe, + .remove = __devexit_p(tpa6130a2_remove), + .id_table = tpa6130a2_id, +}; + +static int __init tpa6130a2_init(void) +{ + return i2c_add_driver(&tpa6130a2_i2c_driver); +} + +static void __exit tpa6130a2_exit(void) +{ + i2c_del_driver(&tpa6130a2_i2c_driver); +} + +MODULE_AUTHOR("Peter Ujfalusi"); +MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); +MODULE_LICENSE("GPL"); + +module_init(tpa6130a2_init); +module_exit(tpa6130a2_exit); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h new file mode 100644 index 000000000000..6a794f16cee9 --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.h @@ -0,0 +1,62 @@ +/* + * ALSA SoC TPA6130A2 amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TPA6130A2_H__ +#define __TPA6130A2_H__ + +/* Register addresses */ +#define TPA6130A2_REG_CONTROL 0x01 +#define TPA6130A2_REG_VOL_MUTE 0x02 +#define TPA6130A2_REG_OUT_IMPEDANCE 0x03 +#define TPA6130A2_REG_VERSION 0x04 + +#define TPA6130A2_CACHEREGNUM (TPA6130A2_REG_VERSION + 1) + +/* Register bits */ +/* TPA6130A2_REG_CONTROL (0x01) */ +#define TPA6130A2_SWS (0x01 << 0) +#define TPA6130A2_TERMAL (0x01 << 1) +#define TPA6130A2_MODE(x) (x << 4) +#define TPA6130A2_MODE_STEREO (0x00) +#define TPA6130A2_MODE_DUAL_MONO (0x01) +#define TPA6130A2_MODE_BRIDGE (0x02) +#define TPA6130A2_MODE_MASK (0x03) +#define TPA6130A2_HP_EN_R (0x01 << 6) +#define TPA6130A2_HP_EN_L (0x01 << 7) + +/* TPA6130A2_REG_VOL_MUTE (0x02) */ +#define TPA6130A2_VOLUME(x) ((x & 0x3f) << 0) +#define TPA6130A2_MUTE_R (0x01 << 6) +#define TPA6130A2_MUTE_L (0x01 << 7) + +/* TPA6130A2_REG_OUT_IMPEDANCE (0x03) */ +#define TPA6130A2_HIZ_R (0x01 << 0) +#define TPA6130A2_HIZ_L (0x01 << 1) + +/* TPA6130A2_REG_VERSION (0x04) */ +#define TPA6130A2_VERSION_MASK (0x0f) + +extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); +extern void tpa6130a2_power(int power); + +#endif /* __TPA6130A2_H__ */ -- cgit v1.2.2 From ebab1b1d07266ab8ca9f65065e68b02f05504c4e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 9 Oct 2009 19:13:47 +0100 Subject: ASoC: Minor fixups to tpa6130a2 driver - Staticise ttpa6130a2_client. - Remove unneeded cast from void. - Use explict NULL rather than 0. Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 1b77c959e2dc..0a6e7b4ace60 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -32,7 +32,7 @@ #include "tpa6130a2.h" -struct i2c_client *tpa6130a2_client; +static struct i2c_client *tpa6130a2_client; /* This struct is used to save the context */ struct tpa6130a2_data { @@ -372,7 +372,7 @@ static int tpa6130a2_probe(struct i2c_client *client, i2c_set_clientdata(tpa6130a2_client, data); - pdata = (struct tpa6130a2_platform_data *)client->dev.platform_data; + pdata = client->dev.platform_data; data->power_gpio = pdata->power_gpio; mutex_init(&data->mutex); @@ -410,7 +410,7 @@ static int tpa6130a2_probe(struct i2c_client *client, fail: kfree(data); i2c_set_clientdata(tpa6130a2_client, NULL); - tpa6130a2_client = 0; + tpa6130a2_client = NULL; return ret; } @@ -424,7 +424,7 @@ static int tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); kfree(data); - tpa6130a2_client = 0; + tpa6130a2_client = NULL; return 0; } -- cgit v1.2.2 From 6fcfa3959a5f5ecb7c333f54f401575d94eb8172 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 10 Oct 2009 10:27:58 +0200 Subject: ALSA: sscape: coding style fixes Fix coding style errors in the driver. Also, add missing argument for CMD_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 169 ++++++++++++++++++++++++++--------------------------- 1 file changed, 83 insertions(+), 86 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 579a59b9e470..e2d5d2d3ed96 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -109,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -166,10 +166,12 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { snd_printk(KERN_ERR "sscape: Failed to allocate " "%lu bytes for DMA\n", @@ -190,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -206,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -219,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -248,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -292,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -311,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -319,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -329,9 +332,10 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) * The AD1845 detection fails if we *don't* do this, so I * think that this is a good idea ... */ -static inline void activate_ad1845_unsafe(unsigned io_base) +static void activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -350,24 +354,27 @@ static void soundscape_free(struct snd_card *c) * Tell the SoundScape to begin a DMA tranfer using the given channel. * All locking issues are left to the caller. */ -static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) +static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -427,13 +434,13 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) /* * Upload a byte-stream into the SoundScape using DMA channel A. */ -static int upload_dma_data(struct soundscape *s, - const unsigned char *data, - size_t size) +static int upload_dma_data(struct soundscape *s, const unsigned char *data, + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; @@ -443,18 +450,21 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* * Upload the firmware to the SoundScape @@ -472,7 +482,7 @@ static int upload_dma_data(struct soundscape *s, sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); @@ -489,7 +499,8 @@ static int upload_dma_data(struct soundscape *s, /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -591,7 +602,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -601,7 +612,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -615,16 +626,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -637,15 +650,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -738,7 +752,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type == SSCAPE_VIVO) wss_io += 4; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ @@ -762,7 +776,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if ((inb(wss_io) & 0x80) != 0) s->type = MEDIA_FX; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { @@ -778,7 +792,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -789,7 +803,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { snd_printk(KERN_ERR "sscape: MIDI disabled, " @@ -803,18 +817,18 @@ static int mpu401_open(struct snd_mpu401 * mpu) /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; @@ -866,19 +880,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -928,7 +929,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1034,7 +1035,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1055,6 +1055,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); + /* + * Enable the master IRQ ... + */ + sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); spin_unlock_irqrestore(&sscape->lock, flags); @@ -1093,11 +1097,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) goto _release_dma; } - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - /* * Initialize mixer */ @@ -1155,7 +1154,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1183,7 +1183,8 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } @@ -1236,20 +1237,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; - /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - */ - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { @@ -1298,7 +1294,8 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- cgit v1.2.2 From abd134db940ddccaf6a61d88cf0841a62b917ab3 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 10 Oct 2009 10:25:39 +0200 Subject: ALSA: wss: convert CS4231 mixer to dB scale Convert CS4231 mixer to dB scale after AD1848 mixer. Also, add missing microphone boost control for the AD1848 and correct wrong bits for loopback volume on the AD1848. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 43 ++++++++++++++++++++++++++----------------- 1 file changed, 26 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5d2ba1b749ab..754a2089c650 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2198,6 +2198,7 @@ EXPORT_SYMBOL(snd_wss_put_double); static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); static struct snd_kcontrol_new snd_ad1848_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, @@ -2224,38 +2225,45 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, +WSS_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0, +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, db_scale_6bit), }; static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE_TLV("Mono Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), WSS_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -2267,15 +2275,16 @@ WSS_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE("Loopback Capture Volume", 0, - CS4231_LOOPBACK, 2, 63, 1) +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, + db_scale_6bit), }; static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Playback Volume", 0, -- cgit v1.2.2 From b6153e1175a46db9dde17d12609adba7d72330b9 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:23 +0800 Subject: ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro IS_VT17*_VENDORID macros are used nowhere, so clean them up. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index ee89db90c9b6..9dfe1b55970c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -76,14 +76,6 @@ #define VT1702_HP_NID 0x17 #define VT1702_DIGOUT_NID 0x11 -#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) -#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) -#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) -#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) -#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) -#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397) -#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398) - enum VIA_HDA_CODEC { UNKNOWN = -1, VT1708, -- cgit v1.2.2 From 744ff5f487925223beb6e21460c8cec468b54ab4 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:26 +0800 Subject: ALSA: HDA VIA: Change get_codec_type argument to hda_codec type Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9dfe1b55970c..e7d739f12247 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -88,8 +88,9 @@ enum VIA_HDA_CODEC { CODEC_TYPES, }; -static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { + u32 vendor_id = codec->vendor_id; u16 ven_id = vendor_id >> 16; u16 dev_id = vendor_id & 0xffff; enum VIA_HDA_CODEC codec_type; @@ -141,7 +142,7 @@ static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { if (size < 4 * sizeof(unsigned int)) return -ENOMEM; @@ -163,7 +164,7 @@ static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; -- cgit v1.2.2 From 518bf3ba753ad93644e7c6cf95c043c918d9429b Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:29 +0800 Subject: ALSA: HDA VIA: Add VT1708B-CE codec support. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e7d739f12247..4d9ffd6f190b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -84,6 +84,7 @@ enum VIA_HDA_CODEC { VT1708B_8CH, VT1708B_4CH, VT1708S, + VT1708BCE, VT1702, CODEC_TYPES, }; @@ -104,9 +105,11 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) codec_type = VT1709_10CH; else if (dev_id >= 0xe714 && dev_id <= 0xe717) codec_type = VT1709_6CH; - else if (dev_id >= 0xe720 && dev_id <= 0xe723) + else if (dev_id >= 0xe720 && dev_id <= 0xe723) { codec_type = VT1708B_8CH; - else if (dev_id >= 0xe724 && dev_id <= 0xe727) + if (snd_hda_param_read(codec, 0x16, AC_PAR_CONNLIST_LEN) == 0x7) + codec_type = VT1708BCE; + } else if (dev_id >= 0xe724 && dev_id <= 0xe727) codec_type = VT1708B_4CH; else if ((dev_id & 0xfff) == 0x397 && (dev_id >> 12) < 8) @@ -224,6 +227,8 @@ struct via_spec { const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + enum VIA_HDA_CODEC codec_type; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -979,6 +984,10 @@ static int via_init(struct hda_codec *codec) for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + spec->codec_type = get_codec_type(codec); + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost + same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) { @@ -2369,12 +2378,14 @@ static struct hda_amp_list vt1708B_loopbacks[] = { { } /* end */ }; #endif - +static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { struct via_spec *spec; int err; + if (get_codec_type(codec) == VT1708BCE) + return patch_vt1708S(codec); /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2906,6 +2917,16 @@ static int patch_vt1708S(struct hda_codec *codec) spec->loopback.amplist = vt1708S_loopbacks; #endif + /* correct names for VT1708BCE */ + if (get_codec_type(codec) == VT1708BCE) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + spec->stream_name_analog = "VT1708BCE Analog"; + spec->stream_name_digital = "VT1708BCE Digital"; + } return 0; } -- cgit v1.2.2 From c2c02ea326d3683f551120e74a297b354a223357 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:32 +0800 Subject: ALSA: HDA VIA: Limit VT1702 AA-Path max volume according to customer request, VT1702 AA-Path max volume (12 dB) is too high, so limit to 0 dB. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4d9ffd6f190b..e62698984287 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3166,6 +3166,12 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; + /* limit AA path volume to 0 dB */ + snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; -- cgit v1.2.2 From f5271101faf1655d862849f42518c2a88ef394fb Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:35 +0800 Subject: ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type Enter low power state if AA-Path volume is muted. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 240 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 239 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e62698984287..d6bee620ced6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -128,6 +128,7 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, + VIA_CTL_WIDGET_ANALOG_MUTE, }; enum { @@ -177,9 +178,34 @@ static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, return 0; } +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); +static void set_jack_power_state(struct hda_codec *codec); + +static int analog_input_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + set_jack_power_state(codec); + analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + return change; +} + +/* modify .put = snd_hda_mixer_amp_switch_put */ +#define ANALOG_INPUT_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = analog_input_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + ANALOG_INPUT_MUTE, }; @@ -303,7 +329,7 @@ static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, if (err < 0) return err; sprintf(name, "%s Playback Switch", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -362,6 +388,131 @@ static void via_auto_init_analog_input(struct hda_codec *codec) } } + +static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int *affected_parm) +{ + unsigned parm; + unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); + unsigned no_presence = (def_conf & AC_DEFCFG_MISC) + >> AC_DEFCFG_MISC_SHIFT + & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ + unsigned present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31; + + if ((no_presence || present) && get_defcfg_connect(def_conf) + != AC_JACK_PORT_NONE) { + *affected_parm = AC_PWRST_D0; /* if it's connected */ + parm = AC_PWRST_D0; + } else + parm = AC_PWRST_D3; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + +static void set_jack_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + + if (spec->codec_type == VT1702) { + imux_is_smixer = snd_hda_codec_read( + codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x16, &parm); + set_pin_power_state(codec, 0x17, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, + parm); + } else if (spec->codec_type == VT1708B_8CH + || spec->codec_type == VT1708B_4CH + || spec->codec_type == VT1708S) { + /* SW0 (17h) = stereo mixer */ + int is_8ch = spec->codec_type != VT1708B_4CH; + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } + } +} + /* * input MUX handling */ @@ -504,6 +655,93 @@ static struct snd_kcontrol_new vt1708_capture_mixer[] = { }, { } /* end */ }; + +/* check AA path's mute statue */ +static int is_aa_path_mute(struct hda_codec *codec) +{ + int mute = 1; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + struct via_spec *spec = codec->spec; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + case VT1702: + nid_mixer = 0x1a; + start_idx = 1; + end_idx = 3; + break; + default: + return 0; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + unsigned int con_list = snd_hda_codec_read( + codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); + int shift = 8 * (i % 4); + hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; + unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); + if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { + /* check mute status while the pin is connected */ + int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, + HDA_INPUT, i) >> 7; + int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, + HDA_INPUT, i) >> 7; + if (!mute_l || !mute_r) { + mute = 0; + break; + } + } + } + return mute; +} + +/* enter/exit analog low-current mode */ +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) +{ + struct via_spec *spec = codec->spec; + static int saved_stream_idle = 1; /* saved stream idle status */ + int enable = is_aa_path_mute(codec); + unsigned int verb = 0; + unsigned int parm = 0; + + if (stream_idle == -1) /* stream status did not change */ + enable = enable && saved_stream_idle; + else { + enable = enable && stream_idle; + saved_stream_idle = stream_idle; + } + + /* decide low current mode's verb & parameter */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + verb = 0xf70; + parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ + break; + case VT1708S: + verb = 0xf73; + parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ + break; + case VT1702: + verb = 0xf73; + parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ + break; + default: + return; /* other codecs are not supported */ + } + /* send verb */ + snd_hda_codec_write(codec, codec->afg, 0, verb, parm); +} + /* * generic initialization of ADC, input mixers and output mixers */ -- cgit v1.2.2 From 173143791068ac9f155c378a591d0b3d6c4a45ca Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:37 +0800 Subject: ALSA: HDA VIA: Add low current mode for power saving. For VT1708B, VT1708S and VT1702, enter low current mode if no analog stream is opened and all aa path mute. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 41 +++++++++++++++++++++++++++++++++++------ 1 file changed, 35 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d6bee620ced6..7ace0fca933d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -783,6 +783,10 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -1089,6 +1093,11 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* init power states */ + set_jack_power_state(codec); + analog_low_current_mode(codec, 1); + via_free_kctls(codec); /* no longer needed */ return 0; } @@ -2312,6 +2321,17 @@ static struct hda_verb vt1708B_uniwill_init_verbs[] = { { } }; +static int via_pcm_open_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); + return 0; +} + static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -2320,7 +2340,8 @@ static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2342,8 +2363,10 @@ static struct hda_pcm_stream vt1708B_pcm_analog_capture = { .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2800,7 +2823,8 @@ static struct hda_pcm_stream vt1708S_pcm_analog_playback = { .ops = { .open = via_playback_pcm_open, .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .cleanup = via_playback_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2810,8 +2834,10 @@ static struct hda_pcm_stream vt1708S_pcm_analog_capture = { .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3236,7 +3262,8 @@ static struct hda_pcm_stream vt1702_pcm_analog_playback = { .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3246,8 +3273,10 @@ static struct hda_pcm_stream vt1702_pcm_analog_capture = { .channels_max = 2, .nid = 0x12, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; -- cgit v1.2.2 From 9510e8dd9cb4469d146953270364af6dd86a39be Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:39 +0800 Subject: ALSA: HDA VIA: Remove unused argument of via_new_analog_input Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 7ace0fca933d..0da57db3a691 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -317,8 +317,8 @@ static void via_free_kctls(struct hda_codec *codec) } /* create input playback/capture controls for the given pin */ -static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, - const char *ctlname, int idx, int mix_nid) +static int via_new_analog_input(struct via_spec *spec, const char *ctlname, + int idx, int mix_nid) { char name[32]; int err; @@ -1480,8 +1480,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x17); + err = via_new_analog_input(spec, labels[i], idx, 0x17); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2014,8 +2013,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x18); + err = via_new_analog_input(spec, labels[i], idx, 0x18); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2576,8 +2574,7 @@ static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3048,8 +3045,7 @@ static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3402,8 +3398,7 @@ static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec, idx = 3; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], - labels[i], idx, 0x1A); + err = via_new_analog_input(spec, labels[i], idx, 0x1A); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; -- cgit v1.2.2 From 0713efebfa1a1878feeeb17cbadc3d2d2c9e9ed2 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:43 +0800 Subject: ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls For VT1708S and VT1702, deactivate "Headphone Playback Volume" and "Headphone Playback Mute" control if "Independent HP" mode is OFF. and rename VT1702 "Independent HP" text. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 38 ++++++++++++++++++++++++++++++++++---- 1 file changed, 34 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0da57db3a691..9e8dd57e8d5c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -572,6 +572,18 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, return 0; } +static void activate_ctl(struct hda_codec *codec, const char *name, int active) +{ + struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); + if (ctl) { + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access |= active + ? 0 : SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(codec->bus->card, + SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id); + } +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -620,6 +632,14 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, spec->multiout.hp_nid, 0, 0, 0); + /* update HP volume/swtich active state */ + if (spec->codec_type == VT1708S + || spec->codec_type == VT1702) { + activate_ctl(codec, "Headphone Playback Volume", + spec->hp_independent_mode); + activate_ctl(codec, "Headphone Playback Switch", + spec->hp_independent_mode); + } return 0; } @@ -3342,11 +3362,11 @@ static int vt1702_auto_create_line_out_ctls(struct via_spec *spec, static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { - int err; - + int err, i; + struct hda_input_mux *imux; + static const char *texts[] = { "ON", "OFF", NULL}; if (!pin) return 0; - spec->multiout.hp_nid = 0x1D; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -3361,8 +3381,18 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) if (err < 0) return err; - create_hp_imux(spec); + imux = &spec->private_imux[1]; + /* for hp mode select */ + i = 0; + while (texts[i] != NULL) { + imux->items[imux->num_items].label = texts[i]; + imux->items[imux->num_items].index = i; + imux->num_items++; + i++; + } + + spec->hp_mux = &spec->private_imux[1]; return 0; } -- cgit v1.2.2 From cdc1784d49258198df600fbc1d37c07d7eee5ed6 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:47 +0800 Subject: ALSA: HDA VIA: Rewrite via_independent_hp_put Use hp_independent_mode_index to store hp index, and simplify function via_independent_hp_put with it. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 85 +++++++++++++++++++++++++---------------------- 1 file changed, 46 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9e8dd57e8d5c..e3bd5261986e 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -252,6 +252,7 @@ struct via_spec { /* HP mode source */ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; enum VIA_HDA_CODEC codec_type; @@ -584,6 +585,36 @@ static void activate_ctl(struct hda_codec *codec, const char *name, int active) } } +static int update_side_mute_status(struct hda_codec *codec) +{ + /* mute side channel */ + struct via_spec *spec = codec->spec; + unsigned int parm = spec->hp_independent_mode + ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + hda_nid_t sw3; + + switch (spec->codec_type) { + case VT1708: + sw3 = 0x1b; + break; + case VT1709_10CH: + sw3 = 0x29; + break; + case VT1708B_8CH: + case VT1708S: + sw3 = 0x27; + break; + default: + sw3 = 0; + break; + } + + if (sw3) + snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, + parm); + return 0; +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -591,47 +622,18 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; hda_nid_t nid = spec->autocfg.hp_pins[0]; unsigned int pinsel = ucontrol->value.enumerated.item[0]; - unsigned int con_nid = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - - if (con_nid == spec->multiout.hp_nid) { - if (pinsel == 0) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } else if (pinsel == 1) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } - } else { - if (pinsel == 0) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } else if (pinsel == 1) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } - } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - pinsel); + /* Get Independent Mode index of headphone pin widget */ + spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel + ? 1 : 0; - if (spec->multiout.hp_nid && - spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) - snd_hda_codec_setup_stream(codec, - spec->multiout.hp_nid, - 0, 0, 0); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + + if (spec->multiout.hp_nid && spec->multiout.hp_nid + != spec->multiout.dac_nids[HDA_FRONT]) + snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, + 0, 0, 0); + update_side_mute_status(codec); /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S || spec->codec_type == VT1702) { @@ -1447,6 +1449,7 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) return 0; spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -1982,6 +1985,7 @@ static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) spec->multiout.hp_nid = VT1709_HP_DAC_NID; else if (spec->multiout.num_dacs == 3) /* 6 channels */ spec->multiout.hp_nid = 0; + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -2541,6 +2545,7 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) return 0; spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3011,6 +3016,7 @@ static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) return 0; spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3368,6 +3374,7 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) if (!pin) return 0; spec->multiout.hp_nid = 0x1D; + spec->hp_independent_mode_index = 0; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", -- cgit v1.2.2 From 1564b2878f5cf160f60af99d4dbca1dd7809ee8a Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:52 +0800 Subject: ALSA: HDA VIA: Add smart5.1 function. Smart 5.1 is for 3-jacks model, to reuse input pins as outputs. While off, they act as "line out" / "line in" / "mic in". While on, they acts as "line out" / "back left/right" / "center/lfe". Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 177 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 173 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e3bd5261986e..26ee1c3a4d16 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -211,7 +211,7 @@ static struct snd_kcontrol_new vt1708_control_templates[] = { struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[3]; + struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -253,6 +253,7 @@ struct via_spec { const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; enum VIA_HDA_CODEC codec_type; @@ -390,6 +391,8 @@ static void via_auto_init_analog_input(struct hda_codec *codec) } } +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -400,9 +403,10 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ unsigned present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) >> 31; - - if ((no_presence || present) && get_defcfg_connect(def_conf) - != AC_JACK_PORT_NONE) { + struct via_spec *spec = codec->spec; + if ((spec->smart51_enabled && is_smart51_pins(spec, nid)) + || ((no_presence || present) + && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { *affected_parm = AC_PWRST_D0; /* if it's connected */ parm = AC_PWRST_D0; } else @@ -657,6 +661,167 @@ static struct snd_kcontrol_new via_hp_mixer[] = { { } /* end */ }; +static void notify_aa_path_ctls(struct hda_codec *codec) +{ + int i; + struct snd_ctl_elem_id id; + const char *labels[] = {"Mic", "Front Mic", "Line"}; + + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + for (i = 0; i < ARRAY_SIZE(labels); i++) { + sprintf(id.name, "%s Playback Volume", labels[i]); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +static void mute_aa_path(struct hda_codec *codec, int mute) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708: + nid_mixer = 0x17; + start_idx = 2; + end_idx = 4; + break; + case VT1709_10CH: + case VT1709_6CH: + nid_mixer = 0x18; + start_idx = 2; + end_idx = 4; + break; + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + default: + return; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE; + snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i, + HDA_AMP_MUTE, val); + } +} +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) +{ + int res = 0; + int index; + for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) { + if (pin == spec->autocfg.input_pins[index]) { + res = 1; + break; + } + } + return res; +} + +static int via_smart51_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int via_smart51_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int on = 1; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + int ctl = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0); + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* ignore FMic for independent HP */ + if (ctl & AC_PINCTL_IN_EN + && !(ctl & AC_PINCTL_OUT_EN)) + on = 0; + } + } + *ucontrol->value.integer.value = on; + return 0; +} + +static int via_smart51_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int out_in = *ucontrol->value.integer.value + ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* don't retask FMic for independent HP */ + if (nid) { + unsigned int parm = snd_hda_codec_read( + codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + parm |= out_in; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + parm); + if (out_in == AC_PINCTL_OUT_EN) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + } + if (i == AUTO_PIN_FRONT_MIC) { + if (spec->codec_type == VT1708S) { + /* input = index 1 (AOW3) */ + snd_hda_codec_write( + codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, 1); + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, + 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); + } + } + } + spec->smart51_enabled = *ucontrol->value.integer.value; + set_jack_power_state(codec); + return 1; +} + +static struct snd_kcontrol_new via_smart51_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Smart 5.1", + .count = 1, + .info = via_smart51_info, + .get = via_smart51_get, + .put = via_smart51_put, + }, + {} /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1587,6 +1752,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2087,6 +2253,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2649,6 +2816,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -3142,6 +3310,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } -- cgit v1.2.2 From a80e6e3c8c21ca50837e2e42fa438a4ff4a9788e Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:07:55 +0800 Subject: ALSA: HDA VIA: When changing input source, update power state. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 26ee1c3a4d16..c5e99944990a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -549,6 +549,14 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, if (!spec->mux_nids[adc_idx]) return -EINVAL; + /* switch to D0 beofre change index */ + if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) + snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + /* update jack power state */ + set_jack_power_state(codec); + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); -- cgit v1.2.2 From a34df19a658170fb7125e8017ee46ba54b1ad495 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:01 +0800 Subject: ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 38 +++++++++++++++++++++++++++++++------- 1 file changed, 31 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c5e99944990a..cd62c88b5246 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -124,6 +124,7 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 +#define VIA_JACK_EVENT 0x04 enum { VIA_CTL_WIDGET_VOL, @@ -1413,10 +1414,12 @@ static void via_unsol_event(struct hda_codec *codec, unsigned int res) { res >>= 26; - if (res == VIA_HP_EVENT) + if (res & VIA_HP_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) + if (res & VIA_GPIO_EVENT) via_gpio_control(codec); + if (res & VIA_JACK_EVENT) + set_jack_power_state(codec); } static int via_init(struct hda_codec *codec) @@ -1878,7 +1881,8 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = { }; static struct hda_verb vt1709_uniwill_init_verbs[] = { - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, { } }; @@ -2514,7 +2518,15 @@ static struct hda_verb vt1708B_4ch_volume_init_verbs[] = { }; static struct hda_verb vt1708B_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3009,7 +3021,15 @@ static struct hda_verb vt1708S_volume_init_verbs[] = { }; static struct hda_verb vt1708S_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3448,8 +3468,12 @@ static struct hda_verb vt1702_volume_init_verbs[] = { }; static struct hda_verb vt1702_uniwill_init_verbs[] = { - {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; -- cgit v1.2.2 From dcf34c8cc685781cebbe1f4c75272a3269eba3a1 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:15 +0800 Subject: ALSA: HDA VIA: Refresh front playback mute in via_hp_automute. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index cd62c88b5246..c1f4307feaae 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1351,14 +1351,25 @@ static void via_free(struct hda_codec *codec) /* mute internal speaker if HP is plugged */ static void via_hp_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = 0; struct via_spec *spec = codec->spec; present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], - HDA_OUTPUT, 0, HDA_AMP_MUTE, - present ? HDA_AMP_MUTE : 0); + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + /* auto mute */ + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Front Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } } static void via_gpio_control(struct hda_codec *codec) -- cgit v1.2.2 From 1f2e99febd5dd0c91f0d0752674029a4376649e5 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:17 +0800 Subject: ALSA: HDA VIA: Add Jack detect feature for VT1708. VT1708 does not support unsolicited response, but we need hp detect to automute speaker. Implemented in workqueue. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 230 ++++++++++++++++++++++++++++++++++------------ 1 file changed, 173 insertions(+), 57 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c1f4307feaae..38418a53acd7 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -89,6 +89,64 @@ enum VIA_HDA_CODEC { CODEC_TYPES, }; +struct via_spec { + /* codec parameterization */ + struct snd_kcontrol_new *mixers[4]; + unsigned int num_mixers; + + struct hda_verb *init_verbs[5]; + unsigned int num_iverbs; + + char *stream_name_analog; + struct hda_pcm_stream *stream_analog_playback; + struct hda_pcm_stream *stream_analog_capture; + + char *stream_name_digital; + struct hda_pcm_stream *stream_digital_playback; + struct hda_pcm_stream *stream_digital_capture; + + /* playback */ + struct hda_multi_out multiout; + hda_nid_t slave_dig_outs[2]; + + /* capture */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; + hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[3]; + + /* PCM information */ + struct hda_pcm pcm_rec[3]; + + /* dynamic controls, init_verbs and input_mux */ + struct auto_pin_cfg autocfg; + struct snd_array kctls; + struct hda_input_mux private_imux[2]; + hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + + /* HP mode source */ + const struct hda_input_mux *hp_mux; + unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; + + enum VIA_HDA_CODEC codec_type; + + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work vt1708_hp_work; + int vt1708_jack_detectect; + int vt1708_hp_present; +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif +}; + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -181,6 +239,31 @@ static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); +static int is_aa_path_mute(struct hda_codec *codec); + +static void vt1708_start_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + if (!delayed_work_pending(&spec->vt1708_hp_work)) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); +} + +static void vt1708_stop_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 + && !is_aa_path_mute(spec->codec)) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + cancel_delayed_work(&spec->vt1708_hp_work); + flush_scheduled_work(); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -190,6 +273,12 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol, set_jack_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { + if (is_aa_path_mute(codec)) + vt1708_start_hp_work(codec->spec); + else + vt1708_stop_hp_work(codec->spec); + } return change; } @@ -210,59 +299,6 @@ static struct snd_kcontrol_new vt1708_control_templates[] = { }; -struct via_spec { - /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; - unsigned int num_mixers; - - struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; - - char *stream_name_analog; - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; - - char *stream_name_digital; - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; - - /* playback */ - struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; - - /* capture */ - unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t mux_nids[3]; - hda_nid_t dig_in_nid; - hda_nid_t dig_in_pin; - - /* capture source */ - const struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; - - /* PCM information */ - struct hda_pcm pcm_rec[3]; - - /* dynamic controls, init_verbs and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_imux[2]; - hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; - - /* HP mode source */ - const struct hda_input_mux *hp_mux; - unsigned int hp_independent_mode; - unsigned int hp_independent_mode_index; - unsigned int smart51_enabled; - - enum VIA_HDA_CODEC codec_type; - -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct hda_loopback_check loopback; -#endif -}; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -981,7 +1017,6 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo, struct via_spec *spec = codec->spec; int idle = substream->pstr->substream_opened == 1 && substream->ref_count == 0; - analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); @@ -994,6 +1029,7 @@ static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_start_hp_work(spec); return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -1003,6 +1039,7 @@ static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } @@ -1094,7 +1131,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo, snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); } - + vt1708_start_hp_work(spec); return 0; } @@ -1134,7 +1171,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo, snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); } - + vt1708_stop_hp_work(spec); return 0; } @@ -1345,6 +1382,7 @@ static void via_free(struct hda_codec *codec) return; via_free_kctls(codec); + vt1708_stop_hp_work(spec); kfree(codec->spec); } @@ -1464,6 +1502,15 @@ static int via_init(struct hda_codec *codec) return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static int via_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); + return 0; +} +#endif + #ifdef CONFIG_SND_HDA_POWER_SAVE static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -1479,6 +1526,9 @@ static struct hda_codec_ops via_patch_ops = { .build_pcms = via_build_pcms, .init = via_init, .free = via_free, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = via_suspend, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = via_check_power_status, #endif @@ -1728,6 +1778,51 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1753,6 +1848,10 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); if (err < 0) return err; @@ -1788,6 +1887,22 @@ static int via_auto_init(struct hda_codec *codec) return 0; } +static void vt1708_update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + vt1708_hp_work.work); + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (spec->vt1708_hp_present + != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + spec->vt1708_hp_present ^= 1; + via_hp_automute(spec->codec); + } + vt1708_start_hp_work(spec); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1864,7 +1979,8 @@ static int patch_vt1708(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; } -- cgit v1.2.2 From 82ef9e45c48634af5e3f6ab9ac75b6642c538020 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:19 +0800 Subject: ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function. like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port Connectivity field into 'AC_JACK_PORT_COMPLEX' Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 38418a53acd7..dc416ec0c6d4 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1768,11 +1768,10 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { - if (seqassoc == 0xff) { - def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_set_pincfg(codec, nid, def_conf); - } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE + && (seqassoc == 0xf0 || seqassoc == 0xff)) { + def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } return; -- cgit v1.2.2 From c873cc25280113d71463ad5075413d283be6b766 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:21 +0800 Subject: ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup Replaced with via_playback_multi_pcm_prepare/cleanup to support multi-stream operations Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 40 +++++++++------------------------------- 1 file changed, 9 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index dc416ec0c6d4..4d3c447342b0 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1022,28 +1022,6 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo, hinfo); } -static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - vt1708_start_hp_work(spec); - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - vt1708_stop_hp_work(spec); - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - - static void playback_multi_pcm_prep_0(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, @@ -1252,7 +1230,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = { }; static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ @@ -1263,8 +1241,8 @@ static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -2062,8 +2040,8 @@ static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = { .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -2074,8 +2052,8 @@ static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = { .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -3166,8 +3144,8 @@ static struct hda_pcm_stream vt1708S_pcm_analog_playback = { .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, .close = via_pcm_open_close }, }; -- cgit v1.2.2 From 9645c2039d5cfdbdcebe297420e180b6cd262836 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:27 +0800 Subject: ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4d3c447342b0..efadacd60835 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1553,7 +1553,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid, nid_vol = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -1562,8 +1562,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, if (!nid) continue; - if (i != AUTO_SEQ_FRONT) - nid_vol = 0x18 + i; + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -1595,13 +1594,13 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; -- cgit v1.2.2 From 4483a2f5907fa824bd6384c36fdcee9777cab1b9 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:29 +0800 Subject: ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index efadacd60835..f9702a17fc16 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2160,7 +2160,7 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -2169,43 +2169,45 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, if (!nid) continue; + nid_vol = nid_vols[i]; + if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT){ - /* add control to mixer index 0 */ + /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; @@ -2226,26 +2228,26 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, } else if (i == AUTO_SEQ_SURROUND) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SIDE) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; -- cgit v1.2.2 From 6369bcfccb57da28ad3e09b25fecd841a415ae95 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:31 +0800 Subject: ALSA: HDA VIA: Replace MIC_BOOST_VOLUME. With snd_hda_override_amp_caps. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 72 ++++++++++------------------------------------- 1 file changed, 15 insertions(+), 57 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index f9702a17fc16..4b7cd5971701 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -197,46 +197,6 @@ enum { AUTO_SEQ_SIDE }; -/* Some VT1708S based boards gets the micboost setting wrong, so we have - * to apply some brute-force and re-write the TLV's by software. */ -static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *_tlv) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - if (size < 4 * sizeof(unsigned int)) - return -ENOMEM; - if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */ - return -EFAULT; - if (put_user(2 * sizeof(unsigned int), _tlv + 1)) - return -EFAULT; - if (put_user(0, _tlv + 2)) /* offset = 0 */ - return -EFAULT; - if (put_user(1000, _tlv + 3)) /* step size = 10 dB */ - return -EFAULT; - } - return 0; -} - -static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 3; - } - return 0; -} - static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); @@ -3063,29 +3023,15 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) /* Patch for VT1708S */ -/* VT1708S software backdoor based override for buggy hardware micboost - * setting */ -#define MIC_BOOST_VOLUME(xname, nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ - .info = mic_boost_volume_info, \ - .get = snd_hda_mixer_amp_volume_get, \ - .put = snd_hda_mixer_amp_volume_put, \ - .tlv = { .c = mic_boost_tlv }, \ - .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) } - /* capture mixer elements */ static struct snd_kcontrol_new vt1708S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), - MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A), - MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -3457,6 +3403,16 @@ static struct hda_amp_list vt1708S_loopbacks[] = { }; #endif +static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, + int offset, int num_steps, int step_size) +{ + snd_hda_override_amp_caps(codec, pin, HDA_INPUT, + (offset << AC_AMPCAP_OFFSET_SHIFT) | + (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | + (step_size << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + static int patch_vt1708S(struct hda_codec *codec) { struct via_spec *spec; @@ -3493,6 +3449,8 @@ static int patch_vt1708S(struct hda_codec *codec) spec->adc_nids = vt1708S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; spec->num_mixers++; } -- cgit v1.2.2 From bc7e7e5ce05047e16633a94d36fa144af1d2b4c7 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:32 +0800 Subject: ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb As init verbs, vt17xx_volume_init_verb is a better place to hold them. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4b7cd5971701..1c87231fa7e5 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3068,6 +3068,8 @@ static struct hda_verb vt1708S_volume_init_verbs[] = { {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* Enable Mic Boost Volume backdoor */ {0x1, 0xf98, 0x1}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, { } }; @@ -3527,6 +3529,10 @@ static struct hda_verb vt1702_volume_init_verbs[] = { /* PW6 PW7 Output enable */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* mixer enable */ + {0x1, 0xF88, 0x3}, + /* GPIO 0~2 */ + {0x1, 0xF82, 0x3F}, { } }; @@ -3768,8 +3774,6 @@ static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; int err; - unsigned int response; - unsigned char control; /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -3814,18 +3818,6 @@ static int patch_vt1702(struct hda_codec *codec) spec->loopback.amplist = vt1702_loopbacks; #endif - /* Open backdoor */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0); - control = (unsigned char)(response & 0xff); - control |= 0x3; - snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control); - - /* Enable GPIO 0&1 for volume&mute control */ - /* Enable GPIO 2 for DMIC-DATA */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0); - control = (unsigned char)((response >> 16) & 0x3f); - snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control); - return 0; } -- cgit v1.2.2 From eb7188cafcb7aa1419b8889494cdbd4e6a01da1c Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:34 +0800 Subject: ALSA: HDA VIA: Add VT1718S support. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 554 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 545 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 1c87231fa7e5..c78385340694 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -86,6 +86,7 @@ enum VIA_HDA_CODEC { VT1708S, VT1708BCE, VT1702, + VT1718S, CODEC_TYPES, }; @@ -175,6 +176,9 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) else if ((dev_id & 0xfff) == 0x398 && (dev_id >> 12) < 8) codec_type = VT1702; + else if ((dev_id & 0xfff) == 0x428 + && (dev_id >> 12) < 8) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -284,6 +288,11 @@ static hda_nid_t vt1702_adc_nids[3] = { 0x12, 0x20, 0x1F }; +static hda_nid_t vt1718S_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -512,6 +521,67 @@ static void set_jack_power_state(struct hda_codec *codec) snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1718S) { + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW0/1 (24h/25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + set_pin_power_state(codec, 0x25, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, + parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } } } @@ -572,11 +642,21 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.hp_pins[0]; - unsigned int pinsel = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_SEL, - 0x00); + hda_nid_t nid; + unsigned int pinsel; + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + /* use !! to translate conn sel 2 for VT1718S */ + pinsel = !!snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_SEL, + 0x00); ucontrol->value.enumerated.item[0] = pinsel; return 0; @@ -635,6 +715,16 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ + spec->multiout.num_dacs = 4; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid @@ -645,7 +735,8 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, update_side_mute_status(codec); /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S - || spec->codec_type == VT1702) { + || spec->codec_type == VT1702 + || spec->codec_type == VT1718S) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -758,7 +849,8 @@ static int via_smart51_get(struct snd_kcontrol *kcontrol, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* ignore FMic for independent HP */ if (ctl & AC_PINCTL_IN_EN && !(ctl & AC_PINCTL_OUT_EN)) @@ -782,7 +874,8 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, for (i = 0; i < ARRAY_SIZE(index); i++) { hda_nid_t nid = spec->autocfg.input_pins[index[i]]; if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* don't retask FMic for independent HP */ if (nid) { unsigned int parm = snd_hda_codec_read( @@ -797,6 +890,10 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, mute_aa_path(codec, 1); notify_aa_path_ctls(codec); } + if (spec->codec_type == VT1718S) + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, + HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { if (spec->codec_type == VT1708S) { @@ -871,6 +968,11 @@ static int is_aa_path_mute(struct hda_codec *codec) start_idx = 1; end_idx = 3; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return 0; } @@ -920,6 +1022,7 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ break; case VT1708S: + case VT1718S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -1026,8 +1129,8 @@ static void playback_multi_pcm_prep_0(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); - if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && - !spec->hp_independent_mode) + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] + && !spec->hp_independent_mode) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); @@ -3821,6 +3924,435 @@ static int patch_vt1702(struct hda_codec *codec) return 0; } +/* Patch for VT1718S */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1718S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1718S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + + /* Setup default input of Front HP to MW9 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* PW9 PW10 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + /* PW11 Input enable */ + {0x2f, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_IN_EN}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf88, 0x8}, + /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Unmute MW4's index 0 */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + + +static struct hda_verb vt1718S_uniwill_init_verbs[] = { + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x27, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 10, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1718S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 4; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x8; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0xa; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x9; + break; + case AUTO_SEQ_SIDE: + spec->multiout.dac_nids[i] = 0xb; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb}; + hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27}; + hda_nid_t nid, nid_vol, nid_mute = 0; + int i, err; + + for (i = 0; i <= AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* Front */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0xc; /* AOW4 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0xc, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 1; + break; + + case 0x2a: /* Line In */ + idx = 2; + break; + + case 0x29: /* Front Mic */ + idx = 3; + break; + + case 0x2c: /* CD */ + idx = 0; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + return 0; +} + +static int vt1718S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + + if (err < 0) + return err; + err = vt1718S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1718S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1718S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->autocfg.dig_in_pin && codec->vendor_id == 0x11060428) + spec->dig_in_nid = 0x13; + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1718S_loopbacks[] = { + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { 0x21, HDA_INPUT, 3 }, + { 0x21, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1718S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1718S Analog"; + spec->stream_analog_playback = &vt1718S_pcm_analog_playback; + spec->stream_analog_capture = &vt1718S_pcm_analog_capture; + + spec->stream_name_digital = "VT1718S Digital"; + spec->stream_digital_playback = &vt1718S_pcm_digital_playback; + if (codec->vendor_id == 0x11060428) + spec->stream_digital_capture = &vt1718S_pcm_digital_capture; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1718S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1718S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -3893,6 +4425,10 @@ static struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1702}, { .id = 0x11067398, .name = "VT1702", .patch = patch_vt1702}, + { .id = 0x11060428, .name = "VT1718S", + .patch = patch_vt1718S}, + { .id = 0x11064428, .name = "VT1718S", + .patch = patch_vt1718S}, {} /* terminator */ }; -- cgit v1.2.2 From bb3c6bfc3f7a5416d85c5dbc312e2d47fc672eef Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:39 +0800 Subject: ALSA: HDA VIA: Add VT1828S and VT2020 support. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c78385340694..2e7e72c83a52 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -179,6 +179,8 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0441 || dev_id == 0x4441) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -4323,21 +4325,31 @@ static int patch_vt1718S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; - spec->stream_name_analog = "VT1718S Analog"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_analog = "VT2020 Analog"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_analog = "VT1828S Analog"; + else + spec->stream_name_analog = "VT1718S Analog"; spec->stream_analog_playback = &vt1718S_pcm_analog_playback; spec->stream_analog_capture = &vt1718S_pcm_analog_capture; - spec->stream_name_digital = "VT1718S Digital"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_digital = "VT2020 Digital"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_digital = "VT1828S Digital"; + else + spec->stream_name_digital = "VT1718S Digital"; spec->stream_digital_playback = &vt1718S_pcm_digital_playback; - if (codec->vendor_id == 0x11060428) + if (codec->vendor_id == 0x11060428 || codec->vendor_id == 0x11060441) spec->stream_digital_capture = &vt1718S_pcm_digital_capture; if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1718S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); get_mux_nids(codec); - override_mic_boost(codec, 0x1a, 0, 3, 40); - override_mic_boost(codec, 0x1e, 0, 3, 40); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; spec->num_mixers++; } @@ -4429,6 +4441,10 @@ static struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1718S}, { .id = 0x11064428, .name = "VT1718S", .patch = patch_vt1718S}, + { .id = 0x11060441, .name = "VT2020", + .patch = patch_vt1718S}, + { .id = 0x11064441, .name = "VT1828S", + .patch = patch_vt1718S}, {} /* terminator */ }; -- cgit v1.2.2 From f3db423df84570c9950754a5771ad26f0111235f Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:41 +0800 Subject: ALSA: HDA VIA: Add VT1716S support. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 648 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 644 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2e7e72c83a52..2977004677ec 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -87,12 +87,13 @@ enum VIA_HDA_CODEC { VT1708BCE, VT1702, VT1718S, + VT1716S, CODEC_TYPES, }; struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; + struct snd_kcontrol_new *mixers[6]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -135,7 +136,7 @@ struct via_spec { unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; unsigned int smart51_enabled; - + unsigned int dmic_enabled; enum VIA_HDA_CODEC codec_type; /* work to check hp jack state */ @@ -179,6 +180,8 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0433 || dev_id == 0xa721) + codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; else @@ -189,6 +192,7 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 +#define VIA_MONO_EVENT 0x08 enum { VIA_CTL_WIDGET_VOL, @@ -295,6 +299,11 @@ static hda_nid_t vt1718S_adc_nids[2] = { 0x10, 0x11 }; +static hda_nid_t vt1716S_adc_nids[2] = { + /* ADC1-2 */ + 0x13, 0x14 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -584,6 +593,106 @@ static void set_jack_power_state(struct hda_codec *codec) snd_hda_codec_write(codec, 0xc, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1716S) { + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write( + codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_codec_read( + codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) + mono_out = 0; + else { + present = snd_hda_codec_read( + codec, 0x1d, 0, AC_VERB_GET_PIN_SENSE, 0) + & 0x80000000; + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); } } @@ -738,7 +847,8 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S || spec->codec_type == VT1702 - || spec->codec_type == VT1718S) { + || spec->codec_type == VT1718S + || spec->codec_type == VT1716S) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -797,6 +907,7 @@ static void mute_aa_path(struct hda_codec *codec, int mute) case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -898,7 +1009,8 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { - if (spec->codec_type == VT1708S) { + if (spec->codec_type == VT1708S + || spec->codec_type == VT1716S) { /* input = index 1 (AOW3) */ snd_hda_codec_write( codec, nid, 0, @@ -961,6 +1073,7 @@ static int is_aa_path_mute(struct hda_codec *codec) case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -1025,6 +1138,7 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) break; case VT1708S: case VT1718S: + case VT1716S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -1453,6 +1567,36 @@ static void via_hp_automute(struct hda_codec *codec) } } +/* mute mono out if HP or Line out is plugged */ +static void via_mono_automute(struct hda_codec *codec) +{ + unsigned int hp_present, lineout_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1716S) + return; + + lineout_present = snd_hda_codec_read( + codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Mute Mono Out if Line Out is plugged */ + if (lineout_present) { + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE); + return; + } + + hp_present = snd_hda_codec_read( + codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, + hp_present ? HDA_AMP_MUTE : 0); +} + static void via_gpio_control(struct hda_codec *codec) { unsigned int gpio_data; @@ -1512,6 +1656,8 @@ static void via_unsol_event(struct hda_codec *codec, via_gpio_control(codec); if (res & VIA_JACK_EVENT) set_jack_power_state(codec); + if (res & VIA_MONO_EVENT) + via_mono_automute(codec); } static int via_init(struct hda_codec *codec) @@ -4365,6 +4511,496 @@ static int patch_vt1718S(struct hda_codec *codec) return 0; } + +/* Patch for VT1716S */ + +static int vt1716s_dmic_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int vt1716s_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int index = 0; + + index = snd_hda_codec_read(codec, 0x26, 0, + AC_VERB_GET_CONNECT_SEL, 0); + if (index != -1) + *ucontrol->value.integer.value = index; + + return 0; +} + +static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index = *ucontrol->value.integer.value; + + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_CONNECT_SEL, index); + spec->dmic_enabled = index; + set_jack_power_state(codec); + + return 1; +} + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1716S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Mic Capture Switch", + .count = 1, + .info = vt1716s_dmic_info, + .get = vt1716s_dmic_get, + .put = vt1716s_dmic_put, + }, + {} /* end */ +}; + + +/* mono-out mixer elements */ +static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { + HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb vt1716S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Stereo Mixer = 5 */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x5}, + + /* Setup default input of PW4 to MW0 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0}, + + /* Setup default input of SW1 as MW0 */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* Setup default input of SW4 as AOW0 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* PW9 PW10 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + /* Unmute SW1, PW12 */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* PW12 Output enable */ + {0x2a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf8a, 0x80}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, + /* Enable mono output */ + {0x1, 0xf90, 0x08}, + { } +}; + + +static struct hda_verb vt1716S_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_MONO_EVENT | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x13, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1716S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 3; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x10; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0x25; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x11; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[3] = { "Front", "Surround", "C/LFE" }; + hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; + hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; + hda_nid_t nid, nid_vol, nid_mute; + int i, err; + + for (i = 0; i <= AUTO_SEQ_CENLFE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x25; /* AOW3 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x1a: /* Mic */ + idx = 2; + break; + + case 0x1b: /* Line In */ + idx = 3; + break; + + case 0x1e: /* Front Mic */ + idx = 4; + break; + + case 0x1f: /* CD */ + idx = 1; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x16); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx-1; + imux->num_items++; + } + return 0; +} + +static int vt1716S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = vt1716S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1716S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1716S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1716S_loopbacks[] = { + { 0x16, HDA_INPUT, 1 }, + { 0x16, HDA_INPUT, 2 }, + { 0x16, HDA_INPUT, 3 }, + { 0x16, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1716S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1716S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1716S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1716S Analog"; + spec->stream_analog_playback = &vt1716S_pcm_analog_playback; + spec->stream_analog_capture = &vt1716S_pcm_analog_capture; + + spec->stream_name_digital = "VT1716S Digital"; + spec->stream_digital_playback = &vt1716S_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1716S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1716S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1716S_capture_mixer; + spec->num_mixers++; + } + + spec->mixers[spec->num_mixers] = vt1716s_dmic_mixer; + spec->num_mixers++; + + spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1716S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -4445,6 +5081,10 @@ static struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1718S}, { .id = 0x11064441, .name = "VT1828S", .patch = patch_vt1718S}, + { .id = 0x11060433, .name = "VT1716S", + .patch = patch_vt1716S}, + { .id = 0x1106a721, .name = "VT1716S", + .patch = patch_vt1716S}, {} /* terminator */ }; -- cgit v1.2.2 From 25eaba2f8a6877ba6f58197c4723c2433a316e09 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:43 +0800 Subject: ALSA: HDA VIA: Add VT2002P support. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 665 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 660 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2977004677ec..a94cc91c18ff 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -88,6 +88,7 @@ enum VIA_HDA_CODEC { VT1702, VT1718S, VT1716S, + VT2002P, CODEC_TYPES, }; @@ -184,6 +185,8 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; + else if (dev_id == 0x0438 || dev_id == 0x4438) + codec_type = VT2002P; else codec_type = UNKNOWN; return codec_type; @@ -193,11 +196,14 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 #define VIA_MONO_EVENT 0x08 +#define VIA_SPEAKER_EVENT 0x10 +#define VIA_BIND_HP_EVENT 0x20 enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, VIA_CTL_WIDGET_ANALOG_MUTE, + VIA_CTL_WIDGET_BIND_PIN_MUTE, }; enum { @@ -235,6 +241,7 @@ static void vt1708_stop_hp_work(struct via_spec *spec) flush_scheduled_work(); } + static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -262,13 +269,108 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol, .put = analog_input_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } +static void via_hp_bind_automute(struct hda_codec *codec); + +static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int i; + int change = 0; + + long *valp = ucontrol->value.integer.value; + int lmute, rmute; + if (strstr(kcontrol->id.name, "Switch") == NULL) { + snd_printd("Invalid control!\n"); + return change; + } + change = snd_hda_mixer_amp_switch_put(kcontrol, + ucontrol); + /* Get mute value */ + lmute = *valp ? 0 : HDA_AMP_MUTE; + valp++; + rmute = *valp ? 0 : HDA_AMP_MUTE; + + /* Set hp pins */ + if (!spec->hp_independent_mode) { + for (i = 0; i < spec->autocfg.hp_outs; i++) { + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + + if (!lmute && !rmute) { + /* Line Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* unmute */ + via_hp_bind_automute(codec); + + } else { + if (lmute) { + /* Mute all left channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + } + if (rmute) { + /* mute all right channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + return change; +} + +#define BIND_PIN_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = bind_pin_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, + BIND_PIN_MUTE, }; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -304,6 +406,11 @@ static hda_nid_t vt1716S_adc_nids[2] = { 0x13, 0x14 }; +static hda_nid_t vt2002P_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -386,10 +493,13 @@ static void via_auto_init_hp_out(struct hda_codec *codec) { struct via_spec *spec = codec->spec; hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + for (i = 0; i < spec->autocfg.hp_outs; i++) { + pin = spec->autocfg.hp_pins[i]; + if (pin) /* connect to front */ + via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + } } static void via_auto_init_analog_input(struct hda_codec *codec) @@ -693,6 +803,107 @@ static void set_jack_power_state(struct hda_codec *codec) imux_is_smixer ? AC_PWRST_D0 : parm); snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, mono_out ? AC_PWRST_D0 : parm); + } else if (spec->codec_type == VT2002P) { + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, + 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Class-D */ + /* PW0 (24h), MW0(18h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* Mono Out */ + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + present = snd_hda_codec_read( + codec, 0x26, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } } @@ -760,6 +971,9 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, case VT1718S: nid = 0x34; break; + case VT2002P: + nid = 0x35; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -832,6 +1046,9 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ spec->multiout.num_dacs = 4; break; + case VT2002P: + nid = 0x35; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -848,7 +1065,8 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, if (spec->codec_type == VT1708S || spec->codec_type == VT1702 || spec->codec_type == VT1718S - || spec->codec_type == VT1716S) { + || spec->codec_type == VT1716S + || spec->codec_type == VT2002P) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -1088,6 +1306,11 @@ static int is_aa_path_mute(struct hda_codec *codec) start_idx = 1; end_idx = 3; break; + case VT2002P: + nid_mixer = 0x21; + start_idx = 0; + end_idx = 2; + break; default: return 0; } @@ -1146,6 +1369,10 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) verb = 0xf73; parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; + case VT2002P: + verb = 0xf93; + parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ + break; default: return; /* other codecs are not supported */ } @@ -1645,6 +1872,66 @@ static void via_gpio_control(struct hda_codec *codec) } } +/* mute Internal-Speaker if HP is plugged */ +static void via_speaker_automute(struct hda_codec *codec) +{ + unsigned int hp_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT2002P) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Speaker Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +/* mute line-out and internal speaker if HP is plugged */ +static void via_hp_bind_automute(struct hda_codec *codec) +{ + unsigned int hp_present, present = 0; + struct via_spec *spec = codec->spec; + int i; + + if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0]) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + present = snd_hda_codec_read(codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + /* Mute Line-Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + if (hp_present) + present = hp_present; + } + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + /* unsolicited event for jack sensing */ static void via_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1658,6 +1945,10 @@ static void via_unsol_event(struct hda_codec *codec, set_jack_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); + if (res & VIA_SPEAKER_EVENT) + via_speaker_automute(codec); + if (res & VIA_BIND_HP_EVENT) + via_hp_bind_automute(codec); } static int via_init(struct hda_codec *codec) @@ -2067,10 +2358,19 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ static int via_auto_init(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; + via_init(codec); via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); + if (spec->codec_type == VT2002P) { + via_hp_bind_automute(codec); + } else { + via_hp_automute(codec); + via_speaker_automute(codec); + } + return 0; } @@ -5001,6 +5301,359 @@ static int patch_vt1716S(struct hda_codec *codec) return 0; } + +/* for vt2002P */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt2002P_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt2002P_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x37, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3b, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; + + +static struct hda_verb vt2002P_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt2002P_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 3; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 4; + imux->num_items++; + + return 0; +} + +static int vt2002P_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + err = vt2002P_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt2002P_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt2002P_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt2002P_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt2002P */ +static int patch_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt2002P_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + + spec->stream_name_analog = "VT2002P Analog"; + spec->stream_analog_playback = &vt2002P_pcm_analog_playback; + spec->stream_analog_capture = &vt2002P_pcm_analog_capture; + + spec->stream_name_digital = "VT2002P Digital"; + spec->stream_digital_playback = &vt2002P_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt2002P_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt2002P_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt2002P_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt2002P_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -5085,6 +5738,8 @@ static struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1716S}, { .id = 0x1106a721, .name = "VT1716S", .patch = patch_vt1716S}, + { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, {} /* terminator */ }; -- cgit v1.2.2 From ab6734e7ea32e9f9cbe0f55eeddf4aa629ed1c3d Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:46 +0800 Subject: ALSA: HDA VIA: Add VT1812 support. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 494 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 491 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a94cc91c18ff..b3c5e8a78154 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -89,6 +89,7 @@ enum VIA_HDA_CODEC { VT1718S, VT1716S, VT2002P, + VT1812, CODEC_TYPES, }; @@ -187,6 +188,8 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) codec_type = VT1718S; else if (dev_id == 0x0438 || dev_id == 0x4438) codec_type = VT2002P; + else if (dev_id == 0x0448) + codec_type = VT1812; else codec_type = UNKNOWN; return codec_type; @@ -411,6 +414,12 @@ static hda_nid_t vt2002P_adc_nids[2] = { 0x10, 0x11 }; +static hda_nid_t vt1812_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -895,6 +904,120 @@ static void set_jack_power_state(struct hda_codec *codec) AC_VERB_SET_POWER_STATE, AC_PWRST_D0); } + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else if (spec->codec_type == VT1812) { + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_codec_read( + codec, 0x28, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, + AC_VERB_SET_POWER_STATE, parm); + /* MW9 (21h) */ if (imux_is_smixer || !is_aa_path_mute(codec)) snd_hda_codec_write( @@ -974,6 +1097,9 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, case VT2002P: nid = 0x35; break; + case VT1812: + nid = 0x3d; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -1049,6 +1175,9 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, case VT2002P: nid = 0x35; break; + case VT1812: + nid = 0x3d; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -1066,7 +1195,8 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, || spec->codec_type == VT1702 || spec->codec_type == VT1718S || spec->codec_type == VT1716S - || spec->codec_type == VT2002P) { + || spec->codec_type == VT2002P + || spec->codec_type == VT1812) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -1307,6 +1437,7 @@ static int is_aa_path_mute(struct hda_codec *codec) end_idx = 3; break; case VT2002P: + case VT1812: nid_mixer = 0x21; start_idx = 0; end_idx = 2; @@ -1370,6 +1501,7 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; case VT2002P: + case VT1812: verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -1878,7 +2010,7 @@ static void via_speaker_automute(struct hda_codec *codec) unsigned int hp_present; struct via_spec *spec = codec->spec; - if (spec->codec_type != VT2002P) + if (spec->codec_type != VT2002P && spec->codec_type != VT1812) return; hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, @@ -2364,7 +2496,7 @@ static int via_auto_init(struct hda_codec *codec) via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); - if (spec->codec_type == VT2002P) { + if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { via_hp_bind_automute(codec); } else { via_hp_automute(codec); @@ -5654,6 +5786,361 @@ static int patch_vt2002P(struct hda_codec *codec) return 0; } + +/* for vt1812 */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1812_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1812_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/13/15: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/13/15 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0xa8}, + { } +}; + + +static struct hda_verb vt1812_uniwill_init_verbs[] = { + {0x33, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT }, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1812_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1812_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + + +/* add playback controls from the parsed DAC table */ +static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 6; + imux->num_items++; + + return 0; +} + +static int vt1812_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + fill_dig_outs(codec); + err = vt1812_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_outs) + return 0; /* can't find valid BIOS pin config */ + + err = vt1812_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1812_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1812_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt1812 */ +static int patch_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1812_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + + spec->init_verbs[spec->num_iverbs++] = vt1812_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1812_uniwill_init_verbs; + + spec->stream_name_analog = "VT1812 Analog"; + spec->stream_analog_playback = &vt1812_pcm_analog_playback; + spec->stream_analog_capture = &vt1812_pcm_analog_capture; + + spec->stream_name_digital = "VT1812 Digital"; + spec->stream_digital_playback = &vt1812_pcm_digital_playback; + + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1812_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1812_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1812_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1812_loopbacks; +#endif + + return 0; +} + /* * patch entries */ @@ -5740,6 +6227,7 @@ static struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1716S}, { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, {} /* terminator */ }; -- cgit v1.2.2 From 71eb7dccb7d2d22236dbe46db07f8000d09fba01 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:49 +0800 Subject: ALSA: HDA VIA: rename vt1708_control_templates[]. To via_control_templates[]. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b3c5e8a78154..257b51c61422 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -367,7 +367,7 @@ static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, .put = bind_pin_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } -static struct snd_kcontrol_new vt1708_control_templates[] = { +static struct snd_kcontrol_new via_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, @@ -430,7 +430,7 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, knew = snd_array_new(&spec->kctls); if (!knew) return -ENOMEM; - *knew = vt1708_control_templates[type]; + *knew = via_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; -- cgit v1.2.2 From bfdc675a73f7697ead12c07dbf11e2b2632676f4 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:50 +0800 Subject: ALSA: HDA VIA: Change PW4 connect select default to to MW0. According to customer request, hp should be default to redirected mode, i.e. PW4 connect select default to to MW0. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 257b51c61422..4ea18a759a05 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1541,8 +1541,8 @@ static struct hda_verb vt1708_volume_init_verbs[] = { {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Setup default input to PW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Setup default input MW0 to PW4 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -2668,8 +2668,8 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = { {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Set input of PW4 as AOW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Set input of PW4 as MW0 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -3222,7 +3222,7 @@ static struct hda_verb vt1708B_8ch_volume_init_verbs[] = { {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Setup default input to PW4 */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x1d, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* PW10 Input enable */ -- cgit v1.2.2 From 8e86597f3cbd0a58808560116abe1bc0023256b0 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:52 +0800 Subject: ALSA: HDA VIA: comments: update copyright, changeset, etc. Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4ea18a759a05..fab875a21726 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1,10 +1,10 @@ /* * Universal Interface for Intel High Definition Audio Codec * - * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec + * HD audio interface patch for VIA VT17xx/VT18xx/VT20xx codec * - * Copyright (c) 2006-2008 Lydia Wang - * Takashi Iwai + * (C) 2006-2009 VIA Technology, Inc. + * (C) 2006-2008 Takashi Iwai * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -36,6 +36,11 @@ /* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ /* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2009-02-16 Logan Li Add support for VT1718S */ +/* 2009-03-13 Logan Li Add support for VT1716S */ +/* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ +/* 2009-07-08 Lydia Wang Add support for VT2002P */ +/* 2009-07-21 Lydia Wang Add support for VT1812 */ /* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ -- cgit v1.2.2 From 377ff31ae06f0d2644839246cd18c3e17fe62a48 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Sat, 10 Oct 2009 19:08:55 +0800 Subject: ALSA: HDA VIA: Only cosmetic changes Signed-off-by: Lydia Wang Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 64 ++++++++++++++++++++++++----------------------- 1 file changed, 33 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fab875a21726..30260e259181 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -22,26 +22,26 @@ */ /* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */ -/* */ +/* */ /* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */ -/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ -/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ +/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ +/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ -/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ -/* 2007-09-17 Lydia Wang Add VT1708B codec support */ +/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ +/* 2007-09-17 Lydia Wang Add VT1708B codec support */ /* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */ /* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */ -/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ -/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ -/* 2008-04-09 Lydia Wang Add Independent HP feature */ +/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ +/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ +/* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ -/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ /* 2009-02-16 Logan Li Add support for VT1718S */ /* 2009-03-13 Logan Li Add support for VT1716S */ /* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ /* 2009-07-08 Lydia Wang Add support for VT2002P */ /* 2009-07-21 Lydia Wang Add support for VT1812 */ -/* */ +/* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -486,7 +486,7 @@ static void via_auto_set_output_and_unmute(struct hda_codec *codec, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } @@ -1545,7 +1545,7 @@ static struct hda_verb vt1708_volume_init_verbs[] = { {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - + /* Setup default input MW0 to PW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ @@ -1865,8 +1865,10 @@ static int via_build_pcms(struct hda_codec *codec) codec->pcm_info = info; info->name = spec->stream_name_analog; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; @@ -2116,7 +2118,7 @@ static int via_init(struct hda_codec *codec) if (spec->slave_dig_outs[0]) codec->slave_dig_outs = spec->slave_dig_outs; - return 0; + return 0; } #ifdef SND_HDA_NEEDS_RESUME @@ -2161,8 +2163,8 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; - - for(i = 0; i < 4; i++) { + + for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -2200,7 +2202,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, if (!nid) continue; - + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { @@ -2229,7 +2231,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2243,7 +2245,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2343,7 +2345,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -2576,7 +2578,7 @@ static int patch_vt1708(struct hda_codec *codec) "from BIOS. Using genenic mode...\n"); } - + spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; /* disable 32bit format on VT1708 */ @@ -2588,7 +2590,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->stream_digital_playback = &vt1708_pcm_digital_playback; spec->stream_digital_capture = &vt1708_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); @@ -2775,11 +2777,11 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ - for(i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ - switch(i) { + switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ spec->multiout.dac_nids[i] = 0x10; @@ -2814,7 +2816,7 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + if (!nid) continue; nid_vol = nid_vols[i]; @@ -2845,7 +2847,7 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2859,7 +2861,7 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2955,7 +2957,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -3064,7 +3066,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -3158,7 +3160,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); -- cgit v1.2.2 From 633c7e92bdd54ba939f2bd3b78c72e1e1a1dd077 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 11 Oct 2009 12:38:49 +0200 Subject: ALSA: wss: reuse CS4231 controls for AD1848 The C4231 control set is a superset of the AD1848 control set so reuse the CS4231 controls definitions for the AD1848. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 79 ++++++++++++++----------------------------------- 1 file changed, 23 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 754a2089c650..2ba18978b419 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2200,49 +2200,12 @@ static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); -static struct snd_kcontrol_new snd_ad1848_controls[] = { -WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, - 7, 7, 1, 1), -WSS_DOUBLE_TLV("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, - db_scale_6bit), -WSS_DOUBLE("Aux Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE("Aux Playback Switch", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, - 0, 0, 15, 0, db_scale_rec_gain), -{ - .name = "Capture Source", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_wss_info_mux, - .get = snd_wss_get_mux, - .put = snd_wss_put_mux, -}, -WSS_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, - db_scale_6bit), -}; - static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, db_scale_6bit), -WSS_DOUBLE("Line Playback Switch", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, - db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Aux Playback Volume", 0, @@ -2253,15 +2216,6 @@ WSS_DOUBLE("Aux Playback Switch", 1, WSS_DOUBLE_TLV("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, db_scale_5bit_12db_max), -WSS_SINGLE("Mono Playback Switch", 0, - CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE_TLV("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1, - db_scale_4bit), -WSS_SINGLE("Mono Output Playback Switch", 0, - CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, - CS4231_MONO_CTRL, 5, 1, 0), WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0, db_scale_rec_gain), { @@ -2277,6 +2231,20 @@ WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, db_scale_6bit), +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), +WSS_SINGLE("Mono Playback Switch", 0, + CS4231_MONO_CTRL, 7, 1, 1), +WSS_SINGLE_TLV("Mono Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), +WSS_SINGLE("Mono Output Playback Switch", 0, + CS4231_MONO_CTRL, 6, 1, 1), +WSS_SINGLE("Mono Output Playback Bypass", 0, + CS4231_MONO_CTRL, 5, 1, 0), }; static struct snd_kcontrol_new snd_opti93x_controls[] = { @@ -2343,22 +2311,21 @@ int snd_wss_mixer(struct snd_wss *chip) if (err < 0) return err; } - else if (chip->hardware & WSS_HW_AD1848_MASK) - for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_ad1848_controls[idx], - chip)); - if (err < 0) - return err; - } - else - for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) { + else { + int count = ARRAY_SIZE(snd_wss_controls); + + /* Use only the first 11 entries on AD1848 */ + if (chip->hardware & WSS_HW_AD1848_MASK) + count = 11; + + for (idx = 0; idx < count; idx++) { err = snd_ctl_add(card, snd_ctl_new1(&snd_wss_controls[idx], chip)); if (err < 0) return err; } + } return 0; } EXPORT_SYMBOL(snd_wss_mixer); -- cgit v1.2.2 From 8066e51ae7329220f459470a38387f8533e99141 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 11 Oct 2009 12:48:00 +0200 Subject: ALSA: snd_dma_pointer workaround for chipsets with buggy DMA The chipsets with the isa_dma_bridge_buggy set do not stop DMA during DMA counter reads. The DMA counter is read in two 8-bit read steps on x86 platform. Sometimes, such reads happen during higher byte change so the lower byte is already decremented (rolled over) but the higher byte is not. It introduces an error that position is moved 256 bytes ahead of the true position. Thus, the next DMA position read can return a lower value then the previous read. If the DMA position is decreased (reversed) the ALSA subsystem is tricked into the playback underrun error and resets the playback. It results in a "pop" during a playback. Work around the issue by reading the counter twice and choosing a higher value. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/core/isadma.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 79f0f16af339..950e19ba91fc 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -85,16 +85,24 @@ EXPORT_SYMBOL(snd_dma_disable); unsigned int snd_dma_pointer(unsigned long dma, unsigned int size) { unsigned long flags; - unsigned int result; + unsigned int result, result1; flags = claim_dma_lock(); clear_dma_ff(dma); if (!isa_dma_bridge_buggy) disable_dma(dma); result = get_dma_residue(dma); + /* + * HACK - read the counter again and choose higher value in order to + * avoid reading during counter lower byte roll over if the + * isa_dma_bridge_buggy is set. + */ + result1 = get_dma_residue(dma); if (!isa_dma_bridge_buggy) enable_dma(dma); release_dma_lock(flags); + if (unlikely(result < result1)) + result = result1; #ifdef CONFIG_SND_DEBUG if (result > size) snd_printk(KERN_ERR "pointer (0x%x) for DMA #%ld is greater than transfer size (0x%x)\n", result, dma, size); -- cgit v1.2.2 From 0f48327eac5f65ad029d7112cac97577766730ba Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Mon, 12 Oct 2009 15:56:17 +1100 Subject: sound: use semicolons to end statements Fixes: sound/pci/hda/patch_via.c: In function 'patch_vt1718S': sound/pci/hda/patch_via.c:4951: error: expected expression before 'return' sound/pci/hda/patch_via.c: In function 'patch_vt1716S': sound/pci/hda/patch_via.c:5441: error: expected expression before 'return' sound/pci/hda/patch_via.c: In function 'patch_vt2002P': sound/pci/hda/patch_via.c:5794: error: expected expression before 'return' sound/pci/hda/patch_via.c: In function 'patch_vt1812': sound/pci/hda/patch_via.c:6148: error: expected expression before 'return' Signed-off-by: Stephen Rothwell Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 30260e259181..a294060ed684 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4942,7 +4942,7 @@ static int patch_vt1718S(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; - codec->patch_ops.unsol_event = via_unsol_event, + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1718S_loopbacks; @@ -5432,7 +5432,7 @@ static int patch_vt1716S(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; - codec->patch_ops.unsol_event = via_unsol_event, + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1716S_loopbacks; @@ -5785,7 +5785,7 @@ static int patch_vt2002P(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; - codec->patch_ops.unsol_event = via_unsol_event, + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt2002P_loopbacks; @@ -6139,7 +6139,7 @@ static int patch_vt1812(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; - codec->patch_ops.unsol_event = via_unsol_event, + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1812_loopbacks; -- cgit v1.2.2 From 68f139204c1a2b10cc292d9cca036ebdbb6730a8 Mon Sep 17 00:00:00 2001 From: Wu Zhangjin Date: Sat, 10 Oct 2009 23:53:49 +0800 Subject: ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency SND_CS5535AUDIO is available on Loongson(MIPS compatible) family machines, and checked it with ARCH=x86_64, no relative compiling warnings & errors, so, remove the platform dependency directly. Reported-by: rixed@happyleptic.org Acked-by: Andres Salomon Signed-off-by: Wu Zhangjin Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fb5ee3cc3968..75c602b5b132 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -259,7 +259,6 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help -- cgit v1.2.2 From 814b7963e50e331f129acc25ad92bd4db45c300f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 12 Oct 2009 11:43:55 +0300 Subject: ASoC: TPA6130A2: Make tpa6130a2_power as static The power for the amplifier should be handled internally by the tpa6130a2 driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/codecs/tpa6130a2.h | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0a6e7b4ace60..6b650c1aa3d1 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -106,7 +106,7 @@ static void tpa6130a2_initialize(void) tpa6130a2_i2c_write(i, data->regs[i]); } -void tpa6130a2_power(int power) +static void tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h index 6a794f16cee9..57e867fd86d1 100644 --- a/sound/soc/codecs/tpa6130a2.h +++ b/sound/soc/codecs/tpa6130a2.h @@ -57,6 +57,5 @@ #define TPA6130A2_VERSION_MASK (0x0f) extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); -extern void tpa6130a2_power(int power); #endif /* __TPA6130A2_H__ */ -- cgit v1.2.2 From a688e4885c1aa6b88ab5ffa64655bacc01749c9e Mon Sep 17 00:00:00 2001 From: Tobias Hansen Date: Mon, 12 Oct 2009 16:24:15 +0200 Subject: ALSA: snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd This is the correct error number for telling the USB system that this driver is not for the device. Signed-off-by: Tobias Hansen Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 6c7b64a23c13..b54e8ca360d1 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -601,7 +601,7 @@ static int snd_us122l_probe(struct usb_interface *intf, if (device->descriptor.idProduct == USB_ID_US144 && device->speed == USB_SPEED_HIGH) { snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); - return -ENOENT; + return -ENODEV; } snd_printdd(KERN_DEBUG"%p:%i\n", -- cgit v1.2.2 From ed9d040d40942e9c48167f9f37f86fab8e0e5e17 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Mon, 12 Oct 2009 21:17:09 +0100 Subject: ASoC: S3C: Remove Remove the include from arch/arm/plat-s3c/include/plat/audio.h as it provides nothing to the current kernel and is not in any future plans for the system. Signed-off-by: Ben Dooks Signed-off-by: Simtec Linux Team Signed-off-by: Mark Brown --- sound/soc/s3c24xx/neo1973_wm8753.c | 1 - sound/soc/s3c24xx/s3c-i2s-v2.c | 1 - sound/soc/s3c24xx/s3c2412-i2s.c | 1 - sound/soc/s3c24xx/s3c2443-ac97.c | 1 - sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 1 - sound/soc/s3c24xx/s3c64xx-i2s.c | 1 - 7 files changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c9b794843a70..77de6c5127d2 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -29,7 +29,6 @@ #include #include #include -#include #include #include diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 11c45a37c631..28b0ab255096 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,6 @@ #include -#include #include #include "s3c-i2s-v2.h" diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index a587ec40b449..ac5e47b082fb 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -34,7 +34,6 @@ #include -#include #include #include diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index fc1beb0930b9..b25e9f968df9 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -32,7 +32,6 @@ #include #include #include -#include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 40e2c4790f0d..c76b8bb214bc 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -32,7 +32,7 @@ #include #include #include -#include + #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 5cbbdc80fde3..27cf097c2b1d 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -29,7 +29,6 @@ #include #include #include -#include #include "s3c24xx-pcm.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 43fb253a3429..b67eed59666a 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -31,7 +31,6 @@ #include #include #include -#include #include #include -- cgit v1.2.2 From d2ed82a3e7d1f63b2da3f1aa5763667dd17919ac Mon Sep 17 00:00:00 2001 From: Logan Li Date: Wed, 14 Oct 2009 10:10:38 +0800 Subject: ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF 48 kHz limit is for slightly better stability, and sample rates other than 48k (like 96k/192k) are for better sound quality. We choose better quality, so remove the 48k limit. Signed-off-by: Logan Li Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a294060ed684..89e084d45369 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4626,7 +4626,6 @@ static struct hda_pcm_stream vt1718S_pcm_digital_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5124,7 +5123,6 @@ static struct hda_pcm_stream vt1716S_pcm_digital_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5561,7 +5559,6 @@ static struct hda_pcm_stream vt2002P_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5914,7 +5911,6 @@ static struct hda_pcm_stream vt1812_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, -- cgit v1.2.2 From d2058b0cd039aad89b111d83b9c347e9d8f57a84 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Oct 2009 17:39:56 +0100 Subject: ASoC: Remove snd_soc_suspend_device() The PM core will grow pm_link infrastructure in 2.6.33 which can be used to implement the intended functionality of the ASoC-specific device suspend and resume callbacks so drop them. Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 20 -------------------- sound/soc/codecs/wm8350.c | 17 ----------------- sound/soc/codecs/wm8400.c | 17 ----------------- sound/soc/codecs/wm8523.c | 17 ----------------- sound/soc/codecs/wm8580.c | 17 ----------------- sound/soc/codecs/wm8711.c | 17 ----------------- sound/soc/codecs/wm8731.c | 34 ---------------------------------- sound/soc/codecs/wm8753.c | 35 ----------------------------------- sound/soc/codecs/wm8776.c | 34 ---------------------------------- sound/soc/codecs/wm8900.c | 17 ----------------- sound/soc/codecs/wm8903.c | 17 ----------------- sound/soc/codecs/wm8940.c | 17 ----------------- sound/soc/codecs/wm8960.c | 17 ----------------- sound/soc/codecs/wm8961.c | 17 ----------------- sound/soc/codecs/wm8988.c | 34 ---------------------------------- sound/soc/codecs/wm9081.c | 17 ----------------- sound/soc/soc-core.c | 39 --------------------------------------- 17 files changed, 383 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ca1e24a8f12a..59bb16d033d6 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -802,22 +802,6 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); * and all registers are written back to the hardware when resuming. */ -static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) -{ - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; - - return snd_soc_suspend_device(codec->dev); -} - -static int cs4270_i2c_resume(struct i2c_client *client) -{ - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; - - return snd_soc_resume_device(codec->dev); -} - static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { struct snd_soc_codec *codec = cs4270_codec; @@ -853,8 +837,6 @@ static int cs4270_soc_resume(struct platform_device *pdev) return snd_soc_write(codec, CS4270_PWRCTL, reg); } #else -#define cs4270_i2c_suspend NULL -#define cs4270_i2c_resume NULL #define cs4270_soc_suspend NULL #define cs4270_soc_resume NULL #endif /* CONFIG_PM */ @@ -873,8 +855,6 @@ static struct i2c_driver cs4270_i2c_driver = { .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, - .suspend = cs4270_i2c_suspend, - .resume = cs4270_i2c_resume, }; /* diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 72abc5a6d8d8..714114b50d18 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1680,21 +1680,6 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM -static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m) -{ - return snd_soc_suspend_device(&pdev->dev); -} - -static int wm8350_codec_resume(struct platform_device *pdev) -{ - return snd_soc_resume_device(&pdev->dev); -} -#else -#define wm8350_codec_suspend NULL -#define wm8350_codec_resume NULL -#endif - static struct platform_driver wm8350_codec_driver = { .driver = { .name = "wm8350-codec", @@ -1702,8 +1687,6 @@ static struct platform_driver wm8350_codec_driver = { }, .probe = wm8350_codec_probe, .remove = __devexit_p(wm8350_codec_remove), - .suspend = wm8350_codec_suspend, - .resume = wm8350_codec_resume, }; static __init int wm8350_init(void) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 9cb8e50f0fbb..bd7eecba20fe 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1559,21 +1559,6 @@ static int __exit wm8400_codec_remove(struct platform_device *dev) return 0; } -#ifdef CONFIG_PM -static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg) -{ - return snd_soc_suspend_device(&pdev->dev); -} - -static int wm8400_pdev_resume(struct platform_device *pdev) -{ - return snd_soc_resume_device(&pdev->dev); -} -#else -#define wm8400_pdev_suspend NULL -#define wm8400_pdev_resume NULL -#endif - static struct platform_driver wm8400_codec_driver = { .driver = { .name = "wm8400-codec", @@ -1581,8 +1566,6 @@ static struct platform_driver wm8400_codec_driver = { }, .probe = wm8400_codec_probe, .remove = __exit_p(wm8400_codec_remove), - .suspend = wm8400_pdev_suspend, - .resume = wm8400_pdev_resume, }; static int __init wm8400_codec_init(void) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 25870a4652fb..268cab21c2cc 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -638,21 +638,6 @@ static __devexit int wm8523_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8523_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8523_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8523_i2c_suspend NULL -#define wm8523_i2c_resume NULL -#endif - static const struct i2c_device_id wm8523_i2c_id[] = { { "wm8523", 0 }, { } @@ -666,8 +651,6 @@ static struct i2c_driver wm8523_i2c_driver = { }, .probe = wm8523_i2c_probe, .remove = __devexit_p(wm8523_i2c_remove), - .suspend = wm8523_i2c_suspend, - .resume = wm8523_i2c_resume, .id_table = wm8523_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 3be5c0b2552c..a09b23e03664 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -961,21 +961,6 @@ static int wm8580_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8580_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8580_i2c_suspend NULL -#define wm8580_i2c_resume NULL -#endif - static const struct i2c_device_id wm8580_i2c_id[] = { { "wm8580", 0 }, { } @@ -989,8 +974,6 @@ static struct i2c_driver wm8580_i2c_driver = { }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, - .suspend = wm8580_i2c_suspend, - .resume = wm8580_i2c_resume, .id_table = wm8580_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 90ec8c58e2f4..54189fbf9e93 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -548,21 +548,6 @@ static int __devexit wm8711_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8711_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8711_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8711_spi_suspend NULL -#define wm8711_spi_resume NULL -#endif - static struct spi_driver wm8711_spi_driver = { .driver = { .name = "wm8711", @@ -570,8 +555,6 @@ static struct spi_driver wm8711_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8711_spi_probe, - .suspend = wm8711_spi_suspend, - .resume = wm8711_spi_resume, .remove = __devexit_p(wm8711_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index d3fd4f28d96e..0e59219a59f4 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -623,21 +623,6 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8731_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8731_spi_suspend NULL -#define wm8731_spi_resume NULL -#endif - static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", @@ -645,8 +630,6 @@ static struct spi_driver wm8731_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8731_spi_probe, - .suspend = wm8731_spi_suspend, - .resume = wm8731_spi_resume, .remove = __devexit_p(wm8731_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -679,21 +662,6 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8731_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8731_i2c_suspend NULL -#define wm8731_i2c_resume NULL -#endif - static const struct i2c_device_id wm8731_i2c_id[] = { { "wm8731", 0 }, { } @@ -707,8 +675,6 @@ static struct i2c_driver wm8731_i2c_driver = { }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), - .suspend = wm8731_i2c_suspend, - .resume = wm8731_i2c_resume, .id_table = wm8731_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 9b27efb052fe..8f7305257d29 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1767,21 +1767,6 @@ static int wm8753_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8753_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8753_i2c_suspend NULL -#define wm8753_i2c_resume NULL -#endif - static const struct i2c_device_id wm8753_i2c_id[] = { { "wm8753", 0 }, { } @@ -1795,8 +1780,6 @@ static struct i2c_driver wm8753_i2c_driver = { }, .probe = wm8753_i2c_probe, .remove = wm8753_i2c_remove, - .suspend = wm8753_i2c_suspend, - .resume = wm8753_i2c_resume, .id_table = wm8753_i2c_id, }; #endif @@ -1852,22 +1835,6 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8753_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} - -#else -#define wm8753_spi_suspend NULL -#define wm8753_spi_resume NULL -#endif - static struct spi_driver wm8753_spi_driver = { .driver = { .name = "wm8753", @@ -1876,8 +1843,6 @@ static struct spi_driver wm8753_spi_driver = { }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), - .suspend = wm8753_spi_suspend, - .resume = wm8753_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index a9829aa26e53..a0bbb28eed75 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -616,21 +616,6 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8776_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8776_spi_suspend NULL -#define wm8776_spi_resume NULL -#endif - static struct spi_driver wm8776_spi_driver = { .driver = { .name = "wm8776", @@ -638,8 +623,6 @@ static struct spi_driver wm8776_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8776_spi_probe, - .suspend = wm8776_spi_suspend, - .resume = wm8776_spi_resume, .remove = __devexit_p(wm8776_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -673,21 +656,6 @@ static __devexit int wm8776_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8776_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8776_i2c_suspend NULL -#define wm8776_i2c_resume NULL -#endif - static const struct i2c_device_id wm8776_i2c_id[] = { { "wm8776", 0 }, { } @@ -701,8 +669,6 @@ static struct i2c_driver wm8776_i2c_driver = { }, .probe = wm8776_i2c_probe, .remove = __devexit_p(wm8776_i2c_remove), - .suspend = wm8776_i2c_suspend, - .resume = wm8776_i2c_resume, .id_table = wm8776_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 882604ef768c..b48804b5cacd 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1312,21 +1312,6 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8900_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8900_i2c_suspend NULL -#define wm8900_i2c_resume NULL -#endif - static const struct i2c_device_id wm8900_i2c_id[] = { { "wm8900", 0 }, { } @@ -1340,8 +1325,6 @@ static struct i2c_driver wm8900_i2c_driver = { }, .probe = wm8900_i2c_probe, .remove = __devexit_p(wm8900_i2c_remove), - .suspend = wm8900_i2c_suspend, - .resume = wm8900_i2c_resume, .id_table = wm8900_i2c_id, }; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index fe1307b500cf..94cdb8130415 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1655,21 +1655,6 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8903_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8903_i2c_suspend NULL -#define wm8903_i2c_resume NULL -#endif - /* i2c codec control layer */ static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, @@ -1684,8 +1669,6 @@ static struct i2c_driver wm8903_i2c_driver = { }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), - .suspend = wm8903_i2c_suspend, - .resume = wm8903_i2c_resume, .id_table = wm8903_i2c_id, }; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 1685cfb993c6..8d4fd3c08c09 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -877,21 +877,6 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8940_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8940_i2c_suspend NULL -#define wm8940_i2c_resume NULL -#endif - static const struct i2c_device_id wm8940_i2c_id[] = { { "wm8940", 0 }, { } @@ -905,8 +890,6 @@ static struct i2c_driver wm8940_i2c_driver = { }, .probe = wm8940_i2c_probe, .remove = __devexit_p(wm8940_i2c_remove), - .suspend = wm8940_i2c_suspend, - .resume = wm8940_i2c_resume, .id_table = wm8940_i2c_id, }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 416fb3c17018..b9b096a85396 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -883,21 +883,6 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8960_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8960_i2c_suspend NULL -#define wm8960_i2c_resume NULL -#endif - static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -911,8 +896,6 @@ static struct i2c_driver wm8960_i2c_driver = { }, .probe = wm8960_i2c_probe, .remove = __devexit_p(wm8960_i2c_remove), - .suspend = wm8960_i2c_suspend, - .resume = wm8960_i2c_resume, .id_table = wm8960_i2c_id, }; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 503032085899..b5c6f2cd5ae2 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1206,21 +1206,6 @@ static __devexit int wm8961_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8961_i2c_suspend(struct i2c_client *client, pm_message_t state) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8961_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8961_i2c_suspend NULL -#define wm8961_i2c_resume NULL -#endif - static const struct i2c_device_id wm8961_i2c_id[] = { { "wm8961", 0 }, { } @@ -1234,8 +1219,6 @@ static struct i2c_driver wm8961_i2c_driver = { }, .probe = wm8961_i2c_probe, .remove = __devexit_p(wm8961_i2c_remove), - .suspend = wm8961_i2c_suspend, - .resume = wm8961_i2c_resume, .id_table = wm8961_i2c_id, }; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 3f530f8a972a..d8d8f68b81ea 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -944,21 +944,6 @@ static int wm8988_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8988_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8988_i2c_suspend NULL -#define wm8988_i2c_resume NULL -#endif - static const struct i2c_device_id wm8988_i2c_id[] = { { "wm8988", 0 }, { } @@ -972,8 +957,6 @@ static struct i2c_driver wm8988_i2c_driver = { }, .probe = wm8988_i2c_probe, .remove = wm8988_i2c_remove, - .suspend = wm8988_i2c_suspend, - .resume = wm8988_i2c_resume, .id_table = wm8988_i2c_id, }; #endif @@ -1006,21 +989,6 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8988_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8988_spi_suspend NULL -#define wm8988_spi_resume NULL -#endif - static struct spi_driver wm8988_spi_driver = { .driver = { .name = "wm8988", @@ -1029,8 +997,6 @@ static struct spi_driver wm8988_spi_driver = { }, .probe = wm8988_spi_probe, .remove = __devexit_p(wm8988_spi_remove), - .suspend = wm8988_spi_suspend, - .resume = wm8988_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 686e5aa97206..4cb6b104b729 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1452,21 +1452,6 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm9081_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm9081_i2c_suspend NULL -#define wm9081_i2c_resume NULL -#endif - static const struct i2c_device_id wm9081_i2c_id[] = { { "wm9081", 0 }, { } @@ -1480,8 +1465,6 @@ static struct i2c_driver wm9081_i2c_driver = { }, .probe = wm9081_i2c_probe, .remove = __devexit_p(wm9081_i2c_remove), - .suspend = wm9081_i2c_suspend, - .resume = wm9081_i2c_resume, .id_table = wm9081_i2c_id, }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1dec9d21c55e..fa0da3cac705 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -790,45 +790,6 @@ static int soc_resume(struct device *dev) return 0; } - -/** - * snd_soc_suspend_device: Notify core of device suspend - * - * @dev: Device being suspended. - * - * In order to ensure that the entire audio subsystem is suspended in a - * coordinated fashion ASoC devices should suspend themselves when - * called by ASoC. When the standard kernel suspend process asks the - * device to suspend it should call this function to initiate a suspend - * of the entire ASoC card. - * - * \note Currently this function is stubbed out. - */ -int snd_soc_suspend_device(struct device *dev) -{ - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_suspend_device); - -/** - * snd_soc_resume_device: Notify core of device resume - * - * @dev: Device being resumed. - * - * In order to ensure that the entire audio subsystem is resumed in a - * coordinated fashion ASoC devices should resume themselves when called - * by ASoC. When the standard kernel resume process asks the device - * to resume it should call this function. Once all the components of - * the card have notified that they are ready to be resumed the card - * will be resumed. - * - * \note Currently this function is stubbed out. - */ -int snd_soc_resume_device(struct device *dev) -{ - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_resume_device); #else #define soc_suspend NULL #define soc_resume NULL -- cgit v1.2.2 From 640fb39e386a0dac9014e5b8a512de0950e30288 Mon Sep 17 00:00:00 2001 From: Igor Grinberg Date: Wed, 14 Oct 2009 09:20:26 +0200 Subject: ASoC: finally enable support for eXeda and CM-X300 Signed-off-by: Igor Grinberg Signed-off-by: Mike Rapoport CC: Mark Brown CC: alsa-devel@alsa-project.org Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181bb340..d4f4031afa33 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -90,7 +90,8 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && MACH_EM_X270 + depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ + MACH_CM_X300) select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help -- cgit v1.2.2 From c8bf93f0fe8c5a509a29e30f3bac823fa0f6d96e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 15 Oct 2009 09:03:56 +0300 Subject: ASoC: Codec driver for Texas Instruments tlv320dac33 codec Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320dac33.c | 1237 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320dac33.h | 267 +++++++++ 4 files changed, 1510 insertions(+) create mode 100644 sound/soc/codecs/tlv320dac33.c create mode 100644 sound/soc/codecs/tlv320dac33.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fab01c991828..d30fce71cfe8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -30,6 +30,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C @@ -142,6 +143,9 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate +config SND_SOC_TLV320DAC33 + tristate + config SND_SOC_TWL4030 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2f14391b96f9..8f519ee9600d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -17,6 +17,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -70,6 +71,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c new file mode 100644 index 000000000000..3ca8934fc26c --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.c @@ -0,0 +1,1237 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi + * + * Copyright: (C) 2009 Nokia Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "tlv320dac33.h" + +#define DAC33_BUFFER_SIZE_BYTES 24576 /* bytes, 12288 16 bit words, + * 6144 stereo */ +#define DAC33_BUFFER_SIZE_SAMPLES 6144 + +#define NSAMPLE_MAX 5700 + +#define LATENCY_TIME_MS 20 + +static struct snd_soc_codec *tlv320dac33_codec; + +enum dac33_state { + DAC33_IDLE = 0, + DAC33_PREFILL, + DAC33_PLAYBACK, + DAC33_FLUSH, +}; + +struct tlv320dac33_priv { + struct mutex mutex; + struct workqueue_struct *dac33_wq; + struct work_struct work; + struct snd_soc_codec codec; + int power_gpio; + int chip_power; + int irq; + unsigned int refclk; + + unsigned int alarm_threshold; /* set to be half of LATENCY_TIME_MS */ + unsigned int nsample_min; /* nsample should not be lower than + * this */ + unsigned int nsample_max; /* nsample should not be higher than + * this */ + unsigned int nsample_switch; /* Use FIFO or bypass FIFO switch */ + unsigned int nsample; /* burst read amount from host */ + + enum dac33_state state; +}; + +static const u8 dac33_reg[DAC33_CACHEREGNUM] = { +0x00, 0x00, 0x00, 0x00, /* 0x00 - 0x03 */ +0x00, 0x00, 0x00, 0x00, /* 0x04 - 0x07 */ +0x00, 0x00, 0x00, 0x00, /* 0x08 - 0x0b */ +0x00, 0x00, 0x00, 0x00, /* 0x0c - 0x0f */ +0x00, 0x00, 0x00, 0x00, /* 0x10 - 0x13 */ +0x00, 0x00, 0x00, 0x00, /* 0x14 - 0x17 */ +0x00, 0x00, 0x00, 0x00, /* 0x18 - 0x1b */ +0x00, 0x00, 0x00, 0x00, /* 0x1c - 0x1f */ +0x00, 0x00, 0x00, 0x00, /* 0x20 - 0x23 */ +0x00, 0x00, 0x00, 0x00, /* 0x24 - 0x27 */ +0x00, 0x00, 0x00, 0x00, /* 0x28 - 0x2b */ +0x00, 0x00, 0x00, 0x80, /* 0x2c - 0x2f */ +0x80, 0x00, 0x00, 0x00, /* 0x30 - 0x33 */ +0x00, 0x00, 0x00, 0x00, /* 0x34 - 0x37 */ +0x00, 0x00, /* 0x38 - 0x39 */ +/* Registers 0x3a - 0x3f are reserved */ + 0x00, 0x00, /* 0x3a - 0x3b */ +0x00, 0x00, 0x00, 0x00, /* 0x3c - 0x3f */ + +0x00, 0x00, 0x00, 0x00, /* 0x40 - 0x43 */ +0x00, 0x80, /* 0x44 - 0x45 */ +/* Registers 0x46 - 0x47 are reserved */ + 0x80, 0x80, /* 0x46 - 0x47 */ + +0x80, 0x00, 0x00, /* 0x48 - 0x4a */ +/* Registers 0x4b - 0x7c are reserved */ + 0x00, /* 0x4b */ +0x00, 0x00, 0x00, 0x00, /* 0x4c - 0x4f */ +0x00, 0x00, 0x00, 0x00, /* 0x50 - 0x53 */ +0x00, 0x00, 0x00, 0x00, /* 0x54 - 0x57 */ +0x00, 0x00, 0x00, 0x00, /* 0x58 - 0x5b */ +0x00, 0x00, 0x00, 0x00, /* 0x5c - 0x5f */ +0x00, 0x00, 0x00, 0x00, /* 0x60 - 0x63 */ +0x00, 0x00, 0x00, 0x00, /* 0x64 - 0x67 */ +0x00, 0x00, 0x00, 0x00, /* 0x68 - 0x6b */ +0x00, 0x00, 0x00, 0x00, /* 0x6c - 0x6f */ +0x00, 0x00, 0x00, 0x00, /* 0x70 - 0x73 */ +0x00, 0x00, 0x00, 0x00, /* 0x74 - 0x77 */ +0x00, 0x00, 0x00, 0x00, /* 0x78 - 0x7b */ +0x00, /* 0x7c */ + + 0xda, 0x33, 0x03, /* 0x7d - 0x7f */ +}; + +/* Register read and write */ +static inline unsigned int dac33_read_reg_cache(struct snd_soc_codec *codec, + unsigned reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return 0; + + return cache[reg]; +} + +static inline void dac33_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return; + + cache[reg] = value; +} + +static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int val; + + *value = reg & 0xff; + + /* If powered off, return the cached value */ + if (dac33->chip_power) { + val = i2c_smbus_read_byte_data(codec->control_data, value[0]); + if (val < 0) { + dev_err(codec->dev, "Read failed (%d)\n", val); + value[0] = dac33_read_reg_cache(codec, reg); + } else { + value[0] = val; + dac33_write_reg_cache(codec, reg, val); + } + } else { + value[0] = dac33_read_reg_cache(codec, reg); + } + + return 0; +} + +static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[2]; + int ret = 0; + + /* + * data is + * D15..D8 dac33 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + if (dac33->chip_power) { + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +static int dac33_write_locked(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; + + mutex_lock(&dac33->mutex); + ret = dac33_write(codec, reg, value); + mutex_unlock(&dac33->mutex); + + return ret; +} + +#define DAC33_I2C_ADDR_AUTOINC 0x80 +static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[3]; + int ret = 0; + + /* + * data is + * D23..D16 dac33 register offset + * D15..D8 register data MSB + * D7...D0 register data LSB + */ + data[0] = reg & 0xff; + data[1] = (value >> 8) & 0xff; + data[2] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + dac33_write_reg_cache(codec, data[0] + 1, data[2]); + + if (dac33->chip_power) { + /* We need to set autoincrement mode for 16 bit writes */ + data[0] |= DAC33_I2C_ADDR_AUTOINC; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret != 3) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +static void dac33_restore_regs(struct snd_soc_codec *codec) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 *cache = codec->reg_cache; + u8 data[2]; + int i, ret; + + if (!dac33->chip_power) + return; + + for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) { + data[0] = i; + data[1] = cache[i]; + /* Skip the read only registers */ + if ((i >= DAC33_INT_OSC_STATUS && + i <= DAC33_INT_OSC_FREQ_RAT_READ_B) || + (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) || + i == DAC33_DAC_STATUS_FLAGS || + i == DAC33_SRC_EST_REF_CLK_RATIO_A || + i == DAC33_SRC_EST_REF_CLK_RATIO_B) + continue; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } +} + +static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) +{ + u8 reg; + + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + if (power) + reg |= DAC33_PDNALLB; + else + reg &= ~DAC33_PDNALLB; + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + +static void dac33_hard_power(struct snd_soc_codec *codec, int power) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + + mutex_lock(&dac33->mutex); + if (power) { + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 1); + dac33->chip_power = 1; + /* Restore registers */ + dac33_restore_regs(codec); + } + dac33_soft_power(codec, 1); + } else { + dac33_soft_power(codec, 0); + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 0); + dac33->chip_power = 0; + } + } + mutex_unlock(&dac33->mutex); + +} + +static int dac33_get_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample; + + return 0; +} + +static int dac33_set_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < dac33->nsample_min || + ucontrol->value.integer.value[0] > dac33->nsample_max) + ret = -EINVAL; + else + dac33->nsample = ucontrol->value.integer.value[0]; + + return ret; +} + +static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample_switch; + + return 0; +} + +static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample_switch == ucontrol->value.integer.value[0]) + return 0; + /* Do not allow changes while stream is running*/ + if (codec->active) + return -EPERM; + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 1) + ret = -EINVAL; + else + dac33->nsample_switch = ucontrol->value.integer.value[0]; + + return ret; +} + +/* + * DACL/R digital volume control: + * from 0 dB to -63.5 in 0.5 dB steps + * Need to be inverted later on: + * 0x00 == 0 dB + * 0x7f == -63.5 dB + */ +static DECLARE_TLV_DB_SCALE(dac_digivol_tlv, -6350, 50, 0); + +static const struct snd_kcontrol_new dac33_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC Digital Playback Volume", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, + 0, 0x7f, 1, dac_digivol_tlv), + SOC_DOUBLE_R("DAC Digital Playback Switch", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, 7, 1, 1), + SOC_DOUBLE_R("Line to Line Out Volume", + DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), +}; + +static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { + SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, + dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0, + dac33_get_nsample_switch, dac33_set_nsample_switch), +}; + +/* Analog bypass */ +static const struct snd_kcontrol_new dac33_dapm_abypassl_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1); + +static const struct snd_kcontrol_new dac33_dapm_abypassr_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1); + +static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("LEFT_LO"), + SND_SOC_DAPM_OUTPUT("RIGHT_LO"), + + SND_SOC_DAPM_INPUT("LINEL"), + SND_SOC_DAPM_INPUT("LINER"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", DAC33_LDAC_PWR_CTRL, 2, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", DAC33_RDAC_PWR_CTRL, 2, 0), + + /* Analog bypass */ + SND_SOC_DAPM_SWITCH("Analog Left Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassl_control), + SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassr_control), + + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Analog bypass */ + {"Analog Left Bypass", "Switch", "LINEL"}, + {"Analog Right Bypass", "Switch", "LINER"}, + + {"Output Left Amp Power", NULL, "DACL"}, + {"Output Right Amp Power", NULL, "DACR"}, + + {"Output Left Amp Power", NULL, "Analog Left Bypass"}, + {"Output Right Amp Power", NULL, "Analog Right Bypass"}, + + /* output */ + {"LEFT_LO", NULL, "Output Left Amp Power"}, + {"RIGHT_LO", NULL, "Output Right Amp Power"}, +}; + +static int dac33_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dac33_dapm_widgets, + ARRAY_SIZE(dac33_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int dac33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + dac33_soft_power(codec, 1); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) + dac33_hard_power(codec, 1); + dac33_soft_power(codec, 0); + break; + case SND_SOC_BIAS_OFF: + dac33_hard_power(codec, 0); + break; + } + codec->bias_level = level; + + return 0; +} + +static void dac33_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + u8 reg; + + dac33 = container_of(work, struct tlv320dac33_priv, work); + codec = &dac33->codec; + + mutex_lock(&dac33->mutex); + switch (dac33->state) { + case DAC33_PREFILL: + dac33->state = DAC33_PLAYBACK; + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + break; + case DAC33_PLAYBACK: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + break; + case DAC33_IDLE: + break; + case DAC33_FLUSH: + dac33->state = DAC33_IDLE; + /* Mask all interrupts from dac33 */ + dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0); + + /* flush fifo */ + reg = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + reg |= DAC33_FIFOFLUSH; + dac33_write(codec, DAC33_FIFO_CTRL_A, reg); + break; + } + mutex_unlock(&dac33->mutex); +} + +static irqreturn_t dac33_interrupt_handler(int irq, void *dev) +{ + struct snd_soc_codec *codec = dev; + struct tlv320dac33_priv *dac33 = codec->private_data; + + queue_work(dac33->dac33_wq, &dac33->work); + + return IRQ_HANDLED; +} + +static void dac33_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int pwr_ctrl; + + /* Stop pending workqueue */ + if (dac33->nsample_switch) + cancel_work_sync(&dac33->work); + + mutex_lock(&dac33->mutex); + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + mutex_unlock(&dac33->mutex); +} + +static void dac33_oscwait(struct snd_soc_codec *codec) +{ + int timeout = 20; + u8 reg; + + do { + msleep(1); + dac33_read(codec, DAC33_INT_OSC_STATUS, ®); + } while (((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) && timeout--); + if ((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) + dev_err(codec->dev, + "internal oscillator calibration failed\n"); +} + +static int dac33_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Check parameters for validity */ + switch (params_rate(params)) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + params_rate(params)); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + params_format(params)); + return -EINVAL; + } + + return 0; +} + +#define CALC_OSCSET(rate, refclk) ( \ + ((((rate * 10000) / refclk) * 4096) + 5000) / 10000) +#define CALC_RATIOSET(rate, refclk) ( \ + ((((refclk * 100000) / rate) * 16384) + 50000) / 100000) + +/* + * tlv320dac33 is strict on the sequence of the register writes, if the register + * writes happens in different order, than dac33 might end up in unknown state. + * Use the known, working sequence of register writes to initialize the dac33. + */ +static int dac33_prepare_chip(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; + u8 aictrl_a, fifoctrl_a; + + switch (substream->runtime->rate) { + case 44100: + case 48000: + oscset = CALC_OSCSET(substream->runtime->rate, dac33->refclk); + ratioset = CALC_RATIOSET(substream->runtime->rate, + dac33->refclk); + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + substream->runtime->rate); + return -EINVAL; + } + + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK); + fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + fifoctrl_a &= ~DAC33_WIDTH; + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + aictrl_a |= (DAC33_NCYCL_16 | DAC33_WLEN_16); + fifoctrl_a |= DAC33_WIDTH; + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + substream->runtime->format); + return -EINVAL; + } + + mutex_lock(&dac33->mutex); + dac33_soft_power(codec, 1); + + reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + dac33_write(codec, DAC33_INT_OSC_CTRL, reg_tmp); + + /* Write registers 0x08 and 0x09 (MSB, LSB) */ + dac33_write16(codec, DAC33_INT_OSC_FREQ_RAT_A, oscset); + + /* calib time: 128 is a nice number ;) */ + dac33_write(codec, DAC33_CALIB_TIME, 128); + + /* adjustment treshold & step */ + dac33_write(codec, DAC33_INT_OSC_CTRL_B, DAC33_ADJTHRSHLD(2) | + DAC33_ADJSTEP(1)); + + /* div=4 / gain=1 / div */ + dac33_write(codec, DAC33_INT_OSC_CTRL_C, DAC33_REFDIV(4)); + + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl |= DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB; + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + + dac33_oscwait(codec); + + if (dac33->nsample_switch) { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ + + /* Write registers 0x34 and 0x35 (MSB, LSB) */ + dac33_write16(codec, DAC33_SRC_REF_CLK_RATIO_A, ratioset); + + /* Set interrupts to high active */ + dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH); + + dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, + DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); + dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); + } else { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP); + dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ + } + + if (dac33->nsample_switch) + fifoctrl_a &= ~DAC33_FBYPAS; + else + fifoctrl_a |= DAC33_FBYPAS; + dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); + + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + if (dac33->nsample_switch) + reg_tmp &= ~DAC33_BCLKON; + else + reg_tmp |= DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); + + if (dac33->nsample_switch) { + /* 20: BCLK divide ratio */ + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + + dac33_write16(codec, DAC33_ATHR_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + } else { + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + } + + mutex_unlock(&dac33->mutex); + + return 0; +} + +static void dac33_calculate_times(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int nsample_limit; + + /* Number of samples (16bit, stereo) in one period */ + dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4; + + /* Number of samples (16bit, stereo) in ALSA buffer */ + dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4; + /* Subtract one period from the total */ + dac33->nsample_max -= dac33->nsample_min; + + /* Number of samples for LATENCY_TIME_MS / 2 */ + dac33->alarm_threshold = substream->runtime->rate / + (1000 / (LATENCY_TIME_MS / 2)); + + /* Find and fix up the lowest nsmaple limit */ + nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS); + + if (dac33->nsample_min < nsample_limit) + dac33->nsample_min = nsample_limit; + + if (dac33->nsample < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + + /* + * Find and fix up the highest nsmaple limit + * In order to not overflow the DAC33 buffer substract the + * alarm_threshold value from the size of the DAC33 buffer + */ + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold; + + if (dac33->nsample_max > nsample_limit) + dac33->nsample_max = nsample_limit; + + if (dac33->nsample > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; +} + +static int dac33_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + dac33_calculate_times(substream); + dac33_prepare_chip(substream); + + return 0; +} + +static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (dac33->nsample_switch) { + dac33->state = DAC33_PREFILL; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (dac33->nsample_switch) { + dac33->state = DAC33_FLUSH; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int dac33_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 ioc_reg, asrcb_reg; + + ioc_reg = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + asrcb_reg = dac33_read_reg_cache(codec, DAC33_ASRC_CTRL_B); + switch (clk_id) { + case TLV320DAC33_MCLK: + ioc_reg |= DAC33_REFSEL; + asrcb_reg |= DAC33_SRCREFSEL; + break; + case TLV320DAC33_SLEEPCLK: + ioc_reg &= ~DAC33_REFSEL; + asrcb_reg &= ~DAC33_SRCREFSEL; + break; + default: + dev_err(codec->dev, "Invalid clock ID (%d)\n", clk_id); + break; + } + dac33->refclk = freq; + + dac33_write_reg_cache(codec, DAC33_INT_OSC_CTRL, ioc_reg); + dac33_write_reg_cache(codec, DAC33_ASRC_CTRL_B, asrcb_reg); + + return 0; +} + +static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 aictrl_a, aictrl_b; + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* Codec Master */ + aictrl_a |= (DAC33_MSBCLK | DAC33_MSWCLK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* Codec Slave */ + aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); + break; + default: + return -EINVAL; + } + + aictrl_a &= ~DAC33_AFMT_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aictrl_a |= DAC33_AFMT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; + aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */ + break; + case SND_SOC_DAIFMT_DSP_B: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + aictrl_a |= DAC33_AFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + aictrl_a |= DAC33_AFMT_LEFT_J; + break; + default: + dev_err(codec->dev, "Unsupported format (%u)\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); + + return 0; +} + +static void dac33_init_chip(struct snd_soc_codec *codec) +{ + /* 44-46: DAC Control Registers */ + /* A : DAC sample rate Fsref/1.5 */ + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1)); + /* B : DAC src=normal, not muted */ + dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | + DAC33_DACSRCL_LEFT); + /* C : (defaults) */ + dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); + + /* 64-65 : L&R DAC power control + Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ + dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + + /* 73 : volume soft stepping control, + clock source = internal osc (?) */ + dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); + + /* 66 : LOP/LOM Modes */ + dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); + + /* 68 : LOM inverted from LOP */ + dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); + + dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); +} + +static int dac33_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + int ret = 0; + + BUG_ON(!tlv320dac33_codec); + + codec = tlv320dac33_codec; + socdev->card->codec = codec; + dac33 = codec->private_data; + + /* Power up the codec */ + dac33_hard_power(codec, 1); + /* Set default configuration */ + dac33_init_chip(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + goto pcm_err; + } + + snd_soc_add_controls(codec, dac33_snd_controls, + ARRAY_SIZE(dac33_snd_controls)); + /* Only add the nSample controls, if we have valid IRQ number */ + if (dac33->irq >= 0) + snd_soc_add_controls(codec, dac33_nsample_snd_controls, + ARRAY_SIZE(dac33_nsample_snd_controls)); + + dac33_add_widgets(codec); + + /* power on device */ + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + goto card_err; + } + + return 0; +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + dac33_hard_power(codec, 0); + return ret; +} + +static int dac33_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +static int dac33_soc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int dac33_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dac33_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_tlv320dac33 = { + .probe = dac33_soc_probe, + .remove = dac33_soc_remove, + .suspend = dac33_soc_suspend, + .resume = dac33_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33); + +#define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) +#define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops dac33_dai_ops = { + .shutdown = dac33_shutdown, + .hw_params = dac33_hw_params, + .prepare = dac33_pcm_prepare, + .trigger = dac33_pcm_trigger, + .set_sysclk = dac33_set_dai_sysclk, + .set_fmt = dac33_set_dai_fmt, +}; + +struct snd_soc_dai dac33_dai = { + .name = "tlv320dac33", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DAC33_RATES, + .formats = DAC33_FORMATS,}, + .ops = &dac33_dai_ops, +}; +EXPORT_SYMBOL_GPL(dac33_dai); + +static int dac33_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_priv *dac33; + struct snd_soc_codec *codec; + int ret = 0; + + if (client->dev.platform_data == NULL) { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } + pdata = client->dev.platform_data; + + dac33 = kzalloc(sizeof(struct tlv320dac33_priv), GFP_KERNEL); + if (dac33 == NULL) + return -ENOMEM; + + codec = &dac33->codec; + codec->private_data = dac33; + codec->control_data = client; + + mutex_init(&codec->mutex); + mutex_init(&dac33->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "tlv320dac33"; + codec->owner = THIS_MODULE; + codec->read = dac33_read_reg_cache; + codec->write = dac33_write_locked; + codec->hw_write = (hw_write_t) i2c_master_send; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = dac33_set_bias_level; + codec->dai = &dac33_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(dac33_reg); + codec->reg_cache = kmemdup(dac33_reg, ARRAY_SIZE(dac33_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_reg; + } + + i2c_set_clientdata(client, dac33); + + dac33->power_gpio = pdata->power_gpio; + dac33->irq = client->irq; + dac33->nsample = NSAMPLE_MAX; + /* Disable FIFO use by default */ + dac33->nsample_switch = 0; + + tlv320dac33_codec = codec; + + codec->dev = &client->dev; + dac33_dai.dev = codec->dev; + + /* Check if the reset GPIO number is valid and request it */ + if (dac33->power_gpio >= 0) { + ret = gpio_request(dac33->power_gpio, "tlv320dac33 reset"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to request reset GPIO (%d)\n", + dac33->power_gpio); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(codec); + goto error_gpio; + } + gpio_direction_output(dac33->power_gpio, 0); + } else { + dac33->chip_power = 1; + } + + /* Check if the IRQ number is valid and request it */ + if (dac33->irq >= 0) { + ret = request_irq(dac33->irq, dac33_interrupt_handler, + IRQF_TRIGGER_RISING | IRQF_DISABLED, + codec->name, codec); + if (ret < 0) { + dev_err(codec->dev, "Could not request IRQ%d (%d)\n", + dac33->irq, ret); + dac33->irq = -1; + } + if (dac33->irq != -1) { + /* Setup work queue */ + dac33->dac33_wq = create_rt_workqueue("tlv320dac33"); + if (dac33->dac33_wq == NULL) { + free_irq(dac33->irq, &dac33->codec); + ret = -ENOMEM; + goto error_wq; + } + + INIT_WORK(&dac33->work, dac33_work); + } + } + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dai(&dac33_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } + + /* Shut down the codec for now */ + dac33_hard_power(codec, 0); + + return ret; + +error_codec: + if (dac33->irq >= 0) { + free_irq(dac33->irq, &dac33->codec); + destroy_workqueue(dac33->dac33_wq); + } +error_wq: + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); +error_gpio: + kfree(codec->reg_cache); +error_reg: + tlv320dac33_codec = NULL; + kfree(dac33); + + return ret; +} + +static int dac33_i2c_remove(struct i2c_client *client) +{ + struct tlv320dac33_priv *dac33; + + dac33 = i2c_get_clientdata(client); + dac33_hard_power(&dac33->codec, 0); + + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); + if (dac33->irq >= 0) + free_irq(dac33->irq, &dac33->codec); + + destroy_workqueue(dac33->dac33_wq); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(&dac33->codec); + kfree(dac33->codec.reg_cache); + kfree(dac33); + tlv320dac33_codec = NULL; + + return 0; +} + +static const struct i2c_device_id tlv320dac33_i2c_id[] = { + { + .name = "tlv320dac33", + .driver_data = 0, + }, + { }, +}; + +static struct i2c_driver tlv320dac33_i2c_driver = { + .driver = { + .name = "tlv320dac33", + .owner = THIS_MODULE, + }, + .probe = dac33_i2c_probe, + .remove = __devexit_p(dac33_i2c_remove), + .id_table = tlv320dac33_i2c_id, +}; + +static int __init dac33_module_init(void) +{ + int r; + r = i2c_add_driver(&tlv320dac33_i2c_driver); + if (r < 0) { + printk(KERN_ERR "DAC33: driver registration failed\n"); + return r; + } + return 0; +} +module_init(dac33_module_init); + +static void __exit dac33_module_exit(void) +{ + i2c_del_driver(&tlv320dac33_i2c_driver); +} +module_exit(dac33_module_exit); + + +MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); +MODULE_AUTHOR("Peter Ujfalusi "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h new file mode 100644 index 000000000000..0fedd709028e --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.h @@ -0,0 +1,267 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi + * + * Copyright: (C) 2009 Nokia Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TLV320DAC33_H +#define __TLV320DAC33_H + +#define DAC33_PAGE_SELECT 0x00 +#define DAC33_PWR_CTRL 0x01 +#define DAC33_PLL_CTRL_A 0x02 +#define DAC33_PLL_CTRL_B 0x03 +#define DAC33_PLL_CTRL_C 0x04 +#define DAC33_PLL_CTRL_D 0x05 +#define DAC33_PLL_CTRL_E 0x06 +#define DAC33_INT_OSC_CTRL 0x07 +#define DAC33_INT_OSC_FREQ_RAT_A 0x08 +#define DAC33_INT_OSC_FREQ_RAT_B 0x09 +#define DAC33_INT_OSC_DAC_RATIO_SET 0x0A +#define DAC33_CALIB_TIME 0x0B +#define DAC33_INT_OSC_CTRL_B 0x0C +#define DAC33_INT_OSC_CTRL_C 0x0D +#define DAC33_INT_OSC_STATUS 0x0E +#define DAC33_INT_OSC_DAC_RATIO_READ 0x0F +#define DAC33_INT_OSC_FREQ_RAT_READ_A 0x10 +#define DAC33_INT_OSC_FREQ_RAT_READ_B 0x11 +#define DAC33_SER_AUDIOIF_CTRL_A 0x12 +#define DAC33_SER_AUDIOIF_CTRL_B 0x13 +#define DAC33_SER_AUDIOIF_CTRL_C 0x14 +#define DAC33_FIFO_CTRL_A 0x15 +#define DAC33_UTHR_MSB 0x16 +#define DAC33_UTHR_LSB 0x17 +#define DAC33_ATHR_MSB 0x18 +#define DAC33_ATHR_LSB 0x19 +#define DAC33_LTHR_MSB 0x1A +#define DAC33_LTHR_LSB 0x1B +#define DAC33_PREFILL_MSB 0x1C +#define DAC33_PREFILL_LSB 0x1D +#define DAC33_NSAMPLE_MSB 0x1E +#define DAC33_NSAMPLE_LSB 0x1F +#define DAC33_FIFO_WPTR_MSB 0x20 +#define DAC33_FIFO_WPTR_LSB 0x21 +#define DAC33_FIFO_RPTR_MSB 0x22 +#define DAC33_FIFO_RPTR_LSB 0x23 +#define DAC33_FIFO_DEPTH_MSB 0x24 +#define DAC33_FIFO_DEPTH_LSB 0x25 +#define DAC33_SAMPLES_REMAINING_MSB 0x26 +#define DAC33_SAMPLES_REMAINING_LSB 0x27 +#define DAC33_FIFO_IRQ_FLAG 0x28 +#define DAC33_FIFO_IRQ_MASK 0x29 +#define DAC33_FIFO_IRQ_MODE_A 0x2A +#define DAC33_FIFO_IRQ_MODE_B 0x2B +#define DAC33_DAC_CTRL_A 0x2C +#define DAC33_DAC_CTRL_B 0x2D +#define DAC33_DAC_CTRL_C 0x2E +#define DAC33_LDAC_DIG_VOL_CTRL 0x2F +#define DAC33_RDAC_DIG_VOL_CTRL 0x30 +#define DAC33_DAC_STATUS_FLAGS 0x31 +#define DAC33_ASRC_CTRL_A 0x32 +#define DAC33_ASRC_CTRL_B 0x33 +#define DAC33_SRC_REF_CLK_RATIO_A 0x34 +#define DAC33_SRC_REF_CLK_RATIO_B 0x35 +#define DAC33_SRC_EST_REF_CLK_RATIO_A 0x36 +#define DAC33_SRC_EST_REF_CLK_RATIO_B 0x37 +#define DAC33_INTP_CTRL_A 0x38 +#define DAC33_INTP_CTRL_B 0x39 +/* Registers 0x3A - 0x3F Reserved */ +#define DAC33_LDAC_PWR_CTRL 0x40 +#define DAC33_RDAC_PWR_CTRL 0x41 +#define DAC33_OUT_AMP_CM_CTRL 0x42 +#define DAC33_OUT_AMP_PWR_CTRL 0x43 +#define DAC33_OUT_AMP_CTRL 0x44 +#define DAC33_LINEL_TO_LLO_VOL 0x45 +/* Registers 0x45 - 0x47 Reserved */ +#define DAC33_LINER_TO_RLO_VOL 0x48 +#define DAC33_ANA_VOL_SOFT_STEP_CTRL 0x49 +#define DAC33_OSC_TRIM 0x4A +/* Registers 0x4B - 0x7C Reserved */ +#define DAC33_DEVICE_ID_MSB 0x7D +#define DAC33_DEVICE_ID_LSB 0x7E +#define DAC33_DEVICE_REV_ID 0x7F + +#define DAC33_CACHEREGNUM 128 + +/* Bit definitions */ + +/* DAC33_PWR_CTRL (0x01) */ +#define DAC33_DACRPDNB (0x01 << 0) +#define DAC33_DACLPDNB (0x01 << 1) +#define DAC33_OSCPDNB (0x01 << 2) +#define DAC33_PLLPDNB (0x01 << 3) +#define DAC33_PDNALLB (0x01 << 4) +#define DAC33_SOFT_RESET (0x01 << 7) + +/* DAC33_INT_OSC_CTRL (0x07) */ +#define DAC33_REFSEL (0x01 << 1) + +/* DAC33_INT_OSC_CTRL_B (0x0C) */ +#define DAC33_ADJSTEP(x) (x << 0) +#define DAC33_ADJTHRSHLD(x) (x << 4) + +/* DAC33_INT_OSC_CTRL_C (0x0D) */ +#define DAC33_REFDIV(x) (x << 4) + +/* DAC33_INT_OSC_STATUS (0x0E) */ +#define DAC33_OSCSTATUS_IDLE_CALIB (0x00) +#define DAC33_OSCSTATUS_NORMAL (0x01) +#define DAC33_OSCSTATUS_ADJUSTMENT (0x03) +#define DAC33_OSCSTATUS_NOT_USED (0x02) + +/* DAC33_SER_AUDIOIF_CTRL_A (0x12) */ +#define DAC33_MSWCLK (0x01 << 0) +#define DAC33_MSBCLK (0x01 << 1) +#define DAC33_AFMT_MASK (0x03 << 2) +#define DAC33_AFMT_I2S (0x00 << 2) +#define DAC33_AFMT_DSP (0x01 << 2) +#define DAC33_AFMT_RIGHT_J (0x02 << 2) +#define DAC33_AFMT_LEFT_J (0x03 << 2) +#define DAC33_WLEN_MASK (0x03 << 4) +#define DAC33_WLEN_16 (0x00 << 4) +#define DAC33_WLEN_20 (0x01 << 4) +#define DAC33_WLEN_24 (0x02 << 4) +#define DAC33_WLEN_32 (0x03 << 4) +#define DAC33_NCYCL_MASK (0x03 << 6) +#define DAC33_NCYCL_16 (0x00 << 6) +#define DAC33_NCYCL_20 (0x01 << 6) +#define DAC33_NCYCL_24 (0x02 << 6) +#define DAC33_NCYCL_32 (0x03 << 6) + +/* DAC33_SER_AUDIOIF_CTRL_B (0x13) */ +#define DAC33_DATA_DELAY_MASK (0x03 << 2) +#define DAC33_DATA_DELAY(x) (x << 2) +#define DAC33_BCLKON (0x01 << 5) + +/* DAC33_FIFO_CTRL_A (0x15) */ +#define DAC33_WIDTH (0x01 << 0) +#define DAC33_FBYPAS (0x01 << 1) +#define DAC33_FAUTO (0x01 << 2) +#define DAC33_FIFOFLUSH (0x01 << 3) + +/* + * UTHR, ATHR, LTHR, PREFILL, NSAMPLE (0x16 - 0x1F) + * 13-bit values +*/ +#define DAC33_THRREG(x) (((x) & 0x1FFF) << 3) + +/* DAC33_FIFO_IRQ_MASK (0x29) */ +#define DAC33_MNS (0x01 << 0) +#define DAC33_MPS (0x01 << 1) +#define DAC33_MAT (0x01 << 2) +#define DAC33_MLT (0x01 << 3) +#define DAC33_MUT (0x01 << 4) +#define DAC33_MUF (0x01 << 5) +#define DAC33_MOF (0x01 << 6) + +#define DAC33_FIFO_IRQ_MODE_MASK (0x03) +#define DAC33_FIFO_IRQ_MODE_RISING (0x00) +#define DAC33_FIFO_IRQ_MODE_FALLING (0x01) +#define DAC33_FIFO_IRQ_MODE_LEVEL (0x02) +#define DAC33_FIFO_IRQ_MODE_EDGE (0x03) + +/* DAC33_FIFO_IRQ_MODE_A (0x2A) */ +#define DAC33_UTM(x) (x << 0) +#define DAC33_UFM(x) (x << 2) +#define DAC33_OFM(x) (x << 4) + +/* DAC33_FIFO_IRQ_MODE_B (0x2B) */ +#define DAC33_NSM(x) (x << 0) +#define DAC33_PSM(x) (x << 2) +#define DAC33_ATM(x) (x << 4) +#define DAC33_LTM(x) (x << 4) + +/* DAC33_DAC_CTRL_A (0x2C) */ +#define DAC33_DACRATE(x) (x << 0) +#define DAC33_DACDUAL (0x01 << 4) +#define DAC33_DACLKSEL_MASK (0x03 << 5) +#define DAC33_DACLKSEL_INTSOC (0x00 << 5) +#define DAC33_DACLKSEL_PLL (0x01 << 5) +#define DAC33_DACLKSEL_MCLK (0x02 << 5) +#define DAC33_DACLKSEL_BCLK (0x03 << 5) + +/* DAC33_DAC_CTRL_B (0x2D) */ +#define DAC33_DACSRCR_MASK (0x03 << 0) +#define DAC33_DACSRCR_MUTE (0x00 << 0) +#define DAC33_DACSRCR_RIGHT (0x01 << 0) +#define DAC33_DACSRCR_LEFT (0x02 << 0) +#define DAC33_DACSRCR_MONOMIX (0x03 << 0) +#define DAC33_DACSRCL_MASK (0x03 << 2) +#define DAC33_DACSRCL_MUTE (0x00 << 2) +#define DAC33_DACSRCL_LEFT (0x01 << 2) +#define DAC33_DACSRCL_RIGHT (0x02 << 2) +#define DAC33_DACSRCL_MONOMIX (0x03 << 2) +#define DAC33_DVOLSTEP_MASK (0x03 << 4) +#define DAC33_DVOLSTEP_SS_PERFS (0x00 << 4) +#define DAC33_DVOLSTEP_SS_PER2FS (0x01 << 4) +#define DAC33_DVOLSTEP_SS_DISABLED (0x02 << 4) +#define DAC33_DVOLCTRL_MASK (0x03 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT1 (0x00 << 6) +#define DAC33_DVOLCTRL_LR_RIGHT_CONTROL (0x01 << 6) +#define DAC33_DVOLCTRL_LR_LEFT_CONTROL (0x02 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT2 (0x03 << 6) + +/* DAC33_DAC_CTRL_C (0x2E) */ +#define DAC33_DEEMENR (0x01 << 0) +#define DAC33_EFFENR (0x01 << 1) +#define DAC33_DEEMENL (0x01 << 2) +#define DAC33_EFFENL (0x01 << 3) +#define DAC33_EN3D (0x01 << 4) +#define DAC33_RESYNMUTE (0x01 << 5) +#define DAC33_RESYNEN (0x01 << 6) + +/* DAC33_ASRC_CTRL_A (0x32) */ +#define DAC33_SRCBYP (0x01 << 0) +#define DAC33_SRCLKSEL_MASK (0x03 << 1) +#define DAC33_SRCLKSEL_INTSOC (0x00 << 1) +#define DAC33_SRCLKSEL_PLL (0x01 << 1) +#define DAC33_SRCLKSEL_MCLK (0x02 << 1) +#define DAC33_SRCLKSEL_BCLK (0x03 << 1) +#define DAC33_SRCLKDIV(x) (x << 3) + +/* DAC33_ASRC_CTRL_B (0x33) */ +#define DAC33_SRCSETUP(x) (x << 0) +#define DAC33_SRCREFSEL (0x01 << 4) +#define DAC33_SRCREFDIV(x) (x << 5) + +/* DAC33_INTP_CTRL_A (0x38) */ +#define DAC33_INTPSEL (0x01 << 0) +#define DAC33_INTPM_MASK (0x03 << 1) +#define DAC33_INTPM_ALOW_OPENDRAIN (0x00 << 1) +#define DAC33_INTPM_ALOW (0x01 << 1) +#define DAC33_INTPM_AHIGH (0x02 << 1) + +/* DAC33_LDAC_PWR_CTRL (0x40) */ +/* DAC33_RDAC_PWR_CTRL (0x41) */ +#define DAC33_DACLRNUM (0x01 << 2) +#define DAC33_LROUT_GAIN(x) (x << 0) + +/* DAC33_ANA_VOL_SOFT_STEP_CTRL (0x49) */ +#define DAC33_VOLCLKSEL (0x01 << 0) +#define DAC33_VOLCLKEN (0x01 << 1) +#define DAC33_VOLBYPASS (0x01 << 2) + +#define TLV320DAC33_MCLK 0 +#define TLV320DAC33_SLEEPCLK 1 + +extern struct snd_soc_dai dac33_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320dac33; + +#endif /* __TLV320DAC33_H */ -- cgit v1.2.2 From d8707cecdf396bdb506252829d03837b2c67c939 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 19 Oct 2009 15:42:19 +0300 Subject: ASoC: TWL4030: Only update the needed bits in *set_dai_sysclk Do not rewrite the whole register, but only update the needed bits in set_dai_sysclk functions. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4df7c6c61c76..559e9b279289 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1785,19 +1785,21 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = codec->private_data; - u8 infreq; + u8 apll_ctrl; + apll_ctrl = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); + apll_ctrl &= ~TWL4030_APLL_INFREQ; switch (freq) { case 19200000: - infreq = TWL4030_APLL_INFREQ_19200KHZ; + apll_ctrl |= TWL4030_APLL_INFREQ_19200KHZ; twl4030->sysclk = 19200; break; case 26000000: - infreq = TWL4030_APLL_INFREQ_26000KHZ; + apll_ctrl |= TWL4030_APLL_INFREQ_26000KHZ; twl4030->sysclk = 26000; break; case 38400000: - infreq = TWL4030_APLL_INFREQ_38400KHZ; + apll_ctrl |= TWL4030_APLL_INFREQ_38400KHZ; twl4030->sysclk = 38400; break; default: @@ -1806,8 +1808,7 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } - infreq |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + twl4030_write(codec, TWL4030_REG_APLL_CTL, apll_ctrl); return 0; } @@ -1989,11 +1990,13 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - u8 infreq; + u8 apll_ctrl; + apll_ctrl = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); + apll_ctrl &= ~TWL4030_APLL_INFREQ; switch (freq) { case 26000000: - infreq = TWL4030_APLL_INFREQ_26000KHZ; + apll_ctrl |= TWL4030_APLL_INFREQ_26000KHZ; break; default: printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", @@ -2001,8 +2004,7 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } - infreq |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + twl4030_write(codec, TWL4030_REG_APLL_CTL, apll_ctrl); return 0; } -- cgit v1.2.2 From e697cd410a0c3aaea697c9915837e99933d8935b Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 19 Oct 2009 16:10:58 +0200 Subject: ASoC: au1x: psc-ac97: verify correct codec register was read Verify that the correct register has been received from the codec. Signed-off-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/au1x/psc-ac97.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a521aa90ddee..efe2afd4fe24 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -61,7 +61,8 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, retry, tmo; + unsigned short retry, tmo; + unsigned long data; au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); @@ -79,15 +80,19 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, && --tmo) udelay(2); - data = au_readl(AC97_CDC(pscdata)) & 0xffff; + data = au_readl(AC97_CDC(pscdata)); au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); mutex_unlock(&pscdata->lock); + + if (reg != ((data >> 16) & 0x7f)) + tmo = 1; /* wrong register, try again */ + } while (--retry && !tmo); - return retry ? data : 0xffff; + return retry ? data & 0xffff : 0xffff; } /* AC97 controller writes to codec register */ -- cgit v1.2.2 From 8d567b6b441bfcc20e8cbebc0dc376b2e280cd88 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 19 Oct 2009 16:10:59 +0200 Subject: ASoC: au1x: psc-ac97: reorganize timeouts Codec read/write functions: wait 21us between the pokings of hardware. Add timeouts to unbounded loops waiting for bits to change. Signed-off-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/au1x/psc-ac97.c | 38 +++++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index efe2afd4fe24..2a06a9c548af 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -75,10 +75,12 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, AC97_CDC(pscdata)); au_sync(); - tmo = 2000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) - && --tmo) - udelay(2); + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); data = au_readl(AC97_CDC(pscdata)); @@ -114,10 +116,12 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, AC97_CDC(pscdata)); au_sync(); - tmo = 2000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) - && --tmo) - udelay(2); + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); @@ -200,7 +204,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; unsigned long r, ro, stat; - int chans, stype = SUBSTREAM_TYPE(substream); + int chans, t, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); @@ -242,8 +246,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, au_sync(); /* ...wait for it... */ - while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) - asm volatile ("nop"); + t = 100; + while ((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't disable!\n"); /* ...write config... */ au_writel(r, AC97_CFG(pscdata)); @@ -254,8 +262,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, au_sync(); /* ...and wait for ready bit */ - while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) - asm volatile ("nop"); + t = 100; + while ((!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't enable!\n"); mutex_unlock(&pscdata->lock); -- cgit v1.2.2 From 4f066173fe8deb8874f41917e5d26ea2e0c46e3b Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 17 Oct 2009 08:32:56 +0200 Subject: ASoC: Move dereference after NULL test If the NULL test on jack is needed, then the derefernce should be after the NULL test. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 1d455ab79490..12124149601e 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -58,7 +58,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec = jack->card->codec; + struct snd_soc_codec *codec; struct snd_soc_jack_pin *pin; int enable; int oldstatus; @@ -67,6 +67,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) WARN_ON_ONCE(!jack); return; } + codec = jack->card->codec; mutex_lock(&codec->mutex); -- cgit v1.2.2 From ce491cf85466c3377228c5a852ea627ec5136956 Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Tue, 20 Oct 2009 09:40:47 -0700 Subject: omap: headers: Move remaining headers from include/mach to include/plat Move the remaining headers under plat-omap/include/mach to plat-omap/include/plat. Also search and replace the files using these headers to include using the right path. This was done with: #!/bin/bash mach_dir_old="arch/arm/plat-omap/include/mach" plat_dir_new="arch/arm/plat-omap/include/plat" headers=$(cd $mach_dir_old && ls *.h) omap_dirs="arch/arm/*omap*/ \ drivers/video/omap \ sound/soc/omap" other_files="drivers/leds/leds-ams-delta.c \ drivers/mfd/menelaus.c \ drivers/mfd/twl4030-core.c \ drivers/mtd/nand/ams-delta.c" for header in $headers; do old="#include --- sound/soc/omap/ams-delta.c | 4 ++-- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-mcbsp.c | 6 +++--- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 2 +- 11 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 5a5166ac7279..3f1a6c1a0355 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -31,8 +31,8 @@ #include -#include -#include +#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 0a505938e42b..08e09d72790f 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -32,7 +32,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3341f49402ca..e8e63ba40877 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -31,9 +31,9 @@ #include #include -#include -#include -#include +#include +#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5735945788bf..1169d2ec2e24 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -28,7 +28,7 @@ #include #include -#include +#include #include "omap-pcm.h" static const struct snd_pcm_hardware omap_pcm_hardware = { diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 027e1a40f8a1..c7adea38274c 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -31,7 +31,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index b0cff9f33b7e..d88ad5ca526c 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -29,7 +29,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c263077b..41a91b5cf12b 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -27,7 +27,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index a4e149b7f0eb..498ca2e03519 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -31,7 +31,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index ec4f8fd8b3a2..624f40ecc472 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -29,7 +29,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 4a3f62d1f295..c071f9603a38 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -34,7 +34,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index f90b45f56220..f90a2ac888cf 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -29,7 +29,7 @@ #include #include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" -- cgit v1.2.2 From 02624621a58d7030e0e56f1e3df490202e59056c Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 21 Oct 2009 04:40:55 +0200 Subject: ASoC: Amstrad Delta minor cleanups Hi Mark, Here is a patch that corrects small omissions I have found in my code. Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 5a5166ac7279..ae0fc9b135d4 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -40,7 +40,7 @@ /* Board specific DAPM widgets */ - const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { +static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), SND_SOC_DAPM_HP("Earpiece", NULL), @@ -81,7 +81,7 @@ static const char *ams_delta_audio_mode[] = (1 << AMS_DELTA_SPEAKER)) #define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC)) -unsigned short ams_delta_audio_mode_pins[] = { +static const unsigned short ams_delta_audio_mode_pins[] = { AMS_DELTA_MIXED, AMS_DELTA_HANDSET, AMS_DELTA_HANDSFREE, -- cgit v1.2.2 From 017deee63934349a70292666acfedea8e6eb6eb8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 21 Oct 2009 09:58:35 +0300 Subject: ASoC: tlv320dac33: typo fix in the header Fix the definition of DAC33_LTM field, the LTM bits in FIFO_IRQ_MODE_B register are starting at bit 6. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h index 0fedd709028e..eb8ae07f0bd2 100644 --- a/sound/soc/codecs/tlv320dac33.h +++ b/sound/soc/codecs/tlv320dac33.h @@ -186,7 +186,7 @@ #define DAC33_NSM(x) (x << 0) #define DAC33_PSM(x) (x << 2) #define DAC33_ATM(x) (x << 4) -#define DAC33_LTM(x) (x << 4) +#define DAC33_LTM(x) (x << 6) /* DAC33_DAC_CTRL_A (0x2C) */ #define DAC33_DACRATE(x) (x << 0) -- cgit v1.2.2 From 0ffc11800cb2a74b05c2f5b28966ebd50b27f70c Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 21 Oct 2009 23:10:03 +0200 Subject: ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1 After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c, omap_pcm_prepare() unconditionally calls: omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); Current implementation of those two functions found in arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at all, so they both end with BUG() on that machine. That results in ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta. The patch corrects the problem by not calling those two functions when run on OMAP1 class based machines. Created against linux-2.6.32-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5735945788bf..6a829eef2a4f 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) else omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); - omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); - omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); + if (!(cpu_class_is_omap1())) { + omap_set_dma_src_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + omap_set_dma_dest_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + } return 0; } -- cgit v1.2.2 From 1f0f9b67f98a873fca8288ccb7f2a0f3c8f34371 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Oct 2009 13:26:47 +0300 Subject: ASoC: TWL4030: use the twl4030-codec.h for register descriptions Remove the register descriptions from the twl4030.h file and use the linux/mfd/twl4030-codec.h instead, which has the codec related register descriptions also. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.h | 242 ++------------------------------------------- 1 file changed, 6 insertions(+), 236 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 2b4bfa23f985..dd6396ec9c79 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -22,245 +22,13 @@ #ifndef __TWL4030_AUDIO_H__ #define __TWL4030_AUDIO_H__ -#define TWL4030_REG_CODEC_MODE 0x1 -#define TWL4030_REG_OPTION 0x2 -#define TWL4030_REG_UNKNOWN 0x3 -#define TWL4030_REG_MICBIAS_CTL 0x4 -#define TWL4030_REG_ANAMICL 0x5 -#define TWL4030_REG_ANAMICR 0x6 -#define TWL4030_REG_AVADC_CTL 0x7 -#define TWL4030_REG_ADCMICSEL 0x8 -#define TWL4030_REG_DIGMIXING 0x9 -#define TWL4030_REG_ATXL1PGA 0xA -#define TWL4030_REG_ATXR1PGA 0xB -#define TWL4030_REG_AVTXL2PGA 0xC -#define TWL4030_REG_AVTXR2PGA 0xD -#define TWL4030_REG_AUDIO_IF 0xE -#define TWL4030_REG_VOICE_IF 0xF -#define TWL4030_REG_ARXR1PGA 0x10 -#define TWL4030_REG_ARXL1PGA 0x11 -#define TWL4030_REG_ARXR2PGA 0x12 -#define TWL4030_REG_ARXL2PGA 0x13 -#define TWL4030_REG_VRXPGA 0x14 -#define TWL4030_REG_VSTPGA 0x15 -#define TWL4030_REG_VRX2ARXPGA 0x16 -#define TWL4030_REG_AVDAC_CTL 0x17 -#define TWL4030_REG_ARX2VTXPGA 0x18 -#define TWL4030_REG_ARXL1_APGA_CTL 0x19 -#define TWL4030_REG_ARXR1_APGA_CTL 0x1A -#define TWL4030_REG_ARXL2_APGA_CTL 0x1B -#define TWL4030_REG_ARXR2_APGA_CTL 0x1C -#define TWL4030_REG_ATX2ARXPGA 0x1D -#define TWL4030_REG_BT_IF 0x1E -#define TWL4030_REG_BTPGA 0x1F -#define TWL4030_REG_BTSTPGA 0x20 -#define TWL4030_REG_EAR_CTL 0x21 -#define TWL4030_REG_HS_SEL 0x22 -#define TWL4030_REG_HS_GAIN_SET 0x23 -#define TWL4030_REG_HS_POPN_SET 0x24 -#define TWL4030_REG_PREDL_CTL 0x25 -#define TWL4030_REG_PREDR_CTL 0x26 -#define TWL4030_REG_PRECKL_CTL 0x27 -#define TWL4030_REG_PRECKR_CTL 0x28 -#define TWL4030_REG_HFL_CTL 0x29 -#define TWL4030_REG_HFR_CTL 0x2A -#define TWL4030_REG_ALC_CTL 0x2B -#define TWL4030_REG_ALC_SET1 0x2C -#define TWL4030_REG_ALC_SET2 0x2D -#define TWL4030_REG_BOOST_CTL 0x2E -#define TWL4030_REG_SOFTVOL_CTL 0x2F -#define TWL4030_REG_DTMF_FREQSEL 0x30 -#define TWL4030_REG_DTMF_TONEXT1H 0x31 -#define TWL4030_REG_DTMF_TONEXT1L 0x32 -#define TWL4030_REG_DTMF_TONEXT2H 0x33 -#define TWL4030_REG_DTMF_TONEXT2L 0x34 -#define TWL4030_REG_DTMF_TONOFF 0x35 -#define TWL4030_REG_DTMF_WANONOFF 0x36 -#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 -#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 -#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 -#define TWL4030_REG_APLL_CTL 0x3A -#define TWL4030_REG_DTMF_CTL 0x3B -#define TWL4030_REG_DTMF_PGA_CTL2 0x3C -#define TWL4030_REG_DTMF_PGA_CTL1 0x3D -#define TWL4030_REG_MISC_SET_1 0x3E -#define TWL4030_REG_PCMBTMUX 0x3F -#define TWL4030_REG_RX_PATH_SEL 0x43 -#define TWL4030_REG_VDL_APGA_CTL 0x44 -#define TWL4030_REG_VIBRA_CTL 0x45 -#define TWL4030_REG_VIBRA_SET 0x46 -#define TWL4030_REG_VIBRA_PWM_SET 0x47 -#define TWL4030_REG_ANAMIC_GAIN 0x48 -#define TWL4030_REG_MISC_SET_2 0x49 -#define TWL4030_REG_SW_SHADOW 0x4A +/* Register descriptions are here */ +#include +/* Sgadow register used by the audio driver */ +#define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -/* Bitfield Definitions */ - -/* TWL4030_CODEC_MODE (0x01) Fields */ - -#define TWL4030_APLL_RATE 0xF0 -#define TWL4030_APLL_RATE_8000 0x00 -#define TWL4030_APLL_RATE_11025 0x10 -#define TWL4030_APLL_RATE_12000 0x20 -#define TWL4030_APLL_RATE_16000 0x40 -#define TWL4030_APLL_RATE_22050 0x50 -#define TWL4030_APLL_RATE_24000 0x60 -#define TWL4030_APLL_RATE_32000 0x80 -#define TWL4030_APLL_RATE_44100 0x90 -#define TWL4030_APLL_RATE_48000 0xA0 -#define TWL4030_APLL_RATE_96000 0xE0 -#define TWL4030_SEL_16K 0x08 -#define TWL4030_CODECPDZ 0x02 -#define TWL4030_OPT_MODE 0x01 -#define TWL4030_OPTION_1 (1 << 0) -#define TWL4030_OPTION_2 (0 << 0) - -/* TWL4030_OPTION (0x02) Fields */ - -#define TWL4030_ATXL1_EN (1 << 0) -#define TWL4030_ATXR1_EN (1 << 1) -#define TWL4030_ATXL2_VTXL_EN (1 << 2) -#define TWL4030_ATXR2_VTXR_EN (1 << 3) -#define TWL4030_ARXL1_VRX_EN (1 << 4) -#define TWL4030_ARXR1_EN (1 << 5) -#define TWL4030_ARXL2_EN (1 << 6) -#define TWL4030_ARXR2_EN (1 << 7) - -/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ - -#define TWL4030_MICBIAS2_CTL 0x40 -#define TWL4030_MICBIAS1_CTL 0x20 -#define TWL4030_HSMICBIAS_EN 0x04 -#define TWL4030_MICBIAS2_EN 0x02 -#define TWL4030_MICBIAS1_EN 0x01 - -/* ANAMICL (0x05) Fields */ - -#define TWL4030_CNCL_OFFSET_START 0x80 -#define TWL4030_OFFSET_CNCL_SEL 0x60 -#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 -#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 -#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 -#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 -#define TWL4030_MICAMPL_EN 0x10 -#define TWL4030_CKMIC_EN 0x08 -#define TWL4030_AUXL_EN 0x04 -#define TWL4030_HSMIC_EN 0x02 -#define TWL4030_MAINMIC_EN 0x01 - -/* ANAMICR (0x06) Fields */ - -#define TWL4030_MICAMPR_EN 0x10 -#define TWL4030_AUXR_EN 0x04 -#define TWL4030_SUBMIC_EN 0x01 - -/* AVADC_CTL (0x07) Fields */ - -#define TWL4030_ADCL_EN 0x08 -#define TWL4030_AVADC_CLK_PRIORITY 0x04 -#define TWL4030_ADCR_EN 0x02 - -/* TWL4030_REG_ADCMICSEL (0x08) Fields */ - -#define TWL4030_DIGMIC1_EN 0x08 -#define TWL4030_TX2IN_SEL 0x04 -#define TWL4030_DIGMIC0_EN 0x02 -#define TWL4030_TX1IN_SEL 0x01 - -/* AUDIO_IF (0x0E) Fields */ - -#define TWL4030_AIF_SLAVE_EN 0x80 -#define TWL4030_DATA_WIDTH 0x60 -#define TWL4030_DATA_WIDTH_16S_16W 0x00 -#define TWL4030_DATA_WIDTH_32S_16W 0x40 -#define TWL4030_DATA_WIDTH_32S_24W 0x60 -#define TWL4030_AIF_FORMAT 0x18 -#define TWL4030_AIF_FORMAT_CODEC 0x00 -#define TWL4030_AIF_FORMAT_LEFT 0x08 -#define TWL4030_AIF_FORMAT_RIGHT 0x10 -#define TWL4030_AIF_FORMAT_TDM 0x18 -#define TWL4030_AIF_TRI_EN 0x04 -#define TWL4030_CLK256FS_EN 0x02 -#define TWL4030_AIF_EN 0x01 - -/* VOICE_IF (0x0F) Fields */ - -#define TWL4030_VIF_SLAVE_EN 0x80 -#define TWL4030_VIF_DIN_EN 0x40 -#define TWL4030_VIF_DOUT_EN 0x20 -#define TWL4030_VIF_SWAP 0x10 -#define TWL4030_VIF_FORMAT 0x08 -#define TWL4030_VIF_TRI_EN 0x04 -#define TWL4030_VIF_SUB_EN 0x02 -#define TWL4030_VIF_EN 0x01 - -/* EAR_CTL (0x21) */ -#define TWL4030_EAR_GAIN 0x30 - -/* HS_GAIN_SET (0x23) Fields */ - -#define TWL4030_HSR_GAIN 0x0C -#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 -#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 -#define TWL4030_HSR_GAIN_0DB 0x08 -#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C -#define TWL4030_HSL_GAIN 0x03 -#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 -#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 -#define TWL4030_HSL_GAIN_0DB 0x02 -#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 - -/* HS_POPN_SET (0x24) Fields */ - -#define TWL4030_VMID_EN 0x40 -#define TWL4030_EXTMUTE 0x20 -#define TWL4030_RAMP_DELAY 0x1C -#define TWL4030_RAMP_DELAY_20MS 0x00 -#define TWL4030_RAMP_DELAY_40MS 0x04 -#define TWL4030_RAMP_DELAY_81MS 0x08 -#define TWL4030_RAMP_DELAY_161MS 0x0C -#define TWL4030_RAMP_DELAY_323MS 0x10 -#define TWL4030_RAMP_DELAY_645MS 0x14 -#define TWL4030_RAMP_DELAY_1291MS 0x18 -#define TWL4030_RAMP_DELAY_2581MS 0x1C -#define TWL4030_RAMP_EN 0x02 - -/* PREDL_CTL (0x25) */ -#define TWL4030_PREDL_GAIN 0x30 - -/* PREDR_CTL (0x26) */ -#define TWL4030_PREDR_GAIN 0x30 - -/* PRECKL_CTL (0x27) */ -#define TWL4030_PRECKL_GAIN 0x30 - -/* PRECKR_CTL (0x28) */ -#define TWL4030_PRECKR_GAIN 0x30 - -/* HFL_CTL (0x29, 0x2A) Fields */ -#define TWL4030_HF_CTL_HB_EN 0x04 -#define TWL4030_HF_CTL_LOOP_EN 0x08 -#define TWL4030_HF_CTL_RAMP_EN 0x10 -#define TWL4030_HF_CTL_REF_EN 0x20 - -/* APLL_CTL (0x3A) Fields */ - -#define TWL4030_APLL_EN 0x10 -#define TWL4030_APLL_INFREQ 0x0F -#define TWL4030_APLL_INFREQ_19200KHZ 0x05 -#define TWL4030_APLL_INFREQ_26000KHZ 0x06 -#define TWL4030_APLL_INFREQ_38400KHZ 0x0F - -/* REG_MISC_SET_1 (0x3E) Fields */ - -#define TWL4030_CLK64_EN 0x80 -#define TWL4030_SCRAMBLE_EN 0x40 -#define TWL4030_FMLOOP_EN 0x20 -#define TWL4030_SMOOTH_ANAVOL_EN 0x02 -#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 - /* TWL4030_REG_SW_SHADOW (0x4A) Fields */ #define TWL4030_HFL_EN 0x01 #define TWL4030_HFR_EN 0x02 @@ -279,3 +47,5 @@ struct twl4030_setup_data { }; #endif /* End of __TWL4030_AUDIO_H__ */ + + -- cgit v1.2.2 From 7a1fecf57f435e50ed86851cbb701f4b28e65135 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Oct 2009 13:26:48 +0300 Subject: ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/twl4030.c | 203 ++++++++++++++++++++++++++++----------------- 2 files changed, 127 insertions(+), 77 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d30fce71cfe8..3df3497335bf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -147,6 +147,7 @@ config SND_SOC_TLV320DAC33 tristate config SND_SOC_TWL4030 + select TWL4030_CODEC tristate config SND_SOC_UDA134X diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 559e9b279289..5c5a4c0a424f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -120,6 +120,8 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { + struct snd_soc_codec codec; + unsigned int bypass_state; unsigned int codec_powered; unsigned int codec_muted; @@ -183,19 +185,20 @@ static int twl4030_write(struct snd_soc_codec *codec, static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = codec->private_data; - u8 mode; + int mode; if (enable == twl4030->codec_powered) return; - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); if (enable) - mode |= TWL4030_CODECPDZ; + mode = twl4030_codec_enable_resource(TWL4030_CODEC_RES_POWER); else - mode &= ~TWL4030_CODECPDZ; + mode = twl4030_codec_disable_resource(TWL4030_CODEC_RES_POWER); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030->codec_powered = enable; + if (mode >= 0) { + twl4030_write_reg_cache(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030->codec_powered = enable; + } /* REVISIT: this delay is present in TI sample drivers */ /* but there seems to be no TRM requirement for it */ @@ -219,22 +222,20 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) { struct twl4030_priv *twl4030 = codec->private_data; - u8 reg_val; + int status; if (mute == twl4030->codec_muted) return; - if (mute) { + if (mute) /* Disable PLL */ - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - reg_val &= ~TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); - } else { + status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); + else /* Enable PLL */ - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - reg_val |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); - } + status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); + + if (status >= 0) + twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); twl4030->codec_muted = mute; } @@ -2123,7 +2124,7 @@ struct snd_soc_dai twl4030_dai[] = { }; EXPORT_SYMBOL_GPL(twl4030_dai); -static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) +static int twl4030_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -2133,7 +2134,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -static int twl4030_resume(struct platform_device *pdev) +static int twl4030_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -2143,32 +2144,21 @@ static int twl4030_resume(struct platform_device *pdev) return 0; } -/* - * initialize the driver - * register the mixer and dsp interfaces with the kernel - */ +static struct snd_soc_codec *twl4030_codec; -static int twl4030_init(struct snd_soc_device *socdev) +static int twl4030_soc_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct twl4030_setup_data *setup = socdev->codec_data; - struct twl4030_priv *twl4030 = codec->private_data; - int ret = 0; + struct snd_soc_codec *codec; + struct twl4030_priv *twl4030; + int ret; - printk(KERN_INFO "TWL4030 Audio Codec init \n"); + BUG_ON(!twl4030_codec); - codec->name = "twl4030"; - codec->owner = THIS_MODULE; - codec->read = twl4030_read_reg_cache; - codec->write = twl4030_write; - codec->set_bias_level = twl4030_set_bias_level; - codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), - codec->reg_cache_size = sizeof(twl4030_reg); - codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), - GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec = twl4030_codec; + twl4030 = codec->private_data; + socdev->card->codec = codec; /* Configuration for headset ramp delay from setup data */ if (setup) { @@ -2190,100 +2180,159 @@ static int twl4030_init(struct snd_soc_device *socdev) /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "twl4030: failed to create pcms\n"); - goto pcm_err; + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; } - twl4030_init_chip(codec); - - /* power on device */ - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, twl4030_snd_controls, ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "twl4030: failed to register card\n"); + dev_err(&pdev->dev, "failed to register card\n"); goto card_err; } - return ret; + return 0; card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); + return ret; } -static struct snd_soc_device *twl4030_socdev; - -static int twl4030_probe(struct platform_device *pdev) +static int twl4030_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +static int __devinit twl4030_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; struct snd_soc_codec *codec; struct twl4030_priv *twl4030; + int ret; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; + if (!pdata || !(pdata->audio_mclk == 19200000 || + pdata->audio_mclk == 26000000 || + pdata->audio_mclk == 38400000)) { + dev_err(&pdev->dev, "Invalid platform_data\n"); + return -EINVAL; + } twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); if (twl4030 == NULL) { - kfree(codec); + dev_err(&pdev->dev, "Can not allocate memroy\n"); return -ENOMEM; } + codec = &twl4030->codec; codec->private_data = twl4030; - socdev->card->codec = codec; + codec->dev = &pdev->dev; + twl4030_dai[0].dev = &pdev->dev; + twl4030_dai[1].dev = &pdev->dev; + mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - twl4030_socdev = socdev; - twl4030_init(socdev); + codec->name = "twl4030"; + codec->owner = THIS_MODULE; + codec->read = twl4030_read_reg_cache; + codec->write = twl4030_write; + codec->set_bias_level = twl4030_set_bias_level; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->reg_cache_size = sizeof(twl4030_reg); + codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_cache; + } + + platform_set_drvdata(pdev, twl4030); + twl4030_codec = codec; + + /* Set the defaults, and power up the codec */ + twl4030_init_chip(codec); + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } return 0; + +error_codec: + twl4030_power_down(codec); + kfree(codec->reg_cache); +error_cache: + kfree(twl4030); + return ret; } -static int twl4030_remove(struct platform_device *pdev) +static int __devexit twl4030_codec_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = platform_get_drvdata(pdev); - printk(KERN_INFO "TWL4030 Audio Codec remove\n"); - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - kfree(codec->private_data); - kfree(codec); + kfree(twl4030); + twl4030_codec = NULL; return 0; } -struct snd_soc_codec_device soc_codec_dev_twl4030 = { - .probe = twl4030_probe, - .remove = twl4030_remove, - .suspend = twl4030_suspend, - .resume = twl4030_resume, +MODULE_ALIAS("platform:twl4030_codec_audio"); + +static struct platform_driver twl4030_codec_driver = { + .probe = twl4030_codec_probe, + .remove = __devexit_p(twl4030_codec_remove), + .driver = { + .name = "twl4030_codec_audio", + .owner = THIS_MODULE, + }, }; -EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + return platform_driver_register(&twl4030_codec_driver); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + platform_driver_unregister(&twl4030_codec_driver); } module_exit(twl4030_exit); +struct snd_soc_codec_device soc_codec_dev_twl4030 = { + .probe = twl4030_soc_probe, + .remove = twl4030_soc_remove, + .suspend = twl4030_soc_suspend, + .resume = twl4030_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); + MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); MODULE_LICENSE("GPL"); -- cgit v1.2.2 From 7dea7c01dac9b74faa9afa93fc9bb5f2d37521dc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 26 Oct 2009 15:20:17 +0000 Subject: ASoC: Add regulator support for WM8731 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 51 +++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 47 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0e59219a59f4..bb95af950971 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -33,9 +34,18 @@ static struct snd_soc_codec *wm8731_codec; struct snd_soc_codec_device soc_codec_dev_wm8731; +#define WM8731_NUM_SUPPLIES 4 +static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { + "AVDD", + "HPVDD", + "DCVDD", + "DBVDD", +}; + /* codec private data */ struct wm8731_priv { struct snd_soc_codec codec; + struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; u16 reg_cache[WM8731_CACHEREGNUM]; unsigned int sysclk; }; @@ -422,9 +432,12 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + struct wm8731_priv *wm8731 = codec->private_data; snd_soc_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); return 0; } @@ -432,10 +445,16 @@ static int wm8731_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; + struct wm8731_priv *wm8731 = codec->private_data; + int i, ret; u8 data[2]; u16 *cache = codec->reg_cache; + ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) + return ret; + /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); @@ -444,6 +463,7 @@ static int wm8731_resume(struct platform_device *pdev) } wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8731_set_bias_level(codec, codec->suspend_bias_level); + return 0; } #else @@ -512,7 +532,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); static int wm8731_register(struct wm8731_priv *wm8731, enum snd_soc_control_type control) { - int ret; + int ret, i; struct snd_soc_codec *codec = &wm8731->codec; if (wm8731_codec) { @@ -543,10 +563,27 @@ static int wm8731_register(struct wm8731_priv *wm8731, goto err; } + for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) + wm8731->supplies[i].supply = wm8731_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_regulator_get; + } + ret = wm8731_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - goto err; + goto err_regulator_enable; } wm8731_dai.dev = codec->dev; @@ -567,7 +604,7 @@ static int wm8731_register(struct wm8731_priv *wm8731, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto err; + goto err_regulator_enable; } ret = snd_soc_register_dai(&wm8731_dai); @@ -581,6 +618,10 @@ static int wm8731_register(struct wm8731_priv *wm8731, err_codec: snd_soc_unregister_codec(codec); +err_regulator_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); +err_regulator_get: + regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); err: kfree(wm8731); return ret; @@ -591,6 +632,8 @@ static void wm8731_unregister(struct wm8731_priv *wm8731) wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&wm8731_dai); snd_soc_unregister_codec(&wm8731->codec); + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); kfree(wm8731); wm8731_codec = NULL; } -- cgit v1.2.2 From 78e08e2f209e5e7777e81919d32cfcddad126cfa Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 28 Oct 2009 10:57:04 +0200 Subject: ASoC: TWL4030: Remove bypass tracking Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 128 +++++++++++---------------------------------- 1 file changed, 30 insertions(+), 98 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 5c5a4c0a424f..24002269f03a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -122,7 +122,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { struct twl4030_priv { struct snd_soc_codec codec; - unsigned int bypass_state; unsigned int codec_powered; unsigned int codec_muted; @@ -725,67 +724,6 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } -static int bypass_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct soc_mixer_control *m = - (struct soc_mixer_control *)w->kcontrols->private_value; - struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg, misc; - - reg = twl4030_read_reg_cache(w->codec, m->reg); - - /* - * bypass_state[0:3] - analog HiFi bypass - * bypass_state[4] - analog voice bypass - * bypass_state[5] - digital voice bypass - * bypass_state[6:7] - digital HiFi bypass - */ - if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); - } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { - /* Analog bypass */ - if (reg & (1 << m->shift)) - twl4030->bypass_state |= - (1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); - else - twl4030->bypass_state &= - ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); - } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { - /* Analog voice bypass */ - if (reg & (1 << m->shift)) - twl4030->bypass_state |= (1 << 4); - else - twl4030->bypass_state &= ~(1 << 4); - } else { - /* Digital bypass */ - if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); - else - twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); - } - - /* Enable master analog loopback mode if any analog switch is enabled*/ - misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); - if (twl4030->bypass_state & 0x1F) - misc |= TWL4030_FMLOOP_EN; - else - misc &= ~TWL4030_FMLOOP_EN; - twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); - - if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { - if (twl4030->bypass_state) - twl4030_codec_mute(w->codec, 0); - else - twl4030_codec_mute(w->codec, 1); - } - return 0; -} - /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1193,32 +1131,28 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), /* Analog bypasses */ - SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassr1_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassl1_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassr2_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassl2_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassv_control, - bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr1_control), + SND_SOC_DAPM_SWITCH("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl1_control), + SND_SOC_DAPM_SWITCH("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr2_control), + SND_SOC_DAPM_SWITCH("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl2_control), + SND_SOC_DAPM_SWITCH("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control), + + /* Master analog loopback switch */ + SND_SOC_DAPM_SUPPLY("FM Loop Enable", TWL4030_REG_MISC_SET_1, 5, 0, + NULL, 0), /* Digital bypasses */ - SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassl_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassr_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassv_control, bypass_event, - SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH("Left Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassl_control), + SND_SOC_DAPM_SWITCH("Right Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassr_control), + SND_SOC_DAPM_SWITCH("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control), /* Digital mixers, power control for the physical DACs */ SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer", @@ -1490,6 +1424,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left2 Analog Loopback", "Switch", "Analog Left"}, {"Voice Analog Loopback", "Switch", "Analog Left"}, + /* Supply for the Analog loopbacks */ + {"Right1 Analog Loopback", NULL, "FM Loop Enable"}, + {"Left1 Analog Loopback", NULL, "FM Loop Enable"}, + {"Right2 Analog Loopback", NULL, "FM Loop Enable"}, + {"Left2 Analog Loopback", NULL, "FM Loop Enable"}, + {"Voice Analog Loopback", NULL, "FM Loop Enable"}, + {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, @@ -1521,25 +1462,16 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct twl4030_priv *twl4030 = codec->private_data; - switch (level) { case SND_SOC_BIAS_ON: twl4030_codec_mute(codec, 0); break; case SND_SOC_BIAS_PREPARE: - twl4030_power_up(codec); - if (twl4030->bypass_state) - twl4030_codec_mute(codec, 0); - else - twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_STANDBY: - twl4030_power_up(codec); - if (twl4030->bypass_state) - twl4030_codec_mute(codec, 0); - else - twl4030_codec_mute(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) + twl4030_power_up(codec); + twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_OFF: twl4030_power_down(codec); -- cgit v1.2.2 From 2845fa13e5cbe708ece7fafe29c91f32c66e4f59 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 28 Oct 2009 10:57:05 +0200 Subject: ASoC: TWL4030: Change codec_muted to apll_enabled codec_muted is missleading, change it to apll_enabled, which is what it is doing: enabing and disabling the APLL. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 24002269f03a..9163713a0307 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -123,7 +123,7 @@ struct twl4030_priv { struct snd_soc_codec codec; unsigned int codec_powered; - unsigned int codec_muted; + unsigned int apll_enabled; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -218,25 +218,25 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } -static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) +static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = codec->private_data; int status; - if (mute == twl4030->codec_muted) + if (enable == twl4030->apll_enabled) return; - if (mute) - /* Disable PLL */ - status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); - else + if (enable) /* Enable PLL */ status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); + else + /* Disable PLL */ + status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); if (status >= 0) twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); - twl4030->codec_muted = mute; + twl4030->apll_enabled = enable; } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -1464,14 +1464,14 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_ON: - twl4030_codec_mute(codec, 0); + twl4030_apll_enable(codec, 1); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) twl4030_power_up(codec); - twl4030_codec_mute(codec, 1); + twl4030_apll_enable(codec, 0); break; case SND_SOC_BIAS_OFF: twl4030_power_down(codec); -- cgit v1.2.2 From 26d95b6e300c4847be6ec8bfe817dbd531e94d9a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Oct 2009 15:47:48 +0000 Subject: ASoC: Minor SMDK64xx WM8580 cleanups Fix up some comments, remove all enable_pin() calls (edge widgets are all enabled by default) and mark the microphone as disabled by default since it requires a resistor fit to connect it. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/smdk64xx_wm8580.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index 482aaf10eff6..cb8a9161b643 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -103,7 +103,7 @@ static int smdk64xx_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Set WM8580 to drive MCLK from it's PLLA */ + /* Set WM8580 to drive MCLK from its PLLA */ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, WM8580_CLKSRC_PLLA); if (ret < 0) @@ -115,7 +115,6 @@ static int smdk64xx_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Assuming the CODEC driver evaluates it's rfs too from this call */ ret = snd_soc_dai_set_pll(codec_dai, 0, WM8580_PLLA, SMDK64XX_WM8580_FREQ, pll_out); if (ret < 0) @@ -186,9 +185,10 @@ static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) /* Set up PAIFTX audio path */ snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); - /* All enabled by default */ - snd_soc_dapm_enable_pin(codec, "MicIn"); - snd_soc_dapm_enable_pin(codec, "LineIn"); + /* Enabling the microphone requires the fitting of a 0R + * resistor to connect the line from the microphone jack. + */ + snd_soc_dapm_disable_pin(codec, "MicIn"); /* signal a DAPM event */ snd_soc_dapm_sync(codec); @@ -205,11 +205,6 @@ static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) /* Set up PAIFRX audio path */ snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - /* All enabled by default */ - snd_soc_dapm_enable_pin(codec, "Front-L/R"); - snd_soc_dapm_enable_pin(codec, "Center/Sub"); - snd_soc_dapm_enable_pin(codec, "Rear-L/R"); - /* signal a DAPM event */ snd_soc_dapm_sync(codec); -- cgit v1.2.2 From 7e1aa1dcd0d886df72586e3a94b1a7382952f21f Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 29 Oct 2009 02:24:32 +0100 Subject: ASoC: CS4270: export de-emphasis filter as ALSA control The CS4270 codec features an de-emphasis filter for compensation of audio material filtered by an 50/15 uS algorithm. Not sure whether we should always enable it for 44100Hz sampling frequency, but it should at least be configurable by the user. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 59bb16d033d6..565842dcfc65 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -520,6 +520,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("De-emphasis filter", CS4270_TRANS, 0, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1), -- cgit v1.2.2 From 86139a13ced74b3911c33940f0049b8f97bae07a Mon Sep 17 00:00:00 2001 From: Jari Vanhala Date: Thu, 29 Oct 2009 11:58:09 +0200 Subject: ASoC: TWL4030: Vibra motor stop fix when it is driven with audio This patch fixes vibrator playing incoherently, when driven with audio. There is something wrong in switch 3 at H-bridge and VIBRA_SET still affects PWM generator. Slowest value fixes things. Signed-off-by: Jari Vanhala Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 9163713a0307..ccaeb366eb7c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -613,6 +613,13 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int vibramux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff); + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -1243,8 +1250,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 0, 0, NULL, 0, handsfreerpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Vibra */ - SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, - &twl4030_dapm_vibra_control), + SND_SOC_DAPM_MUX_E("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, + &twl4030_dapm_vibra_control, vibramux_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0, &twl4030_dapm_vibrapath_control), -- cgit v1.2.2 From 7729cf749350b04c80ee1652961de238afc9d5b1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 29 Oct 2009 11:58:10 +0200 Subject: ASoC: TWL4030: Change APLL powering sequence It seams that certain part of the twl4030 codec needs the APLL enabled before they are enabled. Paths which has any digital processing needs need the APLL enabled before they can function. For example the vibra output will have some random data after it is enabled and before the APLL also enabled. If only analog components are in use (analog bypass), than it seams, that the APLL does not need to be enabled. This lowers the power consumption with around ~0.005A. Adding DAPM_SUPPLY to the Digital playback route and also to the capture route to enable and disable the APLL. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ccaeb366eb7c..277e99ce5558 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -620,6 +620,20 @@ static int vibramux_event(struct snd_soc_dapm_widget *w, return 0; } +static int apll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + twl4030_apll_enable(w->codec, 1); + break; + case SND_SOC_DAPM_POST_PMD: + twl4030_apll_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -1185,6 +1199,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1312,6 +1329,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital R2 Playback Mixer", NULL, "DAC Right2"}, {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, + /* Supply for the digital part (APLL) */ + {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, + {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, + {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, + {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, + {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1472,14 +1496,12 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_ON: - twl4030_apll_enable(codec, 1); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) twl4030_power_up(codec); - twl4030_apll_enable(codec, 0); break; case SND_SOC_BIAS_OFF: twl4030_power_down(codec); -- cgit v1.2.2 From 1c3d20027133f145523a072e84ab55d9132920c9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 29 Oct 2009 13:05:52 +0200 Subject: ASoC: TWL4030: Add APLL supply for the capture path Capture path also need the APLL enabled, adding DAPM_SUPPLY for the Virtual ADCs. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 277e99ce5558..f9121ef7fe5c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1449,6 +1449,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, + {"ADC Virtual Left1", NULL, "APLL Enable"}, + {"ADC Virtual Right1", NULL, "APLL Enable"}, + {"ADC Virtual Left2", NULL, "APLL Enable"}, + {"ADC Virtual Right2", NULL, "APLL Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, -- cgit v1.2.2 From ed146aeb68b6b240a015f3c24c9eea9266d845ec Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Wed, 23 Sep 2009 12:40:31 +0530 Subject: ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal The pop-removal specific values are configured for TWL4030 codec for OMAP3EVM through this patch. Signed-off-by: Anuj Aggarwal Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap3evm.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c263077b..8deb59bb10b1 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = { .num_links = 1, }; +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 4, + .sysclk = 26000, +}; + /* Audio subsystem */ static struct snd_soc_device omap3evm_snd_devdata = { .card = &snd_soc_omap3evm, .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, }; static struct platform_device *omap3evm_snd_device; -- cgit v1.2.2 From 89e9abe78151de4d62fefe3976f6ef9f1f086e53 Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 30 Oct 2009 00:22:30 +0530 Subject: ASoC: Adding OMAP3517 / AM3517 EVM support in ASOC Adding support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal Signed-off-by: Mark Brown --- sound/soc/omap/am3517evm.c | 202 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 202 insertions(+) create mode 100644 sound/soc/omap/am3517evm.c (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c new file mode 100644 index 000000000000..135901b2ea11 --- /dev/null +++ b/sound/soc/omap/am3517evm.c @@ -0,0 +1,202 @@ +/* + * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM + * + * Author: Anuj Aggarwal + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2009 Texas Instruments Incorporated + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "omap-mcbsp.h" +#include "omap-pcm.h" + +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static int am3517evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n"); + return ret; + } + + snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops am3517evm_ops = { + .hw_params = am3517evm_hw_params, +}; + +/* am3517evm machine dapm widgets */ +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LOUT"}, + {"Line Out", NULL, "ROUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic In"}, +}; + +static int am3517evm_aic23_init(struct snd_soc_codec *codec) +{ + /* Add am3517-evm specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic In"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link am3517evm_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = am3517evm_aic23_init, + .ops = &am3517evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_am3517evm = { + .name = "am3517evm", + .platform = &omap_soc_platform, + .dai_link = &am3517evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device am3517evm_snd_devdata = { + .card = &snd_soc_am3517evm, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *am3517evm_snd_device; + +static int __init am3517evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3517evm()) { + pr_err("Not OMAP3517 / AM3517 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3517 / AM3517 EVM SoC init\n"); + + am3517evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!am3517evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(am3517evm_snd_device, &am3517evm_snd_devdata); + am3517evm_snd_devdata.dev = &am3517evm_snd_device->dev; + *(unsigned int *)am3517evm_dai.cpu_dai->private_data = 0; /* McBSP1 */ + + ret = platform_device_add(am3517evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(am3517evm_snd_device); + + return ret; +} + +static void __exit am3517evm_soc_exit(void) +{ + platform_device_unregister(am3517evm_snd_device); +} + +module_init(am3517evm_soc_init); +module_exit(am3517evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal "); +MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.2 From 67e646cd7b51e1d5847fb506d4419d436ea25fda Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 30 Oct 2009 00:22:39 +0530 Subject: ASoC: Modifying Kconfig/Makefile for AM3517 EVM Modifying the Kconfig and Makefile in sound/soc/omap folder to add support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 9 +++++++++ sound/soc/omap/Makefile | 2 ++ 2 files changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2dee9839be86..6344456e7a09 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -55,6 +55,15 @@ config SND_OMAP_SOC_OMAP3EVM help Say Y if you want to add support for SoC audio on the omap3evm board. +config SND_OMAP_SOC_AM3517EVM + tristate "SoC Audio support for OMAP3517 / AM3517 EVM" + depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 + EVM. + config SND_OMAP_SOC_SDP3430 tristate "SoC Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d69471dcb5..0c78ae4e6b97 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,6 +12,7 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o +snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o @@ -23,6 +24,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- cgit v1.2.2 From 739b47f1e5aa3b36eadd7906cc6b41f0175c6ed1 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:34:19 +0100 Subject: ALSA: hda - select IbexPeak handler for Calpella An earlier patch merely adds id for 0x80862804. It has 2/3 cvt/pin nodes and shall be tied to the IbexPeak handler. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 01a18ed475ac..7c23016fe8fa 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -684,7 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, - { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, + { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ -- cgit v1.2.2 From 9ddc9aa910687a8787dbbdc53dcd48e738b197d9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 30 Oct 2009 12:02:39 +0900 Subject: ASoC: sh: FSI: Remove DMA support SuperH FSI device have the hardware limitation to use DMA. If DMA is used, LCD output will be broken. Maybe there are some solution. But I don't know how to do it now. This patch remove DMA support and use soft transfer. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 - sound/soc/sh/fsi.c | 141 ++++++++------------------------------------------- 2 files changed, 20 insertions(+), 122 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9154b4363db3..9e6976586554 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -23,7 +23,6 @@ config SND_SOC_SH4_SSI config SND_SOC_SH4_FSI tristate "SH4 FSI support" depends on CPU_SUBTYPE_SH7724 - select SH_DMA help This option enables FSI sound support diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 44123248b630..9742a280ba15 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -26,8 +26,6 @@ #include #include #include -#include -#include #define DO_FMT 0x0000 #define DOFF_CTL 0x0004 @@ -97,7 +95,6 @@ struct fsi_priv { int fifo_max; int chan; - int dma_chan; int byte_offset; int period_len; @@ -308,62 +305,6 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) return residue; } -static int fsi_get_residue(struct fsi_priv *fsi, int is_play) -{ - int residue; - int width; - struct snd_pcm_runtime *runtime; - - runtime = fsi->substream->runtime; - - /* get 1 channel data width */ - width = frames_to_bytes(runtime, 1) / fsi->chan; - - if (2 == width) - residue = fsi_get_fifo_residue(fsi, is_play); - else - residue = get_dma_residue(fsi->dma_chan); - - return residue; -} - -/************************************************************************ - - - basic dma function - - -************************************************************************/ -#define PORTA_DMA 0 -#define PORTB_DMA 1 - -static int fsi_get_dma_chan(void) -{ - if (0 != request_dma(PORTA_DMA, "fsia")) - return -EIO; - - if (0 != request_dma(PORTB_DMA, "fsib")) { - free_dma(PORTA_DMA); - return -EIO; - } - - master->fsia.dma_chan = PORTA_DMA; - master->fsib.dma_chan = PORTB_DMA; - - return 0; -} - -static void fsi_free_dma_chan(void) -{ - dma_wait_for_completion(PORTA_DMA); - dma_wait_for_completion(PORTB_DMA); - free_dma(PORTA_DMA); - free_dma(PORTB_DMA); - - master->fsia.dma_chan = -1; - master->fsib.dma_chan = -1; -} - /************************************************************************ @@ -435,44 +376,6 @@ static void fsi_soft_all_reset(void) mdelay(10); } -static void fsi_16data_push(struct fsi_priv *fsi, - struct snd_pcm_runtime *runtime, - int send) -{ - u16 *dma_start; - u32 snd; - int i; - - /* get dma start position for FSI */ - dma_start = (u16 *)runtime->dma_area; - dma_start += fsi->byte_offset / 2; - - /* - * soft dma - * FSI can not use DMA when 16bpp - */ - for (i = 0; i < send; i++) { - snd = (u32)dma_start[i]; - fsi_reg_write(fsi, DODT, snd << 8); - } -} - -static void fsi_32data_push(struct fsi_priv *fsi, - struct snd_pcm_runtime *runtime, - int send) -{ - u32 *dma_start; - - /* get dma start position for FSI */ - dma_start = (u32 *)runtime->dma_area; - dma_start += fsi->byte_offset / 4; - - dma_wait_for_completion(fsi->dma_chan); - dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); - dma_write(fsi->dma_chan, (u32)dma_start, - (u32)(fsi->base + DODT), send * 4); -} - /* playback interrupt */ static int fsi_data_push(struct fsi_priv *fsi) { @@ -481,6 +384,8 @@ static int fsi_data_push(struct fsi_priv *fsi) int send; int fifo_free; int width; + u8 *start; + int i; if (!fsi || !fsi->substream || @@ -515,12 +420,22 @@ static int fsi_data_push(struct fsi_priv *fsi) if (fifo_free < send) send = fifo_free; - if (2 == width) - fsi_16data_push(fsi, runtime, send); - else if (4 == width) - fsi_32data_push(fsi, runtime, send); - else + start = runtime->dma_area; + start += fsi->byte_offset; + + switch (width) { + case 2: + for (i = 0; i < send; i++) + fsi_reg_write(fsi, DODT, + ((u32)*((u16 *)start + i) << 8)); + break; + case 4: + for (i = 0; i < send; i++) + fsi_reg_write(fsi, DODT, *((u32 *)start + i)); + break; + default: return -EINVAL; + } fsi->byte_offset += send * width; @@ -664,8 +579,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, } fsi_reg_write(fsi, reg, data); - dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n", - msg, fsi->chan, fsi->dma_chan); /* * clear clk reset if master mode @@ -780,10 +693,9 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsi_priv *fsi = fsi_get(substream); - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; long location; - location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + location = (fsi->byte_offset - 1); if (location < 0) location = 0; @@ -912,22 +824,13 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - master->fsia.dma_chan = -1; - master->fsib.dma_chan = -1; - - ret = fsi_get_dma_chan(); - if (ret < 0) { - dev_err(&pdev->dev, "cannot get dma api\n"); - goto exit_iounmap; - } - /* FSI is based on SPU mstp */ snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); master->clk = clk_get(NULL, clk_name); if (IS_ERR(master->clk)) { dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); ret = -EIO; - goto exit_free_dma; + goto exit_iounmap; } fsi_soc_dai[0].dev = &pdev->dev; @@ -938,7 +841,7 @@ static int fsi_probe(struct platform_device *pdev) ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); - goto exit_free_dma; + goto exit_iounmap; } ret = snd_soc_register_platform(&fsi_soc_platform); @@ -951,8 +854,6 @@ static int fsi_probe(struct platform_device *pdev) exit_free_irq: free_irq(irq, master); -exit_free_dma: - fsi_free_dma_chan(); exit_iounmap: iounmap(master->base); exit_kfree: @@ -969,8 +870,6 @@ static int fsi_remove(struct platform_device *pdev) clk_put(master->clk); - fsi_free_dma_chan(); - free_irq(master->irq, master); iounmap(master->base); -- cgit v1.2.2 From 07102f3cefc93aa742af91186830e282c0347e41 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 30 Oct 2009 12:02:44 +0900 Subject: ASoC: sh: FSI: Add capture support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 93 ++++++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 86 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9742a280ba15..e1a3d1a2b4c8 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -447,6 +447,75 @@ static int fsi_data_push(struct fsi_priv *fsi) return 0; } +static int fsi_data_pop(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int free; + int fifo_fill; + int width; + u8 *start; + int i; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get free space for alsa */ + free = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get recv size */ + fifo_fill = fsi_get_fifo_residue(fsi, 0); + + if (free < fifo_fill) + fifo_fill = free; + + start = runtime->dma_area; + start += fsi->byte_offset; + + switch (width) { + case 2: + for (i = 0; i < fifo_fill; i++) + *((u16 *)start + i) = + (u16)(fsi_reg_read(fsi, DIDT) >> 8); + break; + case 4: + for (i = 0; i < fifo_fill; i++) + *((u32 *)start + i) = fsi_reg_read(fsi, DIDT); + break; + default: + return -EINVAL; + } + + fsi->byte_offset += fifo_fill * width; + + fsi_irq_enable(fsi, 0); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + static irqreturn_t fsi_interrupt(int irq, void *data) { u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; @@ -460,6 +529,10 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_data_push(&master->fsia); if (int_st & INT_B_OUT) fsi_data_push(&master->fsib); + if (int_st & INT_A_IN) + fsi_data_pop(&master->fsia); + if (int_st & INT_B_IN) + fsi_data_pop(&master->fsib); fsi_master_write(INT_ST, 0x0000000); @@ -612,16 +685,12 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; - /* capture not supported */ - if (!is_play) - return -ENODEV; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = fsi_data_push(fsi); + ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); @@ -757,7 +826,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, { @@ -769,7 +843,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, }; -- cgit v1.2.2 From f5d6def5c642587434c42722c57fb65642f61038 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:38:26 +0100 Subject: ALSA: hda - vectorize get_empty_pcm_device() This unifies the code and data structure, and makes it easy to add more HDMI devices. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 49 ++++++++++++++++------------------------------- 1 file changed, 16 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index af989f660cca..49289cd50697 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2885,43 +2885,26 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) static const char *dev_name[HDA_PCM_NTYPES] = { "Audio", "SPDIF", "HDMI", "Modem" }; - /* starting device index for each PCM type */ - static int dev_idx[HDA_PCM_NTYPES] = { - [HDA_PCM_TYPE_AUDIO] = 0, - [HDA_PCM_TYPE_SPDIF] = 1, - [HDA_PCM_TYPE_HDMI] = 3, - [HDA_PCM_TYPE_MODEM] = 6 + /* audio device indices; not linear to keep compatibility */ + static int audio_idx[HDA_PCM_NTYPES][5] = { + [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, + [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, + [HDA_PCM_TYPE_HDMI] = { 3, -1 }, + [HDA_PCM_TYPE_MODEM] = { 6, -1 }, }; - /* normal audio device indices; not linear to keep compatibility */ - static int audio_idx[4] = { 0, 2, 4, 5 }; - int i, dev; - - switch (type) { - case HDA_PCM_TYPE_AUDIO: - for (i = 0; i < ARRAY_SIZE(audio_idx); i++) { - dev = audio_idx[i]; - if (!test_bit(dev, bus->pcm_dev_bits)) - goto ok; - } - snd_printk(KERN_WARNING "Too many audio devices\n"); - return -EAGAIN; - case HDA_PCM_TYPE_SPDIF: - case HDA_PCM_TYPE_HDMI: - case HDA_PCM_TYPE_MODEM: - dev = dev_idx[type]; - if (test_bit(dev, bus->pcm_dev_bits)) { - snd_printk(KERN_WARNING "%s already defined\n", - dev_name[type]); - return -EAGAIN; - } - break; - default: + int i; + + if (type >= HDA_PCM_NTYPES) { snd_printk(KERN_WARNING "Invalid PCM type %d\n", type); return -EINVAL; } - ok: - set_bit(dev, bus->pcm_dev_bits); - return dev; + + for (i = 0; audio_idx[type][i] >= 0 ; i++) + if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) + return audio_idx[type][i]; + + snd_printk(KERN_WARNING "Too many %s devices\n", dev_name[type]); + return -EAGAIN; } /* -- cgit v1.2.2 From 92608badc519a8c1f65d93743396517aaa582b53 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:40:03 +0100 Subject: ALSA: hda - allow up to 4 HDMI devices The new Intel HDMI codec supports 2 independant HDMI/DisplayPort pipes. We'll be exporting them as 2 pcm devices. So bump up the allowed number of HDMI devices. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 49289cd50697..2c1366343335 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2889,7 +2889,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, - [HDA_PCM_TYPE_HDMI] = { 3, -1 }, + [HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 }, [HDA_PCM_TYPE_MODEM] = { 6, -1 }, }; int i; -- cgit v1.2.2 From 6797cf2bfcbf2fa1fd05c0b785dc1402f73e2ce5 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:40:40 +0100 Subject: ALSA: hda - convert intelhdmi global references to local parameters No behavior change. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 80 ++++++++++++++++++++++------------------- 1 file changed, 43 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 7c23016fe8fa..2dfb1efc2d08 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -189,35 +189,36 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { */ #ifdef BE_PARANOID -static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t nid, +static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) { int val; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_INDEX, 0); + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_INDEX, 0); *packet_index = val >> 5; *byte_index = val & 0x1f; } #endif -static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t nid, +static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int packet_index, int byte_index) { int val; val = (packet_index << 5) | (byte_index & 0x1f); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); } -static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, +static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, unsigned char val) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); } -static void hdmi_enable_output(struct hda_codec *codec) +static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) { /* Unmute */ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) @@ -231,7 +232,8 @@ static void hdmi_enable_output(struct hda_codec *codec) /* * Enable Audio InfoFrame Transmission */ -static void hdmi_start_infoframe_trans(struct hda_codec *codec) +static void hdmi_start_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) { hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, @@ -241,37 +243,40 @@ static void hdmi_start_infoframe_trans(struct hda_codec *codec) /* * Disable Audio InfoFrame Transmission */ -static void hdmi_stop_infoframe_trans(struct hda_codec *codec) +static void hdmi_stop_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) { hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); } -static int hdmi_get_channel_count(struct hda_codec *codec) +static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) { - return 1 + snd_hda_codec_read(codec, cvt_nid, 0, + return 1 + snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CVT_CHAN_COUNT, 0); } -static void hdmi_set_channel_count(struct hda_codec *codec, int chs) +static void hdmi_set_channel_count(struct hda_codec *codec, + hda_nid_t nid, int chs) { - snd_hda_codec_write(codec, cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); - if (chs != hdmi_get_channel_count(codec)) +#ifdef CONFIG_SND_DEBUG_VERBOSE + if (chs != hdmi_get_channel_count(codec, nid)) snd_printd(KERN_INFO "HDMI channel count: expect %d, get %d\n", - chs, hdmi_get_channel_count(codec)); + chs, hdmi_get_channel_count(codec, nid)); +#endif } -static void hdmi_debug_channel_mapping(struct hda_codec *codec) +static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid) { #ifdef CONFIG_SND_DEBUG_VERBOSE int i; int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, cvt_nid, 0, + slot = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_CHAN_SLOT, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", slot >> 4, slot & 0x7); @@ -293,7 +298,7 @@ static void hdmi_parse_eld(struct hda_codec *codec) * Audio InfoFrame routines */ -static void hdmi_debug_dip_size(struct hda_codec *codec) +static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) { #ifdef CONFIG_SND_DEBUG_VERBOSE int i; @@ -310,7 +315,7 @@ static void hdmi_debug_dip_size(struct hda_codec *codec) #endif } -static void hdmi_clear_dip_buffers(struct hda_codec *codec) +static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) { #ifdef BE_PARANOID int i, j; @@ -340,14 +345,15 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec) } static void hdmi_fill_audio_infoframe(struct hda_codec *codec, - struct hdmi_audio_infoframe *ai) + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) { u8 *params = (u8 *)ai; u8 sum = 0; int i; - hdmi_debug_dip_size(codec); - hdmi_clear_dip_buffers(codec); /* be paranoid */ + hdmi_debug_dip_size(codec, pin_nid); + hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ for (i = 0; i < sizeof(ai); i++) sum += params[i]; @@ -386,7 +392,7 @@ static void init_channel_allocations(void) * * TODO: it could select the wrong CA from multiple candidates. */ -static int hdmi_setup_channel_allocation(struct hda_codec *codec, +static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, struct hdmi_audio_infoframe *ai) { struct intel_hdmi_spec *spec = codec->spec; @@ -439,8 +445,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, return ai->CA; } -static void hdmi_setup_channel_mapping(struct hda_codec *codec, - struct hdmi_audio_infoframe *ai) +static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t nid, + struct hdmi_audio_infoframe *ai) { int i; @@ -453,15 +459,15 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, */ for (i = 0; i < 8; i++) - snd_hda_codec_write(codec, cvt_nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, (i << 4) | i); - hdmi_debug_channel_mapping(codec); + hdmi_debug_channel_mapping(codec, nid); } -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, struct snd_pcm_substream *substream) { struct hdmi_audio_infoframe ai = { @@ -471,11 +477,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, .CC02_CT47 = substream->runtime->channels - 1, }; - hdmi_setup_channel_allocation(codec, &ai); - hdmi_setup_channel_mapping(codec, &ai); + hdmi_setup_channel_allocation(codec, nid, &ai); + hdmi_setup_channel_mapping(codec, nid, &ai); - hdmi_fill_audio_infoframe(codec, &ai); - hdmi_start_infoframe_trans(codec); + hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_start_infoframe_trans(codec, pin_nid); } @@ -553,7 +559,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, { struct intel_hdmi_spec *spec = codec->spec; - hdmi_stop_infoframe_trans(codec); + hdmi_stop_infoframe_trans(codec, pin_nid); return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -569,9 +575,9 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, format, substream); - hdmi_set_channel_count(codec, substream->runtime->channels); + hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, substream); + hdmi_setup_audio_infoframe(codec, cvt_nid, substream); return 0; } @@ -619,7 +625,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) static int intel_hdmi_init(struct hda_codec *codec) { - hdmi_enable_output(codec); + hdmi_enable_output(codec, pin_nid); snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, -- cgit v1.2.2 From 7bedb011ef4db93b15049ece8d50b29d6fe6af9d Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:41:44 +0100 Subject: ALSA: hda - remove intelhdmi dependency on multiout We'll be managing multiple HDMI audio sources/sinks on our own. So remove multiout dependency from intelhdmi. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 23 +++++++---------------- 1 file changed, 7 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 2dfb1efc2d08..02be428be667 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -39,7 +39,6 @@ static hda_nid_t pin_nid; /* HDMI output pin */ #define INTEL_HDMI_EVENT_TAG 0x08 struct intel_hdmi_spec { - struct hda_multi_out multiout; struct hda_pcm pcm_rec; struct hdmi_eld sink_eld; }; @@ -548,9 +547,7 @@ static int intel_hdmi_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct intel_hdmi_spec *spec = codec->spec; - - return snd_hda_multi_out_dig_open(codec, &spec->multiout); + return 0; } static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, @@ -561,7 +558,8 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, hdmi_stop_infoframe_trans(codec, pin_nid); - return snd_hda_multi_out_dig_close(codec, &spec->multiout); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); + return 0; } static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -570,15 +568,12 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int format, struct snd_pcm_substream *substream) { - struct intel_hdmi_spec *spec = codec->spec; - - snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); - - hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); + hdmi_set_channel_count(codec, cvt_nid, + substream->runtime->channels); hdmi_setup_audio_infoframe(codec, cvt_nid, substream); + snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); return 0; } @@ -616,7 +611,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) struct intel_hdmi_spec *spec = codec->spec; int err; - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + err = snd_hda_create_spdif_out_ctls(codec, cvt_nid); if (err < 0) return err; @@ -657,10 +652,6 @@ static int do_patch_intel_hdmi(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 8; - spec->multiout.dig_out_nid = cvt_nid; - codec->spec = spec; codec->patch_ops = intel_hdmi_patch_ops; -- cgit v1.2.2 From 70ca35fb42680fc4315d4a01f6c77c9a9962199c Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:42:18 +0100 Subject: ALSA: hda - use pcm prepare/cleanup callbacks for intelhdmi Remove pcm callbacks open/close in favor of the prepare/cleanup. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 02be428be667..c17feacab754 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -543,16 +543,9 @@ static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) * Callbacks */ -static int intel_hdmi_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - -static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct intel_hdmi_spec *spec = codec->spec; @@ -582,9 +575,8 @@ static struct hda_pcm_stream intel_hdmi_pcm_playback = { .channels_min = 2, .channels_max = 8, .ops = { - .open = intel_hdmi_playback_pcm_open, - .close = intel_hdmi_playback_pcm_close, - .prepare = intel_hdmi_playback_pcm_prepare + .prepare = intel_hdmi_playback_pcm_prepare, + .cleanup = intel_hdmi_playback_pcm_cleanup, }, }; -- cgit v1.2.2 From ddb8152b054e357907f57fb5ae65d494a3c79065 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:43:03 +0100 Subject: ALSA: hda - reorder intelhdmi prepare/cleanup callbacks No behavior change. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index c17feacab754..6be5ca44a83b 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -543,30 +543,30 @@ static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) * Callbacks */ -static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, +static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, struct snd_pcm_substream *substream) { - struct intel_hdmi_spec *spec = codec->spec; + hdmi_set_channel_count(codec, cvt_nid, + substream->runtime->channels); - hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_setup_audio_infoframe(codec, cvt_nid, substream); - snd_hda_codec_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); return 0; } -static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, +static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, struct snd_pcm_substream *substream) { - hdmi_set_channel_count(codec, cvt_nid, - substream->runtime->channels); + struct intel_hdmi_spec *spec = codec->spec; - hdmi_setup_audio_infoframe(codec, cvt_nid, substream); + hdmi_stop_infoframe_trans(codec, pin_nid); - snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } -- cgit v1.2.2 From 54a25f87e943fc77f57e86849897ad6602519286 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:44:26 +0100 Subject: ALSA: hda - vectorize intelhdmi The Intel IbexPeak HDMI codec supports 2 converters and 3 pins, which requires converting the cvt_nid/pin_nid to arrays. The active pin number (the one connected with a live HDMI monitor/sink) will be dynamically identified on hotplug events. It exports two HDMI devices, so that user space can choose the A/V pipe for sending the audio samples. It's still undefined behavior when there are two active monitors connected and routed to the same audio converter. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 5 +- sound/pci/hda/hda_local.h | 6 +- sound/pci/hda/patch_intelhdmi.c | 191 +++++++++++++++++++++++++++++++--------- 3 files changed, 155 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 9446a5abea13..20fa6aee29c0 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -560,13 +560,14 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, } -int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld) +int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, + int index) { char name[32]; struct snd_info_entry *entry; int err; - snprintf(name, sizeof(name), "eld#%d", codec->addr); + snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index); err = snd_card_proc_new(codec->bus->card, name, &entry); if (err < 0) return err; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 5f1dcc59002b..461e0c15c77a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -541,11 +541,13 @@ int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t); void snd_hdmi_show_eld(struct hdmi_eld *eld); #ifdef CONFIG_PROC_FS -int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld); +int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, + int index); void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld); #else static inline int snd_hda_eld_proc_new(struct hda_codec *codec, - struct hdmi_eld *eld) + struct hdmi_eld *eld, + int index) { return 0; } diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 6be5ca44a83b..08ea88deba6f 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -33,14 +33,43 @@ #include "hda_codec.h" #include "hda_local.h" -static hda_nid_t cvt_nid; /* audio converter */ -static hda_nid_t pin_nid; /* HDMI output pin */ +/* + * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device + * could support two independent pipes, each of them can be connected to one or + * more ports (DVI, HDMI or DisplayPort). + * + * The HDA correspondence of pipes/ports are converter/pin nodes. + */ +#define INTEL_HDMI_CVTS 2 +#define INTEL_HDMI_PINS 3 -#define INTEL_HDMI_EVENT_TAG 0x08 +static char *intel_hdmi_pcm_names[INTEL_HDMI_CVTS] = { + "INTEL HDMI 0", + "INTEL HDMI 1", +}; struct intel_hdmi_spec { - struct hda_pcm pcm_rec; - struct hdmi_eld sink_eld; + int num_cvts; + int num_pins; + hda_nid_t cvt[INTEL_HDMI_CVTS+1]; /* audio sources */ + hda_nid_t pin[INTEL_HDMI_PINS+1]; /* audio sinks */ + + /* + * source connection for each pin + */ + hda_nid_t pin_cvt[INTEL_HDMI_PINS+1]; + + /* + * HDMI sink attached to each pin + */ + bool sink_present[INTEL_HDMI_PINS]; + bool sink_eldv[INTEL_HDMI_PINS]; + struct hdmi_eld sink_eld[INTEL_HDMI_PINS]; + + /* + * export one pcm per pipe + */ + struct hda_pcm pcm_rec[INTEL_HDMI_CVTS]; }; struct hdmi_audio_infoframe { @@ -183,6 +212,19 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, }; + +static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) +{ + int i; + + for (i = 0; nids[i]; i++) + if (nids[i] == nid) + return i; + + snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); + return -EINVAL; +} + /* * HDMI routines */ @@ -283,12 +325,12 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid) #endif } -static void hdmi_parse_eld(struct hda_codec *codec) +static void hdmi_parse_eld(struct hda_codec *codec, int index) { struct intel_hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld = &spec->sink_eld; + struct hdmi_eld *eld = &spec->sink_eld[index]; - if (!snd_hdmi_get_eld(eld, codec, pin_nid)) + if (!snd_hdmi_get_eld(eld, codec, spec->pin[index])) snd_hdmi_show_eld(eld); } @@ -395,7 +437,7 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, struct hdmi_audio_infoframe *ai) { struct intel_hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld = &spec->sink_eld; + struct hdmi_eld *eld; int i; int spk_mask = 0; int channels = 1 + (ai->CC02_CT47 & 0x7); @@ -407,6 +449,11 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, if (channels <= 2) return 0; + i = hda_node_index(spec->pin_cvt, nid); + if (i < 0) + return 0; + eld = &spec->sink_eld[i]; + /* * HDMI sink's ELD info cannot always be retrieved for now, e.g. * in console or for audio devices. Assume the highest speakers @@ -469,6 +516,9 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t nid, static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, struct snd_pcm_substream *substream) { + struct intel_hdmi_spec *spec = codec->spec; + hda_nid_t pin_nid; + int i; struct hdmi_audio_infoframe ai = { .type = 0x84, .ver = 0x01, @@ -479,8 +529,16 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, hdmi_setup_channel_allocation(codec, nid, &ai); hdmi_setup_channel_mapping(codec, nid, &ai); - hdmi_fill_audio_infoframe(codec, pin_nid, &ai); - hdmi_start_infoframe_trans(codec, pin_nid); + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != nid) + continue; + if (spec->sink_present[i] != true) + continue; + + pin_nid = spec->pin[i]; + hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_start_infoframe_trans(codec, pin_nid); + } } @@ -490,27 +548,39 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { + struct intel_hdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int pind = !!(res & AC_UNSOL_RES_PD); int eldv = !!(res & AC_UNSOL_RES_ELDV); + int index; printk(KERN_INFO - "HDMI hot plug event: Presence_Detect=%d ELD_Valid=%d\n", - pind, eldv); + "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", + tag, pind, eldv); + + index = hda_node_index(spec->pin, tag); + if (index < 0) + return; + + spec->sink_present[index] = pind; + spec->sink_eldv[index] = eldv; if (pind && eldv) { - hdmi_parse_eld(codec); + hdmi_parse_eld(codec, index); /* TODO: do real things about ELD */ } } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) { + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); printk(KERN_INFO - "HDMI content protection event: SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + tag, subtag, cp_state, cp_ready); @@ -525,10 +595,11 @@ static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) { + struct intel_hdmi_spec *spec = codec->spec; int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - if (tag != INTEL_HDMI_EVENT_TAG) { + if (hda_node_index(spec->pin, tag) < 0) { snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); return; } @@ -549,10 +620,10 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int format, struct snd_pcm_substream *substream) { - hdmi_set_channel_count(codec, cvt_nid, + hdmi_set_channel_count(codec, hinfo->nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, cvt_nid, substream); + hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); return 0; @@ -563,8 +634,14 @@ static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct intel_hdmi_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != hinfo->nid) + continue; - hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_stop_infoframe_trans(codec, spec->pin[i]); + } snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; @@ -583,17 +660,19 @@ static struct hda_pcm_stream intel_hdmi_pcm_playback = { static int intel_hdmi_build_pcms(struct hda_codec *codec) { struct intel_hdmi_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm_rec; + struct hda_pcm *info = spec->pcm_rec; + int i; - codec->num_pcms = 1; + codec->num_pcms = spec->num_cvts; codec->pcm_info = info; - /* NID to query formats and rates and setup streams */ - intel_hdmi_pcm_playback.nid = cvt_nid; - - info->name = "INTEL HDMI"; - info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback; + for (i = 0; i < codec->num_pcms; i++, info++) { + info->name = intel_hdmi_pcm_names[i]; + info->pcm_type = HDA_PCM_TYPE_HDMI; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + intel_hdmi_pcm_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->cvt[i]; + } return 0; } @@ -602,29 +681,39 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) { struct intel_hdmi_spec *spec = codec->spec; int err; + int i; - err = snd_hda_create_spdif_out_ctls(codec, cvt_nid); - if (err < 0) - return err; + for (i = 0; i < codec->num_pcms; i++) { + err = snd_hda_create_spdif_out_ctls(codec, spec->cvt[i]); + if (err < 0) + return err; + } return 0; } static int intel_hdmi_init(struct hda_codec *codec) { - hdmi_enable_output(codec, pin_nid); + struct intel_hdmi_spec *spec = codec->spec; + int i; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | INTEL_HDMI_EVENT_TAG); + for (i = 0; spec->pin[i]; i++) { + hdmi_enable_output(codec, spec->pin[i]); + snd_hda_codec_write(codec, spec->pin[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | spec->pin[i]); + } return 0; } static void intel_hdmi_free(struct hda_codec *codec) { struct intel_hdmi_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); - snd_hda_eld_proc_free(codec, &spec->sink_eld); kfree(spec); } @@ -636,18 +725,38 @@ static struct hda_codec_ops intel_hdmi_patch_ops = { .unsol_event = intel_hdmi_unsol_event, }; -static int do_patch_intel_hdmi(struct hda_codec *codec) +static struct intel_hdmi_spec static_specs[] = { + { + .num_cvts = 1, + .num_pins = 1, + .cvt = { 0x2 }, + .pin = { 0x3 }, + .pin_cvt = { 0x2 }, + }, + { + .num_cvts = 2, + .num_pins = 3, + .cvt = { 0x2, 0x3 }, + .pin = { 0x4, 0x5, 0x6 }, + .pin_cvt = { 0x2, 0x2, 0x2 }, + }, +}; + +static int do_patch_intel_hdmi(struct hda_codec *codec, int spec_id) { struct intel_hdmi_spec *spec; + int i; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; + *spec = static_specs[spec_id]; codec->spec = spec; codec->patch_ops = intel_hdmi_patch_ops; - snd_hda_eld_proc_new(codec, &spec->sink_eld); + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i); init_channel_allocations(); @@ -656,16 +765,12 @@ static int do_patch_intel_hdmi(struct hda_codec *codec) static int patch_intel_hdmi(struct hda_codec *codec) { - cvt_nid = 0x02; - pin_nid = 0x03; - return do_patch_intel_hdmi(codec); + return do_patch_intel_hdmi(codec, 0); } static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec) { - cvt_nid = 0x02; - pin_nid = 0x04; - return do_patch_intel_hdmi(codec); + return do_patch_intel_hdmi(codec, 1); } static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { -- cgit v1.2.2 From 69fb346896b4265c0cbcbd2fdd1a97ae09fe198d Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:45:04 +0100 Subject: ALSA: hda - get intelhdmi max channels from widget caps Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 08ea88deba6f..3c68aa9742d7 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -650,7 +650,6 @@ static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, static struct hda_pcm_stream intel_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, - .channels_max = 8, .ops = { .prepare = intel_hdmi_playback_pcm_prepare, .cleanup = intel_hdmi_playback_pcm_cleanup, @@ -667,11 +666,17 @@ static int intel_hdmi_build_pcms(struct hda_codec *codec) codec->pcm_info = info; for (i = 0; i < codec->num_pcms; i++, info++) { + unsigned int chans; + + chans = get_wcaps(codec, spec->cvt[i]); + chans = get_wcaps_channels(chans); + info->name = intel_hdmi_pcm_names[i]; info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->cvt[i]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; } return 0; -- cgit v1.2.2 From f424367c3a393ca8b9669ceaa5b7f959d83bb14c Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:45:35 +0100 Subject: ALSA: hda - auto parse intelhdmi cvt/pin configurations Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 120 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 119 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 3c68aa9742d7..1c374f11ed07 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -213,6 +213,10 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { }; +/* + * HDA/HDMI auto parsing + */ + static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) { int i; @@ -225,6 +229,113 @@ static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) return -EINVAL; } +static int intel_hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct intel_hdmi_spec *spec = codec->spec; + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int conn_len, curr; + int index; + + if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { + snd_printk(KERN_WARNING + "HDMI: pin %d wcaps %#x " + "does not support connection list\n", + pin_nid, get_wcaps(codec, pin_nid)); + return -EINVAL; + } + + conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, + HDA_MAX_CONNECTIONS); + if (conn_len > 1) + curr = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + else + curr = 0; + + index = hda_node_index(spec->pin, pin_nid); + if (index < 0) + return -EINVAL; + + spec->pin_cvt[index] = conn_list[curr]; + + return 0; +} + +static int intel_hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct intel_hdmi_spec *spec = codec->spec; + + if (spec->num_pins >= INTEL_HDMI_PINS) { + snd_printk(KERN_WARNING + "HDMI: no space for pin %d \n", pin_nid); + return -EINVAL; + } + + spec->pin[spec->num_pins] = pin_nid; + spec->num_pins++; + + /* + * It is assumed that converter nodes come first in the node list and + * hence have been registered and usable now. + */ + return intel_hdmi_read_pin_conn(codec, pin_nid); +} + +static int intel_hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) +{ + struct intel_hdmi_spec *spec = codec->spec; + + if (spec->num_cvts >= INTEL_HDMI_CVTS) { + snd_printk(KERN_WARNING + "HDMI: no space for converter %d \n", nid); + return -EINVAL; + } + + spec->cvt[spec->num_cvts] = nid; + spec->num_cvts++; + + return 0; +} + +static int intel_hdmi_parse_codec(struct hda_codec *codec) +{ + hda_nid_t nid; + int i, nodes; + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + if (!nid || nodes < 0) { + snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); + return -EINVAL; + } + + for (i = 0; i < nodes; i++, nid++) { + unsigned int caps; + unsigned int type; + + caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + type = get_wcaps_type(caps); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + switch (type) { + case AC_WID_AUD_OUT: + if (intel_hdmi_add_cvt(codec, nid) < 0) + return -EINVAL; + break; + case AC_WID_PIN: + caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (!(caps & AC_PINCAP_HDMI)) + continue; + if (intel_hdmi_add_pin(codec, nid) < 0) + return -EINVAL; + break; + } + } + + return 0; +} + /* * HDMI routines */ @@ -756,8 +867,15 @@ static int do_patch_intel_hdmi(struct hda_codec *codec, int spec_id) if (spec == NULL) return -ENOMEM; - *spec = static_specs[spec_id]; codec->spec = spec; + if (intel_hdmi_parse_codec(codec) < 0) { + codec->spec = NULL; + kfree(spec); + return -EINVAL; + } + if (memcmp(spec, static_specs + spec_id, sizeof(*spec))) + snd_printk(KERN_WARNING + "HDMI: auto parse disagree with known config\n"); codec->patch_ops = intel_hdmi_patch_ops; for (i = 0; i < spec->num_pins; i++) -- cgit v1.2.2 From fd080b2d8a6a13992b4b1b6300e1befdb9e089f2 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 30 Oct 2009 11:46:22 +0100 Subject: ALSA: hda - remove static intelhdmi configurations Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 36 +++--------------------------------- 1 file changed, 3 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 1c374f11ed07..650de1b4ea8d 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -841,24 +841,7 @@ static struct hda_codec_ops intel_hdmi_patch_ops = { .unsol_event = intel_hdmi_unsol_event, }; -static struct intel_hdmi_spec static_specs[] = { - { - .num_cvts = 1, - .num_pins = 1, - .cvt = { 0x2 }, - .pin = { 0x3 }, - .pin_cvt = { 0x2 }, - }, - { - .num_cvts = 2, - .num_pins = 3, - .cvt = { 0x2, 0x3 }, - .pin = { 0x4, 0x5, 0x6 }, - .pin_cvt = { 0x2, 0x2, 0x2 }, - }, -}; - -static int do_patch_intel_hdmi(struct hda_codec *codec, int spec_id) +static int patch_intel_hdmi(struct hda_codec *codec) { struct intel_hdmi_spec *spec; int i; @@ -873,9 +856,6 @@ static int do_patch_intel_hdmi(struct hda_codec *codec, int spec_id) kfree(spec); return -EINVAL; } - if (memcmp(spec, static_specs + spec_id, sizeof(*spec))) - snd_printk(KERN_WARNING - "HDMI: auto parse disagree with known config\n"); codec->patch_ops = intel_hdmi_patch_ops; for (i = 0; i < spec->num_pins; i++) @@ -886,23 +866,13 @@ static int do_patch_intel_hdmi(struct hda_codec *codec, int spec_id) return 0; } -static int patch_intel_hdmi(struct hda_codec *codec) -{ - return do_patch_intel_hdmi(codec, 0); -} - -static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec) -{ - return do_patch_intel_hdmi(codec, 1); -} - static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi }, { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, - { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, - { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, + { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, + { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; -- cgit v1.2.2 From 36dd5c4afff825fca1b6ccde678f51d6933a6850 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 20 Oct 2009 13:18:04 +0800 Subject: ALSA: VIA HDA: Add support for VT1818S. Add support for VT1818S codec, which is similiar with VT1708S. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 89e084d45369..5ec0e39593b5 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -41,6 +41,7 @@ /* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ /* 2009-07-08 Lydia Wang Add support for VT2002P */ /* 2009-07-21 Lydia Wang Add support for VT1812 */ +/* 2009-09-19 Lydia Wang Add support for VT1818S */ /* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -195,6 +196,8 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) codec_type = VT2002P; else if (dev_id == 0x0448) codec_type = VT1812; + else if (dev_id == 0x0440) + codec_type = VT1708S; else codec_type = UNKNOWN; return codec_type; @@ -4130,11 +4133,17 @@ static int patch_vt1708S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs; spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs; - spec->stream_name_analog = "VT1708S Analog"; + if (codec->vendor_id == 0x11060440) + spec->stream_name_analog = "VT1818S Analog"; + else + spec->stream_name_analog = "VT1708S Analog"; spec->stream_analog_playback = &vt1708S_pcm_analog_playback; spec->stream_analog_capture = &vt1708S_pcm_analog_capture; - spec->stream_name_digital = "VT1708S Digital"; + if (codec->vendor_id == 0x11060440) + spec->stream_name_digital = "VT1818S Digital"; + else + spec->stream_name_digital = "VT1708S Digital"; spec->stream_digital_playback = &vt1708S_pcm_digital_playback; if (!spec->adc_nids && spec->input_mux) { @@ -6231,6 +6240,8 @@ static struct hda_codec_preset snd_hda_preset_via[] = { { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, + { .id = 0x11060440, .name = "VT1818S", + .patch = patch_vt1708S}, {} /* terminator */ }; -- cgit v1.2.2 From 84ed1a1942e8c28fb4c23a6235ec48672fc43e49 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Fri, 23 Oct 2009 16:03:08 +0200 Subject: ALSA: Cleanup redundant tests on unsigned The variables are unsigned so the test `>= 0' is always true, the `< 0' test always fails. In these cases the other part of the test catches wrapped values. In dac_audio_write() there does not occur a test for wrapped values, but the test appears redundant. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/sh_dac_audio.c | 3 --- sound/pci/ca0106/ca0106_proc.c | 4 ++-- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +-- sound/pci/emu10k1/emuproc.c | 4 ++-- sound/pci/emu10k1/io.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- 7 files changed, 8 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index b2ed8757542a..4153752507e3 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -164,9 +164,6 @@ static ssize_t dac_audio_write(struct file *file, const char *buf, size_t count, int free; int nbytes; - if (count < 0) - return -EINVAL; - if (!count) { dac_audio_sync(); return 0; diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d10ec61..15523e60351c 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if (reg < 0x40 && val <= 0xffffffff) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -405,7 +405,7 @@ static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) ) + if (reg < 0x80 && val <= 0xffffffff && channel_id <= 3) snd_ca0106_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index b1b3a644f738..6bfce99b42a2 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -240,7 +240,7 @@ static int select_rom(unsigned int pitch) } else if (pitch == 0x02000000) { /* pitch == 2 */ return 3; - } else if (pitch >= 0x0 && pitch <= 0x08000000) { + } else if (pitch <= 0x08000000) { /* 0 <= pitch <= 8 */ return 0; } else { diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 36e08bd2b3cc..6b8ae7b5cd54 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f9748aff5..baa7cd508cd8 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -451,7 +451,7 @@ static void snd_emu_proc_io_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; - if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff) ) { + if (reg < 0x40 && val <= 0xffffffff) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if (reg < 0xa0 && val <= 0xffffffff && channel_id <= 3) snd_ptr_write(emu, iobase, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index c1a5aa15af8f..5ef7080e14d0 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -256,7 +256,7 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ - if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */ + if (value > 0x3f) /* 0 to 0x3f are values */ return 1; spin_lock_irqsave(&emu->emu_lock, flags); outl(reg, emu->port + A_IOCFG); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5cd729..35606ae60868 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, * of data into val */ - if ((reg < 0 || reg > 9) && (reg != 15)) { + if (reg > 9 && reg != 15) { printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; } -- cgit v1.2.2 From 6a5f96ce72b9e1a4bf06422df53fa819947d2293 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 Oct 2009 12:31:39 +0100 Subject: ALSA: hda - Add a proper ifdef to a debug code MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Added a proper ifdef CONFIG_SND_DEBUG_VERBOSE to avoid a compile warning: sound/pci/hda/patch_intelhdmi.c:406: warning: ‘hdmi_get_channel_count’ defined but not used Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 650de1b4ea8d..4f25f08d332b 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -403,11 +403,13 @@ static void hdmi_stop_infoframe_trans(struct hda_codec *codec, AC_DIPXMIT_DISABLE); } +#ifdef CONFIG_SND_DEBUG_VERBOSE static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) { return 1 + snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CVT_CHAN_COUNT, 0); } +#endif static void hdmi_set_channel_count(struct hda_codec *codec, hda_nid_t nid, int chs) -- cgit v1.2.2 From b7d5d946e50116f4150542f881ac90ac74c28165 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 24 Oct 2009 17:47:33 +0200 Subject: sound: remove OSS Ensoniq SoundScape driver The OSS driver for Ensoniq SoundScape cards is broken after conversion to mutexes and a new ALSA snd-sscape driver handles all devices handled by the OSS one. The ALSA driver was tested with these cards: Spea V7 MediaFX Ensoniq Soundscape Elite Ensoniq Soundscape VIVO (this card is not handled by the OSS driver) Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/oss/Kconfig | 12 - sound/oss/Makefile | 1 - sound/oss/sscape.c | 1480 ---------------------------------------------------- 3 files changed, 1493 deletions(-) delete mode 100644 sound/oss/sscape.c (limited to 'sound') diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index bcf2a0698d54..135a2b77cc4a 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -287,18 +287,6 @@ config SOUND_DMAP Say Y unless you have 16MB or more RAM or a PCI sound card. -config SOUND_SSCAPE - tristate "Ensoniq SoundScape support" - help - Answer Y if you have a sound card based on the Ensoniq SoundScape - chipset. Such cards are being manufactured at least by Ensoniq, Spea - and Reveal (Reveal makes also other cards). - - If you compile the driver into the kernel, you have to add - "sscape=,,,," to the kernel command - line. - - config SOUND_VMIDI tristate "Loopback MIDI device support" help diff --git a/sound/oss/Makefile b/sound/oss/Makefile index e0ae4d4d6a5c..567b8a74178a 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -13,7 +13,6 @@ obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o -obj-$(CONFIG_SOUND_SSCAPE) += sscape.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_MSS) += ad1848.o obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c deleted file mode 100644 index 30c36d1f35d7..000000000000 --- a/sound/oss/sscape.c +++ /dev/null @@ -1,1480 +0,0 @@ -/* - * sound/oss/sscape.c - * - * Low level driver for Ensoniq SoundScape - * - * - * Copyright (C) by Hannu Savolainen 1993-1997 - * - * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) - * Version 2 (June 1991). See the "COPYING" file distributed with this software - * for more info. - * - * - * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) - * Sergey Smitienko : ensoniq p'n'p support - * Christoph Hellwig : adapted to module_init/module_exit - * Bartlomiej Zolnierkiewicz : added __init to attach_sscape() - * Chris Rankin : Specify that this module owns the coprocessor - * Arnaldo C. de Melo : added missing restore_flags in sscape_pnp_upload_file - */ - -#include -#include - -#include "sound_config.h" -#include "sound_firmware.h" - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "coproc.h" - -#include "ad1848.h" -#include "mpu401.h" - -/* - * I/O ports - */ -#define MIDI_DATA 0 -#define MIDI_CTRL 1 -#define HOST_CTRL 2 -#define TX_READY 0x02 -#define RX_READY 0x01 -#define HOST_DATA 3 -#define ODIE_ADDR 4 -#define ODIE_DATA 5 - -/* - * Indirect registers - */ - -#define GA_INTSTAT_REG 0 -#define GA_INTENA_REG 1 -#define GA_DMAA_REG 2 -#define GA_DMAB_REG 3 -#define GA_INTCFG_REG 4 -#define GA_DMACFG_REG 5 -#define GA_CDCFG_REG 6 -#define GA_SMCFGA_REG 7 -#define GA_SMCFGB_REG 8 -#define GA_HMCTL_REG 9 - -/* - * DMA channel identifiers (A and B) - */ - -#define SSCAPE_DMA_A 0 -#define SSCAPE_DMA_B 1 - -#define PORT(name) (devc->base+name) - -/* - * Host commands recognized by the OBP microcode - */ - -#define CMD_GEN_HOST_ACK 0x80 -#define CMD_GEN_MPU_ACK 0x81 -#define CMD_GET_BOARD_TYPE 0x82 -#define CMD_SET_CONTROL 0x88 /* Old firmware only */ -#define CMD_GET_CONTROL 0x89 /* Old firmware only */ -#define CTL_MASTER_VOL 0 -#define CTL_MIC_MODE 2 -#define CTL_SYNTH_VOL 4 -#define CTL_WAVE_VOL 7 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d - -#define CMD_ACK 0x80 - -#define IC_ODIE 1 -#define IC_OPUS 2 - -typedef struct sscape_info -{ - int base, irq, dma; - - int codec, codec_irq; /* required to setup pnp cards*/ - int codec_type; - int ic_type; - char* raw_buf; - unsigned long raw_buf_phys; - int buffsize; /* -------------------------- */ - spinlock_t lock; - int ok; /* Properly detected */ - int failed; - int dma_allocated; - int codec_audiodev; - int opened; - int *osp; - int my_audiodev; -} sscape_info; - -static struct sscape_info adev_info = { - 0 -}; - -static struct sscape_info *devc = &adev_info; -static int sscape_mididev = -1; - -/* Some older cards have assigned interrupt bits differently than new ones */ -static char valid_interrupts_old[] = { - 9, 7, 5, 15 -}; - -static char valid_interrupts_new[] = { - 9, 5, 7, 10 -}; - -static char *valid_interrupts = valid_interrupts_new; - -/* - * See the bottom of the driver. This can be set by spea =0/1. - */ - -#ifdef REVEAL_SPEA -static char old_hardware = 1; -#else -static char old_hardware; -#endif - -static void sleep(unsigned howlong) -{ - current->state = TASK_INTERRUPTIBLE; - schedule_timeout(howlong); -} - -static unsigned char sscape_read(struct sscape_info *devc, int reg) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&devc->lock,flags); - outb(reg, PORT(ODIE_ADDR)); - val = inb(PORT(ODIE_DATA)); - spin_unlock_irqrestore(&devc->lock,flags); - return val; -} - -static void __sscape_write(int reg, int data) -{ - outb(reg, PORT(ODIE_ADDR)); - outb(data, PORT(ODIE_DATA)); -} - -static void sscape_write(struct sscape_info *devc, int reg, int data) -{ - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - __sscape_write(reg, data); - spin_unlock_irqrestore(&devc->lock,flags); -} - -static unsigned char sscape_pnp_read_codec(sscape_info* devc, unsigned char reg) -{ - unsigned char res; - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - outb( reg, devc -> codec); - res = inb (devc -> codec + 1); - spin_unlock_irqrestore(&devc->lock,flags); - return res; - -} - -static void sscape_pnp_write_codec(sscape_info* devc, unsigned char reg, unsigned char data) -{ - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - outb( reg, devc -> codec); - outb( data, devc -> codec + 1); - spin_unlock_irqrestore(&devc->lock,flags); -} - -static void host_open(struct sscape_info *devc) -{ - outb((0x00), PORT(HOST_CTRL)); /* Put the board to the host mode */ -} - -static void host_close(struct sscape_info *devc) -{ - outb((0x03), PORT(HOST_CTRL)); /* Put the board to the MIDI mode */ -} - -static int host_write(struct sscape_info *devc, unsigned char *data, int count) -{ - unsigned long flags; - int i, timeout_val; - - spin_lock_irqsave(&devc->lock,flags); - /* - * Send the command and data bytes - */ - - for (i = 0; i < count; i++) - { - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - if (inb(PORT(HOST_CTRL)) & TX_READY) - break; - - if (timeout_val <= 0) - { - spin_unlock_irqrestore(&devc->lock,flags); - return 0; - } - outb(data[i], PORT(HOST_DATA)); - } - spin_unlock_irqrestore(&devc->lock,flags); - return 1; -} - -static int host_read(struct sscape_info *devc) -{ - unsigned long flags; - int timeout_val; - unsigned char data; - - spin_lock_irqsave(&devc->lock,flags); - /* - * Read a byte - */ - - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - if (inb(PORT(HOST_CTRL)) & RX_READY) - break; - - if (timeout_val <= 0) - { - spin_unlock_irqrestore(&devc->lock,flags); - return -1; - } - data = inb(PORT(HOST_DATA)); - spin_unlock_irqrestore(&devc->lock,flags); - return data; -} - -#if 0 /* unused */ -static int host_command1(struct sscape_info *devc, int cmd) -{ - unsigned char buf[10]; - buf[0] = (unsigned char) (cmd & 0xff); - return host_write(devc, buf, 1); -} -#endif /* unused */ - - -static int host_command2(struct sscape_info *devc, int cmd, int parm1) -{ - unsigned char buf[10]; - - buf[0] = (unsigned char) (cmd & 0xff); - buf[1] = (unsigned char) (parm1 & 0xff); - - return host_write(devc, buf, 2); -} - -static int host_command3(struct sscape_info *devc, int cmd, int parm1, int parm2) -{ - unsigned char buf[10]; - - buf[0] = (unsigned char) (cmd & 0xff); - buf[1] = (unsigned char) (parm1 & 0xff); - buf[2] = (unsigned char) (parm2 & 0xff); - return host_write(devc, buf, 3); -} - -static void set_mt32(struct sscape_info *devc, int value) -{ - host_open(devc); - host_command2(devc, CMD_SET_MT32, value ? 1 : 0); - if (host_read(devc) != CMD_ACK) - { - /* printk( "SNDSCAPE: Setting MT32 mode failed\n"); */ - } - host_close(devc); -} - -static void set_control(struct sscape_info *devc, int ctrl, int value) -{ - host_open(devc); - host_command3(devc, CMD_SET_CONTROL, ctrl, value); - if (host_read(devc) != CMD_ACK) - { - /* printk( "SNDSCAPE: Setting control (%d) failed\n", ctrl); */ - } - host_close(devc); -} - -static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, int blk_size, int mode) -{ - unsigned char temp; - - if (dma_chan != SSCAPE_DMA_A) - { - printk(KERN_WARNING "soundscape: Tried to use DMA channel != A. Why?\n"); - return; - } - audio_devs[devc->codec_audiodev]->flags &= ~DMA_AUTOMODE; - DMAbuf_start_dma(devc->codec_audiodev, buf, blk_size, mode); - audio_devs[devc->codec_audiodev]->flags |= DMA_AUTOMODE; - - temp = devc->dma << 4; /* Setup DMA channel select bits */ - if (devc->dma <= 3) - temp |= 0x80; /* 8 bit DMA channel */ - - temp |= 1; /* Trigger DMA */ - sscape_write(devc, GA_DMAA_REG, temp); - temp &= 0xfe; /* Clear DMA trigger */ - sscape_write(devc, GA_DMAA_REG, temp); -} - -static int verify_mpu(struct sscape_info *devc) -{ - /* - * The SoundScape board could be in three modes (MPU, 8250 and host). - * If the card is not in the MPU mode, enabling the MPU driver will - * cause infinite loop (the driver believes that there is always some - * received data in the buffer. - * - * Detect this by looking if there are more than 10 received MIDI bytes - * (0x00) in the buffer. - */ - - int i; - - for (i = 0; i < 10; i++) - { - if (inb(devc->base + HOST_CTRL) & 0x80) - return 1; - - if (inb(devc->base) != 0x00) - return 1; - } - printk(KERN_WARNING "SoundScape: The device is not in the MPU-401 mode\n"); - return 0; -} - -static int sscape_coproc_open(void *dev_info, int sub_device) -{ - if (sub_device == COPR_MIDI) - { - set_mt32(devc, 0); - if (!verify_mpu(devc)) - return -EIO; - } - return 0; -} - -static void sscape_coproc_close(void *dev_info, int sub_device) -{ - struct sscape_info *devc = dev_info; - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - if (devc->dma_allocated) - { - __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */ - devc->dma_allocated = 0; - } - spin_unlock_irqrestore(&devc->lock,flags); - return; -} - -static void sscape_coproc_reset(void *dev_info) -{ -} - -static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, int size, int flag) -{ - unsigned long flags; - unsigned char temp; - volatile int done, timeout_val; - static unsigned char codec_dma_bits; - - if (flag & CPF_FIRST) - { - /* - * First block. Have to allocate DMA and to reset the board - * before continuing. - */ - - spin_lock_irqsave(&devc->lock,flags); - codec_dma_bits = sscape_read(devc, GA_CDCFG_REG); - - if (devc->dma_allocated == 0) - devc->dma_allocated = 1; - - spin_unlock_irqrestore(&devc->lock,flags); - - sscape_write(devc, GA_HMCTL_REG, - (temp = sscape_read(devc, GA_HMCTL_REG)) & 0x3f); /*Reset */ - - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - sscape_read(devc, GA_HMCTL_REG); /* Delay */ - - /* Take board out of reset */ - sscape_write(devc, GA_HMCTL_REG, - (temp = sscape_read(devc, GA_HMCTL_REG)) | 0x80); - } - /* - * Transfer one code block using DMA - */ - if (audio_devs[devc->codec_audiodev]->dmap_out->raw_buf == NULL) - { - printk(KERN_WARNING "soundscape: DMA buffer not available\n"); - return 0; - } - memcpy(audio_devs[devc->codec_audiodev]->dmap_out->raw_buf, block, size); - - spin_lock_irqsave(&devc->lock,flags); - - /******** INTERRUPTS DISABLED NOW ********/ - - do_dma(devc, SSCAPE_DMA_A, - audio_devs[devc->codec_audiodev]->dmap_out->raw_buf_phys, - size, DMA_MODE_WRITE); - - /* - * Wait until transfer completes. - */ - - done = 0; - timeout_val = 30; - while (!done && timeout_val-- > 0) - { - int resid; - - if (HZ / 50) - sleep(HZ / 50); - clear_dma_ff(devc->dma); - if ((resid = get_dma_residue(devc->dma)) == 0) - done = 1; - } - - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - return 0; - - if (flag & CPF_LAST) - { - /* - * Take the board out of reset - */ - outb((0x00), PORT(HOST_CTRL)); - outb((0x00), PORT(MIDI_CTRL)); - - temp = sscape_read(devc, GA_HMCTL_REG); - temp |= 0x40; - sscape_write(devc, GA_HMCTL_REG, temp); /* Kickstart the board */ - - /* - * Wait until the ODB wakes up - */ - spin_lock_irqsave(&devc->lock,flags); - done = 0; - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - unsigned char x; - - sleep(1); - x = inb(PORT(HOST_DATA)); - if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ - { - DDB(printk("Soundscape: Acknowledge = %x\n", x)); - done = 1; - } - } - sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); - - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - { - printk(KERN_ERR "soundscape: The OBP didn't respond after code download\n"); - return 0; - } - spin_lock_irqsave(&devc->lock,flags); - done = 0; - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - sleep(1); - if (inb(PORT(HOST_DATA)) == 0xfe) /* Host startup acknowledge */ - done = 1; - } - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - { - printk(KERN_ERR "soundscape: OBP Initialization failed.\n"); - return 0; - } - printk(KERN_INFO "SoundScape board initialized OK\n"); - set_control(devc, CTL_MASTER_VOL, 100); - set_control(devc, CTL_SYNTH_VOL, 100); - -#ifdef SSCAPE_DEBUG3 - /* - * Temporary debugging aid. Print contents of the registers after - * downloading the code. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (new value)\n", i, sscape_read(devc, i)); - } -#endif - - } - return 1; -} - -static int download_boot_block(void *dev_info, copr_buffer * buf) -{ - if (buf->len <= 0 || buf->len > sizeof(buf->data)) - return -EINVAL; - - if (!sscape_download_boot(devc, buf->data, buf->len, buf->flags)) - { - printk(KERN_ERR "soundscape: Unable to load microcode block to the OBP.\n"); - return -EIO; - } - return 0; -} - -static int sscape_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, int local) -{ - copr_buffer *buf; - int err; - - switch (cmd) - { - case SNDCTL_COPR_RESET: - sscape_coproc_reset(dev_info); - return 0; - - case SNDCTL_COPR_LOAD: - buf = (copr_buffer *) vmalloc(sizeof(copr_buffer)); - if (buf == NULL) - return -ENOSPC; - if (copy_from_user(buf, arg, sizeof(copr_buffer))) - { - vfree(buf); - return -EFAULT; - } - err = download_boot_block(dev_info, buf); - vfree(buf); - return err; - - default: - return -EINVAL; - } -} - -static coproc_operations sscape_coproc_operations = -{ - "SoundScape M68K", - THIS_MODULE, - sscape_coproc_open, - sscape_coproc_close, - sscape_coproc_ioctl, - sscape_coproc_reset, - &adev_info -}; - -static struct resource *sscape_ports; -static int sscape_is_pnp; - -static void __init attach_sscape(struct address_info *hw_config) -{ -#ifndef SSCAPE_REGS - /* - * Config register values for Spea/V7 Media FX and Ensoniq S-2000. - * These values are card - * dependent. If you have another SoundScape based card, you have to - * find the correct values. Do the following: - * - Compile this driver with SSCAPE_DEBUG1 defined. - * - Shut down and power off your machine. - * - Boot with DOS so that the SSINIT.EXE program is run. - * - Warm boot to {Linux|SYSV|BSD} and write down the lines displayed - * when detecting the SoundScape. - * - Modify the following list to use the values printed during boot. - * Undefine the SSCAPE_DEBUG1 - */ -#define SSCAPE_REGS { \ -/* I0 */ 0x00, \ -/* I1 */ 0xf0, /* Note! Ignored. Set always to 0xf0 */ \ -/* I2 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \ -/* I3 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \ -/* I4 */ 0xf5, /* Ignored */ \ -/* I5 */ 0x10, \ -/* I6 */ 0x00, \ -/* I7 */ 0x2e, /* I7 MEM config A. Likely to vary between models */ \ -/* I8 */ 0x00, /* I8 MEM config B. Likely to vary between models */ \ -/* I9 */ 0x40 /* Ignored */ \ - } -#endif - - unsigned long flags; - static unsigned char regs[10] = SSCAPE_REGS; - - int i, irq_bits = 0xff; - - if (old_hardware) - { - valid_interrupts = valid_interrupts_old; - conf_printf("Ensoniq SoundScape (old)", hw_config); - } - else - conf_printf("Ensoniq SoundScape", hw_config); - - for (i = 0; i < 4; i++) - { - if (hw_config->irq == valid_interrupts[i]) - { - irq_bits = i; - break; - } - } - if (hw_config->irq > 15 || (regs[4] = irq_bits == 0xff)) - { - printk(KERN_ERR "Invalid IRQ%d\n", hw_config->irq); - release_region(devc->base, 2); - release_region(devc->base + 2, 6); - if (sscape_is_pnp) - release_region(devc->codec, 2); - return; - } - - if (!sscape_is_pnp) { - - spin_lock_irqsave(&devc->lock,flags); - /* Host interrupt enable */ - sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ - /* DMA A status/trigger register */ - sscape_write(devc, 2, 0x20); /* DMA channel disabled */ - /* DMA B status/trigger register */ - sscape_write(devc, 3, 0x20); /* DMA channel disabled */ - /* Host interrupt config reg */ - sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); - /* Don't destroy CD-ROM DMA config bits (0xc0) */ - sscape_write(devc, 5, (regs[5] & 0x3f) | (sscape_read(devc, 5) & 0xc0)); - /* CD-ROM config (WSS codec actually) */ - sscape_write(devc, 6, regs[6]); - sscape_write(devc, 7, regs[7]); - sscape_write(devc, 8, regs[8]); - /* Master control reg. Don't modify CR-ROM bits. Disable SB emul */ - sscape_write(devc, 9, (sscape_read(devc, 9) & 0xf0) | 0x08); - spin_unlock_irqrestore(&devc->lock,flags); - } -#ifdef SSCAPE_DEBUG2 - /* - * Temporary debugging aid. Print contents of the registers after - * changing them. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (new value)\n", i, sscape_read(devc, i)); - } -#endif - - if (probe_mpu401(hw_config, sscape_ports)) - hw_config->always_detect = 1; - hw_config->name = "SoundScape"; - - hw_config->irq *= -1; /* Negative value signals IRQ sharing */ - attach_mpu401(hw_config, THIS_MODULE); - hw_config->irq *= -1; /* Restore it */ - - if (hw_config->slots[1] != -1) /* The MPU driver installed itself */ - { - sscape_mididev = hw_config->slots[1]; - midi_devs[hw_config->slots[1]]->coproc = &sscape_coproc_operations; - } - sscape_write(devc, GA_INTENA_REG, 0x80); /* Master IRQ enable */ - devc->ok = 1; - devc->failed = 0; -} - -static int detect_ga(sscape_info * devc) -{ - unsigned char save; - - DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base)); - - /* - * First check that the address register of "ODIE" is - * there and that it has exactly 4 writable bits. - * First 4 bits - */ - - if ((save = inb(PORT(ODIE_ADDR))) & 0xf0) - { - DDB(printk("soundscape: Detect error A\n")); - return 0; - } - outb((0x00), PORT(ODIE_ADDR)); - if (inb(PORT(ODIE_ADDR)) != 0x00) - { - DDB(printk("soundscape: Detect error B\n")); - return 0; - } - outb((0xff), PORT(ODIE_ADDR)); - if (inb(PORT(ODIE_ADDR)) != 0x0f) - { - DDB(printk("soundscape: Detect error C\n")); - return 0; - } - outb((save), PORT(ODIE_ADDR)); - - /* - * Now verify that some indirect registers return zero on some bits. - * This may break the driver with some future revisions of "ODIE" but... - */ - - if (sscape_read(devc, 0) & 0x0c) - { - DDB(printk("soundscape: Detect error D (%x)\n", sscape_read(devc, 0))); - return 0; - } - if (sscape_read(devc, 1) & 0x0f) - { - DDB(printk("soundscape: Detect error E\n")); - return 0; - } - if (sscape_read(devc, 5) & 0x0f) - { - DDB(printk("soundscape: Detect error F\n")); - return 0; - } - return 1; -} - -static int sscape_read_host_ctrl(sscape_info* devc) -{ - return host_read(devc); -} - -static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b) -{ - host_command2(devc, a, b); -} - -static int sscape_alloc_dma(sscape_info *devc) -{ - char *start_addr, *end_addr; - int dma_pagesize; - int sz, size; - struct page *page; - - if (devc->raw_buf != NULL) return 0; /* Already done */ - dma_pagesize = (devc->dma < 4) ? (64 * 1024) : (128 * 1024); - devc->raw_buf = NULL; - devc->buffsize = 8192*4; - if (devc->buffsize > dma_pagesize) devc->buffsize = dma_pagesize; - start_addr = NULL; - /* - * Now loop until we get a free buffer. Try to get smaller buffer if - * it fails. Don't accept smaller than 8k buffer for performance - * reasons. - */ - while (start_addr == NULL && devc->buffsize > PAGE_SIZE) { - for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1); - devc->buffsize = PAGE_SIZE * (1 << sz); - start_addr = (char *) __get_free_pages(GFP_ATOMIC|GFP_DMA, sz); - if (start_addr == NULL) devc->buffsize /= 2; - } - - if (start_addr == NULL) { - printk(KERN_ERR "sscape pnp init error: Couldn't allocate DMA buffer\n"); - return 0; - } else { - /* make some checks */ - end_addr = start_addr + devc->buffsize - 1; - /* now check if it fits into the same dma-pagesize */ - - if (((long) start_addr & ~(dma_pagesize - 1)) != ((long) end_addr & ~(dma_pagesize - 1)) - || end_addr >= (char *) (MAX_DMA_ADDRESS)) { - printk(KERN_ERR "sscape pnp: Got invalid address 0x%lx for %db DMA-buffer\n", (long) start_addr, devc->buffsize); - return 0; - } - } - devc->raw_buf = start_addr; - devc->raw_buf_phys = virt_to_bus(start_addr); - - for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) - SetPageReserved(page); - return 1; -} - -static void sscape_free_dma(sscape_info *devc) -{ - int sz, size; - unsigned long start_addr, end_addr; - struct page *page; - - if (devc->raw_buf == NULL) return; - for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1); - start_addr = (unsigned long) devc->raw_buf; - end_addr = start_addr + devc->buffsize; - - for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) - ClearPageReserved(page); - - free_pages((unsigned long) devc->raw_buf, sz); - devc->raw_buf = NULL; -} - -/* Intel version !!!!!!!!! */ - -static int sscape_start_dma(int chan, unsigned long physaddr, int count, int dma_mode) -{ - unsigned long flags; - - flags = claim_dma_lock(); - disable_dma(chan); - clear_dma_ff(chan); - set_dma_mode(chan, dma_mode); - set_dma_addr(chan, physaddr); - set_dma_count(chan, count); - enable_dma(chan); - release_dma_lock(flags); - return 0; -} - -static void sscape_pnp_start_dma(sscape_info* devc, int arg ) -{ - int reg; - if (arg == 0) reg = 2; - else reg = 3; - - sscape_write(devc, reg, sscape_read( devc, reg) | 0x01); - sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE); -} - -static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) -{ - int reg; - unsigned long i; - unsigned char d; - - if (arg == 0) reg = 2; - else reg = 3; - - sleep ( 1 ); - i = 0; - do { - d = sscape_read(devc, reg) & 1; - if ( d == 1) break; - i++; - } while (i < 500000); - d = sscape_read(devc, reg) & 1; - return d; -} - -static int sscape_pnp_alloc_dma(sscape_info* devc) -{ - /* printk(KERN_INFO "sscape: requesting dma\n"); */ - if (request_dma(devc -> dma, "sscape")) return 0; - /* printk(KERN_INFO "sscape: dma channel allocated\n"); */ - if (!sscape_alloc_dma(devc)) { - free_dma(devc -> dma); - return 0; - }; - return 1; -} - -static void sscape_pnp_free_dma(sscape_info* devc) -{ - sscape_free_dma( devc); - free_dma(devc -> dma ); - /* printk(KERN_INFO "sscape: dma released\n"); */ -} - -static int sscape_pnp_upload_file(sscape_info* devc, char* fn) -{ - int done = 0; - int timeout_val; - char* data,*dt; - int len,l; - unsigned long flags; - - sscape_write( devc, 9, sscape_read(devc, 9 ) & 0x3F ); - sscape_write( devc, 2, (devc -> dma << 4) | 0x80 ); - sscape_write( devc, 3, 0x20 ); - sscape_write( devc, 9, sscape_read( devc, 9 ) | 0x80 ); - - len = mod_firmware_load(fn, &data); - if (len == 0) { - printk(KERN_ERR "sscape: file not found: %s\n", fn); - return 0; - } - dt = data; - spin_lock_irqsave(&devc->lock,flags); - while ( len > 0 ) { - if (len > devc -> buffsize) l = devc->buffsize; - else l = len; - len -= l; - memcpy(devc->raw_buf, dt, l); dt += l; - sscape_start_dma(devc->dma, devc->raw_buf_phys, l, 0x48); - sscape_pnp_start_dma ( devc, 0 ); - if (sscape_pnp_wait_dma ( devc, 0 ) == 0) { - spin_unlock_irqrestore(&devc->lock,flags); - return 0; - } - } - - spin_unlock_irqrestore(&devc->lock,flags); - vfree(data); - - outb(0, devc -> base + 2); - outb(0, devc -> base); - - sscape_write ( devc, 9, sscape_read( devc, 9 ) | 0x40); - - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - unsigned char x; - sleep(1); - x = inb( devc -> base + 3); - if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ - { - //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x); - done = 1; - } - } - timeout_val = 5 * HZ; - done = 0; - while (!done && timeout_val-- > 0) - { - unsigned char x; - sleep(1); - x = inb( devc -> base + 3); - if (x == 0xfe) /* OBP startup acknowledge */ - { - //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x); - done = 1; - } - } - - if ( !done ) printk(KERN_ERR "soundscape: OBP Initialization failed.\n"); - - sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40); - sscape_write( devc, 3, (devc -> dma << 4) + 0x80); - return 1; -} - -static void __init sscape_pnp_init_hw(sscape_info* devc) -{ - unsigned char midi_irq = 0, sb_irq = 0; - unsigned i; - static char code_file_name[23] = "/sndscape/sndscape.cox"; - - int sscape_joystic_enable = 0x7f; - int sscape_mic_enable = 0; - int sscape_ext_midi = 0; - - if ( !sscape_pnp_alloc_dma(devc) ) { - printk(KERN_ERR "sscape: faild to allocate dma\n"); - return; - } - - for (i = 0; i < 4; i++) { - if ( devc -> irq == valid_interrupts[i] ) - midi_irq = i; - if ( devc -> codec_irq == valid_interrupts[i] ) - sb_irq = i; - } - - sscape_write( devc, 5, 0x50); - sscape_write( devc, 7, 0x2e); - sscape_write( devc, 8, 0x00); - - sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40); - sscape_write( devc, 3, ( devc -> dma << 4) | 0x80); - - sscape_write (devc, 4, 0xF0 | (midi_irq<<2) | midi_irq); - - i = 0x10; //sscape_read(devc, 9) & (devc->ic_type == IC_ODIE ? 0xf0 : 0xc0); - if (sscape_joystic_enable) i |= 8; - - sscape_write (devc, 9, i); - sscape_write (devc, 6, 0x80); - sscape_write (devc, 1, 0x80); - - if (devc -> codec_type == 2) { - sscape_pnp_write_codec( devc, 0x0C, 0x50); - sscape_pnp_write_codec( devc, 0x10, sscape_pnp_read_codec( devc, 0x10) & 0x3F); - sscape_pnp_write_codec( devc, 0x11, sscape_pnp_read_codec( devc, 0x11) | 0xC0); - sscape_pnp_write_codec( devc, 29, 0x20); - } - - if (sscape_pnp_upload_file(devc, "/sndscape/scope.cod") == 0 ) { - printk(KERN_ERR "sscape: faild to upload file /sndscape/scope.cod\n"); - sscape_pnp_free_dma(devc); - return; - } - - i = sscape_read_host_ctrl( devc ); - - if ( (i & 0x0F) > 7 ) { - printk(KERN_ERR "sscape: scope.cod faild\n"); - sscape_pnp_free_dma(devc); - return; - } - if ( i & 0x10 ) sscape_write( devc, 7, 0x2F); - code_file_name[21] = (char) ( i & 0x0F) + 0x30; - if (sscape_pnp_upload_file( devc, code_file_name) == 0) { - printk(KERN_ERR "sscape: faild to upload file %s\n", code_file_name); - sscape_pnp_free_dma(devc); - return; - } - - if (devc->ic_type != IC_ODIE) { - sscape_pnp_write_codec( devc, 10, (sscape_pnp_read_codec(devc, 10) & 0x7f) | - ( sscape_mic_enable == 0 ? 0x00 : 0x80) ); - } - sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */ - sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */ - sscape_write_host_ctrl2( devc, 0x8A, sscape_ext_midi); - - sscape_pnp_write_codec ( devc, 6, 0x3f ); //WAV_VOL - sscape_pnp_write_codec ( devc, 7, 0x3f ); //WAV_VOL - sscape_pnp_write_codec ( devc, 2, 0x1F ); //WD_CDXVOLL - sscape_pnp_write_codec ( devc, 3, 0x1F ); //WD_CDXVOLR - - if (devc -> codec_type == 1) { - sscape_pnp_write_codec ( devc, 4, 0x1F ); - sscape_pnp_write_codec ( devc, 5, 0x1F ); - sscape_write_host_ctrl2( devc, 0x88, sscape_mic_enable); - } else { - int t; - sscape_pnp_write_codec ( devc, 0x10, 0x1F << 1); - sscape_pnp_write_codec ( devc, 0x11, 0xC0 | (0x1F << 1)); - - t = sscape_pnp_read_codec( devc, 0x00) & 0xDF; - if ( (sscape_mic_enable == 0)) t |= 0; - else t |= 0x20; - sscape_pnp_write_codec ( devc, 0x00, t); - t = sscape_pnp_read_codec( devc, 0x01) & 0xDF; - if ( (sscape_mic_enable == 0) ) t |= 0; - else t |= 0x20; - sscape_pnp_write_codec ( devc, 0x01, t); - sscape_pnp_write_codec ( devc, 0x40 | 29 , 0x20); - outb(0, devc -> codec); - } - if (devc -> ic_type == IC_OPUS ) { - int i = sscape_read( devc, 9 ); - sscape_write( devc, 9, i | 3 ); - sscape_write( devc, 3, 0x40); - - if (request_region(0x228, 1, "sscape setup junk")) { - outb(0, 0x228); - release_region(0x228,1); - } - sscape_write( devc, 3, (devc -> dma << 4) | 0x80); - sscape_write( devc, 9, i ); - } - - host_close ( devc ); - sscape_pnp_free_dma(devc); -} - -static int __init detect_sscape_pnp(sscape_info* devc) -{ - long i, irq_bits = 0xff; - unsigned int d; - - DDB(printk("Entered detect_sscape_pnp(%x)\n", devc->base)); - - if (!request_region(devc->codec, 2, "sscape codec")) { - printk(KERN_ERR "detect_sscape_pnp: port %x is not free\n", devc->codec); - return 0; - } - - if ((inb(devc->base + 2) & 0x78) != 0) - goto fail; - - d = inb ( devc -> base + 4) & 0xF0; - if (d & 0x80) - goto fail; - - if (d == 0) { - devc->codec_type = 1; - devc->ic_type = IC_ODIE; - } else if ( (d & 0x60) != 0) { - devc->codec_type = 2; - devc->ic_type = IC_OPUS; - } else if ( (d & 0x40) != 0) { /* WTF? */ - devc->codec_type = 2; - devc->ic_type = IC_ODIE; - } else - goto fail; - - sscape_is_pnp = 1; - - outb(0xFA, devc -> base+4); - if ((inb( devc -> base+4) & 0x9F) != 0x0A) - goto fail; - outb(0xFE, devc -> base+4); - if ( (inb(devc -> base+4) & 0x9F) != 0x0E) - goto fail; - if ( (inb(devc -> base+5) & 0x9F) != 0x0E) - goto fail; - - if (devc->codec_type == 2) { - if (devc->codec != devc->base + 8) { - printk("soundscape warning: incorrect codec port specified\n"); - goto fail; - } - d = 0x10 | (sscape_read(devc, 9) & 0xCF); - sscape_write(devc, 9, d); - sscape_write(devc, 6, 0x80); - } else { - //todo: check codec is not base + 8 - } - - d = (sscape_read(devc, 9) & 0x3F) | 0xC0; - sscape_write(devc, 9, d); - - for (i = 0; i < 550000; i++) - if ( !(inb(devc -> codec) & 0x80) ) break; - - d = inb(devc -> codec); - if (d & 0x80) - goto fail; - if ( inb(devc -> codec + 2) == 0xFF) - goto fail; - - sscape_write(devc, 9, sscape_read(devc, 9) & 0x3F ); - - d = inb(devc -> codec) & 0x80; - if ( d == 0) { - printk(KERN_INFO "soundscape: hardware detected\n"); - valid_interrupts = valid_interrupts_new; - } else { - printk(KERN_INFO "soundscape: board looks like media fx\n"); - valid_interrupts = valid_interrupts_old; - old_hardware = 1; - } - - sscape_write( devc, 9, 0xC0 | (sscape_read(devc, 9) & 0x3F) ); - - for (i = 0; i < 550000; i++) - if ( !(inb(devc -> codec) & 0x80)) - break; - - sscape_pnp_init_hw(devc); - - for (i = 0; i < 4; i++) - { - if (devc->codec_irq == valid_interrupts[i]) { - irq_bits = i; - break; - } - } - sscape_write(devc, GA_INTENA_REG, 0x00); - sscape_write(devc, GA_DMACFG_REG, 0x50); - sscape_write(devc, GA_DMAA_REG, 0x70); - sscape_write(devc, GA_DMAB_REG, 0x20); - sscape_write(devc, GA_INTCFG_REG, 0xf0); - sscape_write(devc, GA_CDCFG_REG, 0x89 | (devc->dma << 4) | (irq_bits << 1)); - - sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 0) | 0x20); - sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 1) | 0x20); - - return 1; -fail: - release_region(devc->codec, 2); - return 0; -} - -static int __init probe_sscape(struct address_info *hw_config) -{ - devc->base = hw_config->io_base; - devc->irq = hw_config->irq; - devc->dma = hw_config->dma; - devc->osp = hw_config->osp; - -#ifdef SSCAPE_DEBUG1 - /* - * Temporary debugging aid. Print contents of the registers before - * changing them. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (old value)\n", i, sscape_read(devc, i)); - } -#endif - devc->failed = 1; - - sscape_ports = request_region(devc->base, 2, "mpu401"); - if (!sscape_ports) - return 0; - - if (!request_region(devc->base + 2, 6, "SoundScape")) { - release_region(devc->base, 2); - return 0; - } - - if (!detect_ga(devc)) { - if (detect_sscape_pnp(devc)) - return 1; - release_region(devc->base, 2); - release_region(devc->base + 2, 6); - return 0; - } - - if (old_hardware) /* Check that it's really an old Spea/Reveal card. */ - { - unsigned char tmp; - int cc; - - if (!((tmp = sscape_read(devc, GA_HMCTL_REG)) & 0xc0)) - { - sscape_write(devc, GA_HMCTL_REG, tmp | 0x80); - for (cc = 0; cc < 200000; ++cc) - inb(devc->base + ODIE_ADDR); - } - } - return 1; -} - -static int __init init_ss_ms_sound(struct address_info *hw_config) -{ - int i, irq_bits = 0xff; - int ad_flags = 0; - struct resource *ports; - - if (devc->failed) - { - printk(KERN_ERR "soundscape: Card not detected\n"); - return 0; - } - if (devc->ok == 0) - { - printk(KERN_ERR "soundscape: Invalid initialization order.\n"); - return 0; - } - for (i = 0; i < 4; i++) - { - if (hw_config->irq == valid_interrupts[i]) - { - irq_bits = i; - break; - } - } - if (irq_bits == 0xff) { - printk(KERN_ERR "soundscape: Invalid MSS IRQ%d\n", hw_config->irq); - return 0; - } - - if (old_hardware) - ad_flags = 0x12345677; /* Tell that we may have a CS4248 chip (Spea-V7 Media FX) */ - else if (sscape_is_pnp) - ad_flags = 0x87654321; /* Tell that we have a soundscape pnp with 1845 chip */ - - ports = request_region(hw_config->io_base, 4, "ad1848"); - if (!ports) { - printk(KERN_ERR "soundscape: ports busy\n"); - return 0; - } - - if (!ad1848_detect(ports, &ad_flags, hw_config->osp)) { - release_region(hw_config->io_base, 4); - return 0; - } - - if (!sscape_is_pnp) /*pnp is already setup*/ - { - /* - * Setup the DMA polarity. - */ - sscape_write(devc, GA_DMACFG_REG, 0x50); - - /* - * Take the gate-array off of the DMA channel. - */ - sscape_write(devc, GA_DMAB_REG, 0x20); - - /* - * Init the AD1848 (CD-ROM) config reg. - */ - sscape_write(devc, GA_CDCFG_REG, 0x89 | (hw_config->dma << 4) | (irq_bits << 1)); - } - - if (hw_config->irq == devc->irq) - printk(KERN_WARNING "soundscape: Warning! The WSS mode can't share IRQ with MIDI\n"); - - hw_config->slots[0] = ad1848_init( - sscape_is_pnp ? "SoundScape" : "SoundScape PNP", - ports, - hw_config->irq, - hw_config->dma, - hw_config->dma, - 0, - devc->osp, - THIS_MODULE); - - - if (hw_config->slots[0] != -1) /* The AD1848 driver installed itself */ - { - audio_devs[hw_config->slots[0]]->coproc = &sscape_coproc_operations; - devc->codec_audiodev = hw_config->slots[0]; - devc->my_audiodev = hw_config->slots[0]; - - /* Set proper routings here (what are they) */ - AD1848_REROUTE(SOUND_MIXER_LINE1, SOUND_MIXER_LINE); - } - -#ifdef SSCAPE_DEBUG5 - /* - * Temporary debugging aid. Print contents of the registers - * after the AD1848 device has been initialized. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x\n", i, sscape_read(devc, i)); - } -#endif - return 1; -} - -static void __exit unload_sscape(struct address_info *hw_config) -{ - release_region(devc->base + 2, 6); - unload_mpu401(hw_config); - if (sscape_is_pnp) - release_region(devc->codec, 2); -} - -static void __exit unload_ss_ms_sound(struct address_info *hw_config) -{ - ad1848_unload(hw_config->io_base, - hw_config->irq, - devc->dma, - devc->dma, - 0); - sound_unload_audiodev(hw_config->slots[0]); -} - -static struct address_info cfg; -static struct address_info cfg_mpu; - -static int __initdata spea = -1; -static int mss = 0; -static int __initdata dma = -1; -static int __initdata irq = -1; -static int __initdata io = -1; -static int __initdata mpu_irq = -1; -static int __initdata mpu_io = -1; - -module_param(dma, int, 0); -module_param(irq, int, 0); -module_param(io, int, 0); -module_param(spea, int, 0); /* spea=0/1 set the old_hardware */ -module_param(mpu_irq, int, 0); -module_param(mpu_io, int, 0); -module_param(mss, int, 0); - -static int __init init_sscape(void) -{ - printk(KERN_INFO "Soundscape driver Copyright (C) by Hannu Savolainen 1993-1996\n"); - - cfg.irq = irq; - cfg.dma = dma; - cfg.io_base = io; - - cfg_mpu.irq = mpu_irq; - cfg_mpu.io_base = mpu_io; - /* WEH - Try to get right dma channel */ - cfg_mpu.dma = dma; - - devc->codec = cfg.io_base; - devc->codec_irq = cfg.irq; - devc->codec_type = 0; - devc->ic_type = 0; - devc->raw_buf = NULL; - spin_lock_init(&devc->lock); - - if (cfg.dma == -1 || cfg.irq == -1 || cfg.io_base == -1) { - printk(KERN_ERR "DMA, IRQ, and IO port must be specified.\n"); - return -EINVAL; - } - - if (cfg_mpu.irq == -1 && cfg_mpu.io_base != -1) { - printk(KERN_ERR "MPU_IRQ must be specified if MPU_IO is set.\n"); - return -EINVAL; - } - - if(spea != -1) { - old_hardware = spea; - printk(KERN_INFO "Forcing %s hardware support.\n", - spea?"new":"old"); - } - if (probe_sscape(&cfg_mpu) == 0) - return -ENODEV; - - attach_sscape(&cfg_mpu); - - mss = init_ss_ms_sound(&cfg); - - return 0; -} - -static void __exit cleanup_sscape(void) -{ - if (mss) - unload_ss_ms_sound(&cfg); - unload_sscape(&cfg_mpu); -} - -module_init(init_sscape); -module_exit(cleanup_sscape); - -#ifndef MODULE -static int __init setup_sscape(char *str) -{ - /* io, irq, dma, mpu_io, mpu_irq */ - int ints[6]; - - str = get_options(str, ARRAY_SIZE(ints), ints); - - io = ints[1]; - irq = ints[2]; - dma = ints[3]; - mpu_io = ints[4]; - mpu_irq = ints[5]; - - return 1; -} - -__setup("sscape=", setup_sscape); -#endif -MODULE_LICENSE("GPL"); -- cgit v1.2.2 From 3c76b4d69bedde5b9e7e42612a7d2ede4ab7fd8d Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 25 Oct 2009 11:05:19 +0100 Subject: ALSA: es18xx: remove snd_card pointer from snd_es18xx structure The snd_card pointer is redundant and code can be easily changed to work without it. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 75 ++++++++++++++++++++++++++++++++---------------------- 1 file changed, 44 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 8cfbff73a835..160752bc2e8e 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -121,7 +121,6 @@ struct snd_es18xx { unsigned int dma1_shift; unsigned int dma2_shift; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_a_substream; struct snd_pcm_substream *capture_a_substream; @@ -755,7 +754,9 @@ static int snd_es18xx_playback_trigger(struct snd_pcm_substream *substream, static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) { - struct snd_es18xx *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; unsigned char status; if (chip->caps & ES18XX_CONTROL) { @@ -805,12 +806,16 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) int split = 0; if (chip->caps & ES18XX_HWV) { split = snd_es18xx_mixer_read(chip, 0x64) & 0x80; - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->hw_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->hw_volume->id); } if (!split) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } /* ack interrupt */ snd_es18xx_mixer_write(chip, 0x66, 0x00); @@ -1691,8 +1696,11 @@ static struct snd_pcm_ops snd_es18xx_capture_ops = { .pointer = snd_es18xx_capture_pointer, }; -static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct snd_pcm ** rpcm) +static int __devinit snd_es18xx_pcm(struct snd_card *card, int device, + struct snd_pcm **rpcm) { + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; struct snd_pcm *pcm; char str[16]; int err; @@ -1701,9 +1709,9 @@ static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct *rpcm = NULL; sprintf(str, "ES%x", chip->version); if (chip->caps & ES18XX_PCM2) - err = snd_pcm_new(chip->card, str, device, 2, 1, &pcm); + err = snd_pcm_new(card, str, device, 2, 1, &pcm); else - err = snd_pcm_new(chip->card, str, device, 1, 1, &pcm); + err = snd_pcm_new(card, str, device, 1, 1, &pcm); if (err < 0) return err; @@ -1737,7 +1745,7 @@ static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) struct snd_audiodrive *acard = card->private_data; struct snd_es18xx *chip = acard->chip; - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_pcm_suspend_all(chip->pcm); @@ -1758,18 +1766,21 @@ static int snd_es18xx_resume(struct snd_card *card) /* restore PM register, we won't wake till (not 0x07) i/o activity though */ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_FM); - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } #endif /* CONFIG_PM */ -static int snd_es18xx_free(struct snd_es18xx *chip) +static int snd_es18xx_free(struct snd_card *card) { + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; + release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_ctrl_port); release_and_free_resource(chip->res_mpu_port); if (chip->irq >= 0) - free_irq(chip->irq, (void *) chip); + free_irq(chip->irq, (void *) card); if (chip->dma1 >= 0) { disable_dma(chip->dma1); free_dma(chip->dma1); @@ -1784,8 +1795,7 @@ static int snd_es18xx_free(struct snd_es18xx *chip) static int snd_es18xx_dev_free(struct snd_device *device) { - struct snd_es18xx *chip = device->device_data; - return snd_es18xx_free(chip); + return snd_es18xx_free(device->card); } static int __devinit snd_es18xx_new_device(struct snd_card *card, @@ -1808,7 +1818,6 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); spin_lock_init(&chip->ctrl_lock); - chip->card = card; chip->port = port; chip->mpu_port = mpu_port; chip->fm_port = fm_port; @@ -1818,53 +1827,55 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, chip->audio2_vol = 0x00; chip->active = 0; - if ((chip->res_port = request_region(port, 16, "ES18xx")) == NULL) { - snd_es18xx_free(chip); + chip->res_port = request_region(port, 16, "ES18xx"); + if (chip->res_port == NULL) { + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap ports 0x%lx-0x%lx\n", port, port + 16 - 1); return -EBUSY; } - if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", (void *) chip)) { - snd_es18xx_free(chip); + if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", + (void *) card)) { + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq); return -EBUSY; } chip->irq = irq; if (request_dma(dma1, "ES18xx DMA 1")) { - snd_es18xx_free(chip); + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap DMA1 %d\n", dma1); return -EBUSY; } chip->dma1 = dma1; if (dma2 != dma1 && request_dma(dma2, "ES18xx DMA 2")) { - snd_es18xx_free(chip); + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap DMA2 %d\n", dma2); return -EBUSY; } chip->dma2 = dma2; if (snd_es18xx_probe(chip) < 0) { - snd_es18xx_free(chip); + snd_es18xx_free(card); return -ENODEV; } - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_es18xx_free(chip); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, NULL, &ops); + if (err < 0) { + snd_es18xx_free(card); return err; } *rchip = chip; return 0; } -static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) +static int __devinit snd_es18xx_mixer(struct snd_card *card) { - struct snd_card *card; + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; int err; unsigned int idx; - card = chip->card; - strcpy(card->mixername, chip->pcm->name); for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_base_controls); idx++) { @@ -2161,10 +2172,12 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) chip->port, irq[dev], dma1[dev]); - if ((err = snd_es18xx_pcm(chip, 0, NULL)) < 0) + err = snd_es18xx_pcm(card, 0, NULL); + if (err < 0) return err; - if ((err = snd_es18xx_mixer(chip)) < 0) + err = snd_es18xx_mixer(card); + if (err < 0) return err; if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { -- cgit v1.2.2 From b14f5de731ae657d498d18d713c6431bfbeefb4b Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 25 Oct 2009 11:10:01 +0100 Subject: ALSA: es18xx: remove snd_audiodrive structure Remove intermediate snd_audiodrive structure between snd_card structure and snd_es18xx. This reduces size of source code and binary driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 71 +++++++++++++++++++----------------------------------- 1 file changed, 25 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 160752bc2e8e..5cf42b4d65fd 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -139,10 +139,6 @@ struct snd_es18xx { #ifdef CONFIG_PM unsigned char pm_reg; #endif -}; - -struct snd_audiodrive { - struct snd_es18xx *chip; #ifdef CONFIG_PNP struct pnp_dev *dev; struct pnp_dev *devc; @@ -755,8 +751,7 @@ static int snd_es18xx_playback_trigger(struct snd_pcm_substream *substream, static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) { struct snd_card *card = dev_id; - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; unsigned char status; if (chip->caps & ES18XX_CONTROL) { @@ -1699,8 +1694,7 @@ static struct snd_pcm_ops snd_es18xx_capture_ops = { static int __devinit snd_es18xx_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; struct snd_pcm *pcm; char str[16]; int err; @@ -1742,8 +1736,7 @@ static int __devinit snd_es18xx_pcm(struct snd_card *card, int device, #ifdef CONFIG_PM static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1760,8 +1753,7 @@ static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) static int snd_es18xx_resume(struct snd_card *card) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; /* restore PM register, we won't wake till (not 0x07) i/o activity though */ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_FM); @@ -1773,8 +1765,7 @@ static int snd_es18xx_resume(struct snd_card *card) static int snd_es18xx_free(struct snd_card *card) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_ctrl_port); @@ -1789,7 +1780,6 @@ static int snd_es18xx_free(struct snd_card *card) disable_dma(chip->dma2); free_dma(chip->dma2); } - kfree(chip); return 0; } @@ -1802,19 +1792,14 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, unsigned long port, unsigned long mpu_port, unsigned long fm_port, - int irq, int dma1, int dma2, - struct snd_es18xx ** rchip) + int irq, int dma1, int dma2) { - struct snd_es18xx *chip; + struct snd_es18xx *chip = card->private_data; static struct snd_device_ops ops = { .dev_free = snd_es18xx_dev_free, }; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); spin_lock_init(&chip->ctrl_lock); @@ -1865,14 +1850,12 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, snd_es18xx_free(card); return err; } - *rchip = chip; return 0; } static int __devinit snd_es18xx_mixer(struct snd_card *card) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; int err; unsigned int idx; @@ -2074,11 +2057,11 @@ static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev) return 0; } -static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard, +static int __devinit snd_audiodrive_pnp(int dev, struct snd_es18xx *chip, struct pnp_dev *pdev) { - acard->dev = pdev; - if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) + chip->dev = pdev; + if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0) return -EBUSY; return 0; } @@ -2104,26 +2087,26 @@ static struct pnp_card_device_id snd_audiodrive_pnpids[] = { MODULE_DEVICE_TABLE(pnp_card, snd_audiodrive_pnpids); -static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, +static int __devinit snd_audiodrive_pnpc(int dev, struct snd_es18xx *chip, struct pnp_card_link *card, const struct pnp_card_device_id *id) { - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) + chip->dev = pnp_request_card_device(card, id->devs[0].id, NULL); + if (chip->dev == NULL) return -EBUSY; - acard->devc = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devc == NULL) + chip->devc = pnp_request_card_device(card, id->devs[1].id, NULL); + if (chip->devc == NULL) return -EBUSY; /* Control port initialization */ - if (pnp_activate_dev(acard->devc) < 0) { + if (pnp_activate_dev(chip->devc) < 0) { snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n"); return -EAGAIN; } snd_printdd("pnp: port=0x%llx\n", - (unsigned long long)pnp_port_start(acard->devc, 0)); - if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) + (unsigned long long)pnp_port_start(chip->devc, 0)); + if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0) return -EBUSY; return 0; @@ -2139,24 +2122,20 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, static int snd_es18xx_card_new(int dev, struct snd_card **cardp) { return snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_audiodrive), cardp); + sizeof(struct snd_es18xx), cardp); } static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip; + struct snd_es18xx *chip = card->private_data; struct snd_opl3 *opl3; int err; - if ((err = snd_es18xx_new_device(card, - port[dev], - mpu_port[dev], - fm_port[dev], - irq[dev], dma1[dev], dma2[dev], - &chip)) < 0) + err = snd_es18xx_new_device(card, + port[dev], mpu_port[dev], fm_port[dev], + irq[dev], dma1[dev], dma2[dev]); + if (err < 0) return err; - acard->chip = chip; sprintf(card->driver, "ES%x", chip->version); -- cgit v1.2.2 From 23c4a8812a17f0af2b573a63fc991baa7d3570ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 Oct 2009 13:21:49 +0100 Subject: ALSA: hda - Switch to polling mode before disabling MSI When any codec communication error happens, try to switch to the polling mode first before turning off MSI. MSI gets more stable nowadays, thus we should keep it on as much as possible. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d0effa3563e2..a0eface6e99a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -677,6 +677,14 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } + if (!chip->polling_mode) { + snd_printk(KERN_WARNING SFX "azx_get_response timeout, " + "switching to polling mode: last cmd=0x%08x\n", + chip->last_cmd[addr]); + chip->polling_mode = 1; + goto again; + } + if (chip->msi) { snd_printk(KERN_WARNING SFX "No response from codec, " "disabling MSI: last cmd=0x%08x\n", @@ -692,14 +700,6 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, goto again; } - if (!chip->polling_mode) { - snd_printk(KERN_WARNING SFX "azx_get_response timeout, " - "switching to polling mode: last cmd=0x%08x\n", - chip->last_cmd[addr]); - chip->polling_mode = 1; - goto again; - } - if (chip->probing) { /* If this critical timeout happens during the codec probing * phase, this is likely an access to a non-existing codec -- cgit v1.2.2 From 8538a119bfb9031c402a33fc65c276ab9bfafdd5 Mon Sep 17 00:00:00 2001 From: Eero Nurkkala Date: Fri, 30 Oct 2009 13:34:02 +0200 Subject: ASoC: remove io_mutex Remove the io_mutex. It has a drawback of serializing all accesses to snd_soc_update_bits() even when multiple codecs are in use. In addition, it fails to actually do its task - during snd_soc_update_bits(), dapm_update_bits() may also be accessing the same register which may result in an outdated register value. Signed-off-by: Eero Nurkkala Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d190df9fccc..025c5a7f8b72 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -37,7 +37,6 @@ #include static DEFINE_MUTEX(pcm_mutex); -static DEFINE_MUTEX(io_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -1346,14 +1345,12 @@ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, int change; unsigned int old, new; - mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); new = (old & ~mask) | value; change = old != new; if (change) snd_soc_write(codec, reg, new); - mutex_unlock(&io_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_update_bits); @@ -1376,11 +1373,9 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, int change; unsigned int old, new; - mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); new = (old & ~mask) | value; change = old != new; - mutex_unlock(&io_mutex); return change; } -- cgit v1.2.2 From 6c508c62f90240ef58300a5e12093ee769a44364 Mon Sep 17 00:00:00 2001 From: Eero Nurkkala Date: Fri, 30 Oct 2009 13:34:03 +0200 Subject: ASoC: refactor snd_soc_update_bits() Introduce a wrapper call snd_soc_update_bits_locked() that will take the codec mutex. This call is used when the codec mutex is not already taken. Drivers calling snd_soc_update_bits() may wish to make sure the codec mutex is taken from the driver. Signed-off-by: Eero Nurkkala Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 36 ++++++++++++++++++++++++++++++------ 1 file changed, 30 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 025c5a7f8b72..6e24654194ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1355,6 +1355,30 @@ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, } EXPORT_SYMBOL_GPL(snd_soc_update_bits); +/** + * snd_soc_update_bits_locked - update codec register bits + * @codec: audio codec + * @reg: codec register + * @mask: register mask + * @value: new value + * + * Writes new register value, and takes the codec mutex. + * + * Returns 1 for change else 0. + */ +static int snd_soc_update_bits_locked(struct snd_soc_codec *codec, + unsigned short reg, unsigned int mask, + unsigned int value) +{ + int change; + + mutex_lock(&codec->mutex); + change = snd_soc_update_bits(codec, reg, mask, value); + mutex_unlock(&codec->mutex); + + return change; +} + /** * snd_soc_test_bits - test register for change * @codec: audio codec @@ -1706,7 +1730,7 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg, mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); @@ -1780,7 +1804,7 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg, mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); @@ -1941,7 +1965,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val_mask |= mask << rshift; val |= val2 << rshift; } - return snd_soc_update_bits(codec, reg, val_mask, val); + return snd_soc_update_bits_locked(codec, reg, val_mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); @@ -2047,11 +2071,11 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, val = val << shift; val2 = val2 << shift; - err = snd_soc_update_bits(codec, reg, val_mask, val); + err = snd_soc_update_bits_locked(codec, reg, val_mask, val); if (err < 0) return err; - err = snd_soc_update_bits(codec, reg2, val_mask, val2); + err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); return err; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); @@ -2130,7 +2154,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - return snd_soc_update_bits(codec, reg, 0xffff, val); + return snd_soc_update_bits_locked(codec, reg, 0xffff, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); -- cgit v1.2.2 From bcc2c6b7cb320d10c7fcccd87dce87f4384b4332 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Sun, 1 Nov 2009 11:13:19 +0100 Subject: ALSA: snd-pcsp: add nopcm mode Currently, if the high-res timers are unavailable, snd-pcsp does not initialize. People who choose it over pcspkr, loose their console beeps in that case and get annoyed. With this patch, the console beeps remain regardless of the high-res timers. Additionally, the "nopcm" modparam is added to forcibly disable the PCM capabilities of the driver. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 32 ++++++++++++++++++++------------ sound/drivers/pcsp/pcsp.h | 2 +- sound/drivers/pcsp/pcsp_mixer.c | 33 ++++++++++++++++++++++++++------- 3 files changed, 47 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index b60cef257b58..f165c77d6273 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -26,6 +26,7 @@ MODULE_ALIAS("platform:pcspkr"); static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +static int nopcm; /* Disable PCM capability of the driver */ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); @@ -33,6 +34,8 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable PC-Speaker sound."); +module_param(nopcm, bool, 0444); +MODULE_PARM_DESC(nopcm, "Disable PC-Speaker PCM sound. Only beeps remain."); struct snd_pcsp pcsp_chip; @@ -43,13 +46,16 @@ static int __devinit snd_pcsp_create(struct snd_card *card) int err; int div, min_div, order; - hrtimer_get_res(CLOCK_MONOTONIC, &tp); - if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { - printk(KERN_ERR "PCSP: Timer resolution is not sufficient " - "(%linS)\n", tp.tv_nsec); - printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " - "enabled.\n"); - return -EIO; + if (!nopcm) { + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { + printk(KERN_ERR "PCSP: Timer resolution is not sufficient " + "(%linS)\n", tp.tv_nsec); + printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " + "enabled.\n"); + printk(KERN_ERR "PCSP: Turned into nopcm mode.\n"); + nopcm = 1; + } } if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS) @@ -107,12 +113,14 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) snd_card_free(card); return err; } - err = snd_pcsp_new_pcm(&pcsp_chip); - if (err < 0) { - snd_card_free(card); - return err; + if (!nopcm) { + err = snd_pcsp_new_pcm(&pcsp_chip); + if (err < 0) { + snd_card_free(card); + return err; + } } - err = snd_pcsp_new_mixer(&pcsp_chip); + err = snd_pcsp_new_mixer(&pcsp_chip, nopcm); if (err < 0) { snd_card_free(card); return err; diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h index 174dd2ff0f22..1e123077923d 100644 --- a/sound/drivers/pcsp/pcsp.h +++ b/sound/drivers/pcsp/pcsp.h @@ -83,6 +83,6 @@ extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle); extern void pcsp_sync_stop(struct snd_pcsp *chip); extern int snd_pcsp_new_pcm(struct snd_pcsp *chip); -extern int snd_pcsp_new_mixer(struct snd_pcsp *chip); +extern int snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm); #endif diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 903bc846763f..02e05552632b 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -119,24 +119,43 @@ static int pcsp_pcspkr_put(struct snd_kcontrol *kcontrol, .put = pcsp_##ctl_type##_put, \ } -static struct snd_kcontrol_new __devinitdata snd_pcsp_controls[] = { +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_pcm[] = { PCSP_MIXER_CONTROL(enable, "Master Playback Switch"), PCSP_MIXER_CONTROL(treble, "BaseFRQ Playback Volume"), +}; + +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_spkr[] = { PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"), }; -int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip) +static int __devinit snd_pcsp_ctls_add(struct snd_pcsp *chip, + struct snd_kcontrol_new *ctls, int num) { - struct snd_card *card = chip->card; int i, err; + struct snd_card *card = chip->card; + for (i = 0; i < num; i++) { + err = snd_ctl_add(card, snd_ctl_new1(ctls + i, chip)); + if (err < 0) + return err; + } + return 0; +} + +int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm) +{ + int err; + struct snd_card *card = chip->card; - for (i = 0; i < ARRAY_SIZE(snd_pcsp_controls); i++) { - err = snd_ctl_add(card, - snd_ctl_new1(snd_pcsp_controls + i, - chip)); + if (!nopcm) { + err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_pcm, + ARRAY_SIZE(snd_pcsp_controls_pcm)); if (err < 0) return err; } + err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_spkr, + ARRAY_SIZE(snd_pcsp_controls_spkr)); + if (err < 0) + return err; strcpy(card->mixername, "PC-Speaker"); -- cgit v1.2.2 From 0f83d639d84c99a775c60696dbde77372c2cf4ac Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Sat, 31 Oct 2009 20:15:08 +0100 Subject: ASoC: au1x: convert to platform drivers. Convert psc-ac97,i2s to platform drivers similar to the davinci ones. Signed-off-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 117 +++++++++++++++++++++++----- sound/soc/au1x/psc-ac97.c | 194 ++++++++++++++++++++++++++++------------------ sound/soc/au1x/psc-i2s.c | 189 +++++++++++++++++++++++++++----------------- sound/soc/au1x/psc.h | 7 +- 4 files changed, 344 insertions(+), 163 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 594c6c5b7838..fe9f4657c959 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss + * Manuel Lauss * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -332,6 +332,30 @@ static int au1xpsc_pcm_new(struct snd_card *card, } static int au1xpsc_pcm_probe(struct platform_device *pdev) +{ + if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX]) + return -ENODEV; + + return 0; +} + +static int au1xpsc_pcm_remove(struct platform_device *pdev) +{ + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform au1xpsc_soc_platform = { + .name = "au1xpsc-pcm-dbdma", + .probe = au1xpsc_pcm_probe, + .remove = au1xpsc_pcm_remove, + .pcm_ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, + .pcm_free = au1xpsc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); + +static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { struct resource *r; int ret; @@ -365,7 +389,9 @@ static int au1xpsc_pcm_probe(struct platform_device *pdev) } (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; - return 0; + ret = snd_soc_register_platform(&au1xpsc_soc_platform); + if (!ret) + return ret; out2: kfree(au1xpsc_audio_pcmdma[PCM_RX]); @@ -376,10 +402,12 @@ out1: return ret; } -static int au1xpsc_pcm_remove(struct platform_device *pdev) +static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { int i; + snd_soc_unregister_platform(&au1xpsc_soc_platform); + for (i = 0; i < 2; i++) { if (au1xpsc_audio_pcmdma[i]) { au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); @@ -391,32 +419,83 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev) return 0; } -/* au1xpsc audio platform */ -struct snd_soc_platform au1xpsc_soc_platform = { - .name = "au1xpsc-pcm-dbdma", - .probe = au1xpsc_pcm_probe, - .remove = au1xpsc_pcm_remove, - .pcm_ops = &au1xpsc_pcm_ops, - .pcm_new = au1xpsc_pcm_new, - .pcm_free = au1xpsc_pcm_free_dma_buffers, +static struct platform_driver au1xpsc_pcm_driver = { + .driver = { + .name = "au1xpsc-pcm", + .owner = THIS_MODULE, + }, + .probe = au1xpsc_pcm_drvprobe, + .remove = __devexit_p(au1xpsc_pcm_drvremove), }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); -static int __init au1xpsc_audio_dbdma_init(void) +static int __init au1xpsc_audio_dbdma_load(void) { au1xpsc_audio_pcmdma[PCM_TX] = NULL; au1xpsc_audio_pcmdma[PCM_RX] = NULL; - return snd_soc_register_platform(&au1xpsc_soc_platform); + return platform_driver_register(&au1xpsc_pcm_driver); } -static void __exit au1xpsc_audio_dbdma_exit(void) +static void __exit au1xpsc_audio_dbdma_unload(void) { - snd_soc_unregister_platform(&au1xpsc_soc_platform); + platform_driver_unregister(&au1xpsc_pcm_driver); +} + +module_init(au1xpsc_audio_dbdma_load); +module_exit(au1xpsc_audio_dbdma_unload); + + +struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) +{ + struct resource *res, *r; + struct platform_device *pd; + int id[2]; + int ret; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + return NULL; + id[0] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + return NULL; + id[1] = r->start; + + res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL); + if (!res) + return NULL; + + res[0].start = res[0].end = id[0]; + res[1].start = res[1].end = id[1]; + res[0].flags = res[1].flags = IORESOURCE_DMA; + + pd = platform_device_alloc("au1xpsc-pcm", -1); + if (!pd) + goto out; + + pd->resource = res; + pd->num_resources = 2; + + ret = platform_device_add(pd); + if (!ret) + return pd; + +out: + kfree(res); + return NULL; } +EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); -module_init(au1xpsc_audio_dbdma_init); -module_exit(au1xpsc_audio_dbdma_exit); +void au1xpsc_pcm_destroy(struct platform_device *dmapd) +{ + if (dmapd) { + kfree(dmapd->resource); + dmapd->resource = NULL; + platform_device_unregister(dmapd); + } +} +EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); -MODULE_AUTHOR("Manuel Lauss "); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 2a06a9c548af..340311d7fed5 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -316,20 +316,56 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, static int au1xpsc_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) +{ + return au1xpsc_ac97_workdata ? 0 : -ENODEV; +} + +static void au1xpsc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ +} + +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .ac97_control = 1, + .probe = au1xpsc_ac97_probe, + .remove = au1xpsc_ac97_remove, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xpsc_ac97_dai_ops, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + +static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; struct resource *r; unsigned long sel; + struct au1xpsc_audio_data *wd; if (au1xpsc_ac97_workdata) return -EBUSY; - au1xpsc_ac97_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_ac97_workdata) + wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!wd) return -ENOMEM; - mutex_init(&au1xpsc_ac97_workdata->lock); + mutex_init(&wd->lock); r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { @@ -338,81 +374,95 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, } ret = -EBUSY; - au1xpsc_ac97_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, + wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, "au1xpsc_ac97"); - if (!au1xpsc_ac97_workdata->ioarea) + if (!wd->ioarea) goto out0; - au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_ac97_workdata->mmio) + wd->mmio = ioremap(r->start, 0xffff); + if (!wd->mmio) goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; - /* preserve PSC clock source set up by platform (dev.platform_data - * is already occupied by soc layer) - */ - sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + /* preserve PSC clock source set up by platform */ + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - /* next up: cold reset. Dont check for PSC-ready now since - * there may not be any codec clock yet. - */ - return 0; + ret = snd_soc_register_dai(&au1xpsc_ac97_dai); + if (ret) + goto out1; + + wd->dmapd = au1xpsc_pcm_add(pdev); + if (wd->dmapd) { + platform_set_drvdata(pdev, wd); + au1xpsc_ac97_workdata = wd; /* MDEV */ + return 0; + } + snd_soc_unregister_dai(&au1xpsc_ac97_dai); out1: - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); + release_resource(wd->ioarea); + kfree(wd->ioarea); out0: - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; + kfree(wd); return ret; } -static void au1xpsc_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) { + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + if (wd->dmapd) + au1xpsc_pcm_destroy(wd->dmapd); + + snd_soc_unregister_dai(&au1xpsc_ac97_dai); + /* disable PSC completely */ - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_ac97_workdata->mmio); - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; + iounmap(wd->mmio); + release_resource(wd->ioarea); + kfree(wd->ioarea); + kfree(wd); + + au1xpsc_ac97_workdata = NULL; /* MDEV */ + + return 0; } -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) +#ifdef CONFIG_PM +static int au1xpsc_ac97_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting registers and disable PSC */ - au1xpsc_ac97_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_ac97_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) +static int au1xpsc_ac97_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* restore PSC clock config */ - au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, - PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd)); au_sync(); /* after this point the ac97 core will cold-reset the codec. @@ -422,48 +472,44 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, +static struct dev_pm_ops au1xpscac97_pmops = { + .suspend = au1xpsc_ac97_drvsuspend, + .resume = au1xpsc_ac97_drvresume, }; -struct snd_soc_dai au1xpsc_ac97_dai = { - .name = "au1xpsc_ac97", - .ac97_control = 1, - .probe = au1xpsc_ac97_probe, - .remove = au1xpsc_ac97_remove, - .suspend = au1xpsc_ac97_suspend, - .resume = au1xpsc_ac97_resume, - .playback = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, - }, - .capture = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, +#define AU1XPSCAC97_PMOPS &au1xpscac97_pmops + +#else + +#define AU1XPSCAC97_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .owner = THIS_MODULE, + .pm = AU1XPSCAC97_PMOPS, }, - .ops = &au1xpsc_ac97_dai_ops, + .probe = au1xpsc_ac97_drvprobe, + .remove = __devexit_p(au1xpsc_ac97_drvremove), }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); -static int __init au1xpsc_ac97_init(void) +static int __init au1xpsc_ac97_load(void) { au1xpsc_ac97_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_ac97_dai); + return platform_driver_register(&au1xpsc_ac97_driver); } -static void __exit au1xpsc_ac97_exit(void) +static void __exit au1xpsc_ac97_unload(void) { - snd_soc_unregister_dai(&au1xpsc_ac97_dai); + platform_driver_unregister(&au1xpsc_ac97_driver); } -module_init(au1xpsc_ac97_init); -module_exit(au1xpsc_ac97_exit); +module_init(au1xpsc_ac97_load); +module_exit(au1xpsc_ac97_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss "); +MODULE_AUTHOR("Manuel Lauss"); + diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index bb589327ee32..0cf2ca61c776 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss + * Manuel Lauss * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -264,17 +264,53 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static int au1xpsc_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) +{ + return au1xpsc_i2s_workdata ? 0 : -ENODEV; +} + +static void au1xpsc_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ +} + +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .probe = au1xpsc_i2s_probe, + .remove = au1xpsc_i2s_remove, + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = &au1xpsc_i2s_dai_ops, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *r; unsigned long sel; int ret; + struct au1xpsc_audio_data *wd; if (au1xpsc_i2s_workdata) return -EBUSY; - au1xpsc_i2s_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_i2s_workdata) + wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!wd) return -ENOMEM; r = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -284,131 +320,146 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, } ret = -EBUSY; - au1xpsc_i2s_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, + wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, "au1xpsc_i2s"); - if (!au1xpsc_i2s_workdata->ioarea) + if (!wd->ioarea) goto out0; - au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_i2s_workdata->mmio) + wd->mmio = ioremap(r->start, 0xffff); + if (!wd->mmio) goto out1; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) */ - sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd)); + au_writel(0, I2S_CFG(wd)); au_sync(); /* preconfigure: set max rx/tx fifo depths */ - au1xpsc_i2s_workdata->cfg |= - PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; + wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; /* don't wait for I2S core to become ready now; clocks may not * be running yet; depending on clock input for PSC a wait might * time out. */ - return 0; + ret = snd_soc_register_dai(&au1xpsc_i2s_dai); + if (ret) + goto out1; + /* finally add the DMA device for this PSC */ + wd->dmapd = au1xpsc_pcm_add(pdev); + if (wd->dmapd) { + platform_set_drvdata(pdev, wd); + au1xpsc_i2s_workdata = wd; + return 0; + } + + snd_soc_unregister_dai(&au1xpsc_i2s_dai); out1: - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); + release_resource(wd->ioarea); + kfree(wd->ioarea); out0: - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; + kfree(wd); return ret; } -static void au1xpsc_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) { - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + if (wd->dmapd) + au1xpsc_pcm_destroy(wd->dmapd); + + snd_soc_unregister_dai(&au1xpsc_i2s_dai); + + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_i2s_workdata->mmio); - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; + iounmap(wd->mmio); + release_resource(wd->ioarea); + kfree(wd->ioarea); + kfree(wd); + + au1xpsc_i2s_workdata = NULL; /* MDEV */ + + return 0; } -static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) +#ifdef CONFIG_PM +static int au1xpsc_i2s_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting register and disable PSC */ - au1xpsc_i2s_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_i2s_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* select I2S mode and PSC clock */ - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(au1xpsc_i2s_workdata->pm[0], - PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(wd->pm[0], PSC_SEL(wd)); au_sync(); return 0; } -static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, +static struct dev_pm_ops au1xpsci2s_pmops = { + .suspend = au1xpsc_i2s_drvsuspend, + .resume = au1xpsc_i2s_drvresume, }; -struct snd_soc_dai au1xpsc_i2s_dai = { - .name = "au1xpsc_i2s", - .probe = au1xpsc_i2s_probe, - .remove = au1xpsc_i2s_remove, - .suspend = au1xpsc_i2s_suspend, - .resume = au1xpsc_i2s_resume, - .playback = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .capture = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ +#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops + +#else + +#define AU1XPSCI2S_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .owner = THIS_MODULE, + .pm = AU1XPSCI2S_PMOPS, }, - .ops = &au1xpsc_i2s_dai_ops, + .probe = au1xpsc_i2s_drvprobe, + .remove = __devexit_p(au1xpsc_i2s_drvremove), }; -EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_init(void) +static int __init au1xpsc_i2s_load(void) { au1xpsc_i2s_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_i2s_dai); + return platform_driver_register(&au1xpsc_i2s_driver); } -static void __exit au1xpsc_i2s_exit(void) +static void __exit au1xpsc_i2s_unload(void) { - snd_soc_unregister_dai(&au1xpsc_i2s_dai); + platform_driver_unregister(&au1xpsc_i2s_driver); } -module_init(au1xpsc_i2s_init); -module_exit(au1xpsc_i2s_exit); +module_init(au1xpsc_i2s_load); +module_exit(au1xpsc_i2s_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss "); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 3f474e8ed4f6..32d3807d3f5a 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss + * Manuel Lauss * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -21,6 +21,10 @@ extern struct snd_soc_dai au1xpsc_i2s_dai; extern struct snd_soc_platform au1xpsc_soc_platform; extern struct snd_ac97_bus_ops soc_ac97_ops; +/* DBDMA helpers */ +extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); +extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); + struct au1xpsc_audio_data { void __iomem *mmio; @@ -30,6 +34,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; struct mutex lock; + struct platform_device *dmapd; }; #define PCM_TX 0 -- cgit v1.2.2 From 89933dee5b17c09f2673c2bfd853625a848f91f5 Mon Sep 17 00:00:00 2001 From: Neil Jones Date: Mon, 2 Nov 2009 15:14:17 +0000 Subject: ASoC: Add support for the WM8727 DAC. Add support for the Wolfson Microelectronics WM8727 DAC, this is a simple non-configurable DAC. Signed-off-by: Neil Jones Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8727.c | 143 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8727.h | 21 +++++++ 4 files changed, 170 insertions(+) create mode 100644 sound/soc/codecs/wm8727.c create mode 100644 sound/soc/codecs/wm8727.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3df3497335bf..4a3e8dcf24d9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8523 if I2C select SND_SOC_WM8580 if I2C select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8727 select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI @@ -174,6 +175,9 @@ config SND_SOC_WM8580 config SND_SOC_WM8711 tristate +config SND_SOC_WM8727 + tristate + config SND_SOC_WM8728 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8f519ee9600d..cacfc7692d7f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,6 +27,7 @@ snd-soc-wm8510-objs := wm8510.o snd-soc-wm8523-objs := wm8523.o snd-soc-wm8580-objs := wm8580.o snd-soc-wm8711-objs := wm8711.o +snd-soc-wm8727-objs := wm8727.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o @@ -81,6 +82,7 @@ obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o +obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c new file mode 100644 index 000000000000..b3b60dd7bc14 --- /dev/null +++ b/sound/soc/codecs/wm8727.c @@ -0,0 +1,143 @@ +/* + * wm8727.c + * + * Created on: 15-Oct-2009 + * Author: neil.jones@imgtec.com + * + * Copyright (C) 2009 Imagination Technologies Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8727.h" +/* + * Note this is a simple chip with no configuration interface, sample rate is + * determined automatically by examining the Master clock and Bit clock ratios + */ +#define WM8727_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000) + + +struct snd_soc_dai wm8727_dai = { + .name = "WM8727", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8727_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}; +EXPORT_SYMBOL_GPL(wm8727_dai); + +static int wm8727_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "WM8727"; + codec->owner = THIS_MODULE; + codec->dai = &wm8727_dai; + codec->num_dai = 1; + socdev->card->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8727: failed to create pcms\n"); + goto pcm_err; + } + /* register card */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8727: failed to register card\n"); + goto register_err; + } + + return ret; + +register_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int wm8727_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8727 = { + .probe = wm8727_soc_probe, + .remove = wm8727_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8727); + + +static __devinit int wm8727_platform_probe(struct platform_device *pdev) +{ + wm8727_dai.dev = &pdev->dev; + return snd_soc_register_dai(&wm8727_dai); +} + +static int __devexit wm8727_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&wm8727_dai); + return 0; +} + +struct platform_driver wm8727_codec_driver = { + .driver = { + .name = "wm8727-codec", + .owner = THIS_MODULE, + }, + + .probe = wm8727_platform_probe, + .remove = __devexit_p(wm8727_platform_remove), +}; + +static int __init wm8727_init(void) +{ + return platform_driver_register(&wm8727_codec_driver); +} +module_init(wm8727_init); + +static void __exit wm8727_exit(void) +{ + platform_driver_unregister(&wm8727_codec_driver); +} +module_exit(wm8727_exit); + +MODULE_DESCRIPTION("ASoC wm8727 driver"); +MODULE_AUTHOR("Neil Jones"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8727.h b/sound/soc/codecs/wm8727.h new file mode 100644 index 000000000000..ee19aa71bcdc --- /dev/null +++ b/sound/soc/codecs/wm8727.h @@ -0,0 +1,21 @@ +/* + * wm8727.h + * + * Created on: 15-Oct-2009 + * Author: neil.jones@imgtec.com + * + * Copyright (C) 2009 Imagination Technologies Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef WM8727_H_ +#define WM8727_H_ + +extern struct snd_soc_dai wm8727_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8727; + +#endif /* WM8727_H_ */ -- cgit v1.2.2 From b3f5a272a33ef06a37cd44703c46ec916b8a1c93 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 2 Nov 2009 14:34:54 +0200 Subject: ASoC: TWL4030: Make sure, that the codec is powered on startup Set the codec->bias_level to SND_SOC_BIAS_OFF before changing the initial bias level to STANDBY. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f9121ef7fe5c..c0b47dfc3328 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2234,6 +2234,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) /* Set the defaults, and power up the codec */ twl4030_init_chip(codec); + codec->bias_level = SND_SOC_BIAS_OFF; twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); ret = snd_soc_register_codec(codec); -- cgit v1.2.2 From 529697c5463d941445db18e9526e7fc76a18e503 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 3 Nov 2009 22:13:30 +0000 Subject: ASoC: Staticise wm8727 driver structure Signed-off-by: Mark Brown --- sound/soc/codecs/wm8727.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index b3b60dd7bc14..7df5a17eb733 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -116,7 +116,7 @@ static int __devexit wm8727_platform_remove(struct platform_device *pdev) return 0; } -struct platform_driver wm8727_codec_driver = { +static struct platform_driver wm8727_codec_driver = { .driver = { .name = "wm8727-codec", .owner = THIS_MODULE, -- cgit v1.2.2 From 2624d5fa67a5d3d720613a4ab0672e8c387ba806 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 3 Nov 2009 21:56:13 +0000 Subject: ASoC: Move sysfs and debugfs functions to head of soc-core.c A fairly hefty change in diff terms but no actual code changes, will be used by the next commit. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 334 +++++++++++++++++++++++++-------------------------- 1 file changed, 167 insertions(+), 167 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e24654194ee..d81a16187769 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -80,6 +80,173 @@ static int run_delayed_work(struct delayed_work *dwork) return ret; } +/* codec register dump */ +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) +{ + int i, step = 1, count = 0; + + if (!codec->reg_cache_size) + return 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + count += sprintf(buf, "%s registers\n", codec->name); + for (i = 0; i < codec->reg_cache_size; i += step) { + if (codec->readable_register && !codec->readable_register(i)) + continue; + + count += sprintf(buf + count, "%2x: ", i); + if (count >= PAGE_SIZE - 1) + break; + + if (codec->display_register) + count += codec->display_register(codec, buf + count, + PAGE_SIZE - count, i); + else + count += snprintf(buf + count, PAGE_SIZE - count, + "%4x", codec->read(codec, i)); + + if (count >= PAGE_SIZE - 1) + break; + + count += snprintf(buf + count, PAGE_SIZE - count, "\n"); + if (count >= PAGE_SIZE - 1) + break; + } + + /* Truncate count; min() would cause a warning */ + if (count >= PAGE_SIZE) + count = PAGE_SIZE - 1; + + return count; +} +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata->card->codec, buf); +} + +static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); + +#ifdef CONFIG_DEBUG_FS +static int codec_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_codec *codec = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(codec, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t codec_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_codec *codec = file->private_data; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = codec_reg_open_file, + .read = codec_reg_read_file, + .write = codec_reg_write_file, +}; + +static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ + char codec_root[128]; + + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); + + codec->debugfs_codec_root = debugfs_create_dir(codec_root, + debugfs_root); + if (!codec->debugfs_codec_root) { + printk(KERN_WARNING + "ASoC: Failed to create codec debugfs directory\n"); + return; + } + + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, + codec->debugfs_codec_root, + codec, &codec_reg_fops); + if (!codec->debugfs_reg) + printk(KERN_WARNING + "ASoC: Failed to create codec register debugfs file\n"); + + codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, + codec->debugfs_codec_root, + &codec->pop_time); + if (!codec->debugfs_pop_time) + printk(KERN_WARNING + "Failed to create pop time debugfs file\n"); + + codec->debugfs_dapm = debugfs_create_dir("dapm", + codec->debugfs_codec_root); + if (!codec->debugfs_dapm) + printk(KERN_WARNING + "Failed to create DAPM debugfs directory\n"); + + snd_soc_dapm_debugfs_init(codec); +} + +static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ + debugfs_remove_recursive(codec->debugfs_codec_root); +} + +#else + +static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ +} + +static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ +} +#endif + #ifdef CONFIG_SND_SOC_AC97_BUS /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) @@ -1111,173 +1278,6 @@ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) } EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); -/* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) -{ - int i, step = 1, count = 0; - - if (!codec->reg_cache_size) - return 0; - - if (codec->reg_cache_step) - step = codec->reg_cache_step; - - count += sprintf(buf, "%s registers\n", codec->name); - for (i = 0; i < codec->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(i)) - continue; - - count += sprintf(buf + count, "%2x: ", i); - if (count >= PAGE_SIZE - 1) - break; - - if (codec->display_register) - count += codec->display_register(codec, buf + count, - PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); - - if (count >= PAGE_SIZE - 1) - break; - - count += snprintf(buf + count, PAGE_SIZE - count, "\n"); - if (count >= PAGE_SIZE - 1) - break; - } - - /* Truncate count; min() would cause a warning */ - if (count >= PAGE_SIZE) - count = PAGE_SIZE - 1; - - return count; -} -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct snd_soc_device *devdata = dev_get_drvdata(dev); - return soc_codec_reg_show(devdata->card->codec, buf); -} - -static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); - -#ifdef CONFIG_DEBUG_FS -static int codec_reg_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - -static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - ssize_t ret; - struct snd_soc_codec *codec = file->private_data; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!buf) - return -ENOMEM; - ret = soc_codec_reg_show(codec, buf); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - kfree(buf); - return ret; -} - -static ssize_t codec_reg_write_file(struct file *file, - const char __user *user_buf, size_t count, loff_t *ppos) -{ - char buf[32]; - int buf_size; - char *start = buf; - unsigned long reg, value; - int step = 1; - struct snd_soc_codec *codec = file->private_data; - - buf_size = min(count, (sizeof(buf)-1)); - if (copy_from_user(buf, user_buf, buf_size)) - return -EFAULT; - buf[buf_size] = 0; - - if (codec->reg_cache_step) - step = codec->reg_cache_step; - - while (*start == ' ') - start++; - reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->reg_cache_size) || (reg % step)) - return -EINVAL; - while (*start == ' ') - start++; - if (strict_strtoul(start, 16, &value)) - return -EINVAL; - codec->write(codec, reg, value); - return buf_size; -} - -static const struct file_operations codec_reg_fops = { - .open = codec_reg_open_file, - .read = codec_reg_read_file, - .write = codec_reg_write_file, -}; - -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ - char codec_root[128]; - - if (codec->dev) - snprintf(codec_root, sizeof(codec_root), - "%s.%s", codec->name, dev_name(codec->dev)); - else - snprintf(codec_root, sizeof(codec_root), - "%s", codec->name); - - codec->debugfs_codec_root = debugfs_create_dir(codec_root, - debugfs_root); - if (!codec->debugfs_codec_root) { - printk(KERN_WARNING - "ASoC: Failed to create codec debugfs directory\n"); - return; - } - - codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->debugfs_codec_root, - codec, &codec_reg_fops); - if (!codec->debugfs_reg) - printk(KERN_WARNING - "ASoC: Failed to create codec register debugfs file\n"); - - codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, - codec->debugfs_codec_root, - &codec->pop_time); - if (!codec->debugfs_pop_time) - printk(KERN_WARNING - "Failed to create pop time debugfs file\n"); - - codec->debugfs_dapm = debugfs_create_dir("dapm", - codec->debugfs_codec_root); - if (!codec->debugfs_dapm) - printk(KERN_WARNING - "Failed to create DAPM debugfs directory\n"); - - snd_soc_dapm_debugfs_init(codec); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_codec_root); -} - -#else - -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} - -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ -} -#endif - /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec -- cgit v1.2.2 From fe3e78e073d25308756f38019956061153267769 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 3 Nov 2009 22:13:13 +0000 Subject: ASoC: Factor out snd_soc_init_card() snd_soc_init_card() is always called as the last part of the CODEC probe function so we can factor it out into the core card setup rather than have each CODEC replicate the code to do the initialiastation. This will be required to support multiple CODECs per card. Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 3 - sound/soc/codecs/ad1836.c | 6 -- sound/soc/codecs/ad1938.c | 6 -- sound/soc/codecs/ad1980.c | 5 -- sound/soc/codecs/ad73311.c | 8 --- sound/soc/codecs/ak4104.c | 8 --- sound/soc/codecs/ak4535.c | 8 --- sound/soc/codecs/ak4642.c | 9 --- sound/soc/codecs/ak4671.c | 9 --- sound/soc/codecs/cs4270.c | 7 -- sound/soc/codecs/cx20442.c | 6 -- sound/soc/codecs/pcm3008.c | 9 --- sound/soc/codecs/ssm2602.c | 8 --- sound/soc/codecs/stac9766.c | 3 - sound/soc/codecs/tlv320aic23.c | 8 --- sound/soc/codecs/tlv320aic26.c | 11 ---- sound/soc/codecs/tlv320aic3x.c | 10 --- sound/soc/codecs/tlv320dac33.c | 10 +-- sound/soc/codecs/twl4030.c | 12 ---- sound/soc/codecs/uda134x.c | 9 --- sound/soc/codecs/uda1380.c | 8 --- sound/soc/codecs/wm8350.c | 11 ---- sound/soc/codecs/wm8400.c | 6 -- sound/soc/codecs/wm8510.c | 9 +-- sound/soc/codecs/wm8523.c | 8 --- sound/soc/codecs/wm8580.c | 8 --- sound/soc/codecs/wm8711.c | 8 --- sound/soc/codecs/wm8727.c | 8 --- sound/soc/codecs/wm8728.c | 8 --- sound/soc/codecs/wm8731.c | 8 --- sound/soc/codecs/wm8750.c | 8 --- sound/soc/codecs/wm8753.c | 9 --- sound/soc/codecs/wm8776.c | 9 --- sound/soc/codecs/wm8900.c | 6 -- sound/soc/codecs/wm8903.c | 9 --- sound/soc/codecs/wm8940.c | 6 -- sound/soc/codecs/wm8960.c | 8 --- sound/soc/codecs/wm8961.c | 9 --- sound/soc/codecs/wm8971.c | 9 +-- sound/soc/codecs/wm8974.c | 8 --- sound/soc/codecs/wm8988.c | 9 --- sound/soc/codecs/wm8990.c | 9 +-- sound/soc/codecs/wm8993.c | 9 --- sound/soc/codecs/wm9081.c | 9 --- sound/soc/codecs/wm9705.c | 8 --- sound/soc/codecs/wm9712.c | 8 --- sound/soc/codecs/wm9713.c | 7 +- sound/soc/soc-core.c | 141 +++++++++++++++++++---------------------- 48 files changed, 69 insertions(+), 449 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 932299bb5d1e..69bd0acc81c8 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -117,9 +117,6 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto bus_err; return 0; bus_err: diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index c48485f2c55d..2e360c243075 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -387,12 +387,6 @@ static int ad1836_probe(struct platform_device *pdev) snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); snd_soc_dapm_new_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; card_err: diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 34b30efc3cb0..09c008ad1476 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -596,12 +596,6 @@ static int ad1938_probe(struct platform_device *pdev) ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; card_err: diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index d7440a982d22..39c0f7584e65 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -257,11 +257,6 @@ static int ad1980_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, ad1980_snd_ac97_controls, ARRAY_SIZE(ad1980_snd_ac97_controls)); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ad1980: failed to register card\n"); - goto reset_err; - } return 0; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index e61dac5e7b8f..d2fcc601722c 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -64,16 +64,8 @@ static int ad73311_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ad73311: failed to register card\n"); - goto register_err; - } - return ret; -register_err: - snd_soc_free_pcms(socdev); pcm_err: kfree(socdev->card->codec); socdev->card->codec = NULL; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 4d47bc4f7428..3a14c6fc4f5e 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -313,14 +313,6 @@ static int ak4104_probe(struct platform_device *pdev) return ret; } - /* Register the socdev */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - snd_soc_free_pcms(socdev); - return ret; - } - return 0; } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 0abec0d29a96..57a6846a9a1f 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -485,17 +485,9 @@ static int ak4535_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); ak4535_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ak4535: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index e057c7b578df..b69861d52161 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -442,18 +442,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ak4642: failed to register card\n"); - goto card_err; - } - dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index b61214d1c5de..364832ccd748 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -662,19 +662,10 @@ static int ak4671_probe(struct platform_device *pdev) ARRAY_SIZE(ak4671_snd_controls)); ak4671_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 565842dcfc65..ffe122d1cd76 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -599,13 +599,6 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } - /* And finally, register the socdev */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - goto error_free_pcms; - } - return 0; error_free_pcms: diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 38eac9c866e1..d7f9bf18b72e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -355,12 +355,6 @@ static int cx20442_codec_probe(struct platform_device *pdev) cx20442_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - return ret; card_err: diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 5cda9e6b5a74..2afcd0a8669d 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -90,13 +90,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev) goto pcm_err; } - /* Register Card. */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "pcm3008: failed to register card\n"); - goto card_err; - } - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF @@ -136,8 +129,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev) gpio_err: pcm3008_gpio_free(setup); -card_err: - snd_soc_free_pcms(socdev); pcm_err: kfree(socdev->card->codec); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c550750c79c0..b3130339d29a 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -613,17 +613,9 @@ static int ssm2602_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - pr_err("ssm2602: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index befc6488c39a..bbc72c2ddfca 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -418,9 +418,6 @@ static int stac9766_codec_probe(struct platform_device *pdev) snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto reset_err; return 0; reset_err: diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5cd729..ee8cb2c08b87 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -707,17 +707,9 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); tlv320aic23_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "tlv320aic23: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 3387d9e736ea..357b609196e3 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -356,18 +356,7 @@ static int aic26_probe(struct platform_device *pdev) ARRAY_SIZE(aic26_snd_controls)); WARN_ON(err < 0); - /* CODEC is setup, we can register the card now */ - dev_dbg(&pdev->dev, "Registering card\n"); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "aic26: failed to register card\n"); - goto card_err; - } return 0; - - card_err: - snd_soc_free_pcms(socdev); - return ret; } static int aic26_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3395cf945d56..03cad250f58d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1405,18 +1405,8 @@ static int aic3x_probe(struct platform_device *pdev) aic3x_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "aic3x: failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 3ca8934fc26c..bff476d65d05 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -960,16 +960,8 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - goto card_err; - } - return 0; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); + pcm_err: dac33_hard_power(codec, 0); return ret; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c0b47dfc3328..928257b25111 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2155,19 +2155,7 @@ static int twl4030_soc_probe(struct platform_device *pdev) ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - return 0; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - - return ret; } static int twl4030_soc_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c33b92edbded..aa40d985138f 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -562,17 +562,8 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "UDA134X: failed to register card\n"); - goto card_err; - } - return 0; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); reg_err: diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 92ec03442154..a42e47d94630 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -713,17 +713,9 @@ static int uda1380_probe(struct platform_device *pdev) snd_soc_add_controls(codec, uda1380_snd_controls, ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 714114b50d18..2e35a354b166 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1501,18 +1501,7 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - return 0; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - return ret; } static int wm8350_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index bd7eecba20fe..0e30997c8db0 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1400,12 +1400,6 @@ static int wm8400_probe(struct platform_device *pdev) wm8400_add_controls(codec); wm8400_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - return ret; card_err: diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 5702435af81b..e3c21ebcc08e 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -604,16 +604,9 @@ static int wm8510_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8510_snd_controls, ARRAY_SIZE(wm8510_snd_controls)); wm8510_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8510: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 268cab21c2cc..2e2b01d6c82b 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -448,17 +448,9 @@ static int wm8523_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8523_snd_controls, ARRAY_SIZE(wm8523_snd_controls)); wm8523_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index a09b23e03664..dde50d118181 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -800,17 +800,9 @@ static int wm8580_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8580_snd_controls, ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 54189fbf9e93..70e0675b5d4a 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -404,17 +404,9 @@ static int wm8711_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8711_snd_controls, ARRAY_SIZE(wm8711_snd_controls)); wm8711_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 7df5a17eb733..d8ffbd641d71 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -68,17 +68,9 @@ static int wm8727_soc_probe(struct platform_device *pdev) printk(KERN_ERR "wm8727: failed to create pcms\n"); goto pcm_err; } - /* register card */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8727: failed to register card\n"); - goto register_err; - } return ret; -register_err: - snd_soc_free_pcms(socdev); pcm_err: kfree(socdev->card->codec); socdev->card->codec = NULL; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 16e969a762c3..1252a8a486a6 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -287,17 +287,9 @@ static int wm8728_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8728_snd_controls, ARRAY_SIZE(wm8728_snd_controls)); wm8728_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8728: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index bb95af950971..e3675e7a9813 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -495,17 +495,9 @@ static int wm8731_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8731_snd_controls, ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4ba1e7e93fb4..50a3d6590588 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -772,16 +772,8 @@ static int wm8750_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); wm8750_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8f7305257d29..c652bc04cc81 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1583,18 +1583,9 @@ static int wm8753_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8753_snd_controls, ARRAY_SIZE(wm8753_snd_controls)); wm8753_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8753: failed to register card\n"); - goto card_err; - } return 0; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index a0bbb28eed75..ab2c0da18091 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -447,17 +447,8 @@ static int wm8776_probe(struct platform_device *pdev) ARRAY_SIZE(wm8776_dapm_widgets)); snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index b48804b5cacd..0d185cb6418d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1353,12 +1353,6 @@ static int wm8900_probe(struct platform_device *pdev) ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "Failed to register card\n"); - goto card_err; - } - return ret; card_err: diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 94cdb8130415..bfeff4ee5de9 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1695,17 +1695,8 @@ static int wm8903_probe(struct platform_device *pdev) ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(socdev->card->codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "wm8903: failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: return ret; } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 8d4fd3c08c09..fc80aa6c913c 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -731,12 +731,6 @@ static int wm8940_probe(struct platform_device *pdev) if (ret) goto error_free_pcms; - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto error_free_pcms; - } - return ret; error_free_pcms: diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index b9b096a85396..40390afa75f3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -713,17 +713,9 @@ static int wm8960_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index b5c6f2cd5ae2..07e389574db1 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -988,17 +988,8 @@ static int wm8961_probe(struct platform_device *pdev) snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); snd_soc_dapm_new_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index d66efb0546ea..56a66e89ab91 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -703,16 +703,9 @@ static int wm8971_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); wm8971_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8971: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index eff29331235b..c245f0ee0ec2 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -641,17 +641,9 @@ static int wm8974_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8974_snd_controls, ARRAY_SIZE(wm8974_snd_controls)); wm8974_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index d8d8f68b81ea..bee292e37d1b 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -792,17 +792,8 @@ static int wm8988_probe(struct platform_device *pdev) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index f657e9a5fe26..e43cb2c8b915 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1409,16 +1409,9 @@ static int wm8990_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, wm8990_snd_controls, ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8990: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index dac397712147..0d4d2be92b64 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1466,17 +1466,8 @@ static int wm8993_probe(struct platform_device *pdev) snd_soc_dapm_new_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: return ret; } diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 4cb6b104b729..3f1f84421312 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1264,17 +1264,8 @@ static int wm9081_probe(struct platform_device *pdev) snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); snd_soc_dapm_new_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index e7d2840d9e59..0e817b8705cd 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -403,16 +403,8 @@ static int wm9705_soc_probe(struct platform_device *pdev) ARRAY_SIZE(wm9705_snd_ac97_controls)); wm9705_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm9705: failed to register card\n"); - goto reset_err; - } - return 0; -reset_err: - snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fd4e88f50cf..155cacf124ea 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -695,17 +695,9 @@ static int wm9712_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm9712_snd_ac97_controls, ARRAY_SIZE(wm9712_snd_ac97_controls)); wm9712_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm9712: failed to register card\n"); - goto reset_err; - } return 0; -reset_err: - snd_soc_free_pcms(socdev); - pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index ca3d449ed89e..5f81ecd20a81 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1247,13 +1247,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm9713_snd_ac97_controls, ARRAY_SIZE(wm9713_snd_ac97_controls)); wm9713_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto reset_err; - return 0; -reset_err: - snd_soc_free_pcms(socdev); + return 0; pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d81a16187769..e2b6d75f16e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -970,6 +970,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct platform_device, dev); struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev; + struct snd_soc_codec *codec; struct snd_soc_platform *platform; struct snd_soc_dai *dai; int i, found, ret, ac97; @@ -1058,6 +1059,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) goto cpu_dai_err; } + codec = card->codec; if (platform->probe) { ret = platform->probe(pdev); @@ -1072,10 +1074,72 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); #endif + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].init) { + ret = card->dai_link[i].init(codec); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to init %s\n", + card->dai_link[i].stream_name); + continue; + } + } + if (card->dai_link[i].codec_dai->ac97_control) { + ac97 = 1; + snd_ac97_dev_add_pdata(codec->ac97, + card->dai_link[i].cpu_dai->ac97_pdata); + } + } + + snprintf(codec->card->shortname, sizeof(codec->card->shortname), + "%s", card->name); + snprintf(codec->card->longname, sizeof(codec->card->longname), + "%s (%s)", card->name, codec->name); + + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec); + + ret = snd_card_register(codec->card); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", + codec->name); + goto card_err; + } + + mutex_lock(&codec->mutex); +#ifdef CONFIG_SND_SOC_AC97_BUS + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (ac97 && strcmp(codec->name, "AC97") != 0) { + ret = soc_ac97_dev_register(codec); + if (ret < 0) { + printk(KERN_ERR "asoc: AC97 device register failed\n"); + snd_card_free(codec->card); + mutex_unlock(&codec->mutex); + goto card_err; + } + } +#endif + + ret = snd_soc_dapm_sys_add(card->socdev->dev); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); + + ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + + soc_init_codec_debugfs(codec); + mutex_unlock(&codec->mutex); + card->instantiated = 1; return; +card_err: + if (platform->remove) + platform->remove(pdev); + platform_err: if (codec_dev->remove) codec_dev->remove(pdev); @@ -1453,83 +1517,6 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) } EXPORT_SYMBOL_GPL(snd_soc_new_pcms); -/** - * snd_soc_init_card - register sound card - * @socdev: the SoC audio device - * - * Register a SoC sound card. Also registers an AC97 device if the - * codec is AC97 for ad hoc devices. - * - * Returns 0 for success, else error. - */ -int snd_soc_init_card(struct snd_soc_device *socdev) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = card->codec; - int ret = 0, i, ac97 = 0, err = 0; - - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].init) { - err = card->dai_link[i].init(codec); - if (err < 0) { - printk(KERN_ERR "asoc: failed to init %s\n", - card->dai_link[i].stream_name); - continue; - } - } - if (card->dai_link[i].codec_dai->ac97_control) { - ac97 = 1; - snd_ac97_dev_add_pdata(codec->ac97, - card->dai_link[i].cpu_dai->ac97_pdata); - } - } - snprintf(codec->card->shortname, sizeof(codec->card->shortname), - "%s", card->name); - snprintf(codec->card->longname, sizeof(codec->card->longname), - "%s (%s)", card->name, codec->name); - - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(codec); - - ret = snd_card_register(codec->card); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", - codec->name); - goto out; - } - - mutex_lock(&codec->mutex); -#ifdef CONFIG_SND_SOC_AC97_BUS - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (ac97 && strcmp(codec->name, "AC97") != 0) { - ret = soc_ac97_dev_register(codec); - if (ret < 0) { - printk(KERN_ERR "asoc: AC97 device register failed\n"); - snd_card_free(codec->card); - mutex_unlock(&codec->mutex); - goto out; - } - } -#endif - - err = snd_soc_dapm_sys_add(socdev->dev); - if (err < 0) - printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); - - err = device_create_file(socdev->dev, &dev_attr_codec_reg); - if (err < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); - - soc_init_codec_debugfs(codec); - mutex_unlock(&codec->mutex); - -out: - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_init_card); - /** * snd_soc_free_pcms - free sound card and pcms * @socdev: the SoC audio device -- cgit v1.2.2 From 9dcaa7b25f2c8f6a0485854cd3641f585a154072 Mon Sep 17 00:00:00 2001 From: Rafael Ignacio Zurita Date: Tue, 3 Nov 2009 17:16:27 -0300 Subject: ALSA: sh: add SuperH DAC audio driver for ALSA V4 This is a port of the sound/oss/sh_dac_audio.c driver. The driver uses an on-chip 8-bit D/A converter, which has a speaker connected to one of its channels, found in several ancient HP machines. For interrupts it uses a high-resolution timer (hrtimer). Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx). Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver would be obsolete soon, and it could be removed. Signed-off-by: Rafael Ignacio Zurita Acked-by: Paul Mundt Signed-off-by: Takashi Iwai --- sound/sh/Kconfig | 8 + sound/sh/Makefile | 2 + sound/sh/sh_dac_audio.c | 453 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 463 insertions(+) create mode 100644 sound/sh/sh_dac_audio.c (limited to 'sound') diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig index aed0f90c3919..61139f3c1614 100644 --- a/sound/sh/Kconfig +++ b/sound/sh/Kconfig @@ -19,5 +19,13 @@ config SND_AICA help ALSA Sound driver for the SEGA Dreamcast console. +config SND_SH_DAC_AUDIO + tristate "SuperH DAC audio support" + depends on SND + depends on CPU_SH3 && HIGH_RES_TIMERS + select SND_PCM + help + Say Y here to include support for the on-chip DAC. + endif # SND_SUPERH diff --git a/sound/sh/Makefile b/sound/sh/Makefile index 8fdcb6e26f00..7d09b5188cf7 100644 --- a/sound/sh/Makefile +++ b/sound/sh/Makefile @@ -3,6 +3,8 @@ # snd-aica-objs := aica.o +snd-sh_dac_audio-objs := sh_dac_audio.o # Toplevel Module Dependency obj-$(CONFIG_SND_AICA) += snd-aica.o +obj-$(CONFIG_SND_SH_DAC_AUDIO) += snd-sh_dac_audio.o diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c new file mode 100644 index 000000000000..76d9ad27d91c --- /dev/null +++ b/sound/sh/sh_dac_audio.c @@ -0,0 +1,453 @@ +/* + * sh_dac_audio.c - SuperH DAC audio driver for ALSA + * + * Copyright (c) 2009 by Rafael Ignacio Zurita + * + * + * Based on sh_dac_audio.c (Copyright (C) 2004, 2005 by Andriy Skulysh) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Rafael Ignacio Zurita "); +MODULE_DESCRIPTION("SuperH DAC audio driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SuperH DAC audio support}}"); + +/* Module Parameters */ +static int index = SNDRV_DEFAULT_IDX1; +static char *id = SNDRV_DEFAULT_STR1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SuperH DAC audio."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SuperH DAC audio."); + +/* main struct */ +struct snd_sh_dac { + struct snd_card *card; + struct snd_pcm_substream *substream; + struct hrtimer hrtimer; + ktime_t wakeups_per_second; + + int rate; + int empty; + char *data_buffer, *buffer_begin, *buffer_end; + int processed; /* bytes proccesed, to compare with period_size */ + int buffer_size; + struct dac_audio_pdata *pdata; +}; + + +static void dac_audio_start_timer(struct snd_sh_dac *chip) +{ + hrtimer_start(&chip->hrtimer, chip->wakeups_per_second, + HRTIMER_MODE_REL); +} + +static void dac_audio_stop_timer(struct snd_sh_dac *chip) +{ + hrtimer_cancel(&chip->hrtimer); +} + +static void dac_audio_reset(struct snd_sh_dac *chip) +{ + dac_audio_stop_timer(chip); + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; +} + +static void dac_audio_set_rate(struct snd_sh_dac *chip) +{ + chip->wakeups_per_second = ktime_set(0, 1000000000 / chip->rate); +} + + +/* PCM INTERFACE */ + +static struct snd_pcm_hardware snd_sh_dac_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX), + .formats = SNDRV_PCM_FMTBIT_U8, + .rates = SNDRV_PCM_RATE_8000, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = (48*1024), + .period_bytes_min = 1, + .period_bytes_max = (48*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_sh_dac_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sh_dac_pcm_hw; + + chip->substream = substream; + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; + + chip->pdata->start(chip->pdata); + + return 0; +} + +static int snd_sh_dac_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + + chip->substream = NULL; + + dac_audio_stop_timer(chip); + chip->pdata->stop(chip->pdata); + + return 0; +} + +static int snd_sh_dac_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_sh_dac_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_sh_dac_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = chip->substream->runtime; + + chip->buffer_size = runtime->buffer_size; + memset(chip->data_buffer, 0, chip->pdata->buffer_size); + + return 0; +} + +static int snd_sh_dac_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dac_audio_start_timer(chip); + break; + case SNDRV_PCM_TRIGGER_STOP: + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; + dac_audio_stop_timer(chip); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count) +{ + /* channel is not used (interleaved data) */ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + ssize_t b_count = frames_to_bytes(runtime , count); + ssize_t b_pos = frames_to_bytes(runtime , pos); + + if (count < 0) + return -EINVAL; + + if (!count) + return 0; + + memcpy_toio(chip->data_buffer + b_pos, src, b_count); + chip->buffer_end = chip->data_buffer + b_pos + b_count; + + if (chip->empty) { + chip->empty = 0; + dac_audio_start_timer(chip); + } + + return 0; +} + +static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + /* channel is not used (interleaved data) */ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + ssize_t b_count = frames_to_bytes(runtime , count); + ssize_t b_pos = frames_to_bytes(runtime , pos); + + if (count < 0) + return -EINVAL; + + if (!count) + return 0; + + memset_io(chip->data_buffer + b_pos, 0, b_count); + chip->buffer_end = chip->data_buffer + b_pos + b_count; + + if (chip->empty) { + chip->empty = 0; + dac_audio_start_timer(chip); + } + + return 0; +} + +static +snd_pcm_uframes_t snd_sh_dac_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + int pointer = chip->buffer_begin - chip->data_buffer; + + return pointer; +} + +/* pcm ops */ +static struct snd_pcm_ops snd_sh_dac_pcm_ops = { + .open = snd_sh_dac_pcm_open, + .close = snd_sh_dac_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sh_dac_pcm_hw_params, + .hw_free = snd_sh_dac_pcm_hw_free, + .prepare = snd_sh_dac_pcm_prepare, + .trigger = snd_sh_dac_pcm_trigger, + .pointer = snd_sh_dac_pcm_pointer, + .copy = snd_sh_dac_pcm_copy, + .silence = snd_sh_dac_pcm_silence, + .mmap = snd_pcm_lib_mmap_iomem, +}; + +static int __devinit snd_sh_dac_pcm(struct snd_sh_dac *chip, int device) +{ + int err; + struct snd_pcm *pcm; + + /* device should be always 0 for us */ + err = snd_pcm_new(chip->card, "SH_DAC PCM", device, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SH_DAC PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sh_dac_pcm_ops); + + /* buffer size=48K */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 48 * 1024, + 48 * 1024); + + return 0; +} +/* END OF PCM INTERFACE */ + + +/* driver .remove -- destructor */ +static int snd_sh_dac_remove(struct platform_device *devptr) +{ + snd_card_free(platform_get_drvdata(devptr)); + platform_set_drvdata(devptr, NULL); + + return 0; +} + +/* free -- it has been defined by create */ +static int snd_sh_dac_free(struct snd_sh_dac *chip) +{ + /* release the data */ + kfree(chip->data_buffer); + kfree(chip); + + return 0; +} + +static int snd_sh_dac_dev_free(struct snd_device *device) +{ + struct snd_sh_dac *chip = device->device_data; + + return snd_sh_dac_free(chip); +} + +static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle) +{ + struct snd_sh_dac *chip = container_of(handle, struct snd_sh_dac, + hrtimer); + struct snd_pcm_runtime *runtime = chip->substream->runtime; + ssize_t b_ps = frames_to_bytes(runtime, runtime->period_size); + + if (!chip->empty) { + sh_dac_output(*chip->buffer_begin, chip->pdata->channel); + chip->buffer_begin++; + + chip->processed++; + if (chip->processed >= b_ps) { + chip->processed -= b_ps; + snd_pcm_period_elapsed(chip->substream); + } + + if (chip->buffer_begin == (chip->data_buffer + + chip->buffer_size - 1)) + chip->buffer_begin = chip->data_buffer; + + if (chip->buffer_begin == chip->buffer_end) + chip->empty = 1; + + } + + if (!chip->empty) + hrtimer_start(&chip->hrtimer, chip->wakeups_per_second, + HRTIMER_MODE_REL); + + return HRTIMER_NORESTART; +} + +/* create -- chip-specific constructor for the cards components */ +static int __devinit snd_sh_dac_create(struct snd_card *card, + struct platform_device *devptr, + struct snd_sh_dac **rchip) +{ + struct snd_sh_dac *chip; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_sh_dac_dev_free, + }; + + *rchip = NULL; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + hrtimer_init(&chip->hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + chip->hrtimer.function = sh_dac_audio_timer; + + dac_audio_reset(chip); + chip->rate = 8000; + dac_audio_set_rate(chip); + + chip->pdata = devptr->dev.platform_data; + + chip->data_buffer = kmalloc(chip->pdata->buffer_size, GFP_KERNEL); + if (chip->data_buffer == NULL) { + kfree(chip); + return -ENOMEM; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sh_dac_free(chip); + return err; + } + + *rchip = chip; + + return 0; +} + +/* driver .probe -- constructor */ +static int __devinit snd_sh_dac_probe(struct platform_device *devptr) +{ + struct snd_sh_dac *chip; + struct snd_card *card; + int err; + + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR "cannot allocate the card\n"); + return err; + } + + err = snd_sh_dac_create(card, devptr, &chip); + if (err < 0) + goto probe_error; + + err = snd_sh_dac_pcm(chip, 0); + if (err < 0) + goto probe_error; + + strcpy(card->driver, "snd_sh_dac"); + strcpy(card->shortname, "SuperH DAC audio driver"); + printk(KERN_INFO "%s %s", card->longname, card->shortname); + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + snd_printk("ALSA driver for SuperH DAC audio"); + + platform_set_drvdata(devptr, card); + return 0; + +probe_error: + snd_card_free(card); + return err; +} + +/* + * "driver" definition + */ +static struct platform_driver driver = { + .probe = snd_sh_dac_probe, + .remove = snd_sh_dac_remove, + .driver = { + .name = "dac_audio", + }, +}; + +static int __init sh_dac_init(void) +{ + return platform_driver_register(&driver); +} + +static void __exit sh_dac_exit(void) +{ + platform_driver_unregister(&driver); +} + +module_init(sh_dac_init); +module_exit(sh_dac_exit); -- cgit v1.2.2 From 2dcf9fb99d4ecadecb2685a9eb82e6b85511c960 Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Wed, 4 Nov 2009 17:49:22 +0000 Subject: ASoC: ADS117x ADC driver This patch adds support for the TI ADS117x family of multichannel ADCs and was sponsored by Shotspotter Inc. Signed-off-by: Graeme Gregory Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ads117x.c | 127 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ads117x.h | 13 +++++ 4 files changed, 146 insertions(+) create mode 100644 sound/soc/codecs/ads117x.c create mode 100644 sound/soc/codecs/ads117x.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4a3e8dcf24d9..52b005f8fed4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD1938 if SPI_MASTER select SND_SOC_AD1980 if SND_SOC_AC97_BUS + select SND_SOC_ADS117X select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -91,6 +92,9 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate + +config SND_SOC_ADS117X + tristate config SND_SOC_AK4104 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cacfc7692d7f..dbaecb133ac7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1836-objs := ad1836.o snd-soc-ad1938-objs := ad1938.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o @@ -58,6 +59,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c new file mode 100644 index 000000000000..f3230927dc66 --- /dev/null +++ b/sound/soc/codecs/ads117x.c @@ -0,0 +1,127 @@ +/* + * ads117x.c -- Driver for ads1174/8 ADC chips + * + * Copyright 2009 ShotSpotter Inc. + * Author: Graeme Gregory + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "ads117x.h" + +#define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) + +#define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai ads117x_dai = { +/* ADC */ + .name = "ADS117X ADC", + .id = 1, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 32, + .rates = ADS117X_RATES, + .formats = ADS117X_FORMATS,}, +}; +EXPORT_SYMBOL_GPL(ads117x_dai); + +/* + * initialise the ads117x driver + */ +static int ads117x_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + int ret = 0; + + codec->name = "ADS117X"; + codec->owner = THIS_MODULE; + codec->dai = &ads117x_dai; + codec->num_dai = 1; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ads117x: failed to create pcms\n"); + return ret; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ads117x: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + return ret; +} + +static int ads117x_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + pr_info("ads117x ADC\n"); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = ads117x_init(socdev); + if (ret != 0) + kfree(codec); + + return ret; +} + +static int ads117x_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_free_pcms(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ads117x = { + .probe = ads117x_probe, + .remove = ads117x_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ads117x); + +static int __init ads117x_modinit(void) +{ + return snd_soc_register_dai(&ads117x_dai); +} +module_init(ads117x_modinit); + +static void __exit ads117x_exit(void) +{ + snd_soc_unregister_dai(&ads117x_dai); +} +module_exit(ads117x_exit); + +MODULE_DESCRIPTION("ASoC ads117x driver"); +MODULE_AUTHOR("Graeme Gregory"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h new file mode 100644 index 000000000000..dbcf50ec9bd1 --- /dev/null +++ b/sound/soc/codecs/ads117x.h @@ -0,0 +1,13 @@ +/* + * ads117x.h -- Driver for ads1174/8 ADC chips + * + * Copyright 2009 ShotSpotter Inc. + * Author: Graeme Gregory + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +extern struct snd_soc_dai ads117x_dai; +extern struct snd_soc_codec_device soc_codec_dev_ads117x; -- cgit v1.2.2 From f3d0e82fe3cce0dd3ffcd9c59e6caa671a30f929 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Nov 2009 21:43:27 +0000 Subject: ASoC: Update ads117x to current APIs Probe as a platform driver (ads117x) and remove the call to snd_soc_init_card(). Signed-off-by: Mark Brown --- sound/soc/codecs/ads117x.c | 76 ++++++++++++++++++++++------------------------ 1 file changed, 36 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index f3230927dc66..cc96411ca3e6 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -37,46 +37,12 @@ struct snd_soc_dai ads117x_dai = { }; EXPORT_SYMBOL_GPL(ads117x_dai); -/* - * initialise the ads117x driver - */ -static int ads117x_init(struct snd_soc_device *socdev) -{ - struct snd_soc_codec *codec = socdev->card->codec; - int ret = 0; - - codec->name = "ADS117X"; - codec->owner = THIS_MODULE; - codec->dai = &ads117x_dai; - codec->num_dai = 1; - - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "ads117x: failed to create pcms\n"); - return ret; - } - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ads117x: failed to register card\n"); - goto card_err; - } - return ret; - -card_err: - snd_soc_free_pcms(socdev); - return ret; -} - static int ads117x_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; int ret; - pr_info("ads117x ADC\n"); - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; @@ -85,12 +51,20 @@ static int ads117x_probe(struct platform_device *pdev) mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "ADS117X"; + codec->owner = THIS_MODULE; + codec->dai = &ads117x_dai; + codec->num_dai = 1; - ret = ads117x_init(socdev); - if (ret != 0) + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ads117x: failed to create pcms\n"); kfree(codec); + return ret; + } - return ret; + return 0; } static int ads117x_remove(struct platform_device *pdev) @@ -110,15 +84,37 @@ struct snd_soc_codec_device soc_codec_dev_ads117x = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ads117x); -static int __init ads117x_modinit(void) +static __devinit int ads117x_platform_probe(struct platform_device *pdev) { + ads117x_dai.dev = &pdev->dev; return snd_soc_register_dai(&ads117x_dai); } -module_init(ads117x_modinit); -static void __exit ads117x_exit(void) +static int __devexit ads117x_platform_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&ads117x_dai); + return 0; +} + +static struct platform_driver ads117x_codec_driver = { + .driver = { + .name = "ads117x", + .owner = THIS_MODULE, + }, + + .probe = ads117x_platform_probe, + .remove = __devexit_p(ads117x_platform_remove), +}; + +static int __init ads117x_init(void) +{ + return platform_driver_register(&ads117x_codec_driver); +} +module_init(ads117x_init); + +static void __exit ads117x_exit(void) +{ + platform_driver_unregister(&ads117x_codec_driver); } module_exit(ads117x_exit); -- cgit v1.2.2 From d355c82a0191d5a3e971bd5af96cc81fe3ed25b9 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 3 Nov 2009 15:47:25 +0100 Subject: ALSA: rename "PC Speaker" and "PC Beep" controls to "Beep" To avoid confusion in control names for the standard analog PC Beep generator using a small Internal PC Speaker, rename all related "PC Speaker" and "PC Beep" controls to "Beep" only. This name is more universal and can be also used on more platforms without confusion. Introduce also "Internal Speaker" in ControlNames.txt for systems with full-featured build-in internal speaker. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 3 ++- sound/drivers/pcsp/pcsp_mixer.c | 2 +- sound/isa/cmi8330.c | 4 ++-- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 2 +- sound/isa/sb/sb_mixer.c | 4 ++-- sound/pci/ac97/ac97_codec.c | 6 +++--- sound/pci/ac97/ac97_patch.c | 12 ++++++------ sound/pci/azt3328.c | 4 ++-- sound/pci/ca0106/ca0106_mixer.c | 4 ++-- sound/pci/cmipci.c | 4 ++-- sound/pci/emu10k1/emumixer.c | 4 ++-- sound/pci/es1938.c | 2 +- sound/pci/hda/patch_cmedia.c | 4 ++-- sound/pci/hda/patch_realtek.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 6 +++--- sound/soc/codecs/wm9713.c | 22 +++++++++++----------- 17 files changed, 45 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 772423889eb3..b935ac9dce8d 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1251,7 +1251,8 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer) { SOUND_MIXER_SYNTH, "FM", 0 }, /* fallback */ { SOUND_MIXER_SYNTH, "Music", 0 }, /* fallback */ { SOUND_MIXER_PCM, "PCM", 0 }, - { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, + { SOUND_MIXER_SPEAKER, "Beep", 0 }, + { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, /* fallback */ { SOUND_MIXER_LINE, "Line", 0 }, { SOUND_MIXER_MIC, "Mic", 0 }, { SOUND_MIXER_CD, "CD", 0 }, diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 02e05552632b..6f633f4f3b96 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -125,7 +125,7 @@ static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_pcm[] = { }; static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_spkr[] = { - PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"), + PCSP_MIXER_CONTROL(pcspkr, "Beep Playback Switch"), }; static int __devinit snd_pcsp_ctls_add(struct snd_pcsp *chip, diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 02f79d252718..8246aae32ab4 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -237,7 +237,7 @@ WSS_DOUBLE("Wavetable Capture Volume", 0, CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0), WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), -WSS_SINGLE("PC Speaker Playback Volume", 0, +WSS_SINGLE("Beep Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), WSS_DOUBLE("FM Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), @@ -262,7 +262,7 @@ SB_DOUBLE("SB Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, SB_DOUBLE("SB Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31), SB_SINGLE("SB Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), SB_SINGLE("SB Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), -SB_SINGLE("SB PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3), +SB_SINGLE("SB Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3), SB_DOUBLE("SB Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3), SB_DOUBLE("SB Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3), SB_SINGLE("SB Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1), diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 4c6e14f87f2d..c76bb00c9d15 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -982,7 +982,7 @@ ES1688_DOUBLE("CD Playback Volume", 0, ES1688_CD_DEV, ES1688_CD_DEV, 4, 0, 15, 0 ES1688_DOUBLE("FM Playback Volume", 0, ES1688_FM_DEV, ES1688_FM_DEV, 4, 0, 15, 0), ES1688_DOUBLE("Mic Playback Volume", 0, ES1688_MIC_DEV, ES1688_MIC_DEV, 4, 0, 15, 0), ES1688_DOUBLE("Aux Playback Volume", 0, ES1688_AUX_DEV, ES1688_AUX_DEV, 4, 0, 15, 0), -ES1688_SINGLE("PC Speaker Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0), +ES1688_SINGLE("Beep Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0), ES1688_DOUBLE("Capture Volume", 0, ES1688_RECLEV_DEV, ES1688_RECLEV_DEV, 4, 0, 15, 0), ES1688_SINGLE("Capture Switch", 0, ES1688_REC_DEV, 4, 1, 1), { diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 5cf42b4d65fd..e5bf3355d2ca 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1313,7 +1313,7 @@ ES18XX_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0) * The chipset specific mixer controls */ static struct snd_kcontrol_new snd_es18xx_opt_speaker = - ES18XX_SINGLE("PC Speaker Playback Volume", 0, 0x3c, 0, 7, 0); + ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 475220bbcc96..318ff0c823e7 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -631,7 +631,7 @@ static struct sbmix_elem snd_sb16_ctl_mic_play_switch = static struct sbmix_elem snd_sb16_ctl_mic_play_vol = SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31); static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol = - SB_SINGLE("PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3); + SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3); static struct sbmix_elem snd_sb16_ctl_capture_vol = SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3); static struct sbmix_elem snd_sb16_ctl_play_vol = @@ -689,7 +689,7 @@ static struct sbmix_elem snd_dt019x_ctl_cd_play_vol = static struct sbmix_elem snd_dt019x_ctl_mic_play_vol = SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7); static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol = - SB_SINGLE("PC Speaker Volume", SB_DT019X_SPKR_DEV, 0, 7); + SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7); static struct sbmix_elem snd_dt019x_ctl_line_play_vol = SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15); static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch = diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 78288dbfc17a..20cb60afb200 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -603,8 +603,8 @@ AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_pc_beep[2] = { -AC97_SINGLE("PC Speaker Playback Switch", AC97_PC_BEEP, 15, 1, 1), -AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) +AC97_SINGLE("Beep Playback Switch", AC97_PC_BEEP, 15, 1, 1), +AC97_SINGLE("Beep Playback Volume", AC97_PC_BEEP, 1, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = @@ -1393,7 +1393,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } } - /* build PC Speaker controls */ + /* build Beep controls */ if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) && ((ac97->flags & AC97_HAS_PC_BEEP) || snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) { diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 7337abdbe4e3..139cf3b2b9d7 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -800,12 +800,12 @@ AC97_SINGLE("Mono Switch", AC97_MASTER_TONE, 7, 1, 1), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_TONE, 6, 1, 0), AC97_SINGLE("Mono Volume", AC97_MASTER_TONE, 0, 31, 1), -AC97_SINGLE("PC Beep to Headphone Switch", AC97_AUX, 15, 1, 1), -AC97_SINGLE("PC Beep to Headphone Volume", AC97_AUX, 12, 7, 1), -AC97_SINGLE("PC Beep to Master Switch", AC97_AUX, 11, 1, 1), -AC97_SINGLE("PC Beep to Master Volume", AC97_AUX, 8, 7, 1), -AC97_SINGLE("PC Beep to Mono Switch", AC97_AUX, 7, 1, 1), -AC97_SINGLE("PC Beep to Mono Volume", AC97_AUX, 4, 7, 1), +AC97_SINGLE("Beep to Headphone Switch", AC97_AUX, 15, 1, 1), +AC97_SINGLE("Beep to Headphone Volume", AC97_AUX, 12, 7, 1), +AC97_SINGLE("Beep to Master Switch", AC97_AUX, 11, 1, 1), +AC97_SINGLE("Beep to Master Volume", AC97_AUX, 8, 7, 1), +AC97_SINGLE("Beep to Mono Switch", AC97_AUX, 7, 1, 1), +AC97_SINGLE("Beep to Mono Volume", AC97_AUX, 4, 7, 1), AC97_SINGLE("Voice to Headphone Switch", AC97_PCM, 15, 1, 1), AC97_SINGLE("Voice to Headphone Volume", AC97_PCM, 12, 7, 1), diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 8451a0169f32..69867ace7860 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -830,8 +830,8 @@ static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Mic Boost (+20dB)", IDX_MIXER_MIC, 6, 0), AZF3328_MIXER_SWITCH("Line Playback Switch", IDX_MIXER_LINEIN, 15, 1), AZF3328_MIXER_VOL_STEREO("Line Playback Volume", IDX_MIXER_LINEIN, 0x1f, 1), - AZF3328_MIXER_SWITCH("PC Speaker Playback Switch", IDX_MIXER_PCBEEP, 15, 1), - AZF3328_MIXER_VOL_SPECIAL("PC Speaker Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1), + AZF3328_MIXER_SWITCH("Beep Playback Switch", IDX_MIXER_PCBEEP, 15, 1), + AZF3328_MIXER_VOL_SPECIAL("Beep Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1), AZF3328_MIXER_SWITCH("Video Playback Switch", IDX_MIXER_VIDEO, 15, 1), AZF3328_MIXER_VOL_STEREO("Video Playback Volume", IDX_MIXER_VIDEO, 0x1f, 1), AZF3328_MIXER_SWITCH("Aux Playback Switch", IDX_MIXER_AUX, 15, 1), diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index c8c6f437f5b3..8f443a9d61ec 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -792,8 +792,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) "Phone Playback Volume", "Video Playback Switch", "Video Playback Volume", - "PC Speaker Playback Switch", - "PC Speaker Playback Volume", + "Beep Playback Switch", + "Beep Playback Volume", "Mono Output Select", "Capture Source", "Capture Switch", diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index ddcd4a9fd7e6..a312bae08f52 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2302,7 +2302,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_SB_VOL_MONO("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), CMIPCI_SB_SW_MONO("Mic Playback Switch", 0), CMIPCI_DOUBLE("Mic Capture Switch", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0, 1, 0, 0), - CMIPCI_SB_VOL_MONO("PC Speaker Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3), + CMIPCI_SB_VOL_MONO("Beep Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3), CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15), CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0), CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0), @@ -2310,7 +2310,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7), CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7), CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0), - CMIPCI_DOUBLE("PC Speaker Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), + CMIPCI_DOUBLE("Beep Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0), }; diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index b0fb6c917c38..05afe06e353a 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1818,8 +1818,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Master Playback Switch", "Master Capture Switch", "Master Playback Volume", "Master Capture Volume", "Wave Master Playback Volume", "Master Playback Volume", - "PC Speaker Playback Switch", "PC Speaker Capture Switch", - "PC Speaker Playback Volume", "PC Speaker Capture Volume", + "Beep Playback Switch", "Beep Capture Switch", + "Beep Playback Volume", "Beep Capture Volume", "Phone Playback Switch", "Phone Capture Switch", "Phone Playback Volume", "Phone Capture Volume", "Mic Playback Switch", "Mic Capture Switch", diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 820318ee62c1..fb83e1ffa5cb 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1387,7 +1387,7 @@ ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0, db_scale_line), ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0, db_scale_capture), -ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0), +ES1938_SINGLE("Beep Volume", 0, 0x3c, 0, 7, 0), ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0), ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 780e1a72114a..85c81feb10cf 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -197,8 +197,8 @@ static struct snd_kcontrol_new cmi9880_basic_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x23, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x23, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x23, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x23, 0, HDA_OUTPUT), { } /* end */ }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c08ca660daba..08a5b8a55408 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7334,8 +7334,8 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), /* FIXME: this looks suspicious... - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT), */ { } /* end */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 66c0876bf734..426edfa476a2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3221,7 +3221,7 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, /* check for mute support for the the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "PC Beep Playback Switch", + "Beep Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -3230,7 +3230,7 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, /* check to see if there is volume support for the amp */ if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "PC Beep Playback Volume", + "Beep Playback Volume", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -3271,7 +3271,7 @@ static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { static int stac92xx_beep_switch_ctl(struct hda_codec *codec) { return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl, - 0, "PC Beep Playback Switch", 0); + 0, "Beep Playback Switch", 0); } #endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index abed37acf787..60e360b10468 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -165,9 +165,9 @@ SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), -SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), -SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), -SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), +SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), +SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), +SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), @@ -266,7 +266,7 @@ static int mixer_event(struct snd_soc_dapm_widget *w, /* Left Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0), SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), @@ -276,7 +276,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), /* Right Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0), SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), @@ -294,7 +294,7 @@ SOC_DAPM_ENUM("Route", wm9713_enum[0]); /* Speaker Mixer */ static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), +SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 11, 1, 1), SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), @@ -304,7 +304,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), /* Mono Mixer */ static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), +SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 7, 1, 1), SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), @@ -463,7 +463,7 @@ SND_SOC_DAPM_VMID("VMID"), static const struct snd_soc_dapm_route audio_map[] = { /* left HP mixer */ - {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Left HP Mixer", "Beep Playback Switch", "PCBEEP"}, {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, @@ -472,7 +472,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left HP Mixer", NULL, "Capture Headphone Mux"}, /* right HP mixer */ - {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Right HP Mixer", "Beep Playback Switch", "PCBEEP"}, {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, @@ -491,7 +491,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Capture Mixer", NULL, "Right Capture Source"}, /* speaker mixer */ - {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Speaker Mixer", "Beep Playback Switch", "PCBEEP"}, {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, @@ -499,7 +499,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, /* mono mixer */ - {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Mono Mixer", "Beep Playback Switch", "PCBEEP"}, {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, -- cgit v1.2.2 From ad1cd745060ae2f24026b3b3d09da3426df6ab36 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 4 Nov 2009 14:30:36 +0100 Subject: ALSA: rename "PC Speaker" controls to "Speaker" To unify control names, rename "PC Speaker" to "Speaker" for PPC ALSA drivers. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 1 + sound/ppc/awacs.c | 12 ++++++------ sound/ppc/burgundy.c | 8 ++++---- sound/ppc/tumbler.c | 2 +- 4 files changed, 12 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index b935ac9dce8d..54e2eb56e4c2 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1253,6 +1253,7 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer) { SOUND_MIXER_PCM, "PCM", 0 }, { SOUND_MIXER_SPEAKER, "Beep", 0 }, { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, /* fallback */ + { SOUND_MIXER_SPEAKER, "Speaker", 0 }, /* fallback */ { SOUND_MIXER_LINE, "Line", 0 }, { SOUND_MIXER_MIC, "Mic", 0 }, { SOUND_MIXER_CD, "CD", 0 }, diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 2cc0eda4f20e..2e156467b814 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -479,7 +479,7 @@ static int snd_pmac_awacs_put_master_amp(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Volume", + .name = "Speaker Playback Volume", .info = snd_pmac_awacs_info_volume_amp, .get = snd_pmac_awacs_get_volume_amp, .put = snd_pmac_awacs_put_volume_amp, @@ -525,7 +525,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __devinitdata = { static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Switch", + .name = "Speaker Playback Switch", .info = snd_pmac_boolean_stereo_info, .get = snd_pmac_awacs_get_switch_amp, .put = snd_pmac_awacs_put_switch_amp, @@ -696,17 +696,17 @@ static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __devinitdata }; static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __devinitdata = { - AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1), + AWACS_VOLUME("Speaker Playback Volume", 4, 6, 1), }; static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 1); static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); /* diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 16ed240e423c..0accfe49735b 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -505,7 +505,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __devinitdata = { MASK_ADDR_BURGUNDY_GAINLINE, 1, 0), BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0, MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), - BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + BURGUNDY_VOLUME_B("Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), BURGUNDY_VOLUME_B("Line out Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1), @@ -527,7 +527,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __devinitdata = { MASK_ADDR_BURGUNDY_VOLMIC, 16), BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), - BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + BURGUNDY_VOLUME_B("Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1), BURGUNDY_VOLUME_B("Line out Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), @@ -549,11 +549,11 @@ BURGUNDY_SWITCH_B("Master Playback Switch", 0, BURGUNDY_OUTPUT_INTERN | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __devinitdata = -BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, +BURGUNDY_SWITCH_B("Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __devinitdata = -BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, +BURGUNDY_SWITCH_B("Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_INTERN, 0, 0); static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __devinitdata = diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 08e584d1453a..789f44f4ac78 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -905,7 +905,7 @@ static struct snd_kcontrol_new tumbler_hp_sw __devinitdata = { }; static struct snd_kcontrol_new tumbler_speaker_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Switch", + .name = "Speaker Playback Switch", .info = snd_pmac_boolean_mono_info, .get = tumbler_get_mute_switch, .put = tumbler_put_mute_switch, -- cgit v1.2.2 From d114cd84a1c5ce42bb10cd3a2da57b2bbcef909b Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 5 Nov 2009 18:32:41 +0100 Subject: ALSA: cs4236: detect chip in one pass The cs4236 was two step detection with call to the snd_wss_free() between two steps. The snd_wss_free() did not free a sound device created in the snd_wss_create(). This caused an OOPS during module removal as the same sound device was released twice. The same OOPS happened if the cs4236 module loading failed. Fix this by adapting the snd_cs4236_create() to correctly work with chips less capable then cs4236. The snd_cs4236_create() behaves the same as the snd_wss_create() if the chip is less capable than the cs4236. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/cs423x/cs4236.c | 13 +++-------- sound/isa/cs423x/cs4236_lib.c | 50 +++++++++++++++++++++++++++---------------- sound/isa/wss/wss_lib.c | 3 +-- 3 files changed, 35 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index a076a6ce8071..93fa6720d197 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -394,21 +394,15 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) return -EBUSY; } - err = snd_wss_create(card, port[dev], cport[dev], + err = snd_cs4236_create(card, port[dev], cport[dev], irq[dev], dma1[dev], dma2[dev], WSS_HW_DETECT3, 0, &chip); if (err < 0) return err; + + acard->chip = chip; if (chip->hardware & WSS_HW_CS4236B_MASK) { - snd_wss_free(chip); - err = snd_cs4236_create(card, - port[dev], cport[dev], - irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; err = snd_cs4236_pcm(chip, 0, &pcm); if (err < 0) @@ -418,7 +412,6 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) if (err < 0) return err; } else { - acard->chip = chip; err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) return err; diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 38835f31298b..1b1ad1cad328 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -87,6 +87,7 @@ #include #include #include +#include /* * @@ -264,7 +265,10 @@ static void snd_cs4236_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ - +/* + * This function does no fail if the chip is not CS4236B or compatible. + * It just an equivalent to the snd_wss_create() then. + */ int snd_cs4236_create(struct snd_card *card, unsigned long port, unsigned long cport, @@ -281,21 +285,17 @@ int snd_cs4236_create(struct snd_card *card, *rchip = NULL; if (hardware == WSS_HW_DETECT) hardware = WSS_HW_DETECT3; - if (cport < 0x100) { - snd_printk(KERN_ERR "please, specify control port " - "for CS4236+ chips\n"); - return -ENODEV; - } + err = snd_wss_create(card, port, cport, irq, dma1, dma2, hardware, hwshare, &chip); if (err < 0) return err; - if (!(chip->hardware & WSS_HW_CS4236B_MASK)) { - snd_printk(KERN_ERR "CS4236+: MODE3 and extended registers " - "not available, hardware=0x%x\n", chip->hardware); - snd_device_free(card, chip); - return -ENODEV; + if ((chip->hardware & WSS_HW_CS4236B_MASK) == 0) { + snd_printd("chip is not CS4236+, hardware=0x%x\n", + chip->hardware); + *rchip = chip; + return 0; } #if 0 { @@ -308,9 +308,16 @@ int snd_cs4236_create(struct snd_card *card, idx, snd_cs4236_ctrl_in(chip, idx)); } #endif + if (cport < 0x100 || cport == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please, specify control port " + "for CS4236+ chips\n"); + snd_device_free(card, chip); + return -ENODEV; + } ver1 = snd_cs4236_ctrl_in(chip, 1); ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION); - snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2); + snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", + cport, ver1, ver2); if (ver1 != ver2) { snd_printk(KERN_ERR "CS4236+ chip detected, but " "control port 0x%lx is not valid\n", cport); @@ -321,13 +328,17 @@ int snd_cs4236_create(struct snd_card *card, snd_cs4236_ctrl_out(chip, 2, 0xff); snd_cs4236_ctrl_out(chip, 3, 0x00); snd_cs4236_ctrl_out(chip, 4, 0x80); - snd_cs4236_ctrl_out(chip, 5, ((IEC958_AES1_CON_PCM_CODER & 3) << 6) | IEC958_AES0_CON_EMPHASIS_NONE); + reg = ((IEC958_AES1_CON_PCM_CODER & 3) << 6) | + IEC958_AES0_CON_EMPHASIS_NONE; + snd_cs4236_ctrl_out(chip, 5, reg); snd_cs4236_ctrl_out(chip, 6, IEC958_AES1_CON_PCM_CODER >> 2); snd_cs4236_ctrl_out(chip, 7, 0x00); - /* 0x8c for C8 is valid for Turtle Beach Malibu - the IEC-958 output */ - /* is working with this setup, other hardware should have */ - /* different signal paths and this value should be selectable */ - /* in the future */ + /* + * 0x8c for C8 is valid for Turtle Beach Malibu - the IEC-958 + * output is working with this setup, other hardware should + * have different signal paths and this value should be + * selectable in the future + */ snd_cs4236_ctrl_out(chip, 8, 0x8c); chip->rate_constraint = snd_cs4236_xrate; chip->set_playback_format = snd_cs4236_playback_format; @@ -339,9 +350,10 @@ int snd_cs4236_create(struct snd_card *card, /* initialize extended registers */ for (reg = 0; reg < sizeof(snd_cs4236_ext_map); reg++) - snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), snd_cs4236_ext_map[reg]); + snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), + snd_cs4236_ext_map[reg]); - /* initialize compatible but more featured registers */ + /* initialize compatible but more featured registers */ snd_wss_out(chip, CS4231_LEFT_INPUT, 0x40); snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x40); snd_wss_out(chip, CS4231_AUX1_LEFT_INPUT, 0xff); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 2ba18978b419..705db0924375 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1682,7 +1682,7 @@ static void snd_wss_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ -int snd_wss_free(struct snd_wss *chip) +static int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); @@ -1705,7 +1705,6 @@ int snd_wss_free(struct snd_wss *chip) kfree(chip); return 0; } -EXPORT_SYMBOL(snd_wss_free); static int snd_wss_dev_free(struct snd_device *device) { -- cgit v1.2.2 From 31cef7076ed9409a33f19ea372d6dc5fdefe27ae Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Nov 2009 09:34:16 +0100 Subject: control: remove snd_konctrol_volatile::owner_pid field We do not need to save the ID of the process that locked a control because that information is already available in the owner's file data. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/control.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index a8b7fabe645e..814d2cf1a34c 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -672,7 +672,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, info->access |= SNDRV_CTL_ELEM_ACCESS_LOCK; if (vd->owner == ctl) info->access |= SNDRV_CTL_ELEM_ACCESS_OWNER; - info->owner = vd->owner_pid; + info->owner = vd->owner->pid; } else { info->owner = -1; } @@ -827,7 +827,6 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, result = -EBUSY; else { vd->owner = file; - vd->owner_pid = current->pid; result = 0; } } @@ -858,7 +857,6 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, result = -EPERM; else { vd->owner = NULL; - vd->owner_pid = 0; result = 0; } } -- cgit v1.2.2 From 25d27eded1f4fc728e64f443adc339b5229be5d7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Nov 2009 09:35:44 +0100 Subject: control: use reference-counted pid Instead of storing the PID number, take a reference to the task's pid structure. This protects against duplicates due to PID overflows, and using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is correct as seen from the current namespace. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/control.c | 5 +++-- sound/core/pcm.c | 2 +- sound/core/rawmidi.c | 2 +- 3 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 814d2cf1a34c..73dc10ac33f6 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -75,7 +75,7 @@ static int snd_ctl_open(struct inode *inode, struct file *file) ctl->card = card; ctl->prefer_pcm_subdevice = -1; ctl->prefer_rawmidi_subdevice = -1; - ctl->pid = current->pid; + ctl->pid = get_pid(task_pid(current)); file->private_data = ctl; write_lock_irqsave(&card->ctl_files_rwlock, flags); list_add_tail(&ctl->list, &card->ctl_files); @@ -125,6 +125,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); snd_ctl_empty_read_queue(ctl); + put_pid(ctl->pid); kfree(ctl); module_put(card->module); snd_card_file_remove(card, file); @@ -672,7 +673,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, info->access |= SNDRV_CTL_ELEM_ACCESS_LOCK; if (vd->owner == ctl) info->access |= SNDRV_CTL_ELEM_ACCESS_OWNER; - info->owner = vd->owner->pid; + info->owner = pid_vnr(vd->owner->pid); } else { info->owner = -1; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c69c60b2a48a..8e2c7833614c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -809,7 +809,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, card = pcm->card; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { prefer_subdevice = kctl->prefer_pcm_subdevice; if (prefer_subdevice != -1) break; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c0adc14c91f0..8a81bdafce6e 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -415,7 +415,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) subdevice = -1; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { subdevice = kctl->prefer_rawmidi_subdevice; if (subdevice != -1) break; -- cgit v1.2.2 From 167eae5a17b3cd44a324dbb972c338e489413f54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Nov 2009 15:47:50 +0100 Subject: ALSA: hda - Reset pins of IDT/STAC codecs at free Some laptops cause annoying clicks or noises at shutdown/reboot since the speaker pin is set still high. Apply the same procedure used for the suspend to avoid such clicks/noises for freeing the codec, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 42 ++++++++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8eb6508cd991..3087705a8e51 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4327,6 +4327,28 @@ static void stac92xx_free_kctls(struct hda_codec *codec) snd_array_free(&spec->kctls); } +static void stac92xx_shutup(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + hda_nid_t nid; + + /* reset each pin before powering down DAC/ADC to avoid click noise */ + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wcaps); + if (wid_type == AC_WID_PIN) + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } + + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); +} + static void stac92xx_free(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4334,6 +4356,7 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; + stac92xx_shutup(codec); stac92xx_free_jacks(codec); snd_array_free(&spec->events); @@ -4793,24 +4816,7 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { - struct sigmatel_spec *spec = codec->spec; - int i; - hda_nid_t nid; - - /* reset each pin before powering down DAC/ADC to avoid click noise */ - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - unsigned int wid_type = get_wcaps_type(wcaps); - if (wid_type == AC_WID_PIN) - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } - - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); + stac92xx_shutup(codec); return 0; } #endif -- cgit v1.2.2 From 4cae37fa98f4d50778161ec033122444e3c10a01 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 7 Nov 2009 10:18:22 +0100 Subject: ASoC: Remove dead code and labels Remove the dead code and labels "card_err" in the error paths of some codec drivers. Signed-off-by: Takashi Iwai --- sound/soc/codecs/ad1836.c | 5 ----- sound/soc/codecs/ad1938.c | 5 ----- sound/soc/codecs/cx20442.c | 5 ----- sound/soc/codecs/wm8400.c | 5 ----- sound/soc/codecs/wm8900.c | 5 ----- 5 files changed, 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 2e360c243075..b4be96decf32 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -387,11 +387,6 @@ static int ad1836_probe(struct platform_device *pdev) snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); snd_soc_dapm_new_widgets(codec); - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 09c008ad1476..3b2222a0c808 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -596,11 +596,6 @@ static int ad1938_probe(struct platform_device *pdev) ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d7f9bf18b72e..dda751c885cb 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -355,11 +355,6 @@ static int cx20442_codec_probe(struct platform_device *pdev) cx20442_add_widgets(codec); - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 0e30997c8db0..584af68af22a 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1400,11 +1400,6 @@ static int wm8400_probe(struct platform_device *pdev) wm8400_add_controls(codec); wm8400_add_widgets(codec); - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0d185cb6418d..85f67dbe211d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1353,11 +1353,6 @@ static int wm8900_probe(struct platform_device *pdev) ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } -- cgit v1.2.2 From 8f159d720b89f2a6c5ae8a8cc54823933a58120b Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sat, 7 Nov 2009 01:33:53 -0700 Subject: ASoC/mpc5200: Track DMA position by period number instead of bytes All DMA blocks are lined up to period boundaries, but the DMA handling code tracks bytes instead. This patch reworks the code to track the period index into the DMA buffer instead of the physical address pointer. Doing so makes the code simpler and easier to understand. Signed-off-by: Grant Likely Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 28 +++++++++------------------- sound/soc/fsl/mpc5200_dma.h | 8 ++------ 2 files changed, 11 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 6096d22283e6..986d3c8ab6e1 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -58,13 +58,11 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) /* Prepare and enqueue the next buffer descriptor */ bd = bcom_prepare_next_buffer(s->bcom_task); bd->status = s->period_bytes; - bd->data[0] = s->period_next_pt; + bd->data[0] = s->runtime->dma_addr + (s->period_next * s->period_bytes); bcom_submit_next_buffer(s->bcom_task, NULL); /* Update for next period */ - s->period_next_pt += s->period_bytes; - if (s->period_next_pt >= s->period_end) - s->period_next_pt = s->period_start; + s->period_next = (s->period_next + 1) % s->runtime->periods; } static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) @@ -79,7 +77,7 @@ static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) if (bcom_queue_full(s->bcom_task)) return; - s->appl_ptr += s->period_size; + s->appl_ptr += s->runtime->period_size; psc_dma_bcom_enqueue_next_buffer(s); } @@ -91,7 +89,7 @@ static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) if (bcom_queue_full(s->bcom_task)) return; - s->appl_ptr += s->period_size; + s->appl_ptr += s->runtime->period_size; psc_dma_bcom_enqueue_next_buffer(s); } @@ -108,9 +106,7 @@ static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) while (bcom_buffer_done(s->bcom_task)) { bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - s->period_current_pt += s->period_bytes; - if (s->period_current_pt >= s->period_end) - s->period_current_pt = s->period_start; + s->period_current = (s->period_current+1) % s->runtime->periods; } psc_dma_bcom_enqueue_tx(s); spin_unlock(&s->psc_dma->lock); @@ -133,9 +129,7 @@ static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) while (bcom_buffer_done(s->bcom_task)) { bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - s->period_current_pt += s->period_bytes; - if (s->period_current_pt >= s->period_end) - s->period_current_pt = s->period_start; + s->period_current = (s->period_current+1) % s->runtime->periods; psc_dma_bcom_enqueue_next_buffer(s); } @@ -185,12 +179,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: s->period_bytes = frames_to_bytes(runtime, runtime->period_size); - s->period_start = virt_to_phys(runtime->dma_area); - s->period_end = s->period_start + - (s->period_bytes * runtime->periods); - s->period_next_pt = s->period_start; - s->period_current_pt = s->period_start; - s->period_size = runtime->period_size; + s->period_next = 0; + s->period_current = 0; s->active = 1; /* track appl_ptr so that we have a better chance of detecting @@ -343,7 +333,7 @@ psc_dma_pointer(struct snd_pcm_substream *substream) else s = &psc_dma->playback; - count = s->period_current_pt - s->period_start; + count = s->period_current * s->period_bytes; return bytes_to_frames(substream->runtime, count); } diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 8d396bb9d9fe..529f7a094479 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -13,7 +13,6 @@ * @psc_dma: pointer back to parent psc_dma data structure * @bcom_task: bestcomm task structure * @irq: irq number for bestcomm task - * @period_start: physical address of start of DMA region * @period_end: physical address of end of DMA region * @period_next_pt: physical address of next DMA buffer to enqueue * @period_bytes: size of DMA period in bytes @@ -27,12 +26,9 @@ struct psc_dma_stream { struct bcom_task *bcom_task; int irq; struct snd_pcm_substream *stream; - dma_addr_t period_start; - dma_addr_t period_end; - dma_addr_t period_next_pt; - dma_addr_t period_current_pt; + int period_next; + int period_current; int period_bytes; - int period_size; }; /** -- cgit v1.2.2 From d56b6eb6df7f6fb92383a52d640e27f71e6262d0 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sat, 7 Nov 2009 01:34:05 -0700 Subject: ASoC/mpc5200: get rid of the appl_ptr tracking nonsense Sound drivers PCM DMA is supposed to free-run until told to stop by the trigger callback. The current code tries to track appl_ptr, to avoid stale buffer data getting played out at the end of the data stream. Unfortunately it also results in race conditions which can cause the audio to stall. Signed-off-by: Grant Likely Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 52 +++++++-------------------------------------- sound/soc/fsl/mpc5200_dma.h | 2 -- 2 files changed, 8 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 986d3c8ab6e1..4e475861f5db 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -65,36 +65,6 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) s->period_next = (s->period_next + 1) % s->runtime->periods; } -static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) -{ - if (s->appl_ptr > s->runtime->control->appl_ptr) { - /* - * In this case s->runtime->control->appl_ptr has wrapped around. - * Play the data to the end of the boundary, then wrap our own - * appl_ptr back around. - */ - while (s->appl_ptr < s->runtime->boundary) { - if (bcom_queue_full(s->bcom_task)) - return; - - s->appl_ptr += s->runtime->period_size; - - psc_dma_bcom_enqueue_next_buffer(s); - } - s->appl_ptr -= s->runtime->boundary; - } - - while (s->appl_ptr < s->runtime->control->appl_ptr) { - - if (bcom_queue_full(s->bcom_task)) - return; - - s->appl_ptr += s->runtime->period_size; - - psc_dma_bcom_enqueue_next_buffer(s); - } -} - /* Bestcomm DMA irq handler */ static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) { @@ -107,8 +77,9 @@ static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) bcom_retrieve_buffer(s->bcom_task, NULL, NULL); s->period_current = (s->period_current+1) % s->runtime->periods; + + psc_dma_bcom_enqueue_next_buffer(s); } - psc_dma_bcom_enqueue_tx(s); spin_unlock(&s->psc_dma->lock); /* If the stream is active, then also inform the PCM middle layer @@ -182,28 +153,21 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) s->period_next = 0; s->period_current = 0; s->active = 1; - - /* track appl_ptr so that we have a better chance of detecting - * end of stream and not over running it. - */ s->runtime = runtime; - s->appl_ptr = s->runtime->control->appl_ptr - - (runtime->period_size * runtime->periods); /* Fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. */ spin_lock_irqsave(&psc_dma->lock, flags); - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) bcom_gen_bd_rx_reset(s->bcom_task); - for (i = 0; i < runtime->periods; i++) - if (!bcom_queue_full(s->bcom_task)) - psc_dma_bcom_enqueue_next_buffer(s); - } else { + else bcom_gen_bd_tx_reset(s->bcom_task); - psc_dma_bcom_enqueue_tx(s); - } + + for (i = 0; i < runtime->periods; i++) + if (!bcom_queue_full(s->bcom_task)) + psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); spin_unlock_irqrestore(&psc_dma->lock, flags); diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 529f7a094479..d9c741bf9ab6 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -19,8 +19,6 @@ */ struct psc_dma_stream { struct snd_pcm_runtime *runtime; - snd_pcm_uframes_t appl_ptr; - int active; struct psc_dma *psc_dma; struct bcom_task *bcom_task; -- cgit v1.2.2 From c4878274750ae0bb90c351a737ac6cdcb608e546 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sat, 7 Nov 2009 01:34:18 -0700 Subject: ASoC/mpc5200: Improve printk debug output for trigger Signed-off-by: Grant Likely Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 15 ++++++++++----- sound/soc/fsl/mpc5200_dma.h | 1 + 2 files changed, 11 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 4e475861f5db..658e3fa14663 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -77,6 +77,7 @@ static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) bcom_retrieve_buffer(s->bcom_task, NULL, NULL); s->period_current = (s->period_current+1) % s->runtime->periods; + s->period_count++; psc_dma_bcom_enqueue_next_buffer(s); } @@ -101,6 +102,7 @@ static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) bcom_retrieve_buffer(s->bcom_task, NULL, NULL); s->period_current = (s->period_current+1) % s->runtime->periods; + s->period_count++; psc_dma_bcom_enqueue_next_buffer(s); } @@ -142,17 +144,17 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) else s = &psc_dma->playback; - dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)" - " stream_id=%i\n", - substream, cmd, substream->pstr->stream); - switch (cmd) { case SNDRV_PCM_TRIGGER_START: + dev_dbg(psc_dma->dev, "START: stream=%i fbits=%u ps=%u #p=%u\n", + substream->pstr->stream, runtime->frame_bits, + (int)runtime->period_size, runtime->periods); s->period_bytes = frames_to_bytes(runtime, runtime->period_size); s->period_next = 0; s->period_current = 0; s->active = 1; + s->period_count = 0; s->runtime = runtime; /* Fill up the bestcomm bd queue and enable DMA. @@ -177,6 +179,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: + dev_dbg(psc_dma->dev, "STOP: stream=%i periods_count=%i\n", + substream->pstr->stream, s->period_count); s->active = 0; spin_lock_irqsave(&psc_dma->lock, flags); @@ -190,7 +194,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) break; default: - dev_dbg(psc_dma->dev, "invalid command\n"); + dev_dbg(psc_dma->dev, "unhandled trigger: stream=%i cmd=%i\n", + substream->pstr->stream, cmd); return -EINVAL; } diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index d9c741bf9ab6..c6f29e4d093c 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -27,6 +27,7 @@ struct psc_dma_stream { int period_next; int period_current; int period_bytes; + int period_count; }; /** -- cgit v1.2.2 From 1d8222e8df07ce4f86fb7fa80b02bdee03b57985 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sat, 7 Nov 2009 01:34:31 -0700 Subject: ASoC/mpc5200: add to_psc_dma_stream() helper Move the resolving of the psc_dma_stream pointer to a helper function to reduce duplicate code Signed-off-by: Grant Likely Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 7 +------ sound/soc/fsl/mpc5200_dma.h | 9 +++++++++ 2 files changed, 10 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 658e3fa14663..9c88e15ce693 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -133,17 +133,12 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct psc_dma_stream *s; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 imr; unsigned long flags; int i; - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_dma->capture; - else - s = &psc_dma->playback; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: dev_dbg(psc_dma->dev, "START: stream=%i fbits=%u ps=%u #p=%u\n", diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index c6f29e4d093c..956d6a5f5a8c 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -68,6 +68,15 @@ struct psc_dma { } stats; }; +/* Utility for retrieving psc_dma_stream structure from a substream */ +inline struct psc_dma_stream * +to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) +{ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + return &psc_dma->capture; + return &psc_dma->playback; +} + int mpc5200_audio_dma_create(struct of_device *op); int mpc5200_audio_dma_destroy(struct of_device *op); -- cgit v1.2.2 From c939e5c82142978d9d534aca34187a8489fd13f3 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sat, 7 Nov 2009 01:34:43 -0700 Subject: ASoC/mpc5200: fix enable/disable of AC97 slots The MPC5200 AC97 driver is disabling the slots when a stop trigger is received, but not reenabling them if the stream is started again without processing the hw_params again. This patch fixes the problem by caching the slot enable bit settings calculated at hw_params time so that they can be reapplied every time the start trigger is received. Signed-off-by: Grant Likely Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.h | 4 ++++ sound/soc/fsl/mpc5200_psc_ac97.c | 39 +++++++++++++++++++++------------------ 2 files changed, 25 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 956d6a5f5a8c..22208b373fb9 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -16,6 +16,7 @@ * @period_end: physical address of end of DMA region * @period_next_pt: physical address of next DMA buffer to enqueue * @period_bytes: size of DMA period in bytes + * @ac97_slot_bits: Enable bits for turning on the correct AC97 slot */ struct psc_dma_stream { struct snd_pcm_runtime *runtime; @@ -28,6 +29,9 @@ struct psc_dma_stream { int period_current; int period_bytes; int period_count; + + /* AC97 state */ + u32 ac97_slot_bits; }; /** diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index c4ae3e096bb9..3dbc7f7cd7b9 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -130,6 +130,7 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct psc_dma *psc_dma = cpu_dai->private_data; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" @@ -140,20 +141,10 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, params_channels(params), params_rate(params), params_format(params)); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (params_channels(params) == 1) - psc_dma->slots |= 0x00000100; - else - psc_dma->slots |= 0x00000300; - } else { - if (params_channels(params) == 1) - psc_dma->slots |= 0x01000000; - else - psc_dma->slots |= 0x03000000; - } - out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); - + /* Determine the set of enable bits to turn on */ + s->ac97_slot_bits = (params_channels(params) == 1) ? 0x100 : 0x300; + if (substream->pstr->stream != SNDRV_PCM_STREAM_CAPTURE) + s->ac97_slot_bits <<= 16; return 0; } @@ -163,6 +154,8 @@ static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, { struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "%s(substream=%p)\n", __func__, substream); + if (params_channels(params) == 1) out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); else @@ -176,14 +169,24 @@ static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dev_dbg(psc_dma->dev, "AC97 START: stream=%i\n", + substream->pstr->stream); + + /* Set the slot enable bits */ + psc_dma->slots |= s->ac97_slot_bits; + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + break; + case SNDRV_PCM_TRIGGER_STOP: - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - psc_dma->slots &= 0xFFFF0000; - else - psc_dma->slots &= 0x0000FFFF; + dev_dbg(psc_dma->dev, "AC97 STOP: stream=%i\n", + substream->pstr->stream); + /* Clear the slot enable bits */ + psc_dma->slots &= ~(s->ac97_slot_bits); out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); break; } -- cgit v1.2.2 From faa1242c59311525b0f337e95ae3c324a833a8eb Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 8 Nov 2009 11:58:08 +0100 Subject: ALSA: es18xx: code improvements 1. Set the third argument of the snd_device_new to not NULL, so there is no warning about bug during chip detection. The third argument is not used in this driver. It was changed in my previous patch. 2. Remove the fm_port and mpu_port fields from the snd_es18xx structure. They can be converted to function arguments. 3. Remove the dmaN_size fields from the snd_es18xx structure. These values are used only in pointer functions and can be easily calculated. 4. Remove the ctrl_lock spinlock which is used only in one read function which is called once during chip initialization. There are many writes to the same register and they are not protected on purpose (see the comment ina the snd_es18xx_config_write()). 5. Use the first part of the text5Sources string table as the text4Soruces table (they are the same). 6. Merge the same cases for the ES1887 and ES1888 when setting chip's caps. 7. Move the snd_es18xx_reset() to __devinit section. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 91 ++++++++++++++++++++++++------------------------------ 1 file changed, 41 insertions(+), 50 deletions(-) (limited to 'sound') diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 5cf42b4d65fd..06e871e66c97 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -102,8 +102,6 @@ struct snd_es18xx { unsigned long port; /* port of ESS chip */ - unsigned long mpu_port; /* MPU-401 port of ESS chip */ - unsigned long fm_port; /* FM port */ unsigned long ctrl_port; /* Control port of ESS chip */ struct resource *res_port; struct resource *res_mpu_port; @@ -116,8 +114,6 @@ struct snd_es18xx { unsigned short audio2_vol; /* volume level of audio2 */ unsigned short active; /* active channel mask */ - unsigned int dma1_size; - unsigned int dma2_size; unsigned int dma1_shift; unsigned int dma2_shift; @@ -135,7 +131,6 @@ struct snd_es18xx { spinlock_t reg_lock; spinlock_t mixer_lock; - spinlock_t ctrl_lock; #ifdef CONFIG_PM unsigned char pm_reg; #endif @@ -354,7 +349,7 @@ static inline int snd_es18xx_mixer_writable(struct snd_es18xx *chip, unsigned ch } -static int snd_es18xx_reset(struct snd_es18xx *chip) +static int __devinit snd_es18xx_reset(struct snd_es18xx *chip) { int i; outb(0x03, chip->port + 0x06); @@ -490,8 +485,6 @@ static int snd_es18xx_playback1_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma2_size = size; - snd_es18xx_rate_set(chip, substream, DAC2); /* Transfer Count Reload */ @@ -591,8 +584,6 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -659,8 +650,6 @@ static int snd_es18xx_playback2_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -821,17 +810,18 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) { if (!(chip->active & DAC2)) return 0; - pos = snd_dma_pointer(chip->dma2, chip->dma2_size); + pos = snd_dma_pointer(chip->dma2, size); return pos >> chip->dma2_shift; } else { if (!(chip->active & DAC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } } @@ -839,11 +829,12 @@ static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *s static snd_pcm_uframes_t snd_es18xx_capture_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (!(chip->active & ADC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } @@ -974,9 +965,6 @@ static int snd_es18xx_capture_close(struct snd_pcm_substream *substream) static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts4Source[4] = { - "Mic", "CD", "Line", "Master" - }; static char *texts5Source[5] = { "Mic", "CD", "Line", "Master", "Mix" }; @@ -994,7 +982,8 @@ static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_ele uinfo->value.enumerated.items = 4; if (uinfo->value.enumerated.item > 3) uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts4Source[uinfo->value.enumerated.item]); + strcpy(uinfo->value.enumerated.name, + texts5Source[uinfo->value.enumerated.item]); break; case 0x1887: case 0x1888: @@ -1378,11 +1367,9 @@ ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), static int __devinit snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; - unsigned long flags; - spin_lock_irqsave(&chip->ctrl_lock, flags); + outb(reg, chip->ctrl_port); data = inb(chip->ctrl_port + 1); - spin_unlock_irqrestore(&chip->ctrl_lock, flags); return data; } @@ -1398,7 +1385,9 @@ static void __devinit snd_es18xx_config_write(struct snd_es18xx *chip, #endif } -static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) +static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { int mask = 0; @@ -1412,15 +1401,15 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) if (chip->caps & ES18XX_CONTROL) { /* Hardware volume IRQ */ snd_es18xx_config_write(chip, 0x27, chip->irq); - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { /* FM I/O */ - snd_es18xx_config_write(chip, 0x62, chip->fm_port >> 8); - snd_es18xx_config_write(chip, 0x63, chip->fm_port & 0xff); + snd_es18xx_config_write(chip, 0x62, fm_port >> 8); + snd_es18xx_config_write(chip, 0x63, fm_port & 0xff); } - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU-401 I/O */ - snd_es18xx_config_write(chip, 0x64, chip->mpu_port >> 8); - snd_es18xx_config_write(chip, 0x65, chip->mpu_port & 0xff); + snd_es18xx_config_write(chip, 0x64, mpu_port >> 8); + snd_es18xx_config_write(chip, 0x65, mpu_port & 0xff); /* MPU-401 IRQ */ snd_es18xx_config_write(chip, 0x28, chip->irq); } @@ -1507,11 +1496,12 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) snd_es18xx_mixer_write(chip, 0x7A, 0x68); /* Enable and set hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x06); - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU401 share irq with audio Joystick enabled FM enabled */ - snd_es18xx_mixer_write(chip, 0x40, 0x43 | (chip->mpu_port & 0xf0) >> 1); + snd_es18xx_mixer_write(chip, 0x40, + 0x43 | (mpu_port & 0xf0) >> 1); } snd_es18xx_mixer_write(chip, 0x7f, ((irqmask + 1) << 1) | 0x01); } @@ -1629,7 +1619,9 @@ static int __devinit snd_es18xx_identify(struct snd_es18xx *chip) return 0; } -static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) +static int __devinit snd_es18xx_probe(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { if (snd_es18xx_identify(chip) < 0) { snd_printk(KERN_ERR PFX "[0x%lx] ESS chip not found\n", chip->port); @@ -1650,8 +1642,6 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_I2S | ES18XX_CONTROL | ES18XX_HWV; break; case 0x1887: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; - break; case 0x1888: chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; break; @@ -1666,7 +1656,7 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) if (chip->dma1 == chip->dma2) chip->caps &= ~(ES18XX_PCM2 | ES18XX_DUPLEX_SAME); - return snd_es18xx_initialize(chip); + return snd_es18xx_initialize(chip, mpu_port, fm_port); } static struct snd_pcm_ops snd_es18xx_playback_ops = { @@ -1802,10 +1792,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - spin_lock_init(&chip->ctrl_lock); chip->port = port; - chip->mpu_port = mpu_port; - chip->fm_port = fm_port; chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; @@ -1841,11 +1828,11 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, } chip->dma2 = dma2; - if (snd_es18xx_probe(chip) < 0) { + if (snd_es18xx_probe(chip, mpu_port, fm_port) < 0) { snd_es18xx_free(card); - return -ENODEV; - } - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, NULL, &ops); + return -ENODEV; + } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_es18xx_free(card); return err; @@ -1980,7 +1967,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ #ifdef CONFIG_PNP -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; #endif static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */ #ifndef CONFIG_PNP @@ -2160,19 +2147,23 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) return err; if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, chip->fm_port, chip->fm_port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) { - snd_printk(KERN_WARNING PFX "opl3 not detected at 0x%lx\n", chip->fm_port); + if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) { + snd_printk(KERN_WARNING PFX + "opl3 not detected at 0x%lx\n", + fm_port[dev]); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) return err; } } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, - chip->mpu_port, 0, - irq[dev], 0, - &chip->rmidi)) < 0) + err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, + mpu_port[dev], 0, + irq[dev], 0, &chip->rmidi); + if (err < 0) return err; } -- cgit v1.2.2 From 7c5af6ffd69bb2bb3c86b374153627529d67598c Mon Sep 17 00:00:00 2001 From: Dominik Brodowski Date: Sat, 24 Oct 2009 15:55:12 +0200 Subject: pcmcia: use dynamic debug infrastructure, deprecate CS_CHECK (sound) Convert PCMCIA drivers to use the dynamic debug infrastructure, instead of requiring manual settings of PCMCIA_DEBUG. Also, remove all usages of the CS_CHECK macro and replace them with proper Linux style calling and return value checking. The extra error reporting may be dropped, as the PCMCIA core already complains about any (non-driver-author) errors. CC: Jaroslav Kysela CC: alsa-devel@alsa-project.org Signed-off-by: Dominik Brodowski --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 21 ++++++++++++--------- sound/pcmcia/vx/vxpocket.c | 21 ++++++++++++--------- 2 files changed, 24 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 7dea74b71cf1..64b859925c0b 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -217,20 +217,25 @@ static void snd_pdacf_detach(struct pcmcia_device *link) * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int pdacf_config(struct pcmcia_device *link) { struct snd_pdacf *pdacf = link->priv; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "pdacf_config called\n"); link->conf.ConfigIndex = 0x5; - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0) goto failed; @@ -238,8 +243,6 @@ static int pdacf_config(struct pcmcia_device *link) link->dev_node = &pdacf->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7445cc8a47d3..1492744ad67f 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -213,14 +213,11 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int vxpocket_config(struct pcmcia_device *link) { struct vx_core *chip = link->priv; struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "vxpocket_config called\n"); @@ -235,9 +232,17 @@ static int vxpocket_config(struct pcmcia_device *link) strcpy(chip->card->driver, vxp440_hw.name); } - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; chip->dev = &handle_to_dev(link); snd_card_set_dev(chip->card, chip->dev); @@ -248,8 +253,6 @@ static int vxpocket_config(struct pcmcia_device *link) link->dev_node = &vxp->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; -- cgit v1.2.2 From 06fe9fb4182177fb046e6d934f80254dd90956ea Mon Sep 17 00:00:00 2001 From: Dirk Hohndel Date: Mon, 28 Sep 2009 21:43:57 -0400 Subject: tree-wide: fix a very frequent spelling mistake something-bility is spelled as something-blity so a grep for 'blit' would find these lines this is so trivial that I didn't split it by subsystem / copy additional maintainers - all changes are to comments The only purpose is to get fewer false positives when grepping around the kernel sources. Signed-off-by: Dirk Hohndel Signed-off-by: Jiri Kosina --- sound/pci/ice1712/juli.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index fd948bfd9aef..9c0f78ea2c41 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -380,7 +380,7 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = { * inputs) are fed from Xilinx. * * I even checked traces on board and coded a support in driver for - * an alternative possiblity - the unused I2S ICE output channels + * an alternative possibility - the unused I2S ICE output channels * switched to HW-IN/SPDIF-IN and providing the monitoring signal to * the DAC - to no avail. The I2S outputs seem to be unconnected. * -- cgit v1.2.2 From fa3012318bfb395552baef69bb1ebe87e64945c8 Mon Sep 17 00:00:00 2001 From: Michael Roth Date: Sun, 4 Oct 2009 18:14:29 +0200 Subject: Kconfig: Remove useless and sometimes wrong comments Additionally, some excessive newlines removed. Signed-off-by: Michael Roth Signed-off-by: Jiri Kosina --- sound/Kconfig | 4 ---- sound/oss/Kconfig | 2 -- 2 files changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/Kconfig b/sound/Kconfig index 439e15c8faa3..4b5365ad6b46 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,6 +1,3 @@ -# sound/Config.in -# - menuconfig SOUND tristate "Sound card support" depends on HAS_IOMEM @@ -136,4 +133,3 @@ config AC97_BUS sound subsystem and other function drivers completely unrelated to sound although they're sharing the AC97 bus. Concerned drivers should "select" this. - diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index bcf2a0698d54..ea0b1aeffe66 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -1,5 +1,3 @@ -# drivers/sound/Config.in -# # 18 Apr 1998, Michael Elizabeth Chastain, # More hacking for modularisation. # -- cgit v1.2.2 From b71a8eb0fa64ec6d00175f479e3ef851703568af Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 6 Oct 2009 12:42:51 +0200 Subject: tree-wide: fix typos "selct" + "slect" -> "select" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch was generated by git grep -E -i -l 's(le|el)ct' | xargs -r perl -p -i -e 's/([Ss])(le|el)ct/$1elect/ with only skipping net/netfilter/xt_SECMARK.c and include/linux/netfilter/xt_SECMARK.h which have a struct member called selctx. Signed-off-by: Uwe Kleine-König Signed-off-by: Jiri Kosina --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 8ba306856d38..7b0446fa6009 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -947,7 +947,7 @@ static void init_input(struct hda_codec *codec) coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ if (is_active_pin(codec, CS_DMIC1_PIN_NID)) coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 - * No effect if SPDIF_OUT2 is slected in + * No effect if SPDIF_OUT2 is selected in * IDX_SPDIF_CTL. */ cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); -- cgit v1.2.2 From 9e5d86fe6a401f7957f6ea02ee300db0f6c03d03 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 9 Nov 2009 08:44:32 +0200 Subject: ASoC: Pandora: Pass SRG input clock frequency to the OMAP McBSP DAI Upcoming change to omap-mcbsp.c require that machine drivers using OMAP as a DAI master to pass sample rate generator input clock frequency to the omap-mcbsp.c DAI driver. Pandora is using 256*Fs output from the TWL4030 codec as an input clock to the McBSP sample rate generator. Signed-off-by: Jarkko Nikula Tested-by: Grazvydas Ignotas Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 24 ++++++++++-------------- 1 file changed, 10 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index ad219aaf7cb8..cace5f13792d 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -40,9 +40,12 @@ #define PREFIX "ASoC omap3pandora: " -static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai, unsigned int fmt) +static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, unsigned int fmt) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set codec DAI configuration */ @@ -68,8 +71,9 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, } /* Set McBSP clock to external */ - ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, + 256 * params_rate(params), + SND_SOC_CLOCK_IN); if (ret < 0) { pr_err(PREFIX "can't set cpu system clock\n"); return ret; @@ -87,11 +91,7 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -100,11 +100,7 @@ static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); -- cgit v1.2.2 From 5f63ef9909c187581c7f2c28fbc93866a0d59f7f Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Mon, 9 Nov 2009 19:02:15 +0000 Subject: ASoC: omap-mcbsp - add support for upto 16 channels. This patch increases the number of supported audio channels from 4 to 16 and has been sponsored by Shotspotter Inc. It also fixes a FSYNC rate calculation bug when McBSP is FSYNC master. Signed-off-by: Graeme Gregory Signed-off-by: Liam Girdwood Acked-by: Peter Ujfalusi Tested-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 63 ++++++++++++++++++++++++++------------------- 1 file changed, 37 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3341f49402ca..45be94201c89 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -49,6 +49,8 @@ struct omap_mcbsp_data { */ int active; int configured; + unsigned int in_freq; + int clk_div; }; #define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id) @@ -257,7 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; unsigned long port; - unsigned int format; + unsigned int format, div, framesize, master; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; @@ -294,28 +296,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - switch (channels) { - case 2: - if (format == SND_SOC_DAIFMT_I2S) { - /* Use dual-phase frames */ - regs->rcr2 |= RPHASE; - regs->xcr2 |= XPHASE; - /* Set 1 word per (McBSP) frame for phase1 and phase2 */ - wpf--; - regs->rcr2 |= RFRLEN2(wpf - 1); - regs->xcr2 |= XFRLEN2(wpf - 1); - } - case 1: - case 4: - /* Set word per (McBSP) frame for phase1 */ - regs->rcr1 |= RFRLEN1(wpf - 1); - regs->xcr1 |= XFRLEN1(wpf - 1); - break; - default: - /* Unsupported number of channels */ - return -EINVAL; + if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + /* Set 1 word per (McBSP) frame for phase1 and phase2 */ + wpf--; + regs->rcr2 |= RFRLEN2(wpf - 1); + regs->xcr2 |= XFRLEN2(wpf - 1); } + regs->rcr1 |= RFRLEN1(wpf - 1); + regs->xcr1 |= XFRLEN1(wpf - 1); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ @@ -330,15 +323,30 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* In McBSP master modes, FRAME (i.e. sample rate) is generated + * by _counting_ BCLKs. Calculate frame size in BCLKs */ + master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (master == SND_SOC_DAIFMT_CBS_CFS) { + div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1; + framesize = (mcbsp_data->in_freq / div) / params_rate(params); + + if (framesize < wlen * channels) { + printk(KERN_ERR "%s: not enough bandwidth for desired rate and " + "channels\n", __func__); + return -EINVAL; + } + } else + framesize = wlen * channels; + /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: - regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen - 1); + regs->srgr2 |= FPER(framesize - 1); + regs->srgr1 |= FWID((framesize >> 1) - 1); break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - regs->srgr2 |= FPER(wlen * channels - 1); + regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID(0); break; } @@ -454,6 +462,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, if (div_id != OMAP_MCBSP_CLKGDV) return -ENODEV; + mcbsp_data->clk_div = div; regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -554,6 +563,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + mcbsp_data->in_freq = freq; + switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: regs->srgr2 |= CLKSM; @@ -598,13 +609,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .id = (link_id), \ .playback = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ .capture = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ -- cgit v1.2.2 From 68d019553b8cc4ddac7f861e23efbe48a1367490 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 4 Nov 2009 09:58:20 +0200 Subject: ASoC: TWL4030: Do not modify the APLL_CTL register APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 76 +++++++++++++++++++--------------------------- 1 file changed, 31 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 928257b25111..510b8b226f96 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -214,7 +214,8 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) - twl4030_write(codec, i, cache[i]); + if (i != TWL4030_REG_APLL_CTL) + twl4030_write(codec, i, cache[i]); } @@ -1753,30 +1754,23 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = codec->private_data; - u8 apll_ctrl; - apll_ctrl = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - apll_ctrl &= ~TWL4030_APLL_INFREQ; switch (freq) { case 19200000: - apll_ctrl |= TWL4030_APLL_INFREQ_19200KHZ; - twl4030->sysclk = 19200; - break; case 26000000: - apll_ctrl |= TWL4030_APLL_INFREQ_26000KHZ; - twl4030->sysclk = 26000; - break; case 38400000: - apll_ctrl |= TWL4030_APLL_INFREQ_38400KHZ; - twl4030->sysclk = 38400; break; default: - printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", - freq); + dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq); return -EINVAL; } - twl4030_write(codec, TWL4030_REG_APLL_CTL, apll_ctrl); + if ((freq / 1000) != twl4030->sysclk) { + dev_err(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + freq, twl4030->sysclk * 1000); + return -EINVAL; + } return 0; } @@ -1874,18 +1868,16 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u8 infreq; + struct twl4030_priv *twl4030 = codec->private_data; u8 mode; /* If the system master clock is not 26MHz, the voice PCM interface is * not avilable. */ - infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) - & TWL4030_APLL_INFREQ; - - if (infreq != TWL4030_APLL_INFREQ_26000KHZ) { - printk(KERN_ERR "TWL4030 voice startup: " - "MCLK is not 26MHz, call set_sysclk() on init\n"); + if (twl4030->sysclk != 26000) { + dev_err(codec->dev, "The board is configured for %u Hz, while" + "the Voice interface needs 26MHz APLL mclk\n", + twl4030->sysclk * 1000); return -EINVAL; } @@ -1958,22 +1950,19 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - u8 apll_ctrl; + struct twl4030_priv *twl4030 = codec->private_data; - apll_ctrl = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - apll_ctrl &= ~TWL4030_APLL_INFREQ; - switch (freq) { - case 26000000: - apll_ctrl |= TWL4030_APLL_INFREQ_26000KHZ; - break; - default: - printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", - freq); + if (freq != 26000000) { + dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice" + "interface needs 26MHz APLL mclk\n", freq); + return -EINVAL; + } + if ((freq / 1000) != twl4030->sysclk) { + dev_err(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + freq, twl4030->sysclk * 1000); return -EINVAL; } - - twl4030_write(codec, TWL4030_REG_APLL_CTL, apll_ctrl); - return 0; } @@ -2131,17 +2120,15 @@ static int twl4030_soc_probe(struct platform_device *pdev) if (setup) { unsigned char hs_pop; - if (setup->sysclk) - twl4030->sysclk = setup->sysclk; - else - twl4030->sysclk = 26000; + if (setup->sysclk != twl4030->sysclk) + dev_warn(&pdev->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + setup->sysclk, twl4030->sysclk); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); hs_pop &= ~TWL4030_RAMP_DELAY; hs_pop |= (setup->ramp_delay_value << 2); twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - } else { - twl4030->sysclk = 26000; } /* register pcms */ @@ -2179,10 +2166,8 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) struct twl4030_priv *twl4030; int ret; - if (!pdata || !(pdata->audio_mclk == 19200000 || - pdata->audio_mclk == 26000000 || - pdata->audio_mclk == 38400000)) { - dev_err(&pdev->dev, "Invalid platform_data\n"); + if (!pdata) { + dev_err(&pdev->dev, "platform_data is missing\n"); return -EINVAL; } @@ -2221,6 +2206,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) twl4030_codec = codec; /* Set the defaults, and power up the codec */ + twl4030->sysclk = twl4030_codec_get_mclk() / 1000; twl4030_init_chip(codec); codec->bias_level = SND_SOC_BIAS_OFF; twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.2 From a68cc8daebdd8ba7fe457ab4b2a0ccdf3cedc9f8 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Mon, 9 Nov 2009 09:40:09 -0700 Subject: ASoC: mpc5200: remove duplicate identical IRQ handler The TX and RX irq handlers are identical. Merge them Signed-off-by: Grant Likely Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 33 +++------------------------------ 1 file changed, 3 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9c88e15ce693..30ed568afb2e 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -66,32 +66,7 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) } /* Bestcomm DMA irq handler */ -static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) -{ - struct psc_dma_stream *s = _psc_dma_stream; - - spin_lock(&s->psc_dma->lock); - /* For each finished period, dequeue the completed period buffer - * and enqueue a new one in it's place. */ - while (bcom_buffer_done(s->bcom_task)) { - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - - s->period_current = (s->period_current+1) % s->runtime->periods; - s->period_count++; - - psc_dma_bcom_enqueue_next_buffer(s); - } - spin_unlock(&s->psc_dma->lock); - - /* If the stream is active, then also inform the PCM middle layer - * of the period finished event. */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - return IRQ_HANDLED; -} - -static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) +static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream; @@ -486,11 +461,9 @@ int mpc5200_audio_dma_create(struct of_device *op) rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, "psc-dma-status", psc_dma); - rc |= request_irq(psc_dma->capture.irq, - &psc_dma_bcom_irq_rx, IRQF_SHARED, + rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-capture", &psc_dma->capture); - rc |= request_irq(psc_dma->playback.irq, - &psc_dma_bcom_irq_tx, IRQF_SHARED, + rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { ret = -ENODEV; -- cgit v1.2.2 From fb8d1a344dbe963f16249d07eee8415e93f9f3c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Nov 2009 16:02:29 +0100 Subject: ALSA: hda - Add reboot notifier to each codec Add reboot notifier to each codec so that it can do some workarounds needed for reboot. So far, patch_sigmatel.c calls its shutup routine for avoiding noises at reboot on some HP machines. References: Novell bnc#544779 http://bugzilla.novell.com/show_bug.cgi?id=544779 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 17 +++++++++++++++++ sound/pci/hda/hda_codec.h | 2 ++ sound/pci/hda/hda_intel.c | 1 + sound/pci/hda/patch_sigmatel.c | 1 + 4 files changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c1366343335..146f95be8737 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3404,6 +3404,23 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) } } +/* call each reboot notifier */ +void snd_hda_bus_reboot_notify(struct hda_bus *bus) +{ + struct hda_codec *codec; + + if (!bus) + return; + list_for_each_entry(codec, &bus->codec_list, list) { +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!codec->power_on) + continue; +#endif + if (codec->patch_ops.reboot_notify) + codec->patch_ops.reboot_notify(codec); + } +} + /* * open the digital out in the exclusive mode */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 99552fb5f756..624060837653 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -674,6 +674,7 @@ struct hda_codec_ops { #ifdef CONFIG_SND_HDA_POWER_SAVE int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif + void (*reboot_notify)(struct hda_codec *codec); }; /* record for amp information cache */ @@ -910,6 +911,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, * Misc */ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); +void snd_hda_bus_reboot_notify(struct hda_bus *bus); /* * power management diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 55c7da30bb61..0d3e0c9ea812 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2150,6 +2150,7 @@ static int azx_resume(struct pci_dev *pci) static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) { struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); azx_stop_chip(chip); return NOTIFY_OK; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3087705a8e51..9c33700b21a8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4831,6 +4831,7 @@ static struct hda_codec_ops stac92xx_patch_ops = { .suspend = stac92xx_suspend, .resume = stac92xx_resume, #endif + .reboot_notify = stac92xx_shutup, }; static int patch_stac9200(struct hda_codec *codec) -- cgit v1.2.2 From e3303235209c0496b490e10ab131e72a9568c153 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 10 Nov 2009 14:53:02 +0100 Subject: ALSA: hda - proc - show which I/O NID is associated to PCM device Output something like: Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC888 Analog", type="Audio", device=0, substream=0 Converter: stream=0, channel=0 ... Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 9 +++++---- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_proc.c | 16 +++++++++++++++- 3 files changed, 21 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 146f95be8737..480d1ec49c99 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2877,14 +2877,15 @@ static int set_pcm_default_values(struct hda_codec *codec, return 0; } +const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { + "Audio", "SPDIF", "HDMI", "Modem" +}; + /* * get the empty PCM device number to assign */ static int get_empty_pcm_device(struct hda_bus *bus, int type) { - static const char *dev_name[HDA_PCM_NTYPES] = { - "Audio", "SPDIF", "HDMI", "Modem" - }; /* audio device indices; not linear to keep compatibility */ static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, @@ -2903,7 +2904,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; - snd_printk(KERN_WARNING "Too many %s devices\n", dev_name[type]); + snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); return -EAGAIN; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 624060837653..cbf199a98ab2 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -894,6 +894,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec); /* * PCM */ +extern const char *snd_hda_pcm_type_name[]; int snd_hda_build_pcms(struct hda_bus *bus); int snd_hda_codec_build_pcms(struct hda_codec *codec); void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 95f24e4729f8..f5639c2988ab 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -309,7 +309,21 @@ static void print_audio_io(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, unsigned int wid_type) { - int conv = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + int pcm, conv; + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + int type; + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + for (type = 0; type < 2; type++) { + if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) + continue; + snd_iprintf(buffer, " Device: name=\"%s\", type=\"%s\", device=%i, substream=%i\n", + cpcm->name, + snd_hda_pcm_type_name[cpcm->pcm_type], + cpcm->pcm->device, + cpcm->pcm->streams[type].substream->number); + } + } + conv = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); snd_iprintf(buffer, " Converter: stream=%d, channel=%d\n", (conv & AC_CONV_STREAM) >> AC_CONV_STREAM_SHIFT, -- cgit v1.2.2 From 91d12c485b8949cce6c13ab641147c5bc86ce8b9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:12:26 +0200 Subject: sound: rawmidi: fix opened substreams count The substream_opened field is to count the number of opened substreams, not the number of times that any substreams have been opened. Furthermore, all substreams being opened does not imply that the next open would fail, due to the possibility of O_APPEND. With this wrong check, opening a substream multiple times would succeed only if the device had more unopened substreams. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 4e26563431c8..818b1299ed91 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -242,8 +242,6 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, return -ENXIO; if (subdevice >= 0 && subdevice >= s->substream_count) return -ENODEV; - if (s->substream_opened >= s->substream_count) - return -EAGAIN; list_for_each_entry(substream, &s->substreams, list) { if (substream->opened) { @@ -280,9 +278,9 @@ static int open_substream(struct snd_rawmidi *rmidi, substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; + rmidi->streams[substream->stream].substream_opened++; } substream->use_count++; - rmidi->streams[substream->stream].substream_opened++; return 0; } @@ -466,7 +464,6 @@ static void close_substream(struct snd_rawmidi *rmidi, struct snd_rawmidi_substream *substream, int cleanup) { - rmidi->streams[substream->stream].substream_opened--; if (--substream->use_count) return; @@ -491,6 +488,7 @@ static void close_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); substream->opened = 0; substream->append = 0; + rmidi->streams[substream->stream].substream_opened--; } static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) -- cgit v1.2.2 From e7373b702f6eab35f315e016a4159860a7a4d686 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 10 Nov 2009 10:13:30 +0100 Subject: sound: pcm: record a substream's owner process Record the pid of the task that opened a PCM substream. For sound cards with hardware mixing, this allows determining which process is associated with a specific substream's volume control. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8e2c7833614c..6884ae031f6f 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -435,6 +435,7 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, return; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); + snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); snd_iprintf(buffer, "trigger_time: %ld.%09ld\n", status.trigger_tstamp.tv_sec, status.trigger_tstamp.tv_nsec); snd_iprintf(buffer, "tstamp : %ld.%09ld\n", @@ -900,6 +901,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->private_data = pcm->private_data; substream->ref_count = 1; substream->f_flags = file->f_flags; + substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; return 0; @@ -921,6 +923,8 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) kfree(runtime->hw_constraints.rules); kfree(runtime); substream->runtime = NULL; + put_pid(substream->pid); + substream->pid = NULL; substream->pstr->substream_opened--; } -- cgit v1.2.2 From 7584af10cf46e0f4aa1696f1be79fa0f19a945ba Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 10 Nov 2009 10:14:04 +0100 Subject: sound: rawmidi: record a substream's owner process Record the pid of the task that opened a RawMIDI substream. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 818b1299ed91..2f766123b158 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -278,6 +278,7 @@ static int open_substream(struct snd_rawmidi *rmidi, substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; + substream->pid = get_pid(task_pid(current)); rmidi->streams[substream->stream].substream_opened++; } substream->use_count++; @@ -488,6 +489,8 @@ static void close_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); substream->opened = 0; substream->append = 0; + put_pid(substream->pid); + substream->pid = NULL; rmidi->streams[substream->stream].substream_opened--; } @@ -1336,6 +1339,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Mode : %s\n" @@ -1357,6 +1363,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Buffer size : %lu\n" -- cgit v1.2.2 From 8f217a226cfa7b960b8a6c00cef6b4de2c5dd030 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Nov 2009 18:26:12 +0100 Subject: ALSA: hda - Add missing export for snd_hda_bus_reboot_notify ... forgot to add for modules. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 480d1ec49c99..2b787b013e93 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3421,6 +3421,7 @@ void snd_hda_bus_reboot_notify(struct hda_bus *bus) codec->patch_ops.reboot_notify(codec); } } +EXPORT_SYMBOL_HDA(snd_hda_bus_reboot_notify); /* * open the digital out in the exclusive mode -- cgit v1.2.2 From a2f6309e8392e2c14c04594fca8b4876c8c9bc36 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Nov 2009 09:34:25 +0100 Subject: ALSA: hda - Add power on/off counter Added the power on/off counter and expose via sysfs files. The sysfs files, power_on_acct and power_off_acct, are created under each codec hwdep sysfs directory (e.g. /sys/class/sound/hwC0D0). The files show the msec length of the codec power-on and power-off, respectively. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 16 ++++++++++++++++ sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_hwdep.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_local.h | 9 +++++++++ 4 files changed, 67 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2b787b013e93..444d9039c1ac 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -515,6 +515,7 @@ static int snd_hda_bus_dev_register(struct snd_device *device) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { snd_hda_hwdep_add_sysfs(codec); + snd_hda_hwdep_add_power_sysfs(codec); } return 0; } @@ -2452,9 +2453,11 @@ static void hda_call_codec_suspend(struct hda_codec *codec) codec->afg ? codec->afg : codec->mfg, AC_PWRST_D3); #ifdef CONFIG_SND_HDA_POWER_SAVE + snd_hda_update_power_acct(codec); cancel_delayed_work(&codec->power_work); codec->power_on = 0; codec->power_transition = 0; + codec->power_jiffies = jiffies; #endif } @@ -3191,6 +3194,17 @@ static void hda_keep_power_on(struct hda_codec *codec) { codec->power_count++; codec->power_on = 1; + codec->power_jiffies = jiffies; +} + +void snd_hda_update_power_acct(struct hda_codec *codec) +{ + unsigned long delta = jiffies - codec->power_jiffies; + if (codec->power_on) + codec->power_on_acct += delta; + else + codec->power_off_acct += delta; + codec->power_jiffies += delta; } void snd_hda_power_up(struct hda_codec *codec) @@ -3201,7 +3215,9 @@ void snd_hda_power_up(struct hda_codec *codec) if (codec->power_on || codec->power_transition) return; + snd_hda_update_power_acct(codec); codec->power_on = 1; + codec->power_jiffies = jiffies; if (bus->ops.pm_notify) bus->ops.pm_notify(bus); hda_call_codec_resume(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index cbf199a98ab2..b16678cade18 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -812,6 +812,9 @@ struct hda_codec { unsigned int power_transition :1; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ + unsigned long power_on_acct; + unsigned long power_off_acct; + unsigned long power_jiffies; #endif /* codec-specific additional proc output */ @@ -936,6 +939,7 @@ const char *snd_hda_get_jack_location(u32 cfg); void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); #define snd_hda_codec_needs_resume(codec) codec->power_count +void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index cc24e6721d74..d24328661c6a 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -154,6 +154,44 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static ssize_t power_on_acct_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + snd_hda_update_power_acct(codec); + return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_on_acct)); +} + +static ssize_t power_off_acct_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + snd_hda_update_power_acct(codec); + return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_off_acct)); +} + +static struct device_attribute power_attrs[] = { + __ATTR_RO(power_on_acct), + __ATTR_RO(power_off_acct), +}; + +int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec) +{ + struct snd_hwdep *hwdep = codec->hwdep; + int i; + + for (i = 0; i < ARRAY_SIZE(power_attrs); i++) + snd_add_device_sysfs_file(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card, + hwdep->device, &power_attrs[i]); + return 0; +} +#endif /* CONFIG_SND_HDA_POWER_SAVE */ + #ifdef CONFIG_SND_HDA_RECONFIG /* diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 461e0c15c77a..015fbac914b3 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -437,6 +437,15 @@ int snd_hda_create_hwdep(struct hda_codec *codec); static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; } #endif +#ifdef CONFIG_SND_HDA_POWER_SAVE +int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec); +#else +static inline int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec) +{ + return 0; +} +#endif + #ifdef CONFIG_SND_HDA_RECONFIG int snd_hda_hwdep_add_sysfs(struct hda_codec *codec); #else -- cgit v1.2.2 From f8b7163529831ee3ad6a1aeaa0f1256cb527008d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Nov 2009 09:50:28 +0100 Subject: ALSA: hda - Don't access invalid substream in proc file The commit e3303235209c0496b490e10ab131e72a9568c153 "ALSA: hda - proc - show which I/O NID is associated to PCM device" introduces the access to substream pointer. But, PCMs may have no substreams in one or both directions, and this results in NULL dereference. Also, print the first substream number doesn't make sense. This patch removes the access to the substream pointer, and reformat to fit to the standard coding style. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index f5639c2988ab..f5b783ce450d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -316,11 +316,11 @@ static void print_audio_io(struct snd_info_buffer *buffer, for (type = 0; type < 2; type++) { if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) continue; - snd_iprintf(buffer, " Device: name=\"%s\", type=\"%s\", device=%i, substream=%i\n", - cpcm->name, - snd_hda_pcm_type_name[cpcm->pcm_type], - cpcm->pcm->device, - cpcm->pcm->streams[type].substream->number); + snd_iprintf(buffer, " Device: name=\"%s\", " + "type=\"%s\", device=%i\n", + cpcm->name, + snd_hda_pcm_type_name[cpcm->pcm_type], + cpcm->pcm->device); } } conv = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); -- cgit v1.2.2 From 7288561af9a931c59e431336b553d897ee37b67d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Nov 2009 10:01:18 +0100 Subject: ALSA: hda - Fix build error without CONFIG_SND_HDA_HWDEP=y CONFIG_SND_HDA_POWER_SAVE is independent from CONFIG_SND_HDA_HWDEP. Thus snd_hda_hwdep_add_power_sysfs() needs the check of both kconfigs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 015fbac914b3..c1ca3182e6a4 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -437,7 +437,7 @@ int snd_hda_create_hwdep(struct hda_codec *codec); static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; } #endif -#ifdef CONFIG_SND_HDA_POWER_SAVE +#if defined(CONFIG_SND_HDA_POWER_SAVE) && defined(CONFIG_SND_HDA_HWDEP) int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec); #else static inline int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec) -- cgit v1.2.2 From 7aae816dae150caad8880357e42303935c0179a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 Nov 2009 16:08:04 +0000 Subject: ASoC: Add bit clock rate calculator utility functions Many devices need to calculate the bit clock rate desired to work out the clock configuration required for the device. Provide utility functions to do this using both hw_params structures and raw numbers. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/Makefile | 2 +- sound/soc/soc-utils.c | 68 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+), 1 deletion(-) create mode 100644 sound/soc/soc-utils.c (limited to 'sound') diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0c5eac01bf2e..1470141d4167 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c new file mode 100644 index 000000000000..b16aaaeb0aab --- /dev/null +++ b/sound/soc/soc-utils.c @@ -0,0 +1,68 @@ +/* + * soc-util.c -- ALSA SoC Audio Layer utility functions + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown + * Liam Girdwood + * + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include + +int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) +{ + return sample_size * channels * tdm_slots; +} +EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size); + +int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) +{ + int sample_size; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + sample_size = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + sample_size = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + sample_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + sample_size = 32; + break; + default: + return -ENOTSUPP; + } + + return snd_soc_calc_frame_size(sample_size, params_channels(params), + 1); +} +EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size); + +int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) +{ + int ret; + + ret = snd_soc_params_to_frame_size(params); + + if (ret > 0) + return ret * params_rate(params); + else + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); -- cgit v1.2.2 From ba2b87f5a93659a28cc4fb812ccd7b4146ac3aa9 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 11 Nov 2009 14:02:18 +0900 Subject: ASoC: Fixed arguments passed to SMDK64xx set_pll Corrected the order of 'source' and 'pll_id' arguments. Signed-off-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/s3c24xx/smdk64xx_wm8580.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index cb8a9161b643..216dd1e8e378 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -115,7 +115,7 @@ static int smdk64xx_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(codec_dai, 0, WM8580_PLLA, + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, SMDK64XX_WM8580_FREQ, pll_out); if (ret < 0) return ret; -- cgit v1.2.2 From f773205300fa4a5a405f8ed6e3bb97e46c6eefb4 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Thu, 12 Nov 2009 12:01:47 +0800 Subject: ASoC: move setting ac97 platformdata earlier than ac97 read/write While probing, AC97 codec drivers and soc-core generically execute the following sequence: snd_soc_new_ac97_codec -> snd_soc_new_pcms -> reset ac-link/read AC97 ID to detect ->... -> set platform_data to ac97 by soc-core commit 474828a40f6ddab6e2a3475a19c5c84aa3ec7d60 adds platform_data to snd_ac97 instance. But ac97 platform data hasn't given to snd_ac97 before actual ac97 operations. Then while ac97_read access platform_data of snd_ac97 for detecting, NULL pointer oops will fire. That means old platform_data patch doesn't work in real-life cases. This patch moves the operation of setting ac97 platform_data earlier than ac97 reading/writing operations. Then it makes platform_data of AC97 become practically useful. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e2b6d75f16e3..ef8f28284cb9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1083,11 +1083,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) continue; } } - if (card->dai_link[i].codec_dai->ac97_control) { + if (card->dai_link[i].codec_dai->ac97_control) ac97 = 1; - snd_ac97_dev_add_pdata(codec->ac97, - card->dai_link[i].cpu_dai->ac97_pdata); - } } snprintf(codec->card->shortname, sizeof(codec->card->shortname), @@ -1510,6 +1507,10 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) mutex_unlock(&codec->mutex); return ret; } + if (card->dai_link[i].codec_dai->ac97_control) { + snd_ac97_dev_add_pdata(codec->ac97, + card->dai_link[i].cpu_dai->ac97_pdata); + } } mutex_unlock(&codec->mutex); -- cgit v1.2.2 From c871a05315d1a76034ea06feeda92081e1d608bf Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Thu, 12 Nov 2009 17:14:04 +0900 Subject: ASoC: Add jack_status_check callback function for GPIO jacks The jack_status_check callback function is the interface to check the status of the jack. Some target provides the method to distinguish what is the jack inserted - headphone jack, microphone jack, tvout jack, etc, so we can implement it using the jack_status_check function. Signed-off-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 12124149601e..3c07a94c2e30 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -163,6 +163,9 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) else report = 0; + if (gpio->jack_status_check) + report = gpio->jack_status_check(); + snd_soc_jack_report(jack, report, gpio->report); } -- cgit v1.2.2 From 0d26ce3403b3841fa2656df08a819fc7eaebaa17 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Thu, 12 Nov 2009 17:43:11 +0100 Subject: sound: OSS: fix error return in dma_ioctl() The returned error should stay negative Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/audio.c b/sound/oss/audio.c index b69c05b7ea7b..7df48a25c4ee 100644 --- a/sound/oss/audio.c +++ b/sound/oss/audio.c @@ -838,7 +838,7 @@ static int dma_ioctl(int dev, unsigned int cmd, void __user *arg) if ((err = audio_devs[dev]->d->prepare_for_input(dev, dmap_in->fragment_size, dmap_in->nbufs)) < 0) { spin_unlock_irqrestore(&dmap_in->lock,flags); - return -err; + return err; } dmap_in->dma_mode = DMODE_INPUT; audio_devs[dev]->enable_bits |= PCM_ENABLE_INPUT; -- cgit v1.2.2 From 0a3f5e35aae43b20fef09fd800cf52cc9a2d61a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Nov 2009 23:15:08 +0000 Subject: ASoC: Remove redundant snd_soc_dapm_new_widgets() calls The DAPM widgets are now insntantiated by the core when creating the card so there is no need for the individual CODEC drivers to do so. Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/cx20442.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8728.c | 2 -- sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8900.c | 2 -- sound/soc/codecs/wm8903.c | 2 -- sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8971.c | 2 -- sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8993.c | 2 -- sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9705.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - 35 files changed, 1 insertion(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index b4be96decf32..2c18e3d1b71e 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -385,7 +385,6 @@ static int ad1836_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets, ARRAY_SIZE(ad1836_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); pcm_err: return ret; diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 3b2222a0c808..5d489186c05b 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -592,7 +592,6 @@ static int ad1938_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets, ARRAY_SIZE(ad1938_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 57a6846a9a1f..ff966567e2ba 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -294,7 +294,6 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 364832ccd748..82fca284d007 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -441,7 +441,6 @@ static int ak4671_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index dda751c885cb..e000cdfec1ec 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -93,7 +93,6 @@ static int cx20442_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, cx20442_audio_map, ARRAY_SIZE(cx20442_audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index b3130339d29a..d2ff1cde6883 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -210,7 +210,6 @@ static int ssm2602_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index ee8cb2c08b87..1709e3f614a8 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -395,7 +395,6 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 03cad250f58d..2b4dc2b0b017 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -753,7 +753,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index bff476d65d05..2a013e46ae14 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -462,7 +462,6 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 510b8b226f96..5f1681f6ca76 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1493,7 +1493,6 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a42e47d94630..a2763c2e7348 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -378,7 +378,6 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 2e35a354b166..f82125d9e85a 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -800,7 +800,7 @@ static int wm8350_add_widgets(struct snd_soc_codec *codec) return ret; } - return snd_soc_dapm_new_widgets(codec); + return 0; } static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 584af68af22a..b432f4d4a324 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -915,7 +915,6 @@ static int wm8400_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index e3c21ebcc08e..265e68c75df8 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -219,7 +219,6 @@ static int wm8510_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 2e2b01d6c82b..d3a61d7ea0c5 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -117,7 +117,6 @@ static int wm8523_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index dde50d118181..d077df6f5e75 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -315,7 +315,6 @@ static int wm8580_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 70e0675b5d4a..24a35603bcf7 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -99,7 +99,6 @@ static int wm8711_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 1252a8a486a6..3fb653ba363a 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -74,8 +74,6 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); - return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index e3675e7a9813..3a497810f939 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -159,7 +159,6 @@ static int wm8731_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 50a3d6590588..475c67ac7818 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -403,7 +403,6 @@ static int wm8750_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index c652bc04cc81..d6850dacda29 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -673,7 +673,6 @@ static int wm8753_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 85f67dbe211d..c9438dd62df3 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -618,8 +618,6 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - return 0; } diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bfeff4ee5de9..b8cae1758642 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -919,8 +919,6 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); - return 0; } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index fc80aa6c913c..3d850b97037a 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -298,7 +298,6 @@ static int wm8940_add_widgets(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); if (ret) goto error_ret; - ret = snd_soc_dapm_new_widgets(codec); error_ret: return ret; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 40390afa75f3..d07bcc1e1c60 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -307,7 +307,6 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 07e389574db1..a8007d58813f 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -986,7 +986,6 @@ static int wm8961_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets, ARRAY_SIZE(wm8961_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); return ret; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 56a66e89ab91..d9540d55fc89 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -338,8 +338,6 @@ static int wm8971_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - return 0; } diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index c245f0ee0ec2..81c57b5c591c 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -276,7 +276,6 @@ static int wm8974_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index bee292e37d1b..2862e4dced27 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -790,7 +790,6 @@ static int wm8988_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets, ARRAY_SIZE(wm8988_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return ret; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index e43cb2c8b915..341481e0e830 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,6 @@ static int wm8990_add_widgets(struct snd_soc_codec *codec) /* set up the WM8990 audio map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 0d4d2be92b64..5e32f2ed5fc2 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1464,8 +1464,6 @@ static int wm8993_probe(struct platform_device *pdev) wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff, wm8993->pdata.lineout2_diff); - snd_soc_dapm_new_widgets(codec); - return ret; err: diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 3f1f84421312..c468497314ba 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1262,7 +1262,6 @@ static int wm9081_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets, ARRAY_SIZE(wm9081_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); return ret; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 0e817b8705cd..dfffc6c778c0 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -205,7 +205,6 @@ static int wm9705_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, ARRAY_SIZE(wm9705_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 155cacf124ea..2a0872273007 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -436,7 +436,6 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5f81ecd20a81..00bac315fb3b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -625,7 +625,6 @@ static int wm9713_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } -- cgit v1.2.2 From 401de8184a4d94688962b9258fe10ab309ffda9c Mon Sep 17 00:00:00 2001 From: Akinobu Mita Date: Fri, 13 Nov 2009 16:02:56 +0900 Subject: ALSA: ice1712: Use bitrev8 Signed-off-by: Akinobu Mita Signed-off-by: Takashi Iwai --- sound/i2c/cs8427.c | 15 ++------------- sound/pci/Kconfig | 1 + 2 files changed, 3 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 020a5d512472..04ae8704cdcd 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -55,18 +56,6 @@ struct cs8427 { struct cs8427_stream capture; }; -static unsigned char swapbits(unsigned char val) -{ - int bit; - unsigned char res = 0; - for (bit = 0; bit < 8; bit++) { - res <<= 1; - res |= val & 1; - val >>= 1; - } - return res; -} - int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg, unsigned char val) { @@ -149,7 +138,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device, } data[0] = CS8427_REG_AUTOINC | CS8427_REG_CORU_DATABUF; for (idx = 0; idx < count; idx++) - data[idx + 1] = swapbits(ndata[idx]); + data[idx + 1] = bitrev8(ndata[idx]); if (snd_i2c_sendbytes(device, data, count + 1) != count + 1) return -EIO; return 1; diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 75c602b5b132..351654cf7b09 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -570,6 +570,7 @@ config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" select SND_MPU401_UART select SND_AC97_CODEC + select BITREVERSE help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. -- cgit v1.2.2 From 01a1796bc52f625edc23bf995d200e1556eec544 Mon Sep 17 00:00:00 2001 From: "akpm@linux-foundation.org" Date: Fri, 13 Nov 2009 16:47:10 -0800 Subject: sound/pci/hda/patch_via.c: work around gcc-4.0.2 ICE sound/pci/hda/patch_via.c: In function 'via_hp_bind_automute': sound/pci/hda/patch_via.c:2074: internal compiler error: in do_SUBST, at combine.c:462 Please submit a full bug report, with preprocessed source if appropriate. See for instructions. [added a comment by tiwai] Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 5ec0e39593b5..5a856009c916 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2043,7 +2043,10 @@ static void via_speaker_automute(struct hda_codec *codec) /* mute line-out and internal speaker if HP is plugged */ static void via_hp_bind_automute(struct hda_codec *codec) { - unsigned int hp_present, present = 0; + /* use long instead of int below just to avoid an internal compiler + * error with gcc 4.0.x + */ + unsigned long hp_present, present = 0; struct via_spec *spec = codec->spec; int i; -- cgit v1.2.2 From 50d40f187f9182ee8caa1b83f80a0e11e2226baa Mon Sep 17 00:00:00 2001 From: Aleksey Kunitskiy Date: Sat, 14 Nov 2009 15:18:54 +0200 Subject: ALSA: ice1724 - Patch for suspend/resume for ESI Juli@ Add proper suspend/resume code for Juli@ cards. Based on ice1724 suspend/resume work of Igor Chernyshev. Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413 Tested on linux-2.6.31.6 Signed-off-by: Aleksey Kunitskiy Signed-off-by: Takashi Iwai --- sound/pci/ice1712/juli.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) (limited to 'sound') diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index fd948bfd9aef..f5020ad99a10 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -503,6 +503,31 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) return 0; } +/* + * suspend/resume + * */ + +#ifdef CONFIG_PM +static int juli_resume(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + struct juli_spec *spec = ice->spec; + /* akm4358 un-reset, un-mute */ + snd_akm4xxx_reset(ak, 0); + /* reinit ak4114 */ + snd_ak4114_reinit(spec->ak4114); + return 0; +} + +static int juli_suspend(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + /* akm4358 reset and soft-mute */ + snd_akm4xxx_reset(ak, 1); + return 0; +} +#endif + /* * initialize the chip */ @@ -646,6 +671,13 @@ static int __devinit juli_init(struct snd_ice1712 *ice) ice->set_spdif_clock = juli_set_spdif_clock; ice->spdif.ops.open = juli_spdif_in_open; + +#ifdef CONFIG_PM + ice->pm_resume = juli_resume; + ice->pm_suspend = juli_suspend; + ice->pm_suspend_enabled = 1; +#endif + return 0; } -- cgit v1.2.2 From 123c07aeddd71fbb295842a8c19866e780b9a100 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 21 Oct 2009 14:48:23 +0200 Subject: ALSA: hda_intel: Digital PC Beep - change behaviour for input layer Original implementation was keeping registered input device for SND_BEEP and SND_TONE events all time. This patch changes this behaviour: If digital PC Beep is turned off using universal control switch, the input device is unregistered. Explanation: The kd_mksound() send SND_BEEP and SND_TONE only to last registered device acceping those events. It means that the HDA Intel audio driver blocks also the internal PC Speaker device (pcspkr.c driver) even if the HDA Beep is muted. The user can easy disable all beeps using 'setterm -blength 0' or 'xset b off' command. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 88 +++++++++++++++++++++++++++++++++--------- sound/pci/hda/hda_beep.h | 4 ++ sound/pci/hda/hda_codec.c | 12 ++++++ sound/pci/hda/hda_local.h | 15 +++++++ sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 16 ++++---- 7 files changed, 111 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 3f51a981e604..0e986537d570 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -113,23 +113,25 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, return 0; } -int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +static void snd_hda_do_detach(struct hda_beep *beep) +{ + input_unregister_device(beep->dev); + beep->dev = NULL; + cancel_work_sync(&beep->beep_work); + /* turn off beep for sure */ + snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + AC_VERB_SET_BEEP_CONTROL, 0); +} + +static int snd_hda_do_attach(struct hda_beep *beep) { struct input_dev *input_dev; - struct hda_beep *beep; + struct hda_codec *codec = beep->codec; int err; - if (!snd_hda_get_bool_hint(codec, "beep")) - return 0; /* disabled explicitly */ - - beep = kzalloc(sizeof(*beep), GFP_KERNEL); - if (beep == NULL) - return -ENOMEM; - snprintf(beep->phys, sizeof(beep->phys), - "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); input_dev = input_allocate_device(); if (!input_dev) { - kfree(beep); + printk(KERN_INFO "hda_beep: unable to allocate input device\n"); return -ENOMEM; } @@ -151,21 +153,71 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) err = input_register_device(input_dev); if (err < 0) { input_free_device(input_dev); - kfree(beep); + printk(KERN_INFO "hda_beep: unable to register input device\n"); return err; } + beep->dev = input_dev; + return 0; +} + +static void snd_hda_do_register(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, register_work); + int request; + + mutex_lock(&beep->mutex); + request = beep->request_enable; + if (beep->enabled != request) { + if (!request) { + snd_hda_do_detach(beep); + } else { + if (snd_hda_do_attach(beep) < 0) + goto __out; + } + beep->enabled = request; + } + __out: + mutex_unlock(&beep->mutex); +} + +int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) +{ + struct hda_beep *beep = codec->beep; + enable = !!enable; + if (beep && beep->enabled != enable) { + beep->request_enable = enable; + schedule_work(&beep->register_work); + return 1; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_enable_beep_device); + +int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +{ + struct hda_beep *beep; + + if (!snd_hda_get_bool_hint(codec, "beep")) + return 0; /* disabled explicitly */ + beep = kzalloc(sizeof(*beep), GFP_KERNEL); + if (beep == NULL) + return -ENOMEM; + snprintf(beep->phys, sizeof(beep->phys), + "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); /* enable linear scale */ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0x01); beep->nid = nid; - beep->dev = input_dev; beep->codec = codec; - beep->enabled = 1; codec->beep = beep; + INIT_WORK(&beep->register_work, &snd_hda_do_register); INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); + mutex_init(&beep->mutex); + return 0; } EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device); @@ -174,11 +226,11 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) { struct hda_beep *beep = codec->beep; if (beep) { - cancel_work_sync(&beep->beep_work); - - input_unregister_device(beep->dev); - kfree(beep); + cancel_work_sync(&beep->register_work); + if (beep->enabled) + snd_hda_do_detach(beep); codec->beep = NULL; + kfree(beep); } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 0c3de787c717..68465f679d8c 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -32,11 +32,15 @@ struct hda_beep { int tone; hda_nid_t nid; unsigned int enabled:1; + unsigned int request_enable:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ + struct work_struct register_work; /* scheduled task for beep event */ struct work_struct beep_work; /* scheduled task for beep event */ + struct mutex mutex; }; #ifdef CONFIG_SND_HDA_INPUT_BEEP +int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 444d9039c1ac..7fd2abe1129d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -30,6 +30,7 @@ #include #include #include "hda_local.h" +#include "hda_beep.h" #include /* @@ -1734,6 +1735,17 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); +int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + + snd_hda_enable_beep_device(codec, *valp); + return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +} +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); + /* * bound volume controls * diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index c1ca3182e6a4..3001794ad291 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -66,6 +66,19 @@ /* stereo mute switch */ #define HDA_CODEC_MUTE(xname, nid, xindex, direction) \ HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction) +/* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ +#define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = snd_hda_mixer_amp_switch_put_beep, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } +/* special beep mono mute switch */ +#define HDA_CODEC_MUTE_BEEP_MONO(xname, nid, channel, xindex, direction) \ + HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, 0, nid, channel, xindex, direction) +/* special beep stereo mute switch */ +#define HDA_CODEC_MUTE_BEEP(xname, nid, xindex, direction) \ + HDA_CODEC_MUTE_BEEP_MONO(xname, nid, 3, xindex, direction) int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); @@ -81,6 +94,8 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /* lowlevel accessor with caching; use carefully */ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2d603f6aba63..a0293614a0b9 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -159,7 +159,7 @@ static void ad198x_free_kctls(struct hda_codec *codec); /* additional beep mixers; the actual parameters are overwritten at build */ static struct snd_kcontrol_new ad_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_OUTPUT), { } /* end */ }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 49de107db16b..8c04e0e0f655 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2413,7 +2413,7 @@ static void alc_free_kctls(struct hda_codec *codec); /* additional beep mixers; the actual parameters are overwritten at build */ static struct snd_kcontrol_new alc_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), { } /* end */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8d65d2b25234..87ba239ff1c9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2648,6 +2648,7 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, + STAC_CTL_WIDGET_MUTE_BEEP, STAC_CTL_WIDGET_MONO_MUX, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, @@ -2658,6 +2659,7 @@ enum { static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0), STAC_MONO_MUX, STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), @@ -3221,11 +3223,14 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); - int err; + int err, type = STAC_CTL_WIDGET_MUTE_BEEP; + + if (spec->anabeep_nid == nid) + type = STAC_CTL_WIDGET_MUTE; /* check for mute support for the the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + err = stac92xx_add_control(spec, type, "Beep Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) @@ -3258,12 +3263,7 @@ static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int enabled = !!ucontrol->value.integer.value[0]; - if (codec->beep->enabled != enabled) { - codec->beep->enabled = enabled; - return 1; - } - return 0; + return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]); } static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { -- cgit v1.2.2 From 13dab0808bb41b18888e1758a060a685deee1f30 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 3 Nov 2009 14:29:50 +0100 Subject: ALSA: hda_intel: Digital PC Beep - delay input device unregistration The massive register/unregister calls for input device layer might be overkill. Delay unregister call by one HZ as workaround. Also, as benefit, beep->enabled variable is changed immediately now (not from workqueue). Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 42 +++++++++++++++++++++++++++--------------- sound/pci/hda/hda_beep.h | 3 ++- 2 files changed, 29 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0e986537d570..74db40edb336 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -164,20 +164,21 @@ static void snd_hda_do_register(struct work_struct *work) { struct hda_beep *beep = container_of(work, struct hda_beep, register_work); - int request; mutex_lock(&beep->mutex); - request = beep->request_enable; - if (beep->enabled != request) { - if (!request) { - snd_hda_do_detach(beep); - } else { - if (snd_hda_do_attach(beep) < 0) - goto __out; - } - beep->enabled = request; - } - __out: + if (beep->enabled && !beep->dev) + snd_hda_do_attach(beep); + mutex_unlock(&beep->mutex); +} + +static void snd_hda_do_unregister(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, unregister_work.work); + + mutex_lock(&beep->mutex); + if (!beep->enabled && beep->dev) + snd_hda_do_detach(beep); mutex_unlock(&beep->mutex); } @@ -185,9 +186,19 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) { struct hda_beep *beep = codec->beep; enable = !!enable; - if (beep && beep->enabled != enable) { - beep->request_enable = enable; - schedule_work(&beep->register_work); + if (beep == NULL) + return 0; + if (beep->enabled != enable) { + beep->enabled = enable; + if (enable) { + cancel_delayed_work(&beep->unregister_work); + schedule_work(&beep->register_work); + } else { + /* turn off beep */ + snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + AC_VERB_SET_BEEP_CONTROL, 0); + schedule_delayed_work(&beep->unregister_work, HZ); + } return 1; } return 0; @@ -215,6 +226,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) codec->beep = beep; INIT_WORK(&beep->register_work, &snd_hda_do_register); + INIT_DELAYED_WORK(&beep->unregister_work, &snd_hda_do_unregister); INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); mutex_init(&beep->mutex); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 68465f679d8c..53eba8d8414d 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -34,7 +34,8 @@ struct hda_beep { unsigned int enabled:1; unsigned int request_enable:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ - struct work_struct register_work; /* scheduled task for beep event */ + struct work_struct register_work; /* registration work */ + struct delayed_work unregister_work; /* unregistration work */ struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; }; -- cgit v1.2.2 From 5f81669750504b1e7e00acde5068d972af466f29 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 4 Nov 2009 12:46:49 +0100 Subject: ALSA: hda: beep - add missing cancel_delayed_work The unregister work should be also canceled in snd_hda_detach_beep_device() function. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 74db40edb336..c819152de79b 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -239,6 +239,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) struct hda_beep *beep = codec->beep; if (beep) { cancel_work_sync(&beep->register_work); + cancel_delayed_work(&beep->unregister_work); if (beep->enabled) snd_hda_do_detach(beep); codec->beep = NULL; -- cgit v1.2.2 From 2dca0bba70ce3c233be152e384580c134935332d Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 13 Nov 2009 18:41:52 +0100 Subject: ALSA: hda - add beep_mode module parameter The beep_mode parameter for snd-hda-intel module allows to choose among different digital beep device registation to the input layer. 0 = do not register to the input layer 1 = register to the input layer all time 2 = use "Beep Switch" control exported to user space mixer applications Also, introduce CONFIG_SND_HDA_INPUT_BEEP_MODE for default value. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 11 +++++++++++ sound/pci/hda/hda_beep.c | 21 ++++++++++++++++----- sound/pci/hda/hda_beep.h | 5 +++++ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 15 +++++++++++++++ 5 files changed, 48 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 55545e0818b5..25ae10e16f59 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -38,6 +38,17 @@ config SND_HDA_INPUT_BEEP Say Y here to build a digital beep interface for HD-audio driver. This interface is used to generate digital beeps. +config SND_HDA_INPUT_BEEP_MODE + int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)" + depends on SND_HDA_INPUT_BEEP=y + default "1" + range 0 2 + help + Set 0 to disable the digital beep interface for HD-audio by default. + Set 1 to always enable the digital beep interface for HD-audio by + default. Set 2 to control the beep device registration to input + layer using a "Beep Switch" in mixer applications. + config SND_HDA_INPUT_JACK bool "Support jack plugging notification via input layer" depends on INPUT=y || INPUT=SND_HDA_INTEL diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c819152de79b..9e48798b415b 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -190,14 +190,19 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) return 0; if (beep->enabled != enable) { beep->enabled = enable; - if (enable) { - cancel_delayed_work(&beep->unregister_work); - schedule_work(&beep->register_work); - } else { + if (!enable) { /* turn off beep */ snd_hda_codec_write_cache(beep->codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, 0); - schedule_delayed_work(&beep->unregister_work, HZ); + } + if (beep->mode == HDA_BEEP_MODE_SWREG) { + if (enable) { + cancel_delayed_work(&beep->unregister_work); + schedule_work(&beep->register_work); + } else { + schedule_delayed_work(&beep->unregister_work, + HZ); + } } return 1; } @@ -223,6 +228,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) beep->nid = nid; beep->codec = codec; + beep->mode = codec->beep_mode; codec->beep = beep; INIT_WORK(&beep->register_work, &snd_hda_do_register); @@ -230,6 +236,11 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); mutex_init(&beep->mutex); + if (beep->mode == HDA_BEEP_MODE_ON) { + beep->enabled = 1; + snd_hda_do_register(&beep->register_work); + } + return 0; } EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 53eba8d8414d..17dd1c325e32 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -24,10 +24,15 @@ #include "hda_codec.h" +#define HDA_BEEP_MODE_ON 0 +#define HDA_BEEP_MODE_OFF 1 +#define HDA_BEEP_MODE_SWREG 2 + /* beep information */ struct hda_beep { struct input_dev *dev; struct hda_codec *codec; + unsigned int mode; char phys[32]; int tone; hda_nid_t nid; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index b16678cade18..51920563bc7f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -772,6 +772,7 @@ struct hda_codec { /* beep device */ struct hda_beep *beep; + unsigned int beep_mode; /* widget capabilities cache */ unsigned int num_nodes; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e73e395e7601..91bcbdad5af5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -64,6 +64,10 @@ static int enable_msi = -1; #ifdef CONFIG_SND_HDA_PATCH_LOADER static char *patch[SNDRV_CARDS]; #endif +#ifdef CONFIG_SND_HDA_INPUT_BEEP +static int beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = + CONFIG_SND_HDA_INPUT_BEEP_MODE}; +#endif module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -91,6 +95,11 @@ MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); module_param_array(patch, charp, NULL, 0444); MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface."); #endif +#ifdef CONFIG_SND_HDA_INPUT_BEEP +module_param_array(beep_mode, int, NULL, 0444); +MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " + "(0=off, 1=on, 2=mute switch on/off) (default=1)."); +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; @@ -404,6 +413,7 @@ struct azx { unsigned short codec_mask; int codec_probe_mask; /* copied from probe_mask option */ struct hda_bus *bus; + unsigned int beep_mode; /* CORB/RIRB */ struct azx_rb corb; @@ -1404,6 +1414,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) err = snd_hda_codec_new(chip->bus, c, &codec); if (err < 0) continue; + codec->beep_mode = chip->beep_mode; codecs++; } } @@ -2579,6 +2590,10 @@ static int __devinit azx_probe(struct pci_dev *pci, goto out_free; card->private_data = chip; +#ifdef CONFIG_SND_HDA_INPUT_BEEP + chip->beep_mode = beep_mode[dev]; +#endif + /* create codec instances */ err = azx_codec_create(chip, model[dev]); if (err < 0) -- cgit v1.2.2 From 3911a4c19e927738766003839aa447becbdbaa27 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 11 Nov 2009 13:43:01 +0100 Subject: ALSA: hda - proc - introduce Control: lines to show mixer<->NID assignment This is an initial patch to show universal control<->NID assigment in proc codec file. The change helps to debug codec related problems. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 32 ++++++++++++------------ sound/pci/hda/hda_generic.c | 17 ++++++++----- sound/pci/hda/hda_local.h | 11 +++++++-- sound/pci/hda/hda_proc.c | 55 ++++++++++++++++++++++++++++++------------ sound/pci/hda/patch_analog.c | 4 ++- sound/pci/hda/patch_ca0110.c | 4 +-- sound/pci/hda/patch_cirrus.c | 12 ++++----- sound/pci/hda/patch_realtek.c | 3 ++- sound/pci/hda/patch_sigmatel.c | 4 +-- 9 files changed, 92 insertions(+), 50 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7fd2abe1129d..1ed1d88e1834 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -946,7 +946,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 60); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { @@ -1517,18 +1517,20 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); /* Add a control element and assign to the codec */ -int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) +int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, + struct snd_kcontrol *kctl) { int err; - struct snd_kcontrol **knewp; + struct hda_nid_item *item; err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) return err; - knewp = snd_array_new(&codec->mixers); - if (!knewp) + item = snd_array_new(&codec->mixers); + if (!item) return -ENOMEM; - *knewp = kctl; + item->kctl = kctl; + item->nid = nid; return 0; } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); @@ -1537,9 +1539,9 @@ EXPORT_SYMBOL_HDA(snd_hda_ctl_add); void snd_hda_ctls_clear(struct hda_codec *codec) { int i; - struct snd_kcontrol **kctls = codec->mixers.list; + struct hda_nid_item *items = codec->mixers.list; for (i = 0; i < codec->mixers.used; i++) - snd_ctl_remove(codec->bus->card, kctls[i]); + snd_ctl_remove(codec->bus->card, items[i].kctl); snd_array_free(&codec->mixers); } @@ -1645,7 +1647,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, kctl = snd_ctl_make_virtual_master(name, tlv); if (!kctl) return -ENOMEM; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; @@ -2139,7 +2141,7 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) return -ENOMEM; kctl->id.index = idx; kctl->private_value = nid; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; } @@ -2184,8 +2186,8 @@ int snd_hda_create_spdif_share_sw(struct hda_codec *codec, if (!mout->dig_out_nid) return 0; /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_hda_ctl_add(codec, - snd_ctl_new1(&spdif_share_sw, mout)); + return snd_hda_ctl_add(codec, mout->dig_out_nid, + snd_ctl_new1(&spdif_share_sw, mout)); } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw); @@ -2289,7 +2291,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) if (!kctl) return -ENOMEM; kctl->private_value = nid; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; } @@ -3165,7 +3167,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) { if (!codec->addr) return err; @@ -3173,7 +3175,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) if (!kctl) return -ENOMEM; kctl->id.device = codec->addr; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b36f6c5a92df..092c6a7c2ff3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -727,7 +727,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, if (is_loopback) add_input_loopback(codec, node->nid, HDA_INPUT, index); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -737,7 +738,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, if (is_loopback) add_input_loopback(codec, node->nid, HDA_OUTPUT, 0); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -751,7 +753,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) { knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -759,7 +762,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) { knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -857,7 +861,7 @@ static int build_input_controls(struct hda_codec *codec) } /* create input MUX if multiple sources are available */ - err = snd_hda_ctl_add(codec, snd_ctl_new1(&cap_sel, codec)); + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cap_sel, codec)); if (err < 0) return err; @@ -875,7 +879,8 @@ static int build_input_controls(struct hda_codec *codec) HDA_CODEC_VOLUME(name, adc_node->nid, spec->input_mux.items[i].index, HDA_INPUT); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, adc_node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3001794ad291..e6a0918f70d3 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -440,7 +440,13 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl); +struct hda_nid_item { + struct snd_kcontrol *kctl; + hda_nid_t nid; +}; + +int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, + struct snd_kcontrol *kctl); void snd_hda_ctls_clear(struct hda_codec *codec); /* @@ -514,7 +520,8 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, * AMP control callbacks */ /* retrieve parameters from private_value */ -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) +#define get_amp_nid_(pv) ((pv) & 0xffff) +#define get_amp_nid(kc) get_amp_nid_((kc)->private_value) #define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index f5b783ce450d..f465cff28041 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -46,6 +46,41 @@ static const char *get_wid_type_name(unsigned int wid_value) return "UNKNOWN Widget"; } +static void print_nid_mixers(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i; + struct hda_nid_item *items = codec->mixers.list; + struct snd_kcontrol *kctl; + for (i = 0; i < codec->mixers.used; i++) { + if (items[i].nid == nid) { + kctl = items[i].kctl; + snd_iprintf(buffer, + " Control: name=\"%s\", index=%i, device=%i\n", + kctl->id.name, kctl->id.index, kctl->id.device); + } + } +} + +static void print_nid_pcms(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int pcm, type; + struct hda_pcm *cpcm; + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + cpcm = &codec->pcm_info[pcm]; + for (type = 0; type < 2; type++) { + if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) + continue; + snd_iprintf(buffer, " Device: name=\"%s\", " + "type=\"%s\", device=%i\n", + cpcm->name, + snd_hda_pcm_type_name[cpcm->pcm_type], + cpcm->pcm->device); + } + } +} + static void print_amp_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, int dir) { @@ -309,21 +344,7 @@ static void print_audio_io(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, unsigned int wid_type) { - int pcm, conv; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - int type; - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; - for (type = 0; type < 2; type++) { - if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) - continue; - snd_iprintf(buffer, " Device: name=\"%s\", " - "type=\"%s\", device=%i\n", - cpcm->name, - snd_hda_pcm_type_name[cpcm->pcm_type], - cpcm->pcm->device); - } - } - conv = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + int conv = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); snd_iprintf(buffer, " Converter: stream=%d, channel=%d\n", (conv & AC_CONV_STREAM) >> AC_CONV_STREAM_SHIFT, @@ -471,6 +492,7 @@ static void print_gpio(struct snd_info_buffer *buffer, (data & (1<private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, + get_amp_nid_(spec->beep_amp), + kctl); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index d08353d3bb7f..af478019088e 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -144,7 +144,7 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); - return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, @@ -155,7 +155,7 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); - return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } #define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 8ba306856d38..9ac09e4568b3 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -500,7 +500,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); - return snd_hda_ctl_add(codec, *kctlp); + return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } static int add_volume(struct hda_codec *codec, const char *name, @@ -513,7 +513,7 @@ static int add_volume(struct hda_codec *codec, const char *name, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); - return snd_hda_ctl_add(codec, *kctlp); + return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) @@ -536,14 +536,14 @@ static int add_vmaster(struct hda_codec *codec, hda_nid_t dac) spec->vmaster_sw = snd_ctl_make_virtual_master("Master Playback Switch", NULL); - err = snd_hda_ctl_add(codec, spec->vmaster_sw); + err = snd_hda_ctl_add(codec, dac, spec->vmaster_sw); if (err < 0) return err; snd_hda_set_vmaster_tlv(codec, dac, HDA_OUTPUT, tlv); spec->vmaster_vol = snd_ctl_make_virtual_master("Master Playback Volume", tlv); - err = snd_hda_ctl_add(codec, spec->vmaster_vol); + err = snd_hda_ctl_add(codec, dac, spec->vmaster_vol); if (err < 0) return err; return 0; @@ -756,13 +756,13 @@ static int build_input(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = (long)spec->capture_bind[i]; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } if (spec->num_inputs > 1 && !spec->mic_detect) { - err = snd_hda_ctl_add(codec, + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cs_capture_source, codec)); if (err < 0) return err; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c04e0e0f655..fff9de245646 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2461,7 +2461,8 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, + get_amp_nid_(spec->beep_amp), kctl); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 87ba239ff1c9..a3872b90d6ed 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1085,7 +1085,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (!spec->auto_mic && spec->num_dmuxes > 0 && snd_hda_get_bool_hint(codec, "separate_dmux") == 1) { stac_dmux_mixer.count = spec->num_dmuxes; - err = snd_hda_ctl_add(codec, + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&stac_dmux_mixer, codec)); if (err < 0) return err; @@ -1101,7 +1101,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) spec->spdif_mute = 1; } stac_smux_mixer.count = spec->num_smuxes; - err = snd_hda_ctl_add(codec, + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&stac_smux_mixer, codec)); if (err < 0) return err; -- cgit v1.2.2 From 4d02d1b638af580ae3d69367248539a8b3893064 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 12 Nov 2009 10:15:48 +0100 Subject: ALSA: hda - proc - add support for dynamic controls to mixer<->NID mapping This patch adds support for dynamically created controls to proc codec file (Control: lines). Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 7 ++++++- sound/pci/hda/hda_local.h | 3 +++ sound/pci/hda/patch_analog.c | 2 ++ sound/pci/hda/patch_realtek.c | 2 ++ sound/pci/hda/patch_sigmatel.c | 10 +++++++--- sound/pci/hda/patch_via.c | 2 ++ 6 files changed, 22 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1ed1d88e1834..d71e651046eb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1523,6 +1523,11 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, int err; struct hda_nid_item *item; + if (kctl->id.subdevice & (1<<31)) { + if (nid == 0) + nid = kctl->id.subdevice & 0xffff; + kctl->id.subdevice = 0; + } err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) return err; @@ -3160,7 +3165,7 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config); */ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) { - int err; + int err; for (; knew->name; knew++) { struct snd_kcontrol *kctl; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e6a0918f70d3..3bfcf42ff6cf 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -33,6 +33,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .subdevice = (1<<31)|(nid), \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -53,6 +54,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .subdevice = (1<<31)|(nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -69,6 +71,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .subdevice = (1<<31)|(nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index ef3383912b6e..2d345606265b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2571,6 +2571,8 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; + if (get_amp_nid_(val)) + knew->subdevice = (1<<31)|get_amp_nid_(val); knew->private_value = val; return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fff9de245646..c0a98e724a25 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4323,6 +4323,8 @@ static int add_control(struct alc_spec *spec, int type, const char *name, knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; + if (get_amp_nid_(val)) + knew->subdevice = (1<<31)|get_amp_nid_(val); knew->private_value = val; return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a3872b90d6ed..d2ddb959c290 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2671,7 +2671,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { static struct snd_kcontrol_new * stac_control_new(struct sigmatel_spec *spec, struct snd_kcontrol_new *ktemp, - const char *name) + const char *name, + hda_nid_t nid) { struct snd_kcontrol_new *knew; @@ -2687,6 +2688,8 @@ stac_control_new(struct sigmatel_spec *spec, spec->kctls.alloced--; return NULL; } + if (nid) + knew->subdevice = (1<<31)|nid; return knew; } @@ -2695,7 +2698,8 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec, int idx, const char *name, unsigned long val) { - struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name); + struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, + get_amp_nid_(val)); if (!knew) return -ENOMEM; knew->index = idx; @@ -2766,7 +2770,7 @@ static int stac92xx_add_input_source(struct sigmatel_spec *spec) if (!spec->num_adcs || imux->num_items <= 1) return 0; /* no need for input source control */ knew = stac_control_new(spec, &stac_input_src_temp, - stac_input_src_temp.name); + stac_input_src_temp.name, 0); if (!knew) return -ENOMEM; knew->count = spec->num_adcs; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 5a856009c916..14219d898b2e 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -442,6 +442,8 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; + if (get_amp_nid_(val)) + knew->subdevice = (1<<31)|get_amp_nid_(val); knew->private_value = val; return 0; } -- cgit v1.2.2 From 9c96fa599fe4f0ccc6e3e606df6652335afe28e8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Nov 2009 11:25:33 +0100 Subject: ALSA: hda - Get rid of magic digits for subdev hack Define a proper const for a magic 31bit flag for subdev / NID setup with a brief comment. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_local.h | 15 ++++++++++++--- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 2 +- 6 files changed, 17 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d71e651046eb..5e21b35207ab 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1523,7 +1523,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, int err; struct hda_nid_item *item; - if (kctl->id.subdevice & (1<<31)) { + if (kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) { if (nid == 0) nid = kctl->id.subdevice & 0xffff; kctl->id.subdevice = 0; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3bfcf42ff6cf..4e77f4747291 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -23,6 +23,15 @@ #ifndef __SOUND_HDA_LOCAL_H #define __SOUND_HDA_LOCAL_H +/* We abuse kcontrol_new.subdev field to pass the NID corresponding to + * the given new control. If id.subdev has a bit flag HDA_SUBDEV_NID_FLAG, + * snd_hda_ctl_add() takes the lower-bit subdev value as a valid NID. + * + * Note that the subdevice field is cleared again before the real registration + * in snd_hda_ctl_add(), so that this value won't appear in the outside. + */ +#define HDA_SUBDEV_NID_FLAG (1U << 31) + /* * for mixer controls */ @@ -33,7 +42,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = (1<<31)|(nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -54,7 +63,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = (1<<31)|(nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -71,7 +80,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = (1<<31)|(nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2d345606265b..ceb0c603da04 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2572,7 +2572,7 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, if (! knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = (1<<31)|get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); knew->private_value = val; return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0a98e724a25..eee3143eef75 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4324,7 +4324,7 @@ static int add_control(struct alc_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = (1<<31)|get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); knew->private_value = val; return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d2ddb959c290..7f76a97954f9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2689,7 +2689,7 @@ stac_control_new(struct sigmatel_spec *spec, return NULL; } if (nid) - knew->subdevice = (1<<31)|nid; + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; return knew; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 14219d898b2e..0c621d74b165 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -443,7 +443,7 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = (1<<31)|get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); knew->private_value = val; return 0; } -- cgit v1.2.2 From 85dd662ff4d2967084acfc761a33717383297e42 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 11 Nov 2009 13:49:07 +0100 Subject: ALSA: hda - move snd_hda_pcm_type_name from hda_codec.h to hda_local.h The snd_hda_pcm_type_name array is local only. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_local.h | 2 ++ 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 51920563bc7f..be6c5f443cd9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -898,7 +898,6 @@ int snd_hda_codec_build_controls(struct hda_codec *codec); /* * PCM */ -extern const char *snd_hda_pcm_type_name[]; int snd_hda_build_pcms(struct hda_bus *bus); int snd_hda_codec_build_pcms(struct hda_codec *codec); void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4e77f4747291..7c049839ea26 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -92,6 +92,8 @@ #define HDA_CODEC_MUTE_BEEP(xname, nid, xindex, direction) \ HDA_CODEC_MUTE_BEEP_MONO(xname, nid, 3, xindex, direction) +extern const char *snd_hda_pcm_type_name[]; + int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, -- cgit v1.2.2 From d5191e50b251594bdde10d4839a952ff1646ef62 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Nov 2009 14:58:17 +0100 Subject: ALSA: hda - Update / add kerneldoc comments to exported functions Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 432 +++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 391 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5e21b35207ab..e344235da491 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -94,6 +94,13 @@ static void hda_keep_power_on(struct hda_codec *codec); static inline void hda_keep_power_on(struct hda_codec *codec) {} #endif +/** + * snd_hda_get_jack_location - Give a location string of the jack + * @cfg: pin default config value + * + * Parse the pin default config value and returns the string of the + * jack location, e.g. "Rear", "Front", etc. + */ const char *snd_hda_get_jack_location(u32 cfg) { static char *bases[7] = { @@ -121,6 +128,13 @@ const char *snd_hda_get_jack_location(u32 cfg) } EXPORT_SYMBOL_HDA(snd_hda_get_jack_location); +/** + * snd_hda_get_jack_connectivity - Give a connectivity string of the jack + * @cfg: pin default config value + * + * Parse the pin default config value and returns the string of the + * jack connectivity, i.e. external or internal connection. + */ const char *snd_hda_get_jack_connectivity(u32 cfg) { static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; @@ -129,6 +143,13 @@ const char *snd_hda_get_jack_connectivity(u32 cfg) } EXPORT_SYMBOL_HDA(snd_hda_get_jack_connectivity); +/** + * snd_hda_get_jack_type - Give a type string of the jack + * @cfg: pin default config value + * + * Parse the pin default config value and returns the string of the + * jack type, i.e. the purpose of the jack, such as Line-Out or CD. + */ const char *snd_hda_get_jack_type(u32 cfg) { static char *jack_types[16] = { @@ -822,6 +843,16 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, return 0; } +/** + * snd_hda_codec_set_pincfg - Override a pin default configuration + * @codec: the HDA codec + * @nid: NID to set the pin config + * @cfg: the pin default config value + * + * Override a pin default configuration value in the cache. + * This value can be read by snd_hda_codec_get_pincfg() in a higher + * priority than the real hardware value. + */ int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, unsigned int cfg) { @@ -829,7 +860,15 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); -/* get the current pin config value of the given pin NID */ +/** + * snd_hda_codec_get_pincfg - Obtain a pin-default configuration + * @codec: the HDA codec + * @nid: NID to get the pin config + * + * Get the current pin config value of the given pin NID. + * If the pincfg value is cached or overridden via sysfs or driver, + * returns the cached value. + */ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) { struct hda_pincfg *pin; @@ -1028,6 +1067,15 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr } EXPORT_SYMBOL_HDA(snd_hda_codec_new); +/** + * snd_hda_codec_configure - (Re-)configure the HD-audio codec + * @codec: the HDA codec + * + * Start parsing of the given codec tree and (re-)initialize the whole + * patch instance. + * + * Returns 0 if successful or a negative error code. + */ int snd_hda_codec_configure(struct hda_codec *codec) { int err; @@ -1090,6 +1138,11 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream); +/** + * snd_hda_codec_cleanup_stream - clean up the codec for closing + * @codec: the CODEC to clean up + * @nid: the NID to clean up + */ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) { if (!nid) @@ -1165,8 +1218,17 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key) return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key); } -/* - * query AMP capabilities for the given widget and direction +/** + * query_amp_caps - query AMP capabilities + * @codec: the HD-auio codec + * @nid: the NID to query + * @direction: either #HDA_INPUT or #HDA_OUTPUT + * + * Query AMP capabilities for the given widget and direction. + * Returns the obtained capability bits. + * + * When cap bits have been already read, this doesn't read again but + * returns the cached value. */ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { @@ -1189,6 +1251,19 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) } EXPORT_SYMBOL_HDA(query_amp_caps); +/** + * snd_hda_override_amp_caps - Override the AMP capabilities + * @codec: the CODEC to clean up + * @nid: the NID to clean up + * @direction: either #HDA_INPUT or #HDA_OUTPUT + * @caps: the capability bits to set + * + * Override the cached AMP caps bits value by the given one. + * This function is useful if the driver needs to adjust the AMP ranges, + * e.g. limit to 0dB, etc. + * + * Returns zero if successful or a negative error code. + */ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps) { @@ -1224,6 +1299,17 @@ static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); } +/** + * snd_hda_query_pin_caps - Query PIN capabilities + * @codec: the HD-auio codec + * @nid: the NID to query + * + * Query PIN capabilities for the given widget. + * Returns the obtained capability bits. + * + * When cap bits have been already read, this doesn't read again but + * returns the cached value. + */ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) { return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), @@ -1271,8 +1357,15 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, info->vol[ch] = val; } -/* - * read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. +/** + * snd_hda_codec_amp_read - Read AMP value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @index: the index value (only for input direction) + * + * Read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. */ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index) @@ -1285,8 +1378,18 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read); -/* - * update the AMP value, mask = bit mask to set, val = the value +/** + * snd_hda_codec_amp_update - update the AMP value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the AMP value with a bit mask. + * Returns 0 if the value is unchanged, 1 if changed. */ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) @@ -1305,8 +1408,17 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_update); -/* - * update the AMP stereo with the same mask and value +/** + * snd_hda_codec_amp_stereo - update the AMP stereo values + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the AMP values like snd_hda_codec_amp_update(), but for a + * stereo widget with the same mask and value. */ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, int direction, int idx, int mask, int val) @@ -1320,7 +1432,12 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo); #ifdef SND_HDA_NEEDS_RESUME -/* resume the all amp commands from the cache */ +/** + * snd_hda_codec_resume_amp - Resume all AMP commands from the cache + * @codec: HD-audio codec + * + * Resume the all amp commands from the cache. + */ void snd_hda_codec_resume_amp(struct hda_codec *codec) { struct hda_amp_info *buffer = codec->amp_cache.buf.list; @@ -1346,7 +1463,12 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp); #endif /* SND_HDA_NEEDS_RESUME */ -/* volume */ +/** + * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1402,6 +1524,12 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, HDA_AMP_VOLMASK, val); } +/** + * snd_hda_mixer_amp_volume_get - Get callback for a standard AMP mixer volume + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1421,6 +1549,12 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get); +/** + * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1445,6 +1579,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_put); +/** + * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) { @@ -1474,8 +1614,16 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_tlv); -/* - * set (static) TLV for virtual master volume; recalculated as max 0dB +/** + * snd_hda_set_vmaster_tlv - Set TLV for a virtual master control + * @codec: HD-audio codec + * @nid: NID of a reference widget + * @dir: #HDA_INPUT or #HDA_OUTPUT + * @tlv: TLV data to be stored, at least 4 elements + * + * Set (static) TLV data for a virtual master volume using the AMP caps + * obtained from the reference NID. + * The volume range is recalculated as if the max volume is 0dB. */ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int *tlv) @@ -1509,6 +1657,13 @@ _snd_hda_find_mixer_ctl(struct hda_codec *codec, return snd_ctl_find_id(codec->bus->card, &id); } +/** + * snd_hda_find_mixer_ctl - Find a mixer control element with the given name + * @codec: HD-audio codec + * @name: ctl id name string + * + * Get the control element with the given id string and IFACE_MIXER. + */ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name) { @@ -1516,7 +1671,24 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); -/* Add a control element and assign to the codec */ +/** + * snd_hda_ctl-add - Add a control element and assign to the codec + * @codec: HD-audio codec + * @nid: corresponding NID (optional) + * @kctl: the control element to assign + * + * Add the given control element to an array inside the codec instance. + * All control elements belonging to a codec are supposed to be added + * by this function so that a proper clean-up works at the free or + * reconfiguration time. + * + * If non-zero @nid is passed, the NID is assigned to the control element. + * The assignment is shown in the codec proc file. + * + * snd_hda_ctl_add() checks the control subdev id field whether + * #HDA_SUBDEV_NID_FLAG bit is set. If set (and @nid is zero), the lower + * bits value is taken as the NID to assign. + */ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, struct snd_kcontrol *kctl) { @@ -1540,7 +1712,10 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); -/* Clear all controls assigned to the given codec */ +/** + * snd_hda_ctls_clear - Clear all controls assigned to the given codec + * @codec: HD-audio codec + */ void snd_hda_ctls_clear(struct hda_codec *codec) { int i; @@ -1572,6 +1747,16 @@ static void hda_unlock_devices(struct snd_card *card) spin_unlock(&card->files_lock); } +/** + * snd_hda_codec_reset - Clear all objects assigned to the codec + * @codec: HD-audio codec + * + * This frees the all PCM and control elements assigned to the codec, and + * clears the caches and restores the pin default configurations. + * + * When a device is being used, it returns -EBSY. If successfully freed, + * returns zero. + */ int snd_hda_codec_reset(struct hda_codec *codec) { struct snd_card *card = codec->bus->card; @@ -1635,7 +1820,22 @@ int snd_hda_codec_reset(struct hda_codec *codec) return 0; } -/* create a virtual master control and add slaves */ +/** + * snd_hda_add_vmaster - create a virtual master control and add slaves + * @codec: HD-audio codec + * @name: vmaster control name + * @tlv: TLV data (optional) + * @slaves: slave control names (optional) + * + * Create a virtual master control with the given name. The TLV data + * must be either NULL or a valid data. + * + * @slaves is a NULL-terminated array of strings, each of which is a + * slave control name. All controls with these names are assigned to + * the new virtual master control. + * + * This function returns zero if successful or a negative error code. + */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves) { @@ -1677,7 +1877,12 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, } EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); -/* switch */ +/** + * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1691,6 +1896,12 @@ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_info); +/** + * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1711,6 +1922,12 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get); +/** + * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1742,6 +1959,12 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); +/** + * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch + * + * This function calls snd_hda_enable_beep_device(), which behaves differently + * depending on beep_mode option. + */ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1762,6 +1985,12 @@ EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); #define AMP_VAL_IDX_SHIFT 19 #define AMP_VAL_IDX_MASK (0x0f<<19) +/** + * snd_hda_mixer_bind_switch_get - Get callback for a bound volume control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_MUTE*() macros. + */ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1779,6 +2008,12 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get); +/** + * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_MUTE*() macros. + */ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1803,8 +2038,11 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put); -/* - * generic bound volume/swtich controls +/** + * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. */ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1823,6 +2061,12 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info); +/** + * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. + */ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1840,6 +2084,12 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get); +/** + * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. + */ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1863,6 +2113,12 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put); +/** + * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() macro. + */ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { @@ -2185,6 +2441,11 @@ static struct snd_kcontrol_new spdif_share_sw = { .put = spdif_share_sw_put, }; +/** + * snd_hda_create_spdif_share_sw - create Default PCM switch + * @codec: the HDA codec + * @mout: multi-out instance + */ int snd_hda_create_spdif_share_sw(struct hda_codec *codec, struct hda_multi_out *mout) { @@ -2352,7 +2613,12 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); -/* resume the all commands from the cache */ +/** + * snd_hda_codec_resume_cache - Resume the all commands from the cache + * @codec: HD-audio codec + * + * Execute all verbs recorded in the command caches to resume. + */ void snd_hda_codec_resume_cache(struct hda_codec *codec) { struct hda_cache_head *buffer = codec->cmd_cache.buf.list; @@ -2778,8 +3044,12 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, } /** - * snd_hda_is_supported_format - check whether the given node supports - * the format val + * snd_hda_is_supported_format - Check the validity of the format + * @codec: HD-audio codec + * @nid: NID to check + * @format: the HD-audio format value to check + * + * Check whether the given node supports the format value. * * Returns 1 if supported, 0 if not. */ @@ -2899,6 +3169,7 @@ static int set_pcm_default_values(struct hda_codec *codec, return 0; } +/* global */ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { "Audio", "SPDIF", "HDMI", "Modem" }; @@ -3216,6 +3487,7 @@ static void hda_keep_power_on(struct hda_codec *codec) codec->power_jiffies = jiffies; } +/* update the power on/off account with the current jiffies */ void snd_hda_update_power_acct(struct hda_codec *codec) { unsigned long delta = jiffies - codec->power_jiffies; @@ -3226,6 +3498,13 @@ void snd_hda_update_power_acct(struct hda_codec *codec) codec->power_jiffies += delta; } +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ void snd_hda_power_up(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; @@ -3248,9 +3527,13 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - +/** + * snd_hda_power_down - Power-down the codec + * @codec: HD-audio codec + * + * Decrement the power-up counter and schedules the power-off work if + * the counter rearches to zero. + */ void snd_hda_power_down(struct hda_codec *codec) { --codec->power_count; @@ -3264,6 +3547,19 @@ void snd_hda_power_down(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_power_down); +/** + * snd_hda_check_amp_list_power - Check the amp list and update the power + * @codec: HD-audio codec + * @check: the object containing an AMP list and the status + * @nid: NID to check / update + * + * Check whether the given NID is in the amp list. If it's in the list, + * check the current AMP status, and update the the power-status according + * to the mute status. + * + * This function is supposed to be set or called from the check_power_status + * patch ops. + */ int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, hda_nid_t nid) @@ -3305,6 +3601,10 @@ EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); /* * Channel mode helper */ + +/** + * snd_hda_ch_mode_info - Info callback helper for the channel mode enum + */ int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo, const struct hda_channel_mode *chmode, @@ -3321,6 +3621,9 @@ int snd_hda_ch_mode_info(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_ch_mode_info); +/** + * snd_hda_ch_mode_get - Get callback helper for the channel mode enum + */ int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, @@ -3339,6 +3642,9 @@ int snd_hda_ch_mode_get(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_ch_mode_get); +/** + * snd_hda_ch_mode_put - Put callback helper for the channel mode enum + */ int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, @@ -3363,6 +3669,10 @@ EXPORT_SYMBOL_HDA(snd_hda_ch_mode_put); /* * input MUX helper */ + +/** + * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + */ int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo) { @@ -3381,6 +3691,9 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, } EXPORT_SYMBOL_HDA(snd_hda_input_mux_info); +/** + * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + */ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, struct snd_ctl_elem_value *ucontrol, @@ -3440,7 +3753,10 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) } } -/* call each reboot notifier */ +/** + * snd_hda_bus_reboot_notify - call the reboot notifier of each codec + * @bus: HD-audio bus + */ void snd_hda_bus_reboot_notify(struct hda_bus *bus) { struct hda_codec *codec; @@ -3458,8 +3774,8 @@ void snd_hda_bus_reboot_notify(struct hda_bus *bus) } EXPORT_SYMBOL_HDA(snd_hda_bus_reboot_notify); -/* - * open the digital out in the exclusive mode +/** + * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode */ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) @@ -3474,6 +3790,9 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_open); +/** + * snd_hda_multi_out_dig_prepare - prepare the digital out stream + */ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct hda_multi_out *mout, unsigned int stream_tag, @@ -3487,6 +3806,9 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare); +/** + * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream + */ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) { @@ -3497,8 +3819,8 @@ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup); -/* - * release the digital out +/** + * snd_hda_multi_out_dig_close - release the digital out stream */ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) @@ -3510,8 +3832,12 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_close); -/* - * set up more restrictions for analog out +/** + * snd_hda_multi_out_analog_open - open analog outputs + * + * Open analog outputs and set up the hw-constraints. + * If the digital outputs can be opened as slave, open the digital + * outputs, too. */ int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, @@ -3556,9 +3882,11 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_open); -/* - * set up the i/o for analog out - * when the digital out is available, copy the front out to digital out, too. +/** + * snd_hda_multi_out_analog_prepare - Preapre the analog outputs. + * + * Set up the i/o for analog out. + * When the digital out is available, copy the front out to digital out, too. */ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, @@ -3615,8 +3943,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); -/* - * clean up the setting for analog out +/** + * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out */ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -4002,8 +4330,14 @@ EXPORT_SYMBOL_HDA(snd_hda_resume); * generic arrays */ -/* get a new element from the given array - * if it exceeds the pre-allocated array size, re-allocate the array +/** + * snd_array_new - get a new element from the given array + * @array: the array object + * + * Get a new element from the given array. If it exceeds the + * pre-allocated array size, re-allocate the array. + * + * Returns NULL if allocation failed. */ void *snd_array_new(struct snd_array *array) { @@ -4027,7 +4361,10 @@ void *snd_array_new(struct snd_array *array) } EXPORT_SYMBOL_HDA(snd_array_new); -/* free the given array elements */ +/** + * snd_array_free - free the given array elements + * @array: the array object + */ void snd_array_free(struct snd_array *array) { kfree(array->list); @@ -4037,7 +4374,12 @@ void snd_array_free(struct snd_array *array) } EXPORT_SYMBOL_HDA(snd_array_free); -/* +/** + * snd_print_pcm_rates - Print the supported PCM rates to the string buffer + * @pcm: PCM caps bits + * @buf: the string buffer to write + * @buflen: the max buffer length + * * used by hda_proc.c and hda_eld.c */ void snd_print_pcm_rates(int pcm, char *buf, int buflen) @@ -4056,6 +4398,14 @@ void snd_print_pcm_rates(int pcm, char *buf, int buflen) } EXPORT_SYMBOL_HDA(snd_print_pcm_rates); +/** + * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer + * @pcm: PCM caps bits + * @buf: the string buffer to write + * @buflen: the max buffer length + * + * used by hda_proc.c and hda_eld.c + */ void snd_print_pcm_bits(int pcm, char *buf, int buflen) { static unsigned int bits[] = { 8, 16, 20, 24, 32 }; -- cgit v1.2.2 From 9bb1fe390de3e1def0dd162dbdaf62e0981105fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Nov 2009 15:33:49 +0100 Subject: ALSA: hda - Fix beep_mode option value The beep_mode option value was wrongly defined: it must be 0 = off and 1 = on. Also, evaluate the beep_mode value at snd_hda_attach_beep_device() properly so that no device is created when beep_mode=0 is given. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 4 +++- sound/pci/hda/hda_beep.h | 4 ++-- 2 files changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 9e48798b415b..5fe34a8d8c81 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -215,7 +215,9 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) struct hda_beep *beep; if (!snd_hda_get_bool_hint(codec, "beep")) - return 0; /* disabled explicitly */ + return 0; /* disabled explicitly by hints */ + if (codec->beep_mode == HDA_BEEP_MODE_OFF) + return 0; /* disabled by module option */ beep = kzalloc(sizeof(*beep), GFP_KERNEL); if (beep == NULL) diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 17dd1c325e32..f1de1bac042c 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -24,8 +24,8 @@ #include "hda_codec.h" -#define HDA_BEEP_MODE_ON 0 -#define HDA_BEEP_MODE_OFF 1 +#define HDA_BEEP_MODE_OFF 0 +#define HDA_BEEP_MODE_ON 1 #define HDA_BEEP_MODE_SWREG 2 /* beep information */ -- cgit v1.2.2 From 67d634c07afd8f70973d925463e775fdb89ad536 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Nov 2009 15:35:59 +0100 Subject: ALSA: hda - Fix build errors with CONFIG_SND_HDA_INPUT_BEEP=n Disable beep-related codes when CONFIG_SND_HDA_INPUT_BEEP isn't set. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/hda_local.h | 8 ++++++++ sound/pci/hda/patch_analog.c | 6 ++++++ sound/pci/hda/patch_realtek.c | 8 ++++++++ 4 files changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e344235da491..2be61b31fb3c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1959,6 +1959,7 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); +#ifdef CONFIG_SND_HDA_INPUT_BEEP /** * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch * @@ -1975,6 +1976,7 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); +#endif /* CONFIG_SND_HDA_INPUT_BEEP */ /* * bound volume controls diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 7c049839ea26..d4a3d0942c00 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -77,6 +77,7 @@ /* stereo mute switch */ #define HDA_CODEC_MUTE(xname, nid, xindex, direction) \ HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction) +#ifdef CONFIG_SND_HDA_INPUT_BEEP /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ @@ -85,6 +86,11 @@ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } +#else +/* no digital beep - just the standard one */ +#define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, ch, xidx, dir) \ + HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, ch, xidx, dir) +#endif /* CONFIG_SND_HDA_INPUT_BEEP */ /* special beep mono mute switch */ #define HDA_CODEC_MUTE_BEEP_MONO(xname, nid, channel, xindex, direction) \ HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, 0, nid, channel, xindex, direction) @@ -108,8 +114,10 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +#ifdef CONFIG_SND_HDA_INPUT_BEEP int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +#endif /* lowlevel accessor with caching; use carefully */ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index ceb0c603da04..8a1064bdf4c6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -156,6 +156,7 @@ static const char *ad_slave_sws[] = { static void ad198x_free_kctls(struct hda_codec *codec); +#ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ static struct snd_kcontrol_new ad_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), @@ -165,6 +166,9 @@ static struct snd_kcontrol_new ad_beep_mixer[] = { #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ +#else +#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#endif static int ad198x_build_controls(struct hda_codec *codec) { @@ -194,6 +198,7 @@ static int ad198x_build_controls(struct hda_codec *codec) } /* create beep controls if needed */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP if (spec->beep_amp) { struct snd_kcontrol_new *knew; for (knew = ad_beep_mixer; knew->name; knew++) { @@ -209,6 +214,7 @@ static int ad198x_build_controls(struct hda_codec *codec) return err; } } +#endif /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee3143eef75..ef7d21097eeb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2410,12 +2410,14 @@ static const char *alc_slave_sws[] = { static void alc_free_kctls(struct hda_codec *codec); +#ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ static struct snd_kcontrol_new alc_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), { } /* end */ }; +#endif static int alc_build_controls(struct hda_codec *codec) { @@ -2452,6 +2454,7 @@ static int alc_build_controls(struct hda_codec *codec) return err; } +#ifdef CONFIG_SND_HDA_INPUT_BEEP /* create beep controls if needed */ if (spec->beep_amp) { struct snd_kcontrol_new *knew; @@ -2467,6 +2470,7 @@ static int alc_build_controls(struct hda_codec *codec) return err; } } +#endif /* if we have no master control, let's create it */ if (!spec->no_analog && @@ -4780,8 +4784,12 @@ static void set_capture_mixer(struct hda_codec *codec) } } +#ifdef CONFIG_SND_HDA_INPUT_BEEP #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) +#else +#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#endif /* * OK, here we have finally the patch for ALC880 -- cgit v1.2.2 From 8df89bc35c188e389295eaf7917653f13c83ce70 Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Mon, 16 Nov 2009 16:19:25 +0200 Subject: ASoC: OMAP: enable Overo driver for CM-T35 Signed-off-by: Mike Rapoport Acked-by: Liam Girdwood Acked-by: Steve Sakoman Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 7 ++++--- sound/soc/omap/overo.c | 4 ++-- 2 files changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index bb5731a22bed..4dc6b15a852f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -43,12 +43,13 @@ config SND_OMAP_SOC_OSK5912 Say Y if you want to add support for SoC audio on osk5912. config SND_OMAP_SOC_OVERO - tristate "SoC Audio support for Gumstix Overo" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO + tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35" + depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Gumstix Overo. + Say Y if you want to add support for SoC audio on the + Gumstix Overo or CompuLab CM-T35 config SND_OMAP_SOC_OMAP2EVM tristate "SoC Audio support for OMAP2EVM board" diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index ec4f8fd8b3a2..97a4d6308bd6 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -107,8 +107,8 @@ static int __init overo_soc_init(void) { int ret; - if (!machine_is_overo()) { - pr_debug("Not Overo!\n"); + if (!(machine_is_overo() || machine_is_cm_t35())) { + pr_debug("Incomatible machine!\n"); return -ENODEV; } printk(KERN_INFO "overo SoC init\n"); -- cgit v1.2.2 From 02bb57aeb092cbb0dfdb50c6026dbd0c60af7644 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Mon, 16 Nov 2009 17:05:02 +0100 Subject: sound: OSS: keep index within bounds of midi_devs[] When the {orig,midi}_dev equals num_midis, that's one too large already. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/midi_synth.c | 2 +- sound/oss/mpu401.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 9e450988ed36..3bc7104c5379 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -426,7 +426,7 @@ midi_synth_open(int dev, int mode) int err; struct midi_input_info *inc; - if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL) + if (orig_dev < 0 || orig_dev >= num_midis || midi_devs[orig_dev] == NULL) return -ENXIO; midi2synth[orig_dev] = dev; diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 734b8f9e2f78..0af9d24feb8f 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -770,7 +770,7 @@ static int mpu_synth_ioctl(int dev, unsigned int cmd, void __user *arg) midi_dev = synth_devs[dev]->midi_dev; - if (midi_dev < 0 || midi_dev > num_midis || midi_devs[midi_dev] == NULL) + if (midi_dev < 0 || midi_dev >= num_midis || midi_devs[midi_dev] == NULL) return -ENXIO; devc = &dev_conf[midi_dev]; -- cgit v1.2.2 From baac805fc591b562f22d8f1cd0b65cdbbe9e9518 Mon Sep 17 00:00:00 2001 From: Timothy Knoll Date: Mon, 16 Nov 2009 19:55:46 -0500 Subject: sound: Kconfig typo fix Fix a typo in the help text in sound/Kconfig. Signed-off-by: Timothy Knoll Signed-off-by: Takashi Iwai --- sound/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/Kconfig b/sound/Kconfig index 439e15c8faa3..b3e53e616ec9 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -58,7 +58,7 @@ config SOUND_OSS_CORE_PRECLAIM Please read Documentation/feature-removal-schedule.txt for details. - If unusre, say Y. + If unsure, say Y. source "sound/oss/dmasound/Kconfig" -- cgit v1.2.2 From f9ede4eca01cc64ce37549c282b6fde727c0ec84 Mon Sep 17 00:00:00 2001 From: Marin Mitov Date: Mon, 16 Nov 2009 21:39:26 +0200 Subject: ASoC: Use DMA_BIT_MASK(32) instead of deprecated DMA_32BIT_MASK Signed-off-by: Marin Mitov Signed-off-by: Takashi Iwai --- sound/soc/s6000/s6000-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 83b8028e209d..0eb1722f6581 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -423,7 +423,7 @@ static void s6000_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; +static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -435,7 +435,7 @@ static int s6000_pcm_new(struct snd_card *card, if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (params->dma_in) { s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), -- cgit v1.2.2 From c5b5165ce28099484d5fa733abeae48540680440 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Nov 2009 16:01:58 +0100 Subject: ALSA: hda - Disable default quirk for Sony VAIO with ALC262 codec The ALC262 has a quirk entry matching with all Sony Vaio laptops to use model=sony-assamd as default. But, model=auto works much better for new models in the recent driver versions, thus it's safer to disable that default quirk entry. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba339d745aab..578420523606 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11471,8 +11471,10 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO), +#if 0 /* disable the quirk since model=auto works better in recent versions */ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), +#endif SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), -- cgit v1.2.2 From b753e03e5e7c6ee60e81cd6335c80dc26519f9d0 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Tue, 17 Nov 2009 18:34:54 +0100 Subject: ALSA: cs4236: update control names Update control names to be more closer to their meaning. Change the "Mono" name to the "Beep" as this line is usually used to forward the PC beeper signal to sound card's output. Update names for both cs423x and wss. Clean up cs4235 controls according to the cs4235 doc. Rename some of the cs4235 controls to be consistent with the cs4236's ones. Also, delete one misnamed cs4231 register define. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/cs423x/cs4236_lib.c | 49 +++++++++++++++++++------------------------ sound/isa/wss/wss_lib.c | 8 +++---- 2 files changed, 25 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 1b1ad1cad328..4c4024a73c6b 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -777,7 +777,7 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), -CS4236_DOUBLE("Mic Playback Boost", 0, +CS4236_DOUBLE("Mic Playback Boost (+20dB)", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, @@ -798,10 +798,10 @@ WSS_DOUBLE("CD Capture Switch", 0, CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), -CS4236_DOUBLE1("Mono Playback Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -WSS_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), WSS_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), @@ -815,31 +815,27 @@ CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, static struct snd_kcontrol_new snd_cs4235_controls[] = { -WSS_DOUBLE("Master Switch", 0, +WSS_DOUBLE("Master Playback Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -WSS_DOUBLE("Master Volume", 0, +WSS_DOUBLE("Master Playback Volume", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), CS4235_OUTPUT_ACCU("Playback Volume", 0), -CS4236_DOUBLE("Master Digital Playback Switch", 0, - CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), -CS4236_DOUBLE("Master Digital Capture Switch", 0, - CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), - -WSS_DOUBLE("Master Digital Playback Switch", 1, +WSS_DOUBLE("Synth Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Master Digital Capture Switch", 1, +WSS_DOUBLE("Synth Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -WSS_DOUBLE("Master Digital Volume", 1, +WSS_DOUBLE("Synth Volume", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), CS4236_DOUBLE("Capture Volume", 0, CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), -WSS_DOUBLE("PCM Switch", 0, +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("PCM Capture Switch", 0, + CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), WSS_DOUBLE("PCM Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), @@ -855,28 +851,25 @@ CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), -CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0), +CS4236_SINGLE("Mic Boost (+20dB)", 0, CS4236_LEFT_MIC, 5, 1, 0), -WSS_DOUBLE("Aux Playback Switch", 0, +WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 0, +WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 0, +WSS_DOUBLE("Line Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -WSS_DOUBLE("Aux Playback Switch", 1, +WSS_DOUBLE("CD Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 1, +WSS_DOUBLE("CD Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 1, +WSS_DOUBLE("CD Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -CS4236_DOUBLE1("Master Mono Switch", 0, - CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), - -CS4236_DOUBLE1("Mono Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), WSS_DOUBLE("Analog Loopback Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 705db0924375..5b9d6c18bc45 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2224,7 +2224,7 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, -WSS_DOUBLE("Mic Boost", 0, +WSS_DOUBLE("Mic Boost (+20dB)", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), @@ -2235,14 +2235,14 @@ WSS_DOUBLE("Line Playback Switch", 0, WSS_DOUBLE_TLV("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, db_scale_5bit_12db_max), -WSS_SINGLE("Mono Playback Switch", 0, +WSS_SINGLE("Beep Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE_TLV("Mono Playback Volume", 0, +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), }; -- cgit v1.2.2 From b67cad932c4e45edca2f4da2ee4f46001ba17363 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Tue, 17 Nov 2009 18:35:41 +0100 Subject: ALSA: opti-miro: use variables directly in the probe function Use the fm_port and mpu_port variables directly in a probe function. This completely eliminates a need to copy the fm_port value to the snd_miro structure. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 33 +++++++++++++++++---------------- 1 file changed, 17 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 02e30d7c6a93..b8170adeeff6 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -118,8 +118,6 @@ struct snd_miro { int dma1; int dma2; - long fm_port; - long mpu_port; int mpu_irq; @@ -757,7 +755,6 @@ static int __devinit snd_miro_init(struct snd_miro *chip, chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; - chip->fm_port = -1; chip->mpu_port = -1; chip->mpu_irq = -1; @@ -1261,7 +1258,6 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } miro->wss_base = port; - miro->fm_port = fm_port; miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; @@ -1276,11 +1272,12 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } } - if (miro->mpu_port == SNDRV_AUTO_PORT) { - if ((miro->mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2)) < 0) { + if (mpu_port == SNDRV_AUTO_PORT) { + mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); + if (mpu_port < 0) { snd_card_free(card); snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY; + return -EBUSY } } if (miro->irq == SNDRV_AUTO_IRQ) { @@ -1380,20 +1377,24 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) card->shortname, miro->name, pcm->name, miro->wss_base + 4, miro->irq, miro->dma1, miro->dma2); - if (miro->mpu_port <= 0 || miro->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - miro->mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, - &rmidi))) - snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", miro->mpu_port); + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, + &rmidi); + if (error < 0) + snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", + mpu_port); + } - if (miro->fm_port > 0 && miro->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; struct snd_opl4 *opl4; - if (snd_opl4_create(card, miro->fm_port, miro->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) - snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", miro->fm_port); + snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", + fm_port); } if ((error = snd_set_aci_init_values(miro)) < 0) { -- cgit v1.2.2 From 6f539a98614a014a7d6b64ab62b0dddb14e2d8cc Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:37:59 +0800 Subject: ALSA: intelhdmi - fix audio infoframe fill size MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Reported-by: David Härdeman Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 4f25f08d332b..ad1aa5d87dda 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -509,12 +509,12 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec, hdmi_debug_dip_size(codec, pin_nid); hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ - for (i = 0; i < sizeof(ai); i++) + for (i = 0; i < sizeof(*ai); i++) sum += params[i]; ai->checksum = - sum; hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(ai); i++) + for (i = 0; i < sizeof(*ai); i++) hdmi_write_dip_byte(codec, pin_nid, params[i]); } -- cgit v1.2.2 From 1e7c10fefadb42d9300305c7de57bea365855e9b Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:00 +0800 Subject: ALSA: intelhdmi - fix channel mapping slot mask Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index ad1aa5d87dda..82312c67f8dd 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -433,7 +433,7 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid) slot = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_CHAN_SLOT, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", - slot >> 4, slot & 0x7); + slot >> 4, slot & 0xf); } #endif } -- cgit v1.2.2 From 23ccc2bd246a5bdb1ac03dc9040a0585c1890ef3 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:01 +0800 Subject: ALSA: intelhdmi - export monitor-presence and ELD-valid status Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 8 +++++++- sound/pci/hda/hda_local.h | 4 +++- sound/pci/hda/patch_intelhdmi.c | 8 +++----- 3 files changed, 13 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 20fa6aee29c0..de50cfcf644e 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -477,6 +477,8 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry, [4 ... 7] = "reserved" }; + snd_iprintf(buffer, "monitor_present\t\t%d\n", e->monitor_present); + snd_iprintf(buffer, "eld_valid\t\t%d\n", e->eld_valid); snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name); snd_iprintf(buffer, "connection_type\t\t%s\n", eld_connection_type_names[e->conn_type]); @@ -518,7 +520,11 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, * monitor_name manufacture_id product_id * eld_version edid_version */ - if (!strcmp(name, "connection_type")) + if (!strcmp(name, "monitor_present")) + e->monitor_present = val; + else if (!strcmp(name, "eld_valid")) + e->eld_valid = val; + else if (!strcmp(name, "connection_type")) e->conn_type = val; else if (!strcmp(name, "port_id")) e->port_id = val; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d4a3d0942c00..070b74384d43 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -569,9 +569,11 @@ struct cea_sad { * ELD: EDID Like Data */ struct hdmi_eld { + bool monitor_present; + bool eld_valid; int eld_size; int baseline_len; - int eld_ver; /* (eld_ver == 0) indicates invalid ELD */ + int eld_ver; int cea_edid_ver; char monitor_name[ELD_MAX_MNL + 1]; int manufacture_id; diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 82312c67f8dd..095c993f4b76 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -62,8 +62,6 @@ struct intel_hdmi_spec { /* * HDMI sink attached to each pin */ - bool sink_present[INTEL_HDMI_PINS]; - bool sink_eldv[INTEL_HDMI_PINS]; struct hdmi_eld sink_eld[INTEL_HDMI_PINS]; /* @@ -645,7 +643,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < spec->num_pins; i++) { if (spec->pin_cvt[i] != nid) continue; - if (spec->sink_present[i] != true) + if (!spec->sink_eld[i].monitor_present) continue; pin_nid = spec->pin[i]; @@ -675,8 +673,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (index < 0) return; - spec->sink_present[index] = pind; - spec->sink_eldv[index] = eldv; + spec->sink_eld[index].monitor_present = pind; + spec->sink_eld[index].eld_valid = eldv; if (pind && eldv) { hdmi_parse_eld(codec, index); -- cgit v1.2.2 From 864f92be7e8d4a0ba11d912e3f03d1a92a031dee Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:02 +0800 Subject: ALSA: hda - introduce snd_hda_jack_detect() and snd_hda_pin_sense() This helps merge duplicate code. v2: add snd_hda_jack_detect() and comments recommended by Takashi. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 34 +++++++ sound/pci/hda/hda_eld.c | 7 +- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_cirrus.c | 19 +--- sound/pci/hda/patch_realtek.c | 206 ++++++++++-------------------------------- 5 files changed, 91 insertions(+), 177 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2be61b31fb3c..9cfdb771928c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1317,6 +1317,40 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); +/** + * snd_hda_pin_sense - execute pin sense measurement + * @codec: the CODEC to sense + * @nid: the pin NID to sense + * + * Execute necessary pin sense measurement and return its Presence Detect, + * Impedance, ELD Valid etc. status bits. + */ +u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) +{ + u32 pincap = snd_hda_query_pin_caps(codec, nid); + + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + + return snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0); +} +EXPORT_SYMBOL_HDA(snd_hda_pin_sense); + +/** + * snd_hda_jack_detect - query pin Presence Detect status + * @codec: the CODEC to sense + * @nid: the pin NID to sense + * + * Query and return the pin's Presence Detect status. + */ +int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + u32 sense = snd_hda_pin_sense(codec, nid); + return !!(sense & AC_PINSENSE_PRESENCE); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_detect); + /* * read the current volume to info * if the cache exists, read the cache value. diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index de50cfcf644e..4228f2fe5956 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -309,17 +309,12 @@ out_fail: return -EINVAL; } -static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid) -{ - return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); -} - static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid) { int eldv; int present; - present = hdmi_present_sense(codec, nid); + present = snd_hda_pin_sense(codec, nid); eldv = (present & AC_PINSENSE_ELDV); present = (present & AC_PINSENSE_PRESENCE); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 070b74384d43..5778ae882b83 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -461,6 +461,8 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); +u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); struct hda_nid_item { struct snd_kcontrol *kctl; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 9ac09e4568b3..2439e84dcb21 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -807,7 +807,7 @@ static void cs_automute(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int caps, present, hp_present; + unsigned int caps, hp_present; hda_nid_t nid; int i; @@ -817,12 +817,7 @@ static void cs_automute(struct hda_codec *codec) caps = snd_hda_query_pin_caps(codec, nid); if (!(caps & AC_PINCAP_PRES_DETECT)) continue; - if (caps & AC_PINCAP_TRIG_REQ) - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - hp_present |= (present & AC_PINSENSE_PRESENCE) != 0; + hp_present = snd_hda_jack_detect(codec, nid); if (hp_present) break; } @@ -844,15 +839,11 @@ static void cs_automic(struct hda_codec *codec) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; - unsigned int caps, present; + unsigned int present; nid = cfg->input_pins[spec->automic_idx]; - caps = snd_hda_query_pin_caps(codec, nid); - if (caps & AC_PINCAP_TRIG_REQ) - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - if (present & AC_PINSENSE_PRESENCE) + present = snd_hda_jack_detect(codec, nid); + if (present) change_cur_input(codec, spec->automic_idx, 0); else { unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ? diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 578420523606..cbb2d326e6ad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -961,18 +961,12 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, static void alc_automute_pin(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present, pincap; unsigned int nid = spec->autocfg.hp_pins[0]; int i; if (!nid) return; - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, nid); for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { nid = spec->autocfg.speaker_pins[i]; if (!nid) @@ -1012,9 +1006,7 @@ static void alc_mic_automute(struct hda_codec *codec) cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0]; - present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0, - AC_VERB_GET_PIN_SENSE, 0); - present &= AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, spec->ext_mic.pin); if (present) { alive = &spec->ext_mic; dead = &spec->int_mic; @@ -1513,7 +1505,7 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { static void alc_automute_amp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int val, mute, pincap; + unsigned int mute; hda_nid_t nid; int i; @@ -1522,13 +1514,7 @@ static void alc_automute_amp(struct hda_codec *codec) nid = spec->autocfg.hp_pins[i]; if (!nid) break; - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_SENSE, 0); - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - if (val & AC_PINSENSE_PRESENCE) { + if (snd_hda_jack_detect(codec, nid)) { spec->jack_present = 1; break; } @@ -2784,8 +2770,7 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x18); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } @@ -5102,11 +5087,8 @@ static struct hda_verb alc260_hp_unsol_verbs[] = { static void alc260_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x10, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x10); alc260_hp_master_update(codec, 0x0f, 0x10, 0x11); } @@ -5171,11 +5153,8 @@ static struct hda_verb alc260_hp_3013_unsol_verbs[] = { static void alc260_hp_3013_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x15); alc260_hp_master_update(codec, 0x15, 0x10, 0x11); } @@ -5188,12 +5167,8 @@ static void alc260_hp_3013_unsol_event(struct hda_codec *codec, static void alc260_hp_3012_automute(struct hda_codec *codec) { - unsigned int present, bits; + unsigned int bits = snd_hda_jack_detect(codec, 0x10) ? 0 : PIN_OUT; - present = snd_hda_codec_read(codec, 0x10, 0, - AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; - - bits = present ? 0 : PIN_OUT; snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, bits); snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, @@ -5763,8 +5738,7 @@ static void alc260_replacer_672v_automute(struct hda_codec *codec) unsigned int present; /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */ - present = snd_hda_codec_read(codec, 0x0f, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x0f); if (present) { snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 1); @@ -8196,12 +8170,8 @@ static void alc883_mitac_setup(struct hda_codec *codec) /* static void alc883_mitac_mic_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + unsigned char bits = snd_hda_jack_detect(codec, 0x18) ? HDA_AMP_MUTE : 0; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } */ @@ -8423,10 +8393,8 @@ static struct hda_channel_mode alc888_3st_hp_modes[3] = { /* toggle front-jack and RCA according to the hp-jack state */ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = snd_hda_jack_detect(codec, 0x1b); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -8436,10 +8404,8 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) /* toggle RCA according to the front-jack state */ static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = snd_hda_jack_detect(codec, 0x14); - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } @@ -8532,24 +8498,16 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + int bits = snd_hda_jack_detect(codec, 0x14) ? HDA_AMP_MUTE : 0; - present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); } static void alc883_lenovo_101e_all_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + int bits = snd_hda_jack_detect(codec, 0x1b) ? HDA_AMP_MUTE : 0; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -8700,8 +8658,7 @@ static void alc889A_mb31_automute(struct hda_codec *codec) /* Mute only in 2ch or 4ch mode */ if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) == 0x00) { - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x15); snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, @@ -10044,10 +10001,8 @@ static void alc262_hp_master_update(struct hda_codec *codec) static void alc262_hp_bpc_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int presence; - presence = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE); + + spec->jack_present = snd_hda_jack_detect(codec, 0x1b); alc262_hp_master_update(codec); } @@ -10061,10 +10016,8 @@ static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res) static void alc262_hp_wildwest_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int presence; - presence = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE); + + spec->jack_present = snd_hda_jack_detect(codec, 0x15); alc262_hp_master_update(codec); } @@ -10298,13 +10251,8 @@ static void alc262_hippo_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, hp_nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; + spec->jack_present = snd_hda_jack_detect(codec, hp_nid); alc262_hippo_master_update(codec); } @@ -10630,21 +10578,8 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); - /* check laptop HP jack */ - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - /* check docking HP jack */ - present |= snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - if (present & AC_PINSENSE_PRESENCE) - spec->jack_present = 1; - else - spec->jack_present = 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x14) || + snd_hda_jack_detect(codec, 0x1b); spec->sense_updated = 1; } /* unmute internal speaker only if both HPs are unplugged and @@ -10689,12 +10624,7 @@ static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present_int_hp; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present_int_hp = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present_int_hp & 0x80000000) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x1b); spec->sense_updated = 1; } if (spec->jack_present) { @@ -10886,12 +10816,7 @@ static void alc262_ultra_automute(struct hda_codec *codec) mute = 0; /* auto-mute only when HP is used as HP */ if (!spec->cur_mux[0]) { - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x15); if (spec->jack_present) mute = HDA_AMP_MUTE; } @@ -11933,10 +11858,7 @@ static void alc268_acer_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x14); spec->sense_updated = 1; } if (spec->jack_present) @@ -12055,8 +11977,7 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -13039,8 +12960,7 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -13065,12 +12985,10 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec) unsigned char bits; /* Check laptop headphone socket */ - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); /* Check port replicator headphone socket */ - present |= snd_hda_codec_read(codec, 0x1a, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present |= snd_hda_jack_detect(codec, 0x1a); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, @@ -13094,11 +13012,8 @@ static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) unsigned int present_laptop; unsigned int present_dock; - present_laptop = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - - present_dock = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present_laptop = snd_hda_jack_detect(codec, 0x18); + present_dock = snd_hda_jack_detect(codec, 0x1b); /* Laptop mic port overrides dock mic port, design decision */ if (present_dock) @@ -13183,8 +13098,7 @@ static void alc269_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -14162,10 +14076,8 @@ static struct hda_verb alc861_toshiba_init_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc861_toshiba_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = snd_hda_jack_detect(codec, 0x0f); - present = snd_hda_codec_read(codec, 0x0f, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3, @@ -15070,9 +14982,9 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x18); bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } @@ -16383,9 +16295,9 @@ static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x14); bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); } @@ -16395,9 +16307,9 @@ static void alc662_lenovo_101e_all_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x1b); bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -16456,9 +16368,7 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -16471,9 +16381,7 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -16490,9 +16398,7 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -16509,9 +16415,7 @@ static void alc662_f5z_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x1b); bits = present ? 0 : PIN_OUT; snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, bits); @@ -16521,12 +16425,8 @@ static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec) { unsigned int present1, present2; - present1 = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - present2 = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present1 = snd_hda_jack_detect(codec, 0x21); + present2 = snd_hda_jack_detect(codec, 0x15); if (present1 || present2) { snd_hda_codec_write_cache(codec, 0x14, 0, @@ -16541,12 +16441,8 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) { unsigned int present1, present2; - present1 = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - present2 = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present1 = snd_hda_jack_detect(codec, 0x1b); + present2 = snd_hda_jack_detect(codec, 0x15); if (present1 || present2) { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, @@ -16706,9 +16602,7 @@ static void alc663_g71v_hp_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); @@ -16721,9 +16615,7 @@ static void alc663_g71v_front_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); -- cgit v1.2.2 From 3f54aa5091f48e9d8ce6e99b248449d08acccb26 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:03 +0800 Subject: ALSA: intelhdmi - probe for monitor/eld presence at module init time MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This avoids lost of presence info on module reloading. The presence info used to be only updated at the (rare) hotplug events. Proposed by David, thanks! CC: David Härdeman Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 32 ++++++++++++++++++++++---------- 1 file changed, 22 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 095c993f4b76..c5fd011567fb 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -259,6 +259,25 @@ static int intel_hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) return 0; } +static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) + snd_hdmi_show_eld(eld); +} + +static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + int present = snd_hda_pin_sense(codec, pin_nid); + + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + eld->eld_valid = !!(present & AC_PINSENSE_ELDV); + + if (present & AC_PINSENSE_ELDV) + hdmi_get_show_eld(codec, pin_nid, eld); +} + static int intel_hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) { struct intel_hdmi_spec *spec = codec->spec; @@ -269,6 +288,8 @@ static int intel_hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) return -EINVAL; } + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); + spec->pin[spec->num_pins] = pin_nid; spec->num_pins++; @@ -436,15 +457,6 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid) #endif } -static void hdmi_parse_eld(struct hda_codec *codec, int index) -{ - struct intel_hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld = &spec->sink_eld[index]; - - if (!snd_hdmi_get_eld(eld, codec, spec->pin[index])) - snd_hdmi_show_eld(eld); -} - /* * Audio InfoFrame routines @@ -677,7 +689,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) spec->sink_eld[index].eld_valid = eldv; if (pind && eldv) { - hdmi_parse_eld(codec, index); + hdmi_get_show_eld(codec, spec->pin[index], &spec->sink_eld[index]); /* TODO: do real things about ELD */ } } -- cgit v1.2.2 From 978be6d711be237e0344eca21c3922ae88a240bc Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:04 +0800 Subject: ALSA: intelhdmi - separate out infoframe checksum routine And make it right when called for more than one times. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index c5fd011567fb..d68dba9ac113 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -508,24 +508,35 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) #endif } +static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 sum = 0; + int i; + + ai->checksum = 0; + + for (i = 0; i < sizeof(*ai); i++) + sum += bytes[i]; + + ai->checksum = - sum; +} + static void hdmi_fill_audio_infoframe(struct hda_codec *codec, hda_nid_t pin_nid, struct hdmi_audio_infoframe *ai) { - u8 *params = (u8 *)ai; - u8 sum = 0; + u8 *bytes = (u8 *)ai; int i; hdmi_debug_dip_size(codec, pin_nid); hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ - for (i = 0; i < sizeof(*ai); i++) - sum += params[i]; - ai->checksum = - sum; + hdmi_checksum_audio_infoframe(ai); hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < sizeof(*ai); i++) - hdmi_write_dip_byte(codec, pin_nid, params[i]); + hdmi_write_dip_byte(codec, pin_nid, bytes[i]); } /* -- cgit v1.2.2 From 848de598eef9603d6f2c174f90fded4e63ac5e23 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:05 +0800 Subject: ALSA: intelhdmi - sticky infoframe MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Remember the active infoframe, so as to avoid stop/restart infoframe transmission when switching between audio clips of the same format. Proposed by Shang and David. CC: Shane W CC: David Härdeman Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 38 ++++++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index d68dba9ac113..abb056fde67a 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -646,6 +646,27 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t nid, hdmi_debug_channel_mapping(codec, nid); } +static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 val; + int i; + + if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) + != AC_DIPXMIT_BEST) + return false; + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) { + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_DATA, 0); + if (val != bytes[i]) + return false; + } + + return true; +} static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, struct snd_pcm_substream *substream) @@ -670,8 +691,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, continue; pin_nid = spec->pin[i]; - hdmi_fill_audio_infoframe(codec, pin_nid, &ai); - hdmi_start_infoframe_trans(codec, pin_nid); + if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_start_infoframe_trans(codec, pin_nid); + } } } @@ -767,16 +791,6 @@ static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct intel_hdmi_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_cvt[i] != hinfo->nid) - continue; - - hdmi_stop_infoframe_trans(codec, spec->pin[i]); - } - snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } -- cgit v1.2.2 From 5779191e0efd851fb0d54698c13cb4f5325caca6 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:06 +0800 Subject: ALSA: intelhdmi - sticky stream id and format MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We tracked down the first-0.5s-hdmi-audio-samples-lost problem to the AC_VERB_SET_CHANNEL_STREAMID command. It is suspected that many HDMI sinks need some time to adapt to the new state. The workaround is to avoid changing stream id/format whenever possible. Proposed by David. Signed-off-by: David Härdeman Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 28 ++++++++++++++++++++++++++-- 1 file changed, 26 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index abb056fde67a..8a1cf9d7e5ce 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -772,6 +772,31 @@ static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) * Callbacks */ +static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, + u32 stream_tag, int format) +{ + int tag; + int fmt; + + tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; + fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); + + snd_printdd("hdmi_setup_stream: " + "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", + nid, + tag == stream_tag ? "" : "new-", + stream_tag, + fmt == format ? "" : "new-", + format); + + if (tag != stream_tag) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, stream_tag << 4); + if (fmt != format) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, format); +} + static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -783,7 +808,7 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); + hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); return 0; } @@ -791,7 +816,6 @@ static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } -- cgit v1.2.2 From 81bf31e2d0a6a9f5d83da0a757f8ca03db908162 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:07 +0800 Subject: ALSA: intelhdmi - sticky channel count Don't change channel count if not necessary. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 8a1cf9d7e5ce..928df59be5d8 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -422,24 +422,18 @@ static void hdmi_stop_infoframe_trans(struct hda_codec *codec, AC_DIPXMIT_DISABLE); } -#ifdef CONFIG_SND_DEBUG_VERBOSE static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) { return 1 + snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CVT_CHAN_COUNT, 0); } -#endif static void hdmi_set_channel_count(struct hda_codec *codec, hda_nid_t nid, int chs) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); - -#ifdef CONFIG_SND_DEBUG_VERBOSE if (chs != hdmi_get_channel_count(codec, nid)) - snd_printd(KERN_INFO "HDMI channel count: expect %d, get %d\n", - chs, hdmi_get_channel_count(codec, nid)); -#endif + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); } static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid) -- cgit v1.2.2 From 83d605fd63e704419ccb92d48b735c6890ce3d6a Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 18 Nov 2009 12:38:08 +0800 Subject: ALSA: hda - show EPSS capability in proc Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_proc.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 35 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index be6c5f443cd9..2d627613aea3 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -286,6 +286,10 @@ enum { #define AC_PWRST_D1SUP (1<<1) #define AC_PWRST_D2SUP (1<<2) #define AC_PWRST_D3SUP (1<<3) +#define AC_PWRST_D3COLDSUP (1<<4) +#define AC_PWRST_S3D3COLDSUP (1<<29) +#define AC_PWRST_CLKSTOP (1<<30) +#define AC_PWRST_EPSS (1U<<31) /* Power state values */ #define AC_PWRST_SETTING (0xf<<0) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index f465cff28041..09476fc1ab64 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -26,6 +26,21 @@ #include "hda_codec.h" #include "hda_local.h" +static char *bits_names(unsigned int bits, char *names[], int size) +{ + int i, n; + static char buf[128]; + + for (i = 0, n = 0; i < size; i++) { + if (bits & (1U<> -- cgit v1.2.2 From d56757abc11a21996d9839c0d4e3b2c3666cd318 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Nov 2009 08:00:14 +0100 Subject: ALSA: hda - Replace the rest of jack-detections with snd_hda_jack_detect() Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 47 ++++++++++++++---------------------------- sound/pci/hda/patch_conexant.c | 37 ++++++++++----------------------- sound/pci/hda/patch_realtek.c | 3 +-- sound/pci/hda/patch_sigmatel.c | 7 ++----- sound/pci/hda/patch_via.c | 46 ++++++++++++++--------------------------- 5 files changed, 45 insertions(+), 95 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8a1064bdf4c6..455a0494f907 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -720,10 +720,10 @@ static struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { static void ad1986a_automic(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_PIN_SENSE, 0); + present = snd_hda_jack_detect(codec, 0x1f); /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL, - (present & AC_PINSENSE_PRESENCE) ? 0 : 2); + present ? 0 : 2); } #define AD1986A_MIC_EVENT 0x36 @@ -762,10 +762,8 @@ static void ad1986a_update_hp(struct hda_codec *codec) static void ad1986a_hp_automute(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = !!(present & 0x80000000); + spec->jack_present = snd_hda_jack_detect(codec, 0x1a); if (spec->inv_jack_detect) spec->jack_present = !spec->jack_present; ad1986a_update_hp(codec); @@ -1555,8 +1553,7 @@ static void ad1981_hp_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x06, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x06); snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } @@ -1576,8 +1573,7 @@ static void ad1981_hp_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x08, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x08); if (present) snd_hda_sequence_write(codec, mic_jack_on); else @@ -2532,7 +2528,7 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) { if ((res >> 26) != AD1988_HP_EVENT) return; - if (snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) & (1 << 31)) + if (snd_hda_jack_detect(codec, 0x11)) snd_hda_sequence_write(codec, ad1988_laptop_hp_on); else snd_hda_sequence_write(codec, ad1988_laptop_hp_off); @@ -3778,8 +3774,7 @@ static void ad1884a_hp_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x11, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x11); snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, @@ -3791,8 +3786,7 @@ static void ad1884a_hp_automic(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x14); snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, present ? 0 : 1); } @@ -3827,13 +3821,9 @@ static void ad1884a_laptop_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0); - present &= AC_PINSENSE_PRESENCE; - if (!present) { - present = snd_hda_codec_read(codec, 0x12, 0, - AC_VERB_GET_PIN_SENSE, 0); - present &= AC_PINSENSE_PRESENCE; - } + present = snd_hda_jack_detect(codec, 0x11); + if (!present) + present = snd_hda_jack_detect(codec, 0x12); snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, @@ -3845,11 +3835,9 @@ static void ad1884a_laptop_automic(struct hda_codec *codec) { unsigned int idx; - if (snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE) + if (snd_hda_jack_detect(codec, 0x14)) idx = 0; - else if (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE) + else if (snd_hda_jack_detect(codec, 0x1c)) idx = 4; else idx = 1; @@ -4018,8 +4006,7 @@ static void ad1984a_thinkpad_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x11); snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } @@ -4127,14 +4114,12 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* switch to external mic if plugged */ static void ad1984a_touchsmart_automic(struct hda_codec *codec) { - if (snd_hda_codec_read(codec, 0x1c, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + if (snd_hda_jack_detect(codec, 0x1c)) snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, 0x4); - } else { + else snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, 0x5); - } } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 905859d4f4df..0b097fa5421f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -397,9 +397,7 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) for (i = 0; i < spec->jacks.used; i++) { if (jacks->nid == nid) { unsigned int present; - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, nid); present = (present) ? jacks->type : 0 ; @@ -750,8 +748,7 @@ static void cxt5045_hp_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x12, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x12); if (present) snd_hda_sequence_write(codec, mic_jack_on); else @@ -765,8 +762,7 @@ static void cxt5045_hp_automute(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int bits; - spec->hp_present = snd_hda_codec_read(codec, 0x11, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + spec->hp_present = snd_hda_jack_detect(codec, 0x11); bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, @@ -1243,8 +1239,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int bits; - spec->hp_present = snd_hda_codec_read(codec, 0x13, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + spec->hp_present = snd_hda_jack_detect(codec, 0x13); bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; /* See the note in cxt5047_hp_master_sw_put */ @@ -1267,8 +1262,7 @@ static void cxt5047_hp_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); if (present) snd_hda_sequence_write(codec, mic_jack_on); else @@ -1621,9 +1615,7 @@ static void cxt5051_portb_automic(struct hda_codec *codec) if (spec->no_auto_mic) return; - present = snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x17); snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_CONNECT_SEL, present ? 0x01 : 0x00); @@ -1638,9 +1630,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec) if (spec->no_auto_mic) return; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x18); if (present) spec->cur_adc_idx = 1; else @@ -1661,9 +1651,7 @@ static void cxt5051_hp_automute(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - spec->hp_present = snd_hda_codec_read(codec, 0x16, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + spec->hp_present = snd_hda_jack_detect(codec, 0x16); cxt5051_update_speaker(codec); } @@ -2011,8 +1999,7 @@ static void cxt5066_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x1a, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x1a); if (present) { snd_printdd("CXT5066: external microphone detected\n"); snd_hda_sequence_write(codec, ext_mic_present); @@ -2029,12 +2016,10 @@ static void cxt5066_hp_automute(struct hda_codec *codec) unsigned int portA, portD; /* Port A */ - portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + portA = snd_hda_jack_detect(codec, 0x19); /* Port D */ - portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE) << 1; + portD = snd_hda_jack_detect(codec, 0x1c); spec->hp_present = !!(portA | portD); snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cbb2d326e6ad..28acbe63dfc8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8446,8 +8446,7 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x18); snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f76a97954f9..d83649c25fb2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4413,14 +4413,11 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, pin_ctl & ~flag); } -static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) +static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) { if (!nid) return 0; - if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00) - & (1 << 31)) - return 1; - return 0; + return snd_hda_jack_detect(codec, nid); } static void stac92xx_line_out_detect(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0c621d74b165..b70e26ad263f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -547,8 +547,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned no_presence = (def_conf & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ - unsigned present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0) >> 31; + unsigned present = snd_hda_jack_detect(codec, nid); struct via_spec *spec = codec->spec; if ((spec->smart51_enabled && is_smart51_pins(spec, nid)) || ((no_presence || present) @@ -786,14 +785,11 @@ static void set_jack_power_state(struct hda_codec *codec) /* Mono out */ /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_codec_read( - codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x1c); if (present) mono_out = 0; else { - present = snd_hda_codec_read( - codec, 0x1d, 0, AC_VERB_GET_PIN_SENSE, 0) - & 0x80000000; + present = snd_hda_jack_detect(codec, 0x1d); if (!spec->hp_independent_mode && present) mono_out = 0; else @@ -872,8 +868,7 @@ static void set_jack_power_state(struct hda_codec *codec) /* Class-D */ /* PW0 (24h), MW0(18h), MUX0(34h) */ - present = snd_hda_codec_read( - codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x25); parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (present) { @@ -894,8 +889,7 @@ static void set_jack_power_state(struct hda_codec *codec) /* Mono Out */ /* PW15 (31h), MW8(17h), MUX8(3bh) */ - present = snd_hda_codec_read( - codec, 0x26, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x26); parm = AC_PWRST_D3; set_pin_power_state(codec, 0x31, &parm); if (present) { @@ -973,8 +967,7 @@ static void set_jack_power_state(struct hda_codec *codec) /* Internal Speaker */ /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_codec_read( - codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x25); parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (present) { @@ -994,8 +987,7 @@ static void set_jack_power_state(struct hda_codec *codec) } /* Mono Out */ /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_codec_read( - codec, 0x28, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x28); parm = AC_PWRST_D3; set_pin_power_state(codec, 0x31, &parm); if (present) { @@ -1920,8 +1912,7 @@ static void via_hp_automute(struct hda_codec *codec) unsigned int present = 0; struct via_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); if (!spec->hp_independent_mode) { struct snd_ctl_elem_id id; @@ -1947,9 +1938,8 @@ static void via_mono_automute(struct hda_codec *codec) if (spec->codec_type != VT1716S) return; - lineout_present = snd_hda_codec_read( - codec, spec->autocfg.line_out_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + lineout_present = snd_hda_jack_detect(codec, + spec->autocfg.line_out_pins[0]); /* Mute Mono Out if Line Out is plugged */ if (lineout_present) { @@ -1958,9 +1948,7 @@ static void via_mono_automute(struct hda_codec *codec) return; } - hp_present = snd_hda_codec_read( - codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); if (!spec->hp_independent_mode) snd_hda_codec_amp_stereo( @@ -2025,8 +2013,7 @@ static void via_speaker_automute(struct hda_codec *codec) if (spec->codec_type != VT2002P && spec->codec_type != VT1812) return; - hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); if (!spec->hp_independent_mode) { struct snd_ctl_elem_id id; @@ -2055,11 +2042,9 @@ static void via_hp_bind_automute(struct hda_codec *codec) if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0]) return; - hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); - present = snd_hda_codec_read(codec, spec->autocfg.line_out_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, spec->autocfg.line_out_pins[0]); if (!spec->hp_independent_mode) { /* Mute Line-Outs */ @@ -2529,8 +2514,7 @@ static void vt1708_update_hp_jack_state(struct work_struct *work) return; /* if jack state toggled */ if (spec->vt1708_hp_present - != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + != snd_hda_jack_detect(spec->codec, spec->autocfg.hp_pins[0])) { spec->vt1708_hp_present ^= 1; via_hp_automute(spec->codec); } -- cgit v1.2.2 From 67f2db24fbfdb63495d995d6fbbbe42980004ee0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Nov 2009 08:37:59 +0100 Subject: ALSA: opti-miro: Fix missing semicolon To fix a build error sound/isa/opti9xx/miro.c:1281: error: expected ';' before '}' token Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b8170adeeff6..17761030affa 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1277,7 +1277,7 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) if (mpu_port < 0) { snd_card_free(card); snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY + return -EBUSY; } } if (miro->irq == SNDRV_AUTO_IRQ) { -- cgit v1.2.2 From bec145ae6f6978f0319e5600a742f45f76ecc4dd Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 18 Nov 2009 10:31:57 +0200 Subject: ALSA: remove unnecessary null check This function is only called from snd_ctl_ioctl() and the file parameter can never be null so there is no need to check it here. We dereference file at the start of the function: struct snd_card *card = file->card; and it confuses static checkers to dereference a pointer before checking it. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index a8b7fabe645e..b586019faf3f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1120,7 +1120,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, goto __kctl_end; } if (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - if (file && vd->owner != NULL && vd->owner != file) { + if (vd->owner != NULL && vd->owner != file) { err = -EPERM; goto __kctl_end; } -- cgit v1.2.2 From 8af3aeb498197f6fdf5acc913ffe8a392cb921c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Nov 2009 14:23:37 +0100 Subject: ALSA: hda - Fix detection of dual headphones The dual-headphone mode with STAC/IDT codecs is useful only for machines that have two (or more) built-in headphones. But, some HP laptops give multiple headphone pin configs, one for the built-in and another for the separate (likely a docking station) one. This results in a missing speaker volume control. This patch adds more check for the dual-headphone mode to avoid this problem. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d83649c25fb2..39001c47e627 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3635,6 +3635,26 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec) } } +static int is_dual_headphones(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i, valid_hps; + + if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT || + spec->autocfg.hp_outs <= 1) + return 0; + valid_hps = 0; + for (i = 0; i < spec->autocfg.hp_outs; i++) { + hda_nid_t nid = spec->autocfg.hp_pins[i]; + unsigned int cfg = snd_hda_codec_get_pincfg(codec, nid); + if (get_defcfg_location(cfg) & AC_JACK_LOC_SEPARATE) + continue; + valid_hps++; + } + return (valid_hps > 1); +} + + static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) { struct sigmatel_spec *spec = codec->spec; @@ -3651,8 +3671,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out /* If we have no real line-out pin and multiple hp-outs, HPs should * be set up as multi-channel outputs. */ - if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && - spec->autocfg.hp_outs > 1) { + if (is_dual_headphones(codec)) { /* Copy hp_outs to line_outs, backup line_outs in * speaker_outs so that the following routines can handle * HP pins as primary outputs. -- cgit v1.2.2 From faa31776e4c799d631d8cd3a13dd50cd95b0875e Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Tue, 17 Nov 2009 16:53:23 +0900 Subject: ASoC: Rename s3c24xx_pcm prefix to s3c_dma The s3c24xx_pcm prefix for the soc_platform is inappropriate when some Samsung SoCs have PCM controllers which will eventually have drivers and hence namespace ambiguities. To resolve naming ambiguities in future the following have been renamed in order 1) s3c24xx_pcm_dma_params -> s3c_dma_params 2) s3c24xx_pcm_preallocate_dma_buffer -> s3c_preallocate_dma_buffer 3) s3c24xx_pcm_dmamask -> s3c_dma_mask 4) s3c24xx_pcm_XXX -> s3c_dma_XXX Signed-off-by: Jassi Brar Acked-by: Ben Dooks Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.h | 4 +-- sound/soc/s3c24xx/s3c2412-i2s.c | 4 +-- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++--- sound/soc/s3c24xx/s3c24xx-i2s.c | 10 +++--- sound/soc/s3c24xx/s3c24xx-pcm.c | 76 ++++++++++++++++++++-------------------- sound/soc/s3c24xx/s3c24xx-pcm.h | 6 ++-- sound/soc/s3c24xx/s3c64xx-i2s.c | 4 +-- 8 files changed, 58 insertions(+), 58 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 28b0ab255096..5a442aa8b87b 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -394,7 +394,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index f66854a77fb2..ecf8eaaed1db 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -49,8 +49,8 @@ struct s3c_i2sv2_info { unsigned char master; - struct s3c24xx_pcm_dma_params *dma_playback; - struct s3c24xx_pcm_dma_params *dma_capture; + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; u32 suspend_iismod; u32 suspend_iiscon; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ac5e47b082fb..23718ea85182 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -50,14 +50,14 @@ static struct s3c2410_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { .client = &s3c2412_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { .client = &s3c2412_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index b25e9f968df9..678b1763160b 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -188,21 +188,21 @@ static struct s3c2410_dma_client s3c2443_dma_client_micin = { .name = "AC97 Mic Mono in" }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = { .client = &s3c2443_dma_client_out, .channel = DMACH_PCM_OUT, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = { .client = &s3c2443_dma_client_in, .channel = DMACH_PCM_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { +static struct s3c_dma_params s3c2443_ac97_mic_mono_in = { .client = &s3c2443_dma_client_micin, .channel = DMACH_MIC_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, @@ -290,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); @@ -339,7 +339,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index c76b8bb214bc..afb4bc9033c8 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -49,14 +49,14 @@ static struct s3c2410_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { .client = &s3c24xx_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, .dma_size = 2, }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { .client = &s3c24xx_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, @@ -258,12 +258,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; default: @@ -280,7 +280,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 151a69463269..cb49400d8c56 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -32,7 +32,7 @@ #include "s3c24xx-pcm.h" -static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { +static const struct snd_pcm_hardware s3c_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -62,15 +62,15 @@ struct s3c24xx_runtime_data { dma_addr_t dma_start; dma_addr_t dma_pos; dma_addr_t dma_end; - struct s3c24xx_pcm_dma_params *params; + struct s3c_dma_params *params; }; -/* s3c24xx_pcm_enqueue +/* s3c_dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ -static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; @@ -132,19 +132,19 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, spin_lock(&prtd->lock); if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { prtd->dma_loaded--; - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); } spin_unlock(&prtd->lock); } -static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); int ret = 0; @@ -197,7 +197,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; @@ -214,7 +214,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) +static int s3c_dma_prepare(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -247,12 +247,12 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) prtd->dma_pos = prtd->dma_start; /* enqueue dma buffers */ - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); return ret; } -static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -287,7 +287,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } static snd_pcm_uframes_t -s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) +s3c_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -322,7 +322,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(substream->runtime, res); } -static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) +static int s3c_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; @@ -330,7 +330,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); if (prtd == NULL) @@ -342,7 +342,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) +static int s3c_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -350,14 +350,14 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); if (!prtd) - pr_debug("s3c24xx_pcm_close called with prtd == NULL\n"); + pr_debug("s3c_dma_close called with prtd == NULL\n"); kfree(prtd); return 0; } -static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, +static int s3c_dma_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -370,23 +370,23 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -static struct snd_pcm_ops s3c24xx_pcm_ops = { - .open = s3c24xx_pcm_open, - .close = s3c24xx_pcm_close, +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c24xx_pcm_hw_params, - .hw_free = s3c24xx_pcm_hw_free, - .prepare = s3c24xx_pcm_prepare, - .trigger = s3c24xx_pcm_trigger, - .pointer = s3c24xx_pcm_pointer, - .mmap = s3c24xx_pcm_mmap, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, }; -static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; + size_t size = s3c_dma_hardware.buffer_bytes_max; pr_debug("Entered %s\n", __func__); @@ -401,7 +401,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) return 0; } -static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; @@ -424,9 +424,9 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32); +static u64 s3c_dma_mask = DMA_BIT_MASK(32); -static int s3c24xx_pcm_new(struct snd_card *card, +static int s3c_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -434,19 +434,19 @@ static int s3c24xx_pcm_new(struct snd_card *card, pr_debug("Entered %s\n", __func__); if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c24xx_pcm_dmamask; + card->dev->dma_mask = &s3c_dma_mask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } if (dai->capture.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) goto out; @@ -457,9 +457,9 @@ static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_platform s3c24xx_soc_platform = { .name = "s3c24xx-audio", - .pcm_ops = &s3c24xx_pcm_ops, - .pcm_new = s3c24xx_pcm_new, - .pcm_free = s3c24xx_pcm_free_dma_buffers, + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, }; EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); @@ -476,5 +476,5 @@ static void __exit s3c24xx_soc_platform_exit(void) module_exit(s3c24xx_soc_platform_exit); MODULE_AUTHOR("Ben Dooks, "); -MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c24xx-pcm.h index 0088c79822ea..8cbc071124c4 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h @@ -9,13 +9,13 @@ * ALSA PCM interface for the Samsung S3C24xx CPU */ -#ifndef _S3C24XX_PCM_H -#define _S3C24XX_PCM_H +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H #define ST_RUNNING (1<<0) #define ST_OPENED (1<<1) -struct s3c24xx_pcm_dma_params { +struct s3c_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index d68cae15561c..719d63c27fdb 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -46,7 +46,7 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { [0] = { .channel = DMACH_I2S0_OUT, .client = &s3c64xx_dma_client_out, @@ -61,7 +61,7 @@ static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { }, }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { [0] = { .channel = DMACH_I2S0_IN, .client = &s3c64xx_dma_client_in, -- cgit v1.2.2 From d3ff5a3e610d62d9cdad5b7d53749c9381e244ed Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Tue, 17 Nov 2009 16:53:31 +0900 Subject: ASoC: Rename 's3c24xx-pcm' driver to 's3c-dma' Making room for namespace for the PCM Controller driver the platform driver(s3c24xx-pcm) has been renamed to SoC agnostic name 's3c-dma'. Signed-off-by: Jassi Brar Acked-by: Ben Dooks Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Makefile | 2 +- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c-dma.c | 481 +++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-dma.h | 31 ++ sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 480 ------------------------ sound/soc/s3c24xx/s3c24xx-pcm.h | 31 -- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 2 +- 20 files changed, 528 insertions(+), 527 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c-dma.c create mode 100644 sound/soc/s3c24xx/s3c-dma.h delete mode 100644 sound/soc/s3c24xx/s3c24xx-pcm.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-pcm.h (limited to 'sound') diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 7790406f90b7..ff0a10536efc 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,5 +1,5 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c24xx-pcm.o +snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 93e6c87b7399..59dc2c6b56d9 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -25,7 +25,7 @@ #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #include "../codecs/wm8750.h" diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 12c71482d258..d00d359a03e6 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -24,7 +24,7 @@ #include #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card ln2440sbc; diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 26409a9cef9e..dea83d30a5c9 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -32,7 +32,7 @@ #include #include #include "../codecs/wm8753.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973_gta02; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 77de6c5127d2..0cb4f86f6d1e 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -36,7 +36,7 @@ #include "../codecs/wm8753.h" #include "lm4857.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" /* define the scenarios */ diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c new file mode 100644 index 000000000000..7725e26d6c91 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -0,0 +1,481 @@ +/* + * s3c-dma.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include + +#include "s3c-dma.h" + +static const struct snd_pcm_hardware s3c_dma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, + .fifo_size = 32, +}; + +struct s3c24xx_runtime_data { + spinlock_t lock; + int state; + unsigned int dma_loaded; + unsigned int dma_limit; + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + struct s3c_dma_params *params; +}; + +/* s3c_dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. +*/ +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned int limit; + int ret; + + pr_debug("Entered %s\n", __func__); + + if (s3c_dma_has_circular()) + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { + unsigned long len = prtd->dma_period; + + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + len) > prtd->dma_end) { + len = prtd->dma_end - pos; + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", + __func__, len); + } + + ret = s3c2410_dma_enqueue(prtd->params->channel, + substream, pos, len); + + if (ret == 0) { + prtd->dma_loaded++; + pos += prtd->dma_period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } else + break; + } + + prtd->dma_pos = pos; +} + +static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, + void *dev_id, int size, + enum s3c2410_dma_buffresult result) +{ + struct snd_pcm_substream *substream = dev_id; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) + return; + + prtd = substream->runtime->private_data; + + if (substream) + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { + prtd->dma_loaded--; + s3c_dma_enqueue(substream); + } + + spin_unlock(&prtd->lock); +} + +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; + unsigned long totbytes = params_buffer_bytes(params); + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, client %p, channel %d\n", prtd->params, + prtd->params->client, prtd->params->channel); + + ret = s3c2410_dma_request(prtd->params->channel, + prtd->params->client, NULL); + + if (ret < 0) { + printk(KERN_ERR "failed to get dma channel\n"); + return ret; + } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); + } + + s3c2410_dma_set_buffdone_fn(prtd->params->channel, + s3c24xx_audio_buffdone); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_limit = runtime->hw.periods_min; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + /* TODO - do we need to ensure DMA flushed */ + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + s3c2410_dma_free(prtd->params->channel, prtd->params->client); + prtd->params = NULL; + } + + return 0; +} + +static int s3c_dma_prepare(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* channel needs configuring for mem=>device, increment memory addr, + * sync to pclk, half-word transfers to the IIS-FIFO. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_MEM, + prtd->params->dma_addr); + } else { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_HW, + prtd->params->dma_addr); + } + + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size); + + /* flush the DMA channel */ + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + s3c_dma_enqueue(substream); + + return ret; +} + +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t +s3c_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + unsigned long res; + dma_addr_t src, dst; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + s3c2410_dma_getposition(prtd->params->channel, &src, &dst); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + res = dst - prtd->dma_start; + else + res = src - prtd->dma_start; + + spin_unlock(&prtd->lock); + + pr_debug("Pointer %x %x\n", src, dst); + + /* we seem to be getting the odd error from the pcm library due + * to out-of-bounds pointers. this is maybe due to the dma engine + * not having loaded the new values for the channel before being + * callled... (todo - fix ) + */ + + if (res >= snd_pcm_lib_buffer_bytes(substream)) { + if (res == snd_pcm_lib_buffer_bytes(substream)) + res = 0; + } + + return bytes_to_frames(substream->runtime, res); +} + +static int s3c_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); + + prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + return 0; +} + +static int s3c_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (!prtd) + pr_debug("s3c_dma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static int s3c_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + pr_debug("Entered %s\n", __func__); + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, +}; + +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = s3c_dma_hardware.buffer_bytes_max; + + pr_debug("Entered %s\n", __func__); + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + pr_debug("Entered %s\n", __func__); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 s3c_dma_mask = DMA_BIT_MASK(32); + +static int s3c_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s3c_dma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform s3c24xx_soc_platform = { + .name = "s3c24xx-audio", + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); + +static int __init s3c24xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&s3c24xx_soc_platform); +} +module_init(s3c24xx_soc_platform_init); + +static void __exit s3c24xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&s3c24xx_soc_platform); +} +module_exit(s3c24xx_soc_platform_exit); + +MODULE_AUTHOR("Ben Dooks, "); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-dma.h b/sound/soc/s3c24xx/s3c-dma.h new file mode 100644 index 000000000000..69bb6bf6fc1c --- /dev/null +++ b/sound/soc/s3c24xx/s3c-dma.h @@ -0,0 +1,31 @@ +/* + * s3c-dma.h -- + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * ALSA PCM interface for the Samsung S3C24xx CPU + */ + +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +struct s3c_dma_params { + struct s3c2410_dma_client *client; /* stream identifier */ + int channel; /* Channel ID */ + dma_addr_t dma_addr; + int dma_size; /* Size of the DMA transfer */ +}; + +#define S3C24XX_DAI_I2S 0 + +/* platform data */ +extern struct snd_soc_platform s3c24xx_soc_platform; +extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; + +#endif diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 5a442aa8b87b..e994d8374fe6 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -35,7 +35,7 @@ #include #include "s3c-i2s-v2.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #undef S3C_IIS_V2_SUPPORTED diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 23718ea85182..359e59346ba2 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -37,7 +37,7 @@ #include #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 678b1763160b..0191e3acb0b4 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -35,7 +35,7 @@ #include #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" struct s3c24xx_ac97_info { diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index afb4bc9033c8..0bc5950b9f02 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -38,7 +38,7 @@ #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct s3c2410_dma_client s3c24xx_dma_client_out = { diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c deleted file mode 100644 index cb49400d8c56..000000000000 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ /dev/null @@ -1,480 +0,0 @@ -/* - * s3c24xx-pcm.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include -#include -#include - -#include "s3c24xx-pcm.h" - -static const struct snd_pcm_hardware s3c_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct s3c24xx_runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_limit; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -/* s3c_dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. -*/ -static void s3c_dma_enqueue(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - int ret; - - pr_debug("Entered %s\n", __func__); - - if (s3c_dma_has_circular()) { - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - } else - limit = prtd->dma_limit; - - pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit); - - while (prtd->dma_loaded < limit) { - unsigned long len = prtd->dma_period; - - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + len) > prtd->dma_end) { - len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); - } - - ret = s3c2410_dma_enqueue(prtd->params->channel, - substream, pos, len); - - if (ret == 0) { - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } else - break; - } - - prtd->dma_pos = pos; -} - -static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, - void *dev_id, int size, - enum s3c2410_dma_buffresult result) -{ - struct snd_pcm_substream *substream = dev_id; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) - return; - - prtd = substream->runtime->private_data; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { - prtd->dma_loaded--; - s3c_dma_enqueue(substream); - } - - spin_unlock(&prtd->lock); -} - -static int s3c_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; - unsigned long totbytes = params_buffer_bytes(params); - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - ret = s3c2410_dma_request(prtd->params->channel, - prtd->params->client, NULL); - - if (ret < 0) { - printk(KERN_ERR "failed to get dma channel\n"); - return ret; - } - - /* use the circular buffering if we have it available. */ - if (s3c_dma_has_circular()) - s3c2410_dma_setflags(prtd->params->channel, - S3C2410_DMAF_CIRCULAR); - } - - s3c2410_dma_set_buffdone_fn(prtd->params->channel, - s3c24xx_audio_buffdone); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_limit = runtime->hw.periods_min; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int s3c_dma_hw_free(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - /* TODO - do we need to ensure DMA flushed */ - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - s3c2410_dma_free(prtd->params->channel, prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int s3c_dma_prepare(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* channel needs configuring for mem=>device, increment memory addr, - * sync to pclk, half-word transfers to the IIS-FIFO. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_MEM, - prtd->params->dma_addr); - } else { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_HW, - prtd->params->dma_addr); - } - - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size); - - /* flush the DMA channel */ - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - s3c_dma_enqueue(substream); - - return ret; -} - -static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -s3c_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - unsigned long res; - dma_addr_t src, dst; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - s3c2410_dma_getposition(prtd->params->channel, &src, &dst); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - res = dst - prtd->dma_start; - else - res = src - prtd->dma_start; - - spin_unlock(&prtd->lock); - - pr_debug("Pointer %x %x\n", src, dst); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * callled... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int s3c_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); - - prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int s3c_dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("s3c_dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int s3c_dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops s3c_dma_ops = { - .open = s3c_dma_open, - .close = s3c_dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c_dma_hw_params, - .hw_free = s3c_dma_hw_free, - .prepare = s3c_dma_prepare, - .trigger = s3c_dma_trigger, - .pointer = s3c_dma_pointer, - .mmap = s3c_dma_mmap, -}; - -static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c_dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 s3c_dma_mask = DMA_BIT_MASK(32); - -static int s3c_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c_dma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform s3c24xx_soc_platform = { - .name = "s3c24xx-audio", - .pcm_ops = &s3c_dma_ops, - .pcm_new = s3c_dma_new, - .pcm_free = s3c_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); - -static int __init s3c24xx_soc_platform_init(void) -{ - return snd_soc_register_platform(&s3c24xx_soc_platform); -} -module_init(s3c24xx_soc_platform_init); - -static void __exit s3c24xx_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&s3c24xx_soc_platform); -} -module_exit(s3c24xx_soc_platform_exit); - -MODULE_AUTHOR("Ben Dooks, "); -MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c24xx-pcm.h deleted file mode 100644 index 8cbc071124c4..000000000000 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * s3c24xx-pcm.h -- - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * ALSA PCM interface for the Samsung S3C24xx CPU - */ - -#ifndef _S3C_AUDIO_H -#define _S3C_AUDIO_H - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ - int channel; /* Channel ID */ - dma_addr_t dma_addr; - int dma_size; /* Size of the DMA transfer */ -}; - -#define S3C24XX_DAI_I2S 0 - -/* platform data */ -extern struct snd_soc_platform s3c24xx_soc_platform; -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - -#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 1966e0d5652d..507b2ed5d58b 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -21,7 +21,7 @@ #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c index 8346bd96eaf5..bdf8951af8e3 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -18,7 +18,7 @@ #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c index 25797e096175..185c0acb5ce6 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -18,7 +18,7 @@ #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index c215d32d6322..052d59659c29 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -24,7 +24,7 @@ #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "../codecs/uda134x.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 719d63c27fdb..cc7edb5f792d 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -35,7 +35,7 @@ #include #include -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c64xx-i2s.h" static struct s3c2410_dma_client s3c64xx_dma_client_out = { diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index a2a4f5323c17..12b783b12fcb 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -20,7 +20,7 @@ #include #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card smdk2443; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index 216dd1e8e378..efe4901213a3 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -19,7 +19,7 @@ #include #include "../codecs/wm8580.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c64xx-i2s.h" #define S3C64XX_I2S_V4 2 -- cgit v1.2.2 From 357a1db94ecc5b3d605574b164d288cd7dbf8dbd Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Tue, 17 Nov 2009 16:54:03 +0900 Subject: ASoC: Added the CPU driver for PCM controllers Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 3 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c-pcm.c | 552 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-pcm.h | 123 ++++++++++ 4 files changed, 680 insertions(+) create mode 100644 sound/soc/s3c24xx/s3c-pcm.c create mode 100644 sound/soc/s3c24xx/s3c-pcm.h (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index d7912f1e4627..b489f1ae103d 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -24,6 +24,9 @@ config SND_S3C64XX_SOC_I2S select SND_S3C_I2SV2_SOC select S3C64XX_DMA +config SND_S3C_SOC_PCM + tristate + config SND_S3C2443_SOC_AC97 tristate select S3C2410_DMA diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index ff0a10536efc..b744657733d7 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -5,6 +5,7 @@ snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o +snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o @@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o +obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c new file mode 100644 index 000000000000..9e61a7c2d9ac --- /dev/null +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -0,0 +1,552 @@ +/* sound/soc/s3c24xx/s3c-pcm.c + * + * ALSA SoC Audio Layer - S3C PCM-Controller driver + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh + * based upon I2S drivers by Ben Dooks. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "s3c-dma.h" +#include "s3c-pcm.h" + +static struct s3c2410_dma_client s3c_pcm_dma_client_out = { + .name = "PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c_pcm_dma_client_in = { + .name = "PCM Stereo in" +}; + +static struct s3c_dma_params s3c_pcm_stereo_out[] = { + [0] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, +}; + +static struct s3c_dma_params s3c_pcm_stereo_in[] = { + [0] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, +}; + +static struct s3c_pcm_info s3c_pcm[2]; + +static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + clkctl = readl(regs + S3C_PCM_CLKCTL); + ctl = readl(regs + S3C_PCM_CTL); + ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK + << S3C_PCM_CTL_TXDIPSTICK_SHIFT); + + if (on) { + ctl |= S3C_PCM_CTL_TXDMA_EN; + ctl |= S3C_PCM_CTL_TXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + ctl |= (0x20<idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + ctl = readl(regs + S3C_PCM_CTL); + clkctl = readl(regs + S3C_PCM_CLKCTL); + + if (on) { + ctl |= S3C_PCM_CTL_RXDMA_EN; + ctl |= S3C_PCM_CTL_RXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_RXDMA_EN; + ctl &= ~S3C_PCM_CTL_RXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); + unsigned long flags; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 1); + else + s3c_pcm_snd_txctrl(pcm, 1); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 0); + else + s3c_pcm_snd_txctrl(pcm, 0); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + void __iomem *regs = pcm->regs; + struct clk *clk; + int sclk_div, sync_div; + unsigned long flags; + u32 clkctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->cpu_dai->dma_data = pcm->dma_playback; + else + dai->cpu_dai->dma_data = pcm->dma_capture; + + /* Strictly check for sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + spin_lock_irqsave(&pcm->lock, flags); + + /* Get hold of the PCMSOURCE_CLK */ + clkctl = readl(regs + S3C_PCM_CLKCTL); + if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) + clk = pcm->pclk; + else + clk = pcm->cclk; + + /* Set the SCLK divider */ + sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / + params_rate(params) / 2 - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + + /* Set the SYNC divider */ + sync_div = pcm->sclk_per_fs - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + spin_unlock_irqrestore(&pcm->lock, flags); + + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs \ + SCLK_DIV=%d SYNC_DIV=%d\n", + clk_get_rate(clk), pcm->sclk_per_fs, + sclk_div, sync_div); + + return 0; +} + +static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + unsigned long flags; + int ret = 0; + u32 ctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + spin_lock_irqsave(&pcm->lock, flags); + + ctl = readl(regs + S3C_PCM_CTL); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do, NB_NF by default */ + break; + default: + dev_err(pcm->dev, "Unsupported clock inversion!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Nothing to do, Master by default */ + break; + default: + dev_err(pcm->dev, "Unsupported master/slave format!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + pcm->idleclk = 1; + break; + case SND_SOC_DAIFMT_GATED: + pcm->idleclk = 0; + break; + default: + dev_err(pcm->dev, "Invalid Clock gating request!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + case SND_SOC_DAIFMT_DSP_B: + ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + default: + dev_err(pcm->dev, "Unsupported data format!\n"); + ret = -EINVAL; + goto exit; + } + + writel(ctl, regs + S3C_PCM_CTL); + +exit: + spin_unlock_irqrestore(&pcm->lock, flags); + + return ret; +} + +static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + + switch (div_id) { + case S3C_PCM_SCLK_PER_FS: + pcm->sclk_per_fs = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + u32 clkctl = readl(regs + S3C_PCM_CLKCTL); + + switch (clk_id) { + case S3C_PCM_CLKSRC_PCLK: + clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + break; + + case S3C_PCM_CLKSRC_MUX: + clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + + if (clk_get_rate(pcm->cclk) != freq) + clk_set_rate(pcm->cclk, freq); + + break; + + default: + return -EINVAL; + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + return 0; +} + +static struct snd_soc_dai_ops s3c_pcm_dai_ops = { + .set_sysclk = s3c_pcm_set_sysclk, + .set_clkdiv = s3c_pcm_set_clkdiv, + .trigger = s3c_pcm_trigger, + .hw_params = s3c_pcm_hw_params, + .set_fmt = s3c_pcm_set_fmt, +}; + +#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 + +#define S3C_PCM_DECLARE(n) \ +{ \ + .name = "samsung-pcm", \ + .id = (n), \ + .symmetric_rates = 1, \ + .ops = &s3c_pcm_dai_ops, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ +} + +struct snd_soc_dai s3c_pcm_dai[] = { + S3C_PCM_DECLARE(0), + S3C_PCM_DECLARE(1), +}; +EXPORT_SYMBOL_GPL(s3c_pcm_dai); + +static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm; + struct snd_soc_dai *dai; + struct resource *mem_res, *dmatx_res, *dmarx_res; + struct s3c_audio_pdata *pcm_pdata; + int ret; + + /* Check for valid device index */ + if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + pcm_pdata = pdev->dev.platform_data; + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + return -EINVAL; + } + + pcm = &s3c_pcm[pdev->id]; + pcm->dev = &pdev->dev; + + spin_lock_init(&pcm->lock); + + dai = &s3c_pcm_dai[pdev->id]; + dai->dev = &pdev->dev; + + /* Default is 128fs */ + pcm->sclk_per_fs = 128; + + pcm->cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(pcm->cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(pcm->cclk); + goto err1; + } + clk_enable(pcm->cclk); + + /* record our pcm structure for later use in the callbacks */ + dai->private_data = pcm; + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "samsung-pcm")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + ret = -EBUSY; + goto err2; + } + + pcm->regs = ioremap(mem_res->start, 0x100); + if (pcm->regs == NULL) { + dev_err(&pdev->dev, "cannot ioremap registers\n"); + ret = -ENXIO; + goto err3; + } + + pcm->pclk = clk_get(&pdev->dev, "pcm"); + if (IS_ERR(pcm->pclk)) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + ret = -ENOENT; + goto err4; + } + clk_enable(pcm->pclk); + + ret = snd_soc_register_dai(dai); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + goto err5; + } + + s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + + S3C_PCM_RXFIFO; + s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start + + S3C_PCM_TXFIFO; + + s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; + s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; + + pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; + pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + + return 0; + +err5: + clk_disable(pcm->pclk); + clk_put(pcm->pclk); +err4: + iounmap(pcm->regs); +err3: + release_mem_region(mem_res->start, resource_size(mem_res)); +err2: + clk_disable(pcm->cclk); + clk_put(pcm->cclk); +err1: + return ret; +} + +static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; + struct resource *mem_res; + + iounmap(pcm->regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem_res->start, resource_size(mem_res)); + + clk_disable(pcm->cclk); + clk_disable(pcm->pclk); + clk_put(pcm->pclk); + clk_put(pcm->cclk); + + return 0; +} + +static struct platform_driver s3c_pcm_driver = { + .probe = s3c_pcm_dev_probe, + .remove = s3c_pcm_dev_remove, + .driver = { + .name = "samsung-pcm", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_pcm_init(void) +{ + return platform_driver_register(&s3c_pcm_driver); +} +module_init(s3c_pcm_init); + +static void __exit s3c_pcm_exit(void) +{ + platform_driver_unregister(&s3c_pcm_driver); +} +module_exit(s3c_pcm_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, "); +MODULE_DESCRIPTION("S3C PCM Controller Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-pcm.h b/sound/soc/s3c24xx/s3c-pcm.h new file mode 100644 index 000000000000..69ff9971692f --- /dev/null +++ b/sound/soc/s3c24xx/s3c-pcm.h @@ -0,0 +1,123 @@ +/* sound/soc/s3c24xx/s3c-pcm.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __S3C_PCM_H +#define __S3C_PCM_H __FILE__ + +/*Register Offsets */ +#define S3C_PCM_CTL (0x00) +#define S3C_PCM_CLKCTL (0x04) +#define S3C_PCM_TXFIFO (0x08) +#define S3C_PCM_RXFIFO (0x0C) +#define S3C_PCM_IRQCTL (0x10) +#define S3C_PCM_IRQSTAT (0x14) +#define S3C_PCM_FIFOSTAT (0x18) +#define S3C_PCM_CLRINT (0x20) + +/* PCM_CTL Bit-Fields */ +#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) +#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) +#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) +#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) +#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) +#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) +#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) +#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) +#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) +#define S3C_PCM_CTL_ENABLE (0x1<<0) + +/* PCM_CLKCTL Bit-Fields */ +#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) +#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) +#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) +#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) + +/* PCM_TXFIFO Bit-Fields */ +#define S3C_PCM_TXFIFO_DVALID (0x1<<16) +#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_RXFIFO Bit-Fields */ +#define S3C_PCM_RXFIFO_DVALID (0x1<<16) +#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_IRQCTL Bit-Fields */ +#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) +#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) +#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) +#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) +#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) +#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) +#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) +#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) +#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) +#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) +#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) +#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) +#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) +#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) + +/* PCM_IRQSTAT Bit-Fields */ +#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) +#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) +#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) +#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) +#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) +#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) +#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) +#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) +#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) +#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) +#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) +#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) +#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) +#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) + +/* PCM_FIFOSTAT Bit-Fields */ +#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) +#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) +#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) +#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) +#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) +#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) + +#define S3C_PCM_CLKSRC_PCLK 0 +#define S3C_PCM_CLKSRC_MUX 1 + +#define S3C_PCM_SCLK_PER_FS 0 + +/** + * struct s3c_pcm_info - S3C PCM Controller information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device register block. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + */ +struct s3c_pcm_info { + spinlock_t lock; + struct device *dev; + void __iomem *regs; + + unsigned int sclk_per_fs; + + /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ + unsigned int idleclk; + + struct clk *pclk; + struct clk *cclk; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; +}; + +#endif /* __S3C_PCM_H */ -- cgit v1.2.2 From 57512c6432783c9695ef54f875f705584c65c733 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Mon, 16 Nov 2009 16:52:31 -0700 Subject: ASoC: DaVinci: remove requirement that dma_params is 1st in structure Remove requirement that dma_params is 1st in the structures davinci_audio_dev and davinci_mcbsp_dev. Signed-off-by: Troy Kisky Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 6 +----- sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/davinci/davinci-mcasp.h | 5 ----- sound/soc/davinci/davinci-pcm.c | 7 ++++--- 4 files changed, 6 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 2ab809359c08..d336786683b4 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -98,11 +98,6 @@ enum { }; struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -549,6 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 50ad0519a8fa..0a302e1080d9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -904,6 +904,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc88f7b..582c9249ef09 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index fb10f1d63fdb..187ee965bf0b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -253,10 +253,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ -- cgit v1.2.2 From b4e818768d50a5b7aa1635676839682bcf0691b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Nov 2009 17:20:24 +0100 Subject: ALSA: hda - Fix mute-LED sync on HP laptops with IDT92HD83xxx codecs The mute-LED isn't synchronized with the actual mute state on some HP laptops with IDT 92HD83xxx codecs. A similar hack using check_power_status callback is added for this codec, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 39 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 37 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 39001c47e627..2a45375d79f8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -93,6 +93,7 @@ enum { STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, + STAC_92HD83XXX_HP, STAC_92HD83XXX_MODELS }; @@ -1624,6 +1625,7 @@ static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", + [STAC_92HD83XXX_HP] = "hp", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1634,6 +1636,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, "unknown Dell", STAC_DELL_S14), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600, + "HP", STAC_92HD83XXX_HP), {} /* terminator */ }; @@ -4834,6 +4838,23 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, return 0; } + +static int idt92hd83xxx_hp_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + + if (nid != 0x13) + return 0; + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) + spec->gpio_data |= spec->gpio_led; /* mute LED on */ + else + spec->gpio_data &= ~spec->gpio_led; /* mute LED off */ + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); + + return 0; +} + #endif static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) @@ -5199,6 +5220,22 @@ again: break; } + codec->patch_ops = stac92xx_patch_ops; + + if (spec->board_config == STAC_92HD83XXX_HP) + spec->gpio_led = 0x01; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (spec->gpio_led) { + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + spec->gpio_data |= spec->gpio_led; + /* register check_power_status callback. */ + codec->patch_ops.check_power_status = + idt92hd83xxx_hp_check_power_status; + } +#endif + err = stac92xx_parse_auto_config(codec, 0x1d, 0); if (!err) { if (spec->board_config < 0) { @@ -5234,8 +5271,6 @@ again: snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, num_dacs); - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac92hd_proc_hook; return 0; -- cgit v1.2.2 From 0d6c97742993a00ee2cbfbd6d68fba669c17bf50 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Wed, 18 Nov 2009 17:49:51 -0700 Subject: ASoC: DaVinci: i2s, reduce underruns by combining into 1 element Allow the left and right 16 bit samples to be shifted out as 1 32 bit sample. Signed-off-by: Troy Kisky Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 74 ++++++++++++++++++++++++++++++----------- 1 file changed, 55 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d336786683b4..b2a5372ef72c 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,6 +97,23 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + struct davinci_mcbsp_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; @@ -105,6 +122,27 @@ struct davinci_mcbsp_dev { int mode; u32 pcr; struct clk *clk; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned enable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -344,6 +382,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -373,29 +413,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - dma_params->acnt = dma_params->data_type; + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && dev->enable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; - - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -510,7 +545,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->enable_channel_combine = pdata->enable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- cgit v1.2.2 From 1587ea31572e25a0a2c9c491b7f8c937b6c0454e Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Wed, 18 Nov 2009 17:49:52 -0700 Subject: ASoC: DaVinci: pcm, rename variables in prep for ping/pong Rename variable master_lch to asp_channel Rename variable slave_lch to asp_link[0] Rename local variables: lch to link count to asp_count src to asp_src dst to asp_dst Signed-off-by: Troy Kisky Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-pcm.c | 66 ++++++++++++++++++++--------------------- 1 file changed, 33 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 187ee965bf0b..42a657ea49cb 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -51,8 +51,8 @@ static struct snd_pcm_hardware davinci_pcm_hardware = { struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ - int master_lch; /* Master DMA channel */ - int slave_lch; /* linked parameter RAM reload slot */ + int asp_channel; /* Master DMA channel */ + int asp_link[2]; /* asp parameter link channel, ping/pong */ struct davinci_pcm_dma_params *params; /* DMA params */ }; @@ -60,7 +60,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int lch = prtd->slave_lch; + int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; @@ -78,7 +78,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -102,16 +102,16 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) } acnt = prtd->params->acnt; - edma_set_src(lch, src, INCR, W8BIT); - edma_set_dest(lch, dst, INCR, W8BIT); + edma_set_src(link, src, INCR, W8BIT); + edma_set_dest(link, dst, INCR, W8BIT); - edma_set_src_index(lch, src_bidx, src_cidx); - edma_set_dest_index(lch, dst_bidx, dst_cidx); + edma_set_src_index(link, src_bidx, src_cidx); + edma_set_dest_index(link, dst_bidx, dst_cidx); if (!fifo_level) - edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(lch, acnt, fifo_level, count, + edma_set_transfer_params(link, acnt, fifo_level, count, fifo_level, ABSYNC); prtd->period++; @@ -119,12 +119,12 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) prtd->period = 0; } -static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; - pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status); + pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; @@ -150,15 +150,15 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) EVENTQ_0); if (ret < 0) return ret; - prtd->master_lch = ret; + prtd->asp_channel = ret; /* Request parameter RAM reload slot */ - ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY); + ret = edma_alloc_slot(EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); if (ret < 0) { - edma_free_channel(prtd->master_lch); + edma_free_channel(prtd->asp_channel); return ret; } - prtd->slave_lch = ret; + prtd->asp_link[0] = ret; /* Issue transfer completion IRQ when the channel completes a * transfer, then always reload from the same slot (by a kind @@ -169,10 +169,10 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(prtd->slave_lch, &p_ram); - p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch)); - p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5; - edma_write_slot(prtd->slave_lch, &p_ram); + edma_read_slot(prtd->asp_link[0], &p_ram); + p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); + p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &p_ram); return 0; } @@ -188,12 +188,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_start(prtd->master_lch); + edma_start(prtd->asp_channel); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_stop(prtd->master_lch); + edma_stop(prtd->asp_channel); break; default: ret = -EINVAL; @@ -214,8 +214,8 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) davinci_pcm_enqueue_dma(substream); /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->slave_lch, &temp); - edma_write_slot(prtd->master_lch, &temp); + edma_read_slot(prtd->asp_link[0], &temp); + edma_write_slot(prtd->asp_channel, &temp); davinci_pcm_enqueue_dma(substream); return 0; @@ -227,20 +227,20 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; unsigned int offset; - dma_addr_t count; - dma_addr_t src, dst; + int asp_count; + dma_addr_t asp_src, asp_dst; spin_lock(&prtd->lock); - edma_get_position(prtd->master_lch, &src, &dst); + edma_get_position(prtd->asp_channel, &asp_src, &asp_dst); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - count = src - runtime->dma_addr; + asp_count = asp_src - runtime->dma_addr; else - count = dst - runtime->dma_addr; + asp_count = asp_dst - runtime->dma_addr; spin_unlock(&prtd->lock); - offset = bytes_to_frames(runtime, count); + offset = bytes_to_frames(runtime, asp_count); if (offset >= runtime->buffer_size) offset = 0; @@ -289,10 +289,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - edma_unlink(prtd->slave_lch); + edma_unlink(prtd->asp_link[0]); - edma_free_slot(prtd->slave_lch); - edma_free_channel(prtd->master_lch); + edma_free_slot(prtd->asp_link[0]); + edma_free_channel(prtd->asp_channel); kfree(prtd); -- cgit v1.2.2 From 1e224f322bf22280957a5f76164d848526ed9b08 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Wed, 18 Nov 2009 17:49:53 -0700 Subject: ASoC: DaVinci: pcm, fix underrun by using sram Fix underruns by using dma to copy 1st to sram in a ping/pong buffer style and then copying from the sram to the ASP. This also has the advantage of tolerating very long interrupt latency on dma completion. Signed-off-by: Troy Kisky Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 7 +- sound/soc/davinci/davinci-pcm.c | 515 ++++++++++++++++++++++++++++++++++++---- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 479 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b2a5372ef72c..6362ca05506e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -545,8 +545,13 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - if (pdata) + if (pdata) { dev->enable_channel_combine = pdata->enable_channel_combine; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = + pdata->sram_size_playback; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = + pdata->sram_size_capture; + } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 42a657ea49cb..664d49336508 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -3,6 +3,7 @@ * * Author: Vladimir Barinov, * Copyright: (C) 2007 MontaVista Software, Inc., + * added SRAM ping/pong (C) 2008 Troy Kisky * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -23,10 +24,29 @@ #include #include +#include #include "davinci-pcm.h" -static struct snd_pcm_hardware davinci_pcm_hardware = { +#ifdef DEBUG +static void print_buf_info(int slot, char *name) +{ + struct edmacc_param p; + if (slot < 0) + return; + edma_read_slot(slot, &p); + printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", + name, slot, p.opt, p.src, p.a_b_cnt, p.dst); + printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", + p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); +} +#else +static void print_buf_info(int slot, char *name) +{ +} +#endif + +static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE), @@ -48,14 +68,80 @@ static struct snd_pcm_hardware davinci_pcm_hardware = { .fifo_size = 0, }; +static struct snd_pcm_hardware pcm_hardware_capture = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +/* + * How ping/pong works.... + * + * Playback: + * ram_params - copys 2*ping_size from start of SDRAM to iram, + * links to ram_link2 + * ram_link2 - copys rest of SDRAM to iram in ping_size units, + * links to ram_link + * ram_link - copys entire SDRAM to iram in ping_size uints, + * links to self + * + * asp_params - same as asp_link[0] + * asp_link[0] - copys from lower half of iram to asp port + * links to asp_link[1], triggers iram copy event on completion + * asp_link[1] - copys from upper half of iram to asp port + * links to asp_link[0], triggers iram copy event on completion + * triggers interrupt only needed to let upper SOC levels update position + * in stream on completion + * + * When playback is started: + * ram_params started + * asp_params started + * + * Capture: + * ram_params - same as ram_link, + * links to ram_link + * ram_link - same as playback + * links to self + * + * asp_params - same as playback + * asp_link[0] - same as playback + * asp_link[1] - same as playback + * + * When capture is started: + * asp_params started + */ struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ int asp_channel; /* Master DMA channel */ int asp_link[2]; /* asp parameter link channel, ping/pong */ struct davinci_pcm_dma_params *params; /* DMA params */ + int ram_channel; + int ram_link; + int ram_link2; + struct edmacc_param asp_params; + struct edmacc_param ram_params; }; +/* + * Not used with ping/pong + */ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; @@ -124,41 +210,290 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; + print_buf_info(prtd->ram_channel, "i ram_channel"); pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; if (snd_pcm_running(substream)) { + if (prtd->ram_channel < 0) { + /* No ping/pong must fix up link dma data*/ + spin_lock(&prtd->lock); + davinci_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } snd_pcm_period_elapsed(substream); + } +} + +static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + struct snd_dma_buffer *iram_dma = NULL; + dma_addr_t iram_phys = 0; + void *iram_virt = NULL; + + if (buf->private_data || !size) + return 0; + + ppcm->period_bytes_max = size; + iram_virt = sram_alloc(size, &iram_phys); + if (!iram_virt) + goto exit1; + iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); + if (!iram_dma) + goto exit2; + iram_dma->area = iram_virt; + iram_dma->addr = iram_phys; + memset(iram_dma->area, 0, size); + iram_dma->bytes = size; + buf->private_data = iram_dma; + return 0; +exit2: + if (iram_virt) + sram_free(iram_virt, size); +exit1: + return -ENOMEM; +} - spin_lock(&prtd->lock); - davinci_pcm_enqueue_dma(substream); - spin_unlock(&prtd->lock); +/* + * Only used with ping/pong. + * This is called after runtime->dma_addr, period_bytes and data_type are valid + */ +static int ping_pong_dma_setup(struct snd_pcm_substream *substream) +{ + unsigned short ram_src_cidx, ram_dst_cidx; + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + struct snd_dma_buffer *iram_dma = + (struct snd_dma_buffer *)substream->dma_buffer.private_data; + struct davinci_pcm_dma_params *params = prtd->params; + unsigned int data_type = params->data_type; + unsigned int acnt = params->acnt; + /* divide by 2 for ping/pong */ + unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; + int link = prtd->asp_link[1]; + unsigned int fifo_level = prtd->params->fifo_level; + unsigned int count; + if ((data_type == 0) || (data_type > 4)) { + printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_addr_t asp_src_pong = iram_dma->addr + ping_size; + ram_src_cidx = ping_size; + ram_dst_cidx = -ping_size; + edma_set_src(link, asp_src_pong, INCR, W8BIT); + + link = prtd->asp_link[0]; + edma_set_src_index(link, data_type, data_type * fifo_level); + link = prtd->asp_link[1]; + edma_set_src_index(link, data_type, data_type * fifo_level); + + link = prtd->ram_link; + edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + } else { + dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; + ram_src_cidx = -ping_size; + ram_dst_cidx = ping_size; + edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + + link = prtd->asp_link[0]; + edma_set_dest_index(link, data_type, data_type * fifo_level); + link = prtd->asp_link[1]; + edma_set_dest_index(link, data_type, data_type * fifo_level); + + link = prtd->ram_link; + edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + } + + if (!fifo_level) { + count = ping_size / data_type; + edma_set_transfer_params(prtd->asp_link[0], acnt, count, + 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, count, + 1, 0, ASYNC); + } else { + count = ping_size / (data_type * fifo_level); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, + count, fifo_level, ABSYNC); + } + + link = prtd->ram_link; + edma_set_src_index(link, ping_size, ram_src_cidx); + edma_set_dest_index(link, ping_size, ram_dst_cidx); + edma_set_transfer_params(link, ping_size, 2, + runtime->periods, 2, ASYNC); + + /* init master params */ + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_read_slot(prtd->ram_link, &prtd->ram_params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct edmacc_param p_ram; + /* Copy entire iram buffer before playback started */ + prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); + /* 0 dst_bidx */ + prtd->ram_params.src_dst_bidx = (ping_size << 1); + /* 0 dst_cidx */ + prtd->ram_params.src_dst_cidx = (ping_size << 1); + prtd->ram_params.ccnt = 1; + + /* Skip 1st period */ + edma_read_slot(prtd->ram_link, &p_ram); + p_ram.src += (ping_size << 1); + p_ram.ccnt -= 1; + edma_write_slot(prtd->ram_link2, &p_ram); + /* + * When 1st started, ram -> iram dma channel will fill the + * entire iram. Then, whenever a ping/pong asp buffer finishes, + * 1/2 iram will be filled. + */ + prtd->ram_params.link_bcntrld = + EDMA_CHAN_SLOT(prtd->ram_link2) << 5; + } + return 0; +} + +/* 1 asp tx or rx channel using 2 parameter channels + * 1 ram to/from iram channel using 1 parameter channel + * + * Playback + * ram copy channel kicks off first, + * 1st ram copy of entire iram buffer completion kicks off asp channel + * asp tcc always kicks off ram copy of 1/2 iram buffer + * + * Record + * asp channel starts, tcc kicks off ram copy + */ +static int request_ping_pong(struct snd_pcm_substream *substream, + struct davinci_runtime_data *prtd, + struct snd_dma_buffer *iram_dma) +{ + dma_addr_t asp_src_ping; + dma_addr_t asp_dst_ping; + int link; + struct davinci_pcm_dma_params *params = prtd->params; + + /* Request ram master channel */ + link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + davinci_pcm_dma_irq, substream, + EVENTQ_1); + if (link < 0) + goto exit1; + + /* Request ram link channel */ + link = prtd->ram_link = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit2; + + link = prtd->asp_link[1] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit3; + + prtd->ram_link2 = -1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link = prtd->ram_link2 = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit4; + } + /* circle ping-pong buffers */ + edma_link(prtd->asp_link[0], prtd->asp_link[1]); + edma_link(prtd->asp_link[1], prtd->asp_link[0]); + /* circle ram buffers */ + edma_link(prtd->ram_link, prtd->ram_link); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + asp_src_ping = iram_dma->addr; + asp_dst_ping = params->dma_addr; /* fifo */ + } else { + asp_src_ping = params->dma_addr; /* fifo */ + asp_dst_ping = iram_dma->addr; } + /* ping */ + link = prtd->asp_link[0]; + edma_set_src(link, asp_src_ping, INCR, W16BIT); + edma_set_dest(link, asp_dst_ping, INCR, W16BIT); + edma_set_src_index(link, 0, 0); + edma_set_dest_index(link, 0, 0); + + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); + prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); + edma_write_slot(link, &prtd->asp_params); + + /* pong */ + link = prtd->asp_link[1]; + edma_set_src(link, asp_src_ping, INCR, W16BIT); + edma_set_dest(link, asp_dst_ping, INCR, W16BIT); + edma_set_src_index(link, 0, 0); + edma_set_dest_index(link, 0, 0); + + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); + /* interrupt after every pong completion */ + prtd->asp_params.opt |= TCINTEN | TCCHEN | + EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel)); + edma_write_slot(link, &prtd->asp_params); + + /* ram */ + link = prtd->ram_link; + edma_set_src(link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," + "for asp:%u %u %u\n", __func__, + prtd->ram_channel, prtd->ram_link, prtd->ram_link2, + prtd->asp_channel, prtd->asp_link[0], + prtd->asp_link[1]); + return 0; +exit4: + edma_free_channel(prtd->asp_link[1]); + prtd->asp_link[1] = -1; +exit3: + edma_free_channel(prtd->ram_link); + prtd->ram_link = -1; +exit2: + edma_free_channel(prtd->ram_channel); + prtd->ram_channel = -1; +exit1: + return link; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { + struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct edmacc_param p_ram; - int ret; + struct davinci_pcm_dma_params *params = prtd->params; + int link; - /* Request master DMA channel */ - ret = edma_alloc_channel(prtd->params->channel, - davinci_pcm_dma_irq, substream, - EVENTQ_0); - if (ret < 0) - return ret; - prtd->asp_channel = ret; + if (!params) + return -ENODEV; - /* Request parameter RAM reload slot */ - ret = edma_alloc_slot(EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) { - edma_free_channel(prtd->asp_channel); - return ret; + /* Request asp master DMA channel */ + link = prtd->asp_channel = edma_alloc_channel(params->channel, + davinci_pcm_dma_irq, substream, EVENTQ_0); + if (link < 0) + goto exit1; + + /* Request asp link channels */ + link = prtd->asp_link[0] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit2; + + iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; + if (iram_dma) { + if (request_ping_pong(substream, prtd, iram_dma) == 0) + return 0; + printk(KERN_WARNING "%s: dma channel allocation failed," + "not using sram\n", __func__); } - prtd->asp_link[0] = ret; /* Issue transfer completion IRQ when the channel completes a * transfer, then always reload from the same slot (by a kind @@ -169,12 +504,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(prtd->asp_link[0], &p_ram); - p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; - edma_write_slot(prtd->asp_link[0], &p_ram); - + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt |= TCINTEN | + EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; + edma_write_slot(link, &prtd->asp_params); return 0; +exit2: + edma_free_channel(prtd->asp_channel); + prtd->asp_channel = -1; +exit1: + return link; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -208,14 +548,34 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct edmacc_param temp; + if (prtd->ram_channel >= 0) { + int ret = ping_pong_dma_setup(substream); + if (ret < 0) + return ret; + + edma_write_slot(prtd->ram_channel, &prtd->ram_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); + + print_buf_info(prtd->ram_channel, "ram_channel"); + print_buf_info(prtd->ram_link, "ram_link"); + print_buf_info(prtd->ram_link2, "ram_link2"); + print_buf_info(prtd->asp_channel, "asp_channel"); + print_buf_info(prtd->asp_link[0], "asp_link[0]"); + print_buf_info(prtd->asp_link[1], "asp_link[1]"); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* copy 1st iram buffer */ + edma_start(prtd->ram_channel); + } + return 0; + } prtd->period = 0; davinci_pcm_enqueue_dma(substream); /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->asp_link[0], &temp); - edma_write_slot(prtd->asp_channel, &temp); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); davinci_pcm_enqueue_dma(substream); return 0; @@ -231,13 +591,46 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) dma_addr_t asp_src, asp_dst; spin_lock(&prtd->lock); - - edma_get_position(prtd->asp_channel, &asp_src, &asp_dst); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - asp_count = asp_src - runtime->dma_addr; - else - asp_count = asp_dst - runtime->dma_addr; - + if (prtd->ram_channel >= 0) { + int ram_count; + int mod_ram; + dma_addr_t ram_src, ram_dst; + unsigned int period_size = snd_pcm_lib_period_bytes(substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* reading ram before asp should be safe + * as long as the asp transfers less than a ping size + * of bytes between the 2 reads + */ + edma_get_position(prtd->ram_channel, + &ram_src, &ram_dst); + edma_get_position(prtd->asp_channel, + &asp_src, &asp_dst); + asp_count = asp_src - prtd->asp_params.src; + ram_count = ram_src - prtd->ram_params.src; + mod_ram = ram_count % period_size; + mod_ram -= asp_count; + if (mod_ram < 0) + mod_ram += period_size; + else if (mod_ram == 0) { + if (snd_pcm_running(substream)) + mod_ram += period_size; + } + ram_count -= mod_ram; + if (ram_count < 0) + ram_count += period_size * runtime->periods; + } else { + edma_get_position(prtd->ram_channel, + &ram_src, &ram_dst); + ram_count = ram_dst - prtd->ram_params.dst; + } + asp_count = ram_count; + } else { + edma_get_position(prtd->asp_channel, &asp_src, &asp_dst); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + asp_count = asp_src - runtime->dma_addr; + else + asp_count = asp_dst - runtime->dma_addr; + } spin_unlock(&prtd->lock); offset = bytes_to_frames(runtime, asp_count); @@ -251,6 +644,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; + struct snd_pcm_hardware *ppcm; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; @@ -259,7 +653,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENODEV; params = &pa[substream->stream]; - snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &pcm_hardware_playback : &pcm_hardware_capture; + allocate_sram(substream, params->sram_size, ppcm); + snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -272,6 +669,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) spin_lock_init(&prtd->lock); prtd->params = params; + prtd->asp_channel = -1; + prtd->asp_link[0] = prtd->asp_link[1] = -1; + prtd->ram_channel = -1; + prtd->ram_link = -1; + prtd->ram_link2 = -1; runtime->private_data = prtd; @@ -289,10 +691,29 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - edma_unlink(prtd->asp_link[0]); - - edma_free_slot(prtd->asp_link[0]); - edma_free_channel(prtd->asp_channel); + if (prtd->ram_channel >= 0) + edma_stop(prtd->ram_channel); + if (prtd->asp_channel >= 0) + edma_stop(prtd->asp_channel); + if (prtd->asp_link[0] >= 0) + edma_unlink(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_unlink(prtd->asp_link[1]); + if (prtd->ram_link >= 0) + edma_unlink(prtd->ram_link); + + if (prtd->asp_link[0] >= 0) + edma_free_slot(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_free_slot(prtd->asp_link[1]); + if (prtd->asp_channel >= 0) + edma_free_channel(prtd->asp_channel); + if (prtd->ram_link >= 0) + edma_free_slot(prtd->ram_link); + if (prtd->ram_link2 >= 0) + edma_free_slot(prtd->ram_link2); + if (prtd->ram_channel >= 0) + edma_free_channel(prtd->ram_channel); kfree(prtd); @@ -334,11 +755,11 @@ static struct snd_pcm_ops davinci_pcm_ops = { .mmap = davinci_pcm_mmap, }; -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, + size_t size) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = davinci_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -363,6 +784,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) int stream; for (stream = 0; stream < 2; stream++) { + struct snd_dma_buffer *iram_dma; substream = pcm->streams[stream].substream; if (!substream) continue; @@ -374,6 +796,11 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; + iram_dma = (struct snd_dma_buffer *)buf->private_data; + if (iram_dma) { + sram_free(iram_dma->area, iram_dma->bytes); + kfree(iram_dma); + } } } @@ -391,14 +818,16 @@ static int davinci_pcm_new(struct snd_card *card, if (dai->playback.channels_min) { ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); + SNDRV_PCM_STREAM_PLAYBACK, + pcm_hardware_playback.buffer_bytes_max); if (ret) return ret; } if (dai->capture.channels_min) { ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); + SNDRV_PCM_STREAM_CAPTURE, + pcm_hardware_capture.buffer_bytes_max); if (ret) return ret; } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index c8b0d2baf05a..0764944cf10f 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -20,6 +20,7 @@ struct davinci_pcm_dma_params { int channel; /* sync dma channel ID */ unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned sram_size; enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; -- cgit v1.2.2 From 2b7b250df74f1f9e15cdf33fa90f6c98a419842d Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Wed, 18 Nov 2009 17:49:54 -0700 Subject: ASoC: DaVinci: use edma_pause, edma_resume Use edma_pause and edma_resume to make missing dma_events less likely. This may not be needed, but it looks better. Signed-off-by: Troy Kisky Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 664d49336508..ad4d7f47a86b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -528,12 +528,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_start(prtd->asp_channel); + edma_resume(prtd->asp_channel); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_stop(prtd->asp_channel); + edma_pause(prtd->asp_channel); break; default: ret = -EINVAL; @@ -568,6 +568,7 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) /* copy 1st iram buffer */ edma_start(prtd->ram_channel); } + edma_start(prtd->asp_channel); return 0; } prtd->period = 0; @@ -577,6 +578,7 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) edma_read_slot(prtd->asp_link[0], &prtd->asp_params); edma_write_slot(prtd->asp_channel, &prtd->asp_params); davinci_pcm_enqueue_dma(substream); + edma_start(prtd->asp_channel); return 0; } -- cgit v1.2.2 From b2a2236d1f5e7c09c8e74b61f13d8ba3fe82f7be Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Enric=20Balletb=C3=B2=20i=20Serra?= Date: Wed, 18 Nov 2009 15:59:24 +0100 Subject: ASoC: Add support for IGEP v2 Signed-off-by: Enric Balletbo i Serra Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 7 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/igep0020.c | 148 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 157 insertions(+) create mode 100644 sound/soc/omap/igep0020.c (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4dc6b15a852f..61952aa6cd5a 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -109,3 +109,10 @@ config SND_OMAP_SOC_ZOOM2 help Say Y if you want to add support for Soc audio on Zoom2 board. +config SND_OMAP_SOC_IGEP0020 + tristate "SoC Audio support for IGEP v2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on IGEP v2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0c78ae4e6b97..d49458a29bb7 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -17,6 +17,7 @@ snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o +snd-soc-igep0020-objs := igep0020.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o @@ -29,3 +30,4 @@ obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o +obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c new file mode 100644 index 000000000000..3583c429f9be --- /dev/null +++ b/sound/soc/omap/igep0020.c @@ -0,0 +1,148 @@ +/* + * igep0020.c -- SoC audio for IGEP v2 + * + * Based on sound/soc/omap/overo.c by Steve Sakoman + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int igep2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops igep2_ops = { + .hw_params = igep2_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link igep2_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &igep2_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_igep2 = { + .name = "igep2", + .platform = &omap_soc_platform, + .dai_link = &igep2_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device igep2_snd_devdata = { + .card = &snd_soc_card_igep2, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *igep2_snd_device; + +static int __init igep2_soc_init(void) +{ + int ret; + + if (!machine_is_igep0020()) { + pr_debug("Not IGEP v2!\n"); + return -ENODEV; + } + printk(KERN_INFO "IGEP v2 SoC init\n"); + + igep2_snd_device = platform_device_alloc("soc-audio", -1); + if (!igep2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(igep2_snd_device, &igep2_snd_devdata); + igep2_snd_devdata.dev = &igep2_snd_device->dev; + *(unsigned int *)igep2_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(igep2_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(igep2_snd_device); + + return ret; +} +module_init(igep2_soc_init); + +static void __exit igep2_soc_exit(void) +{ + platform_device_unregister(igep2_snd_device); +} +module_exit(igep2_soc_exit); + +MODULE_AUTHOR("Enric Balletbo i Serra "); +MODULE_DESCRIPTION("ALSA SoC IGEP v2"); +MODULE_LICENSE("GPL"); -- cgit v1.2.2 From f2624791a0c2a2d7664b12d75ca327917141fd3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Nov 2009 11:48:44 +0100 Subject: ALSA: hda - Change quirk for Acer Aspire 5930G Change the quirk for Acer Aspire 5930G from model=acer-aspire-4930g to model=acer-aspre-6530g. The tuba bass gets muted along with the other built-in speakers upon headphones insertion, the internal mic works perfectly etc. Reported-by: Claudio Viano Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28acbe63dfc8..d29fa18232ad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8754,7 +8754,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", ALC888_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", -- cgit v1.2.2 From 4b28dca86066596721a6243c94611dab41970079 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 18 Nov 2009 17:29:36 +0100 Subject: ALSA: cs4236: add dB scale for all volume controls Use db scale for all volume controls according to Crystal's datasheets. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/cs423x/cs4236_lib.c | 152 ++++++++++++++++++++++++++++++------------ 1 file changed, 108 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 4c4024a73c6b..c5adca300632 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -88,6 +88,7 @@ #include #include #include +#include /* * @@ -399,6 +400,14 @@ int snd_cs4236_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } +#define CS4236_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_single, \ + .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 16) & 0xff; @@ -502,6 +511,16 @@ static int snd_cs4236_put_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_ .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 24) & 0xff; @@ -572,12 +591,23 @@ static int snd_cs4236_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE1_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_get_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_wss *chip = snd_kcontrol_chip(kcontrol); @@ -631,16 +661,18 @@ static int snd_cs4236_put_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_ return change; } -#define CS4236_MASTER_DIGITAL(xname, xindex) \ +#define CS4236_MASTER_DIGITAL(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_master_digital, .put = snd_cs4236_put_master_digital, \ - .private_value = 71 << 24 } + .private_value = 71 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4236_mixer_master_digital_invert_volume(int vol) { return (vol < 64) ? 63 - vol : 64 + (71 - vol); -} +} static int snd_cs4236_get_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -673,11 +705,13 @@ static int snd_cs4236_put_master_digital(struct snd_kcontrol *kcontrol, struct s return change; } -#define CS4235_OUTPUT_ACCU(xname, xindex) \ +#define CS4235_OUTPUT_ACCU(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4235_get_output_accu, .put = snd_cs4235_put_output_accu, \ - .private_value = 3 << 24 } + .private_value = 3 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4235_mixer_output_accu_get_volume(int vol) { @@ -732,41 +766,56 @@ static int snd_cs4235_put_output_accu(struct snd_kcontrol *kcontrol, struct snd_ return change; } +static const DECLARE_TLV_DB_SCALE(db_scale_7bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit_12db_max, -8250, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_22db_max, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); + static struct snd_kcontrol_new snd_cs4236_controls[] = { CS4236_DOUBLE("Master Digital Playback Switch", 0, CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), CS4236_DOUBLE("Master Digital Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), +CS4236_MASTER_DIGITAL("Master Digital Volume", 0, db_scale_7bit), -CS4236_DOUBLE("Capture Boost Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Boost Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Playback Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), -CS4236_DOUBLE("DSP Playback Volume", 0, - CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("DSP Playback Volume", 0, + CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("FM Playback Switch", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1), -CS4236_DOUBLE("FM Playback Volume", 0, - CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("FM Playback Volume", 0, + CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("Wavetable Playback Switch", 0, CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), -CS4236_DOUBLE("Wavetable Playback Volume", 0, - CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("Wavetable Playback Volume", 0, + CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1, + db_scale_6bit_12db_max), WSS_DOUBLE("Synth Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Synth Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Synth Capture Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), WSS_DOUBLE("Synth Capture Bypass", 0, @@ -776,14 +825,16 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), -CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), +CS4236_DOUBLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, + 0, 0, 31, 1, db_scale_5bit_22db_max), CS4236_DOUBLE("Mic Playback Boost (+20dB)", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Line Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), WSS_DOUBLE("Line Capture Bypass", 0, @@ -791,8 +842,9 @@ WSS_DOUBLE("Line Capture Bypass", 0, WSS_DOUBLE("CD Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("CD Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("CD Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("CD Capture Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), @@ -800,44 +852,53 @@ CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), WSS_DOUBLE("Analog Loopback Capture Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), -WSS_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, - CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) +WSS_SINGLE("Loopback Digital Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +CS4236_DOUBLE1_TLV("Loopback Digital Playback Volume", 0, + CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1, + db_scale_6bit), }; +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_6db_max, -5600, 200, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit_16db_max, -2400, 800, 0); + static struct snd_kcontrol_new snd_cs4235_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -WSS_DOUBLE("Master Playback Volume", 0, - CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1, + db_scale_5bit_6db_max), -CS4235_OUTPUT_ACCU("Playback Volume", 0), +CS4235_OUTPUT_ACCU("Playback Volume", 0, db_scale_2bit_16db_max), WSS_DOUBLE("Synth Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), WSS_DOUBLE("Synth Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -WSS_DOUBLE("Synth Volume", 1, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 1, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), -CS4236_DOUBLE("Capture Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -WSS_DOUBLE("PCM Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), @@ -850,22 +911,25 @@ CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), -CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), +CS4236_SINGLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1, + db_scale_5bit_22db_max), CS4236_SINGLE("Mic Boost (+20dB)", 0, CS4236_LEFT_MIC, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Line Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("CD Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE("CD Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("CD Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("CD Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -- cgit v1.2.2 From d867bba94513cf149cb8462a6e006848acb91d38 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 19 Nov 2009 14:34:33 +0100 Subject: sound: usb-audio: add Roland UA-1G support Add support for the Roland UA-1G audio interface. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f6f201eb24ce..a892bda03df9 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1563,6 +1563,29 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* has ID 0x00ea when not in Advanced Driver mode */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e9), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-1G", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- cgit v1.2.2 From fbc543915ffb8ec5c35403f294ab799f1936f42a Mon Sep 17 00:00:00 2001 From: Akinobu Mita Date: Fri, 20 Nov 2009 14:56:52 +0900 Subject: ALSA: sound: usbmidi: Use hweight16 Use hweight16 instead of Brian Kernighan's/Peter Wegner's method Signed-off-by: Akinobu Mita Signed-off-by: Takashi Iwai --- sound/usb/usbmidi.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19ceb7e1..e5b068996371 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1062,15 +1062,6 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, return 0; } -static unsigned int snd_usbmidi_count_bits(unsigned int x) -{ - unsigned int bits; - - for (bits = 0; x; ++bits) - x &= x - 1; - return bits; -} - /* * Frees an output endpoint. * May be called when ep hasn't been initialized completely. @@ -1914,8 +1905,8 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, out_ports = 0; in_ports = 0; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - out_ports += snd_usbmidi_count_bits(endpoints[i].out_cables); - in_ports += snd_usbmidi_count_bits(endpoints[i].in_cables); + out_ports += hweight16(endpoints[i].out_cables); + in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); if (err < 0) { -- cgit v1.2.2 From 7cef4cf1c5e9d81554137f52b96a5ab7f6241cdd Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=C5=81ukasz=20Wojni=C5=82owicz?= Date: Fri, 20 Nov 2009 12:14:35 +0100 Subject: ALSA: hda - 4930g mute lfe and side when pluging in headphones MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes first issue from comment 0021423 in bug 0004317 for Acer Aspire 5930g Signed-off-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d29fa18232ad..eedbe19306a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1772,6 +1772,8 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; } static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) -- cgit v1.2.2 From fc08722510494e8185e176713de8c47238512591 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 21 Nov 2009 19:57:11 +0100 Subject: ALSA: hda - Fix input and jack Kconfig depenencies CONFIG_SND_JACK needs to be selected explicitly only when INPUT=y or INPUT_SND. The current way, INPUT=SND_HDA_INTEL isn't strict enough. Reported-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 25ae10e16f59..556cff937be7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -51,7 +51,7 @@ config SND_HDA_INPUT_BEEP_MODE config SND_HDA_INPUT_JACK bool "Support jack plugging notification via input layer" - depends on INPUT=y || INPUT=SND_HDA_INTEL + depends on INPUT=y || INPUT=SND select SND_JACK help Say Y here to enable the jack plugging notification via -- cgit v1.2.2 From 616ad593fe37ef265e5cb1282db6ca264197ffb2 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 21 Nov 2009 01:01:18 +0100 Subject: ALSA: opti-miro: remove snd_card pointer from snd_miro structure Remove the snd_card pointer from the snd_miro structure and do some small code improvements. Also, move Opti chipset detection before detection of the ACI mixer, so the mci_base value is set in one place only. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 53 +++++++++++++++++++++--------------------------- 1 file changed, 23 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 17761030affa..db4a4fbdc5ca 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -110,7 +110,6 @@ struct snd_miro { unsigned long pwd_reg; spinlock_t lock; - struct snd_card *card; struct snd_pcm *pcm; long wss_base; @@ -132,8 +131,6 @@ struct snd_miro { struct mutex aci_mutex; }; -static void snd_miro_proc_init(struct snd_miro * miro); - static char * snd_opti9xx_names[] = { "unkown", "82C928", "82C929", @@ -457,11 +454,9 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right = ucontrol->value.integer.value[1]; setreg_right = (kcontrol->private_value >> 8) & 0xff; - if (setreg_right == ACI_SET_MASTER) { - setreg_left = setreg_right + 1; - } else { - setreg_left = setreg_right + 8; - } + setreg_left = setreg_right + 8; + if (setreg_right == ACI_SET_MASTER) + setreg_left -= 7; getreg_right = kcontrol->private_value & 0xff; getreg_left = getreg_right + 1; @@ -667,17 +662,15 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) return 0; } -static int __devinit snd_miro_mixer(struct snd_miro *miro) +static int __devinit snd_miro_mixer(struct snd_card *card, + struct snd_miro *miro) { - struct snd_card *card; unsigned int idx; int err; - if (snd_BUG_ON(!miro || !miro->card)) + if (snd_BUG_ON(!miro || !card)) return -EINVAL; - card = miro->card; - switch (miro->hardware) { case OPTi9XX_HW_82C924: strcpy(card->mixername, "ACI & OPTi924"); @@ -950,11 +943,12 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, " preamp : 0x%x\n", miro->aci_preamp); } -static void __devinit snd_miro_proc_init(struct snd_miro * miro) +static void __devinit snd_miro_proc_init(struct snd_card *card, + struct snd_miro *miro) { struct snd_info_entry *entry; - if (! snd_card_proc_new(miro->card, "miro", &entry)) + if (!snd_card_proc_new(card, "miro", &entry)) snd_info_set_text_ops(entry, miro, snd_miro_proc_read); } @@ -971,20 +965,18 @@ static int __devinit snd_miro_configure(struct snd_miro *chip) unsigned char mpu_irq_bits; unsigned long flags; + snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ + snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); + switch (chip->hardware) { case OPTi9XX_HW_82C924: snd_miro_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; case OPTi9XX_HW_82C929: /* untested init commands for OPTi929 */ - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(4), 0x00, 0x0c); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; default: snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); @@ -1156,7 +1148,6 @@ static int __devinit snd_card_miro_aci_detect(struct snd_card *card, /* get ACI port from OPTi9xx MC 4 */ - miro->mc_base = 0xf8c; regval=inb(miro->mc_base + 4); miro->aci_port = (regval & 0x10) ? 0x344: 0x354; @@ -1232,7 +1223,13 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) card->private_free = snd_card_miro_free; miro = card->private_data; - miro->card = card; + + error = snd_card_miro_detect(card, miro); + if (error < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + return -ENODEV; + } if ((error = snd_card_miro_aci_detect(card, miro)) < 0) { snd_card_free(card); @@ -1241,13 +1238,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } /* init proc interface */ - snd_miro_proc_init(miro); + snd_miro_proc_init(card, miro); - if ((error = snd_card_miro_detect(card, miro)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); - return -ENODEV; - } if (! miro->res_mc_base && (miro->res_mc_base = request_region(miro->mc_base, miro->mc_base_size, @@ -1341,7 +1333,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) miro->pcm = pcm; - if ((error = snd_miro_mixer(miro)) < 0) { + error = snd_miro_mixer(card, miro); + if (error < 0) { snd_card_free(card); return error; } -- cgit v1.2.2 From 9aeba6297151abcb1b34f3237e4c028aae500ce4 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 22 Nov 2009 17:23:45 +0100 Subject: ALSA: opti-miro: make miro.h header available outside the alsa directory Move the miro.h header to the include/sound directory. It can be used in the Miro PCM20 radio driver (v4l). Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/miro.h | 73 ------------------------------------------------ 2 files changed, 1 insertion(+), 74 deletions(-) delete mode 100644 sound/isa/opti9xx/miro.h (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index db4a4fbdc5ca..932a067ef980 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -40,7 +40,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include -#include "miro.h" +#include MODULE_AUTHOR("Martin Langer "); MODULE_LICENSE("GPL"); diff --git a/sound/isa/opti9xx/miro.h b/sound/isa/opti9xx/miro.h deleted file mode 100644 index 6e1385b8e07e..000000000000 --- a/sound/isa/opti9xx/miro.h +++ /dev/null @@ -1,73 +0,0 @@ -#ifndef _MIRO_H_ -#define _MIRO_H_ - -#define ACI_REG_COMMAND 0 /* write register offset */ -#define ACI_REG_STATUS 1 /* read register offset */ -#define ACI_REG_BUSY 2 /* busy register offset */ -#define ACI_REG_RDS 2 /* PCM20: RDS register offset */ -#define ACI_MINTIME 500 /* ACI time out limit */ - -#define ACI_SET_MUTE 0x0d -#define ACI_SET_POWERAMP 0x0f -#define ACI_SET_TUNERMUTE 0xa3 -#define ACI_SET_TUNERMONO 0xa4 -#define ACI_SET_IDE 0xd0 -#define ACI_SET_WSS 0xd1 -#define ACI_SET_SOLOMODE 0xd2 -#define ACI_SET_PREAMP 0x03 -#define ACI_GET_PREAMP 0x21 -#define ACI_WRITE_TUNE 0xa7 -#define ACI_READ_TUNERSTEREO 0xa8 -#define ACI_READ_TUNERSTATION 0xa9 -#define ACI_READ_VERSION 0xf1 -#define ACI_READ_IDCODE 0xf2 -#define ACI_INIT 0xff -#define ACI_STATUS 0xf0 -#define ACI_S_GENERAL 0x00 -#define ACI_ERROR_OP 0xdf - -/* ACI Mixer */ - -/* These are the values for the right channel GET registers. - Add an offset of 0x01 for the left channel register. - (left=right+0x01) */ - -#define ACI_GET_MASTER 0x03 -#define ACI_GET_MIC 0x05 -#define ACI_GET_LINE 0x07 -#define ACI_GET_CD 0x09 -#define ACI_GET_SYNTH 0x0b -#define ACI_GET_PCM 0x0d -#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */ -#define ACI_GET_LINE2 0x12 - -#define ACI_GET_EQ1 0x22 /* from Bass ... */ -#define ACI_GET_EQ2 0x24 -#define ACI_GET_EQ3 0x26 -#define ACI_GET_EQ4 0x28 -#define ACI_GET_EQ5 0x2a -#define ACI_GET_EQ6 0x2c -#define ACI_GET_EQ7 0x2e /* ... to Treble */ - -/* And these are the values for the right channel SET registers. - For left channel access you have to add an offset of 0x08. - MASTER is an exception, which needs an offset of 0x01 */ - -#define ACI_SET_MASTER 0x00 -#define ACI_SET_MIC 0x30 -#define ACI_SET_LINE 0x31 -#define ACI_SET_CD 0x34 -#define ACI_SET_SYNTH 0x33 -#define ACI_SET_PCM 0x32 -#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */ -#define ACI_SET_LINE2 0x36 - -#define ACI_SET_EQ1 0x40 /* from Bass ... */ -#define ACI_SET_EQ2 0x41 -#define ACI_SET_EQ3 0x42 -#define ACI_SET_EQ4 0x43 -#define ACI_SET_EQ5 0x44 -#define ACI_SET_EQ6 0x45 -#define ACI_SET_EQ7 0x46 /* ... to Treble */ - -#endif /* _MIRO_H_ */ -- cgit v1.2.2 From 9dc9120c774e1d7e3d939542200bd44829c0059d Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 22 Nov 2009 17:26:34 +0100 Subject: ALSA: opti-miro: expose ACI mixer to outside drivers The ACI mixer is used to control the radio FM module installed on the Miro PCM20 sound card. Expose ACI mixer outside the sound card driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 250 +++++++++++++++++++++++++++-------------------- 1 file changed, 145 insertions(+), 105 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 932a067ef980..40b64cd54c85 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -96,7 +96,6 @@ MODULE_PARM_DESC(ide, "enable ide port"); #define OPTi9XX_MC_REG(n) n - struct snd_miro { unsigned short hardware; unsigned char password; @@ -120,17 +119,11 @@ struct snd_miro { long mpu_port; int mpu_irq; - unsigned long aci_port; - int aci_vendor; - int aci_product; - int aci_version; - int aci_amp; - int aci_preamp; - int aci_solomode; - - struct mutex aci_mutex; + struct snd_miro_aci *aci; }; +static struct snd_miro_aci aci_device; + static char * snd_opti9xx_names[] = { "unkown", "82C928", "82C929", @@ -142,13 +135,14 @@ static char * snd_opti9xx_names[] = { * ACI control */ -static int aci_busy_wait(struct snd_miro * miro) +static int aci_busy_wait(struct snd_miro_aci *aci) { long timeout; unsigned char byte; - for (timeout = 1; timeout <= ACI_MINTIME+30; timeout++) { - if (((byte=inb(miro->aci_port + ACI_REG_BUSY)) & 1) == 0) { + for (timeout = 1; timeout <= ACI_MINTIME + 30; timeout++) { + byte = inb(aci->aci_port + ACI_REG_BUSY); + if ((byte & 1) == 0) { if (timeout >= ACI_MINTIME) snd_printd("aci ready in round %ld.\n", timeout-ACI_MINTIME); @@ -174,10 +168,10 @@ static int aci_busy_wait(struct snd_miro * miro) return -EBUSY; } -static inline int aci_write(struct snd_miro * miro, unsigned char byte) +static inline int aci_write(struct snd_miro_aci *aci, unsigned char byte) { - if (aci_busy_wait(miro) >= 0) { - outb(byte, miro->aci_port + ACI_REG_COMMAND); + if (aci_busy_wait(aci) >= 0) { + outb(byte, aci->aci_port + ACI_REG_COMMAND); return 0; } else { snd_printk(KERN_ERR "aci busy, aci_write(0x%x) stopped.\n", byte); @@ -185,12 +179,12 @@ static inline int aci_write(struct snd_miro * miro, unsigned char byte) } } -static inline int aci_read(struct snd_miro * miro) +static inline int aci_read(struct snd_miro_aci *aci) { unsigned char byte; - if (aci_busy_wait(miro) >= 0) { - byte=inb(miro->aci_port + ACI_REG_STATUS); + if (aci_busy_wait(aci) >= 0) { + byte = inb(aci->aci_port + ACI_REG_STATUS); return byte; } else { snd_printk(KERN_ERR "aci busy, aci_read() stopped.\n"); @@ -198,39 +192,49 @@ static inline int aci_read(struct snd_miro * miro) } } -static int aci_cmd(struct snd_miro * miro, int write1, int write2, int write3) +int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3) { int write[] = {write1, write2, write3}; int value, i; - if (mutex_lock_interruptible(&miro->aci_mutex)) + if (mutex_lock_interruptible(&aci->aci_mutex)) return -EINTR; for (i=0; i<3; i++) { if (write[i]< 0 || write[i] > 255) break; else { - value = aci_write(miro, write[i]); + value = aci_write(aci, write[i]); if (value < 0) goto out; } } - value = aci_read(miro); + value = aci_read(aci); -out: mutex_unlock(&miro->aci_mutex); +out: mutex_unlock(&aci->aci_mutex); return value; } +EXPORT_SYMBOL(snd_aci_cmd); + +static int aci_getvalue(struct snd_miro_aci *aci, unsigned char index) +{ + return snd_aci_cmd(aci, ACI_STATUS, index, -1); +} -static int aci_getvalue(struct snd_miro * miro, unsigned char index) +static int aci_setvalue(struct snd_miro_aci *aci, unsigned char index, + int value) { - return aci_cmd(miro, ACI_STATUS, index, -1); + return snd_aci_cmd(aci, index, value, -1); } -static int aci_setvalue(struct snd_miro * miro, unsigned char index, int value) +struct snd_miro_aci *snd_aci_get_aci(void) { - return aci_cmd(miro, index, value, -1); + if (aci_device.aci_port == 0) + return NULL; + return &aci_device; } +EXPORT_SYMBOL(snd_aci_get_aci); /* * MIXER part @@ -244,8 +248,10 @@ static int snd_miro_get_capture(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if ((value = aci_getvalue(miro, ACI_S_GENERAL)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_S_GENERAL); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", + value); return value; } @@ -262,13 +268,15 @@ static int snd_miro_put_capture(struct snd_kcontrol *kcontrol, value = !(ucontrol->value.integer.value[0]); - if ((error = aci_setvalue(miro, ACI_SET_SOLOMODE, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_SOLOMODE, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", + error); return error; } - change = (value != miro->aci_solomode); - miro->aci_solomode = value; + change = (value != miro->aci->aci_solomode); + miro->aci->aci_solomode = value; return change; } @@ -290,7 +298,7 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if (miro->aci_version <= 176) { + if (miro->aci->aci_version <= 176) { /* OSS says it's not readable with versions < 176. @@ -298,12 +306,14 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, which is a PCM12 with aci_version = 176. */ - ucontrol->value.integer.value[0] = miro->aci_preamp; + ucontrol->value.integer.value[0] = miro->aci->aci_preamp; return 0; } - if ((value = aci_getvalue(miro, ACI_GET_PREAMP)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_GET_PREAMP); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", + value); return value; } @@ -320,13 +330,15 @@ static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_PREAMP, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_PREAMP, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", + error); return error; } - change = (value != miro->aci_preamp); - miro->aci_preamp = value; + change = (value != miro->aci->aci_preamp); + miro->aci->aci_preamp = value; return change; } @@ -337,7 +349,7 @@ static int snd_miro_get_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = miro->aci_amp; + ucontrol->value.integer.value[0] = miro->aci->aci_amp; return 0; } @@ -350,13 +362,14 @@ static int snd_miro_put_amp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_POWERAMP, value)) < 0) { + error = aci_setvalue(miro->aci, ACI_SET_POWERAMP, value); + if (error < 0) { snd_printk(KERN_ERR "snd_miro_put_amp() to %d failed: %d\n", value, error); return error; } - change = (value != miro->aci_amp); - miro->aci_amp = value; + change = (value != miro->aci->aci_amp); + miro->aci->aci_amp = value; return change; } @@ -405,12 +418,14 @@ static int snd_miro_get_double(struct snd_kcontrol *kcontrol, int right_reg = kcontrol->private_value & 0xff; int left_reg = right_reg + 1; - if ((right_val = aci_getvalue(miro, right_reg)) < 0) { + right_val = aci_getvalue(miro->aci, right_reg); + if (right_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", right_reg, right_val); return right_val; } - if ((left_val = aci_getvalue(miro, left_reg)) < 0) { + left_val = aci_getvalue(miro->aci, left_reg); + if (left_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", left_reg, left_val); return left_val; } @@ -446,6 +461,7 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); + struct snd_miro_aci *aci = miro->aci; int left, right, left_old, right_old; int setreg_left, setreg_right, getreg_left, getreg_right; int change, error; @@ -461,12 +477,14 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, getreg_right = kcontrol->private_value & 0xff; getreg_left = getreg_right + 1; - if ((left_old = aci_getvalue(miro, getreg_left)) < 0) { + left_old = aci_getvalue(aci, getreg_left); + if (left_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_left, left_old); return left_old; } - if ((right_old = aci_getvalue(miro, getreg_right)) < 0) { + right_old = aci_getvalue(aci, getreg_right); + if (right_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_right, right_old); return right_old; } @@ -485,13 +503,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right_old = 0x80 - right_old; if (left >= 0) { - if ((error = aci_setvalue(miro, setreg_left, left)) < 0) { + error = aci_setvalue(aci, setreg_left, left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", left, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_left, 0x80 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x80 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - left, error); return error; @@ -499,13 +519,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, } if (right >= 0) { - if ((error = aci_setvalue(miro, setreg_right, right)) < 0) { + error = aci_setvalue(aci, setreg_right, right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", right, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_right, 0x80 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x80 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - right, error); return error; @@ -523,12 +545,14 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, left_old = 0x20 - left_old; right_old = 0x20 - right_old; - if ((error = aci_setvalue(miro, setreg_left, 0x20 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x20 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - left, error); return error; } - if ((error = aci_setvalue(miro, setreg_right, 0x20 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x20 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - right, error); return error; @@ -626,11 +650,13 @@ static unsigned char aci_init_values[][2] __devinitdata = { static int __devinit snd_set_aci_init_values(struct snd_miro *miro) { int idx, error; + struct snd_miro_aci *aci = miro->aci; /* enable WSS on PCM1 */ - if ((miro->aci_product == 'A') && wss) { - if ((error = aci_setvalue(miro, ACI_SET_WSS, wss)) < 0) { + if ((aci->aci_product == 'A') && wss) { + error = aci_setvalue(aci, ACI_SET_WSS, wss); + if (error < 0) { snd_printk(KERN_ERR "enabling WSS mode failed\n"); return error; } @@ -639,7 +665,8 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* enable IDE port */ if (ide) { - if ((error = aci_setvalue(miro, ACI_SET_IDE, ide)) < 0) { + error = aci_setvalue(aci, ACI_SET_IDE, ide); + if (error < 0) { snd_printk(KERN_ERR "enabling IDE port failed\n"); return error; } @@ -647,17 +674,18 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* set common aci values */ - for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) - if ((error = aci_setvalue(miro, aci_init_values[idx][0], - aci_init_values[idx][1])) < 0) { + for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) { + error = aci_setvalue(aci, aci_init_values[idx][0], + aci_init_values[idx][1]); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", aci_init_values[idx][0], error); return error; } - - miro->aci_amp = 0; - miro->aci_preamp = 0; - miro->aci_solomode = 1; + } + aci->aci_amp = 0; + aci->aci_preamp = 0; + aci->aci_solomode = 1; return 0; } @@ -688,7 +716,8 @@ static int __devinit snd_miro_mixer(struct snd_card *card, return err; } - if ((miro->aci_product == 'A') || (miro->aci_product == 'B')) { + if ((miro->aci->aci_product == 'A') || + (miro->aci->aci_product == 'B')) { /* PCM1/PCM12 with power-amp and Line 2 */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_line_control[0], miro))) < 0) return err; @@ -696,16 +725,17 @@ static int __devinit snd_miro_mixer(struct snd_card *card, return err; } - if ((miro->aci_product == 'B') || (miro->aci_product == 'C')) { + if ((miro->aci->aci_product == 'B') || + (miro->aci->aci_product == 'C')) { /* PCM12/PCM20 with mic-preamp */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_preamp_control[0], miro))) < 0) return err; - if (miro->aci_version >= 176) + if (miro->aci->aci_version >= 176) if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_capture_control[0], miro))) < 0) return err; } - if (miro->aci_product == 'C') { + if (miro->aci->aci_product == 'C') { /* PCM20 with radio and 7 band equalizer */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_radio_control[0], miro))) < 0) return err; @@ -843,14 +873,15 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, struct snd_info_buffer *buffer) { struct snd_miro *miro = (struct snd_miro *) entry->private_data; + struct snd_miro_aci *aci = miro->aci; char* model = "unknown"; /* miroSOUND PCM1 pro, early PCM12 */ if ((miro->hardware == OPTi9XX_HW_82C929) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'A')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'A')) { + switch (aci->aci_version) { case 3: model = "miroSOUND PCM1 pro"; break; @@ -863,9 +894,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM12, PCM12 (Rev. E), PCM12 pnp */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'B')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'B')) { + switch (aci->aci_version) { case 4: model = "miroSOUND PCM12"; break; @@ -881,9 +912,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM20 radio */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'C')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'C')) { + switch (aci->aci_version) { case 7: model = "miroSOUND PCM20 radio (Rev. E)"; break; @@ -907,17 +938,17 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "ACI information:\n"); snd_iprintf(buffer, " vendor : "); - switch(miro->aci_vendor) { + switch (aci->aci_vendor) { case 'm': snd_iprintf(buffer, "Miro\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_vendor); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_vendor); break; } snd_iprintf(buffer, " product : "); - switch(miro->aci_product) { + switch (aci->aci_product) { case 'A': snd_iprintf(buffer, "miroSOUND PCM1 pro / (early) PCM12\n"); break; @@ -928,19 +959,19 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "miroSOUND PCM20 radio\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_product); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_product); break; } snd_iprintf(buffer, " firmware: %d (0x%x)\n", - miro->aci_version, miro->aci_version); + aci->aci_version, aci->aci_version); snd_iprintf(buffer, " port : 0x%lx-0x%lx\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); snd_iprintf(buffer, " wss : 0x%x\n", wss); snd_iprintf(buffer, " ide : 0x%x\n", ide); - snd_iprintf(buffer, " solomode: 0x%x\n", miro->aci_solomode); - snd_iprintf(buffer, " amp : 0x%x\n", miro->aci_amp); - snd_iprintf(buffer, " preamp : 0x%x\n", miro->aci_preamp); + snd_iprintf(buffer, " solomode: 0x%x\n", aci->aci_solomode); + snd_iprintf(buffer, " amp : 0x%x\n", aci->aci_amp); + snd_iprintf(buffer, " preamp : 0x%x\n", aci->aci_preamp); } static void __devinit snd_miro_proc_init(struct snd_card *card, @@ -1139,46 +1170,53 @@ static int __devinit snd_card_miro_detect(struct snd_card *card, } static int __devinit snd_card_miro_aci_detect(struct snd_card *card, - struct snd_miro * miro) + struct snd_miro *miro) { unsigned char regval; int i; + struct snd_miro_aci *aci = &aci_device; + + miro->aci = aci; - mutex_init(&miro->aci_mutex); + mutex_init(&aci->aci_mutex); /* get ACI port from OPTi9xx MC 4 */ regval=inb(miro->mc_base + 4); - miro->aci_port = (regval & 0x10) ? 0x344: 0x354; + aci->aci_port = (regval & 0x10) ? 0x344 : 0x354; - if ((miro->res_aci_port = request_region(miro->aci_port, 3, "miro aci")) == NULL) { + miro->res_aci_port = request_region(aci->aci_port, 3, "miro aci"); + if (miro->res_aci_port == NULL) { snd_printk(KERN_ERR "aci i/o area 0x%lx-0x%lx already used.\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); return -ENOMEM; } /* force ACI into a known state */ for (i = 0; i < 3; i++) - if (aci_cmd(miro, ACI_ERROR_OP, -1, -1) < 0) { + if (snd_aci_cmd(aci, ACI_ERROR_OP, -1, -1) < 0) { snd_printk(KERN_ERR "can't force aci into known state.\n"); return -ENXIO; } - if ((miro->aci_vendor=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0 || - (miro->aci_product=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0) { - snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", miro->aci_port); + aci->aci_vendor = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + aci->aci_product = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + if (aci->aci_vendor < 0 || aci->aci_product < 0) { + snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", + aci->aci_port); return -ENXIO; } - if ((miro->aci_version=aci_cmd(miro, ACI_READ_VERSION, -1, -1)) < 0) { + aci->aci_version = snd_aci_cmd(aci, ACI_READ_VERSION, -1, -1); + if (aci->aci_version < 0) { snd_printk(KERN_ERR "can't read aci version on 0x%lx.\n", - miro->aci_port); + aci->aci_port); return -ENXIO; } - if (aci_cmd(miro, ACI_INIT, -1, -1) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { + if (snd_aci_cmd(aci, ACI_INIT, -1, -1) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { snd_printk(KERN_ERR "can't initialize aci.\n"); return -ENXIO; } @@ -1191,6 +1229,7 @@ static void snd_card_miro_free(struct snd_card *card) struct snd_miro *miro = card->private_data; release_and_free_resource(miro->res_aci_port); + miro->aci->aci_port = 0; release_and_free_resource(miro->res_mc_base); } @@ -1250,7 +1289,6 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } miro->wss_base = port; - miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; miro->dma1 = dma1; @@ -1272,6 +1310,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return -EBUSY; } } + miro->mpu_port = mpu_port; + if (miro->irq == SNDRV_AUTO_IRQ) { if ((miro->irq = snd_legacy_find_free_irq(possible_irqs)) < 0) { snd_card_free(card); @@ -1339,9 +1379,9 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return error; } - if (miro->aci_vendor == 'm') { + if (miro->aci->aci_vendor == 'm') { /* It looks like a miro sound card. */ - switch (miro->aci_product) { + switch (miro->aci->aci_product) { case 'A': sprintf(card->shortname, "miroSOUND PCM1 pro / PCM12"); -- cgit v1.2.2 From 88cdca9c7376f2220171d09dfc2f9e41b4834435 Mon Sep 17 00:00:00 2001 From: Russell King Date: Mon, 23 Nov 2009 09:44:10 +0100 Subject: ALSA: AACI cleanup Fix the buffer size calculation to use the size which ALSA is expecting. Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f8213e2d5..a03fe80a7a73 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -18,10 +18,7 @@ #include #include #include - -#include -#include -#include +#include #include #include @@ -534,7 +531,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct aaci_runtime *aacirun = runtime->private_data; aacirun->start = (void *)runtime->dma_area; - aacirun->end = aacirun->start + runtime->dma_bytes; + aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; aacirun->period = aacirun->bytes = frames_to_bytes(runtime, runtime->period_size); -- cgit v1.2.2 From 83dd7408b59c1945069199d712df8c7c64a76e1f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Nov 2009 08:57:53 +0100 Subject: Revert "ALSA: hda - Change quirk for Acer Aspire 5930G" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit f2624791a0c2a2d7664b12d75ca327917141fd3b. Łukasz Wojniłowicz reported that the change causes both internal and external mics not working any more. The headphone jacking issue was fixed by his previous patch, it's better to revert to acer-aspire-4930g model. Reported-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eedbe19306a0..7e8b17a1769a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8756,7 +8756,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", - ALC888_ACER_ASPIRE_6530G), + ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", ALC888_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", -- cgit v1.2.2 From 95a618bdac29c7b0f1a516aec9fc37626dec1af9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Einar=20R=C3=BCnkaru?= Date: Mon, 23 Nov 2009 22:23:49 +0200 Subject: ALSA: hda - Make Dell Vostro 1015n mic and speaker switching work MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Dell Vostro 1015n uses Conexant CX20583-10Z (0x14f1:5067). Patch is based on "olpc-xo-1_5" branch. Dell uses digital mic. Signed-off-by: Einar Rünkaru Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 134 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 134 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0b097fa5421f..36dd5a6bf874 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2009,6 +2009,46 @@ static void cxt5066_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_vostro_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int present; + + struct hda_verb ext_mic_present[] = { + /* enable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, + + /* switch to external mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0}, + + /* disable internal digital mic */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + /* enable internal mic, port C */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + /* switch to internal mic input */ + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, + + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + + present = snd_hda_jack_detect(codec, 0x1a); + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + /* mute internal speaker if HP is plugged */ static void cxt5066_hp_automute(struct hda_codec *codec) { @@ -2041,6 +2081,20 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_vostro_automic(codec); + break; + } +} + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -2282,6 +2336,67 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_vostro[] = { + /* Port A: headphones */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port B: external microphone */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port C: unused */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port D: unused */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port E: unused, but has primary EAPD */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* Port F: unused */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port G: internal speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + /* Audio input selectors */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable unsolicited events for Port A and B */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + static struct hda_verb cxt5066_init_verbs_portd_lo[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { } /* end */ @@ -2303,6 +2418,7 @@ enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ + CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_MODELS }; @@ -2310,6 +2426,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", + [CXT5066_DELL_VOSTO] = "dell-vostro" }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -2318,6 +2435,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), {} }; @@ -2382,6 +2500,19 @@ static int patch_cxt5066(struct hda_codec *codec) /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + break; + case CXT5066_DELL_VOSTO: + codec->patch_ops.unsol_event = cxt5066_vostro_event; + spec->init_verbs[0] = cxt5066_init_verbs_vostro; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->port_d_mode = 0; + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ spec->input_mux = NULL; break; @@ -2402,6 +2533,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5051 }, { .id = 0x14f15066, .name = "CX20582 (Pebble)", .patch = patch_cxt5066 }, + { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", + .patch = patch_cxt5066 }, {} /* terminator */ }; @@ -2409,6 +2542,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15045"); MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); MODULE_ALIAS("snd-hda-codec-id:14f15066"); +MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- cgit v1.2.2 From 96f61d9ade82f3e9503df36809175325e8f5eaca Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 22 Oct 2009 09:06:19 +0200 Subject: sound: usb-audio: allow switching altsetting on Roland USB MIDI devices Add a mixer control to select between the two altsettings on Roland USB MIDI devices where the input endpoint is either bulk or interrupt. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbmidi.c | 107 +++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 106 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index e5b068996371..80b2845bc486 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1,7 +1,7 @@ /* * usbmidi.c - ALSA USB MIDI driver * - * Copyright (c) 2002-2007 Clemens Ladisch + * Copyright (c) 2002-2009 Clemens Ladisch * All rights reserved. * * Based on the OSS usb-midi driver by NAGANO Daisuke, @@ -47,6 +47,7 @@ #include #include #include +#include #include #include #include "usbaudio.h" @@ -109,13 +110,17 @@ struct snd_usb_midi { struct list_head list; struct timer_list error_timer; spinlock_t disc_lock; + struct mutex mutex; struct snd_usb_midi_endpoint { struct snd_usb_midi_out_endpoint *out; struct snd_usb_midi_in_endpoint *in; } endpoints[MIDI_MAX_ENDPOINTS]; unsigned long input_triggered; + unsigned int opened; unsigned char disconnected; + + struct snd_kcontrol *roland_load_ctl; }; struct snd_usb_midi_out_endpoint { @@ -879,6 +884,50 @@ static struct usb_protocol_ops snd_usbmidi_emagic_ops = { }; +static void update_roland_altsetting(struct snd_usb_midi* umidi) +{ + struct usb_interface *intf; + struct usb_host_interface *hostif; + struct usb_interface_descriptor *intfd; + int is_light_load; + + intf = umidi->iface; + is_light_load = intf->cur_altsetting != intf->altsetting; + if (umidi->roland_load_ctl->private_value == is_light_load) + return; + hostif = &intf->altsetting[umidi->roland_load_ctl->private_value]; + intfd = get_iface_desc(hostif); + snd_usbmidi_input_stop(&umidi->list); + usb_set_interface(umidi->chip->dev, intfd->bInterfaceNumber, + intfd->bAlternateSetting); + snd_usbmidi_input_start(&umidi->list); +} + +static void substream_open(struct snd_rawmidi_substream *substream, int open) +{ + struct snd_usb_midi* umidi = substream->rmidi->private_data; + struct snd_kcontrol *ctl; + + mutex_lock(&umidi->mutex); + if (open) { + if (umidi->opened++ == 0 && umidi->roland_load_ctl) { + ctl = umidi->roland_load_ctl; + ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(umidi->chip->card, + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + update_roland_altsetting(umidi); + } + } else { + if (--umidi->opened == 0 && umidi->roland_load_ctl) { + ctl = umidi->roland_load_ctl; + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(umidi->chip->card, + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + } + } + mutex_unlock(&umidi->mutex); +} + static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) { struct snd_usb_midi* umidi = substream->rmidi->private_data; @@ -898,11 +947,13 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) } substream->runtime->private_data = port; port->state = STATE_UNKNOWN; + substream_open(substream, 1); return 0; } static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + substream_open(substream, 0); return 0; } @@ -954,11 +1005,13 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) static int snd_usbmidi_input_open(struct snd_rawmidi_substream *substream) { + substream_open(substream, 1); return 0; } static int snd_usbmidi_input_close(struct snd_rawmidi_substream *substream) { + substream_open(substream, 0); return 0; } @@ -1163,6 +1216,7 @@ static void snd_usbmidi_free(struct snd_usb_midi* umidi) if (ep->in) snd_usbmidi_in_endpoint_delete(ep->in); } + mutex_destroy(&umidi->mutex); kfree(umidi); } @@ -1524,6 +1578,52 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, return 0; } +static int roland_load_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + static const char *const names[] = { "High Load", "Light Load" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item > 1) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int roland_load_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + value->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +static int roland_load_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct snd_usb_midi* umidi = kcontrol->private_data; + int changed; + + if (value->value.enumerated.item[0] > 1) + return -EINVAL; + mutex_lock(&umidi->mutex); + changed = value->value.enumerated.item[0] != kcontrol->private_value; + if (changed) + kcontrol->private_value = value->value.enumerated.item[0]; + mutex_unlock(&umidi->mutex); + return changed; +} + +static struct snd_kcontrol_new roland_load_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "MIDI Input Mode", + .info = roland_load_info, + .get = roland_load_get, + .put = roland_load_put, + .private_value = 1, +}; + /* * On Roland devices, use the second alternate setting to be able to use * the interrupt input endpoint. @@ -1549,6 +1649,10 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) intfd->bAlternateSetting); usb_set_interface(umidi->chip->dev, intfd->bInterfaceNumber, intfd->bAlternateSetting); + + umidi->roland_load_ctl = snd_ctl_new1(&roland_load_ctl, umidi); + if (snd_ctl_add(umidi->chip->card, umidi->roland_load_ctl) < 0) + umidi->roland_load_ctl = NULL; } /* @@ -1834,6 +1938,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi->usb_protocol_ops = &snd_usbmidi_standard_ops; init_timer(&umidi->error_timer); spin_lock_init(&umidi->disc_lock); + mutex_init(&umidi->mutex); umidi->error_timer.function = snd_usbmidi_error_timer; umidi->error_timer.data = (unsigned long)umidi; -- cgit v1.2.2 From d82af9f9aab69e82b86450272588c861364f8879 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 16 Nov 2009 12:23:46 +0100 Subject: sound: usb: make the USB MIDI module more independent Remove the dependecy from the USB MIDI code on the snd_usb_audio structure. This allows using the USB MIDI module from another driver without having to pretend to be the generic USB audio driver. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 38 +++++++++------- sound/usb/usbaudio.h | 7 +-- sound/usb/usbmidi.c | 96 ++++++++++++++++++++++------------------- sound/usb/usx2y/us122l.c | 22 +++++----- sound/usb/usx2y/us122l.h | 1 + sound/usb/usx2y/usX2Yhwdep.c | 2 +- sound/usb/usx2y/usbusx2y.c | 4 +- sound/usb/usx2y/usbusx2y.h | 1 + sound/usb/usx2y/usbusx2yaudio.c | 4 +- 9 files changed, 96 insertions(+), 79 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8db0374e10d5..b074a594c595 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2893,7 +2893,9 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { - if (snd_usb_create_midi_interface(chip, iface, NULL) < 0) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j); continue; } @@ -3038,12 +3040,11 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &uaxx_ep }; - if (chip->usb_id == USB_ID(0x0582, 0x002b)) - return snd_usb_create_midi_interface(chip, iface, - &ua700_quirk); - else - return snd_usb_create_midi_interface(chip, iface, - &uaxx_quirk); + const struct snd_usb_audio_quirk *quirk = + chip->usb_id == USB_ID(0x0582, 0x002b) + ? &ua700_quirk : &uaxx_quirk; + return snd_usbmidi_create(chip->card, iface, + &chip->midi_list, quirk); } if (altsd->bNumEndpoints != 1) @@ -3370,6 +3371,13 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int create_any_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk) +{ + return snd_usbmidi_create(chip->card, intf, &chip->midi_list, quirk); +} + /* * audio-interface quirks * @@ -3387,14 +3395,14 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, static const quirk_func_t quirk_funcs[] = { [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, [QUIRK_COMPOSITE] = create_composite_quirk, - [QUIRK_MIDI_STANDARD_INTERFACE] = snd_usb_create_midi_interface, - [QUIRK_MIDI_FIXED_ENDPOINT] = snd_usb_create_midi_interface, - [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, - [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, - [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, - [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface, - [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, - [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, + [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, + [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, + [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, + [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, + [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, + [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, + [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, + [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index e9a3a9dca15c..40ba8115fb81 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -132,7 +132,6 @@ struct snd_usb_audio { int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ - int next_midi_device; struct list_head mixer_list; /* list of mixer interfaces */ }; @@ -227,8 +226,10 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error); void snd_usb_mixer_disconnect(struct list_head *p); -int snd_usb_create_midi_interface(struct snd_usb_audio *chip, struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk); +int snd_usbmidi_create(struct snd_card *card, + struct usb_interface *iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk *quirk); void snd_usbmidi_input_stop(struct list_head* p); void snd_usbmidi_input_start(struct list_head* p); void snd_usbmidi_disconnect(struct list_head *p); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 80b2845bc486..6e89b8368d9a 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -102,7 +102,8 @@ struct usb_protocol_ops { }; struct snd_usb_midi { - struct snd_usb_audio *chip; + struct usb_device *dev; + struct snd_card *card; struct usb_interface *iface; const struct snd_usb_audio_quirk *quirk; struct snd_rawmidi *rmidi; @@ -111,6 +112,8 @@ struct snd_usb_midi { struct timer_list error_timer; spinlock_t disc_lock; struct mutex mutex; + u32 usb_id; + int next_midi_device; struct snd_usb_midi_endpoint { struct snd_usb_midi_out_endpoint *out; @@ -260,7 +263,7 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb) } } - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; snd_usbmidi_submit_urb(urb, GFP_ATOMIC); } @@ -301,7 +304,7 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) unsigned long flags; spin_lock_irqsave(&ep->buffer_lock, flags); - if (ep->umidi->chip->shutdown) { + if (ep->umidi->disconnected) { spin_unlock_irqrestore(&ep->buffer_lock, flags); return; } @@ -317,7 +320,7 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) dump_urb("sending", urb->transfer_buffer, urb->transfer_buffer_length); - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; if (snd_usbmidi_submit_urb(urb, GFP_ATOMIC) < 0) break; ep->active_urbs |= 1 << urb_index; @@ -354,7 +357,7 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { - in->urbs[j]->dev = umidi->chip->dev; + in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } } @@ -374,7 +377,7 @@ static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep, return -ENOMEM; dump_urb("sending", buf, len); if (ep->urbs[0].urb) - err = usb_bulk_msg(ep->umidi->chip->dev, ep->urbs[0].urb->pipe, + err = usb_bulk_msg(ep->umidi->dev, ep->urbs[0].urb->pipe, buf, len, NULL, 250); kfree(buf); return err; @@ -729,8 +732,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, if (!ep->ports[0].active) return; - count = snd_usb_get_speed(ep->umidi->chip->dev) == USB_SPEED_HIGH - ? 1 : 2; + count = snd_usb_get_speed(ep->umidi->dev) == USB_SPEED_HIGH ? 1 : 2; count = snd_rawmidi_transmit(ep->ports[0].substream, urb->transfer_buffer, count); @@ -898,7 +900,7 @@ static void update_roland_altsetting(struct snd_usb_midi* umidi) hostif = &intf->altsetting[umidi->roland_load_ctl->private_value]; intfd = get_iface_desc(hostif); snd_usbmidi_input_stop(&umidi->list); - usb_set_interface(umidi->chip->dev, intfd->bInterfaceNumber, + usb_set_interface(umidi->dev, intfd->bInterfaceNumber, intfd->bAlternateSetting); snd_usbmidi_input_start(&umidi->list); } @@ -913,7 +915,7 @@ static void substream_open(struct snd_rawmidi_substream *substream, int open) if (umidi->opened++ == 0 && umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; - snd_ctl_notify(umidi->chip->card, + snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); update_roland_altsetting(umidi); } @@ -921,7 +923,7 @@ static void substream_open(struct snd_rawmidi_substream *substream, int open) if (--umidi->opened == 0 && umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; - snd_ctl_notify(umidi->chip->card, + snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); } } @@ -963,7 +965,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, port->active = up; if (up) { - if (port->ep->umidi->chip->shutdown) { + if (port->ep->umidi->disconnected) { /* gobble up remaining bytes to prevent wait in * snd_rawmidi_drain_output */ while (!snd_rawmidi_transmit_empty(substream)) @@ -1041,7 +1043,7 @@ static struct snd_rawmidi_ops snd_usbmidi_input_ops = { static void free_urb_and_buffer(struct snd_usb_midi *umidi, struct urb *urb, unsigned int buffer_length) { - usb_buffer_free(umidi->chip->dev, buffer_length, + usb_buffer_free(umidi->dev, buffer_length, urb->transfer_buffer, urb->transfer_dma); usb_free_urb(urb); } @@ -1088,24 +1090,24 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, } } if (ep_info->in_interval) - pipe = usb_rcvintpipe(umidi->chip->dev, ep_info->in_ep); + pipe = usb_rcvintpipe(umidi->dev, ep_info->in_ep); else - pipe = usb_rcvbulkpipe(umidi->chip->dev, ep_info->in_ep); - length = usb_maxpacket(umidi->chip->dev, pipe, 0); + pipe = usb_rcvbulkpipe(umidi->dev, ep_info->in_ep); + length = usb_maxpacket(umidi->dev, pipe, 0); for (i = 0; i < INPUT_URBS; ++i) { - buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL, + buffer = usb_buffer_alloc(umidi->dev, length, GFP_KERNEL, &ep->urbs[i]->transfer_dma); if (!buffer) { snd_usbmidi_in_endpoint_delete(ep); return -ENOMEM; } if (ep_info->in_interval) - usb_fill_int_urb(ep->urbs[i], umidi->chip->dev, + usb_fill_int_urb(ep->urbs[i], umidi->dev, pipe, buffer, length, snd_usbmidi_in_urb_complete, ep, ep_info->in_interval); else - usb_fill_bulk_urb(ep->urbs[i], umidi->chip->dev, + usb_fill_bulk_urb(ep->urbs[i], umidi->dev, pipe, buffer, length, snd_usbmidi_in_urb_complete, ep); ep->urbs[i]->transfer_flags = URB_NO_TRANSFER_DMA_MAP; @@ -1157,15 +1159,15 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, ep->urbs[i].ep = ep; } if (ep_info->out_interval) - pipe = usb_sndintpipe(umidi->chip->dev, ep_info->out_ep); + pipe = usb_sndintpipe(umidi->dev, ep_info->out_ep); else - pipe = usb_sndbulkpipe(umidi->chip->dev, ep_info->out_ep); - if (umidi->chip->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ + pipe = usb_sndbulkpipe(umidi->dev, ep_info->out_ep); + if (umidi->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ ep->max_transfer = 4; else - ep->max_transfer = usb_maxpacket(umidi->chip->dev, pipe, 1); + ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); for (i = 0; i < OUTPUT_URBS; ++i) { - buffer = usb_buffer_alloc(umidi->chip->dev, + buffer = usb_buffer_alloc(umidi->dev, ep->max_transfer, GFP_KERNEL, &ep->urbs[i].urb->transfer_dma); if (!buffer) { @@ -1173,12 +1175,12 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, return -ENOMEM; } if (ep_info->out_interval) - usb_fill_int_urb(ep->urbs[i].urb, umidi->chip->dev, + usb_fill_int_urb(ep->urbs[i].urb, umidi->dev, pipe, buffer, ep->max_transfer, snd_usbmidi_out_urb_complete, &ep->urbs[i], ep_info->out_interval); else - usb_fill_bulk_urb(ep->urbs[i].urb, umidi->chip->dev, + usb_fill_bulk_urb(ep->urbs[i].urb, umidi->dev, pipe, buffer, ep->max_transfer, snd_usbmidi_out_urb_complete, &ep->urbs[i]); @@ -1412,7 +1414,7 @@ static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) int i; for (i = 0; i < ARRAY_SIZE(snd_usbmidi_port_info); ++i) { - if (snd_usbmidi_port_info[i].id == umidi->chip->usb_id && + if (snd_usbmidi_port_info[i].id == umidi->usb_id && snd_usbmidi_port_info[i].port == number) return &snd_usbmidi_port_info[i]; } @@ -1450,7 +1452,7 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, port_info = find_port_info(umidi, number); name_format = port_info ? port_info->name : "%s MIDI %d"; snprintf(substream->name, sizeof(substream->name), - name_format, umidi->chip->card->shortname, number + 1); + name_format, umidi->card->shortname, number + 1); *rsubstream = substream; } @@ -1548,7 +1550,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, endpoints[epidx].out_ep = usb_endpoint_num(ep); if (usb_endpoint_xfer_int(ep)) endpoints[epidx].out_interval = ep->bInterval; - else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW) + else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW) /* * Low speed bulk transfers don't exist, so * force interrupt transfers for devices like @@ -1568,7 +1570,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, endpoints[epidx].in_ep = usb_endpoint_num(ep); if (usb_endpoint_xfer_int(ep)) endpoints[epidx].in_interval = ep->bInterval; - else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW) + else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW) endpoints[epidx].in_interval = 1; endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; snd_printdd(KERN_INFO "EP %02X: %d jack(s)\n", @@ -1647,11 +1649,11 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n", intfd->bAlternateSetting); - usb_set_interface(umidi->chip->dev, intfd->bInterfaceNumber, + usb_set_interface(umidi->dev, intfd->bInterfaceNumber, intfd->bAlternateSetting); umidi->roland_load_ctl = snd_ctl_new1(&roland_load_ctl, umidi); - if (snd_ctl_add(umidi->chip->card, umidi->roland_load_ctl) < 0) + if (snd_ctl_add(umidi->card, umidi->roland_load_ctl) < 0) umidi->roland_load_ctl = NULL; } @@ -1668,7 +1670,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, struct usb_endpoint_descriptor* epd; int i, out_eps = 0, in_eps = 0; - if (USB_ID_VENDOR(umidi->chip->usb_id) == 0x0582) + if (USB_ID_VENDOR(umidi->usb_id) == 0x0582) snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) @@ -1855,12 +1857,12 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, struct snd_rawmidi *rmidi; int err; - err = snd_rawmidi_new(umidi->chip->card, "USB MIDI", - umidi->chip->next_midi_device++, + err = snd_rawmidi_new(umidi->card, "USB MIDI", + umidi->next_midi_device++, out_ports, in_ports, &rmidi); if (err < 0) return err; - strcpy(rmidi->name, umidi->chip->card->shortname); + strcpy(rmidi->name, umidi->card->shortname); rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; @@ -1899,7 +1901,7 @@ static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep) return; for (i = 0; i < INPUT_URBS; ++i) { struct urb* urb = ep->urbs[i]; - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; snd_usbmidi_submit_urb(urb, GFP_KERNEL); } } @@ -1920,9 +1922,10 @@ void snd_usbmidi_input_start(struct list_head* p) /* * Creates and registers everything needed for a MIDI streaming interface. */ -int snd_usb_create_midi_interface(struct snd_usb_audio* chip, - struct usb_interface* iface, - const struct snd_usb_audio_quirk* quirk) +int snd_usbmidi_create(struct snd_card *card, + struct usb_interface* iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk* quirk) { struct snd_usb_midi* umidi; struct snd_usb_midi_endpoint_info endpoints[MIDI_MAX_ENDPOINTS]; @@ -1932,13 +1935,16 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi = kzalloc(sizeof(*umidi), GFP_KERNEL); if (!umidi) return -ENOMEM; - umidi->chip = chip; + umidi->dev = interface_to_usbdev(iface); + umidi->card = card; umidi->iface = iface; umidi->quirk = quirk; umidi->usb_protocol_ops = &snd_usbmidi_standard_ops; init_timer(&umidi->error_timer); spin_lock_init(&umidi->disc_lock); mutex_init(&umidi->mutex); + umidi->usb_id = USB_ID(le16_to_cpu(umidi->dev->descriptor.idVendor), + le16_to_cpu(umidi->dev->descriptor.idProduct)); umidi->error_timer.function = snd_usbmidi_error_timer; umidi->error_timer.data = (unsigned long)umidi; @@ -1947,7 +1953,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, switch (quirk ? quirk->type : QUIRK_MIDI_STANDARD_INTERFACE) { case QUIRK_MIDI_STANDARD_INTERFACE: err = snd_usbmidi_get_ms_info(umidi, endpoints); - if (chip->usb_id == USB_ID(0x0763, 0x0150)) /* M-Audio Uno */ + if (umidi->usb_id == USB_ID(0x0763, 0x0150)) /* M-Audio Uno */ umidi->usb_protocol_ops = &snd_usbmidi_maudio_broken_running_status_ops; break; @@ -1983,7 +1989,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, * interface 0, so we have to make sure that the USB core looks * again at interface 0 by calling usb_set_interface() on it. */ - usb_set_interface(umidi->chip->dev, 0, 0); + usb_set_interface(umidi->dev, 0, 0); err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; case QUIRK_MIDI_EMAGIC: @@ -2029,14 +2035,14 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, return err; } - list_add(&umidi->list, &umidi->chip->midi_list); + list_add_tail(&umidi->list, midi_list); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) snd_usbmidi_input_start_ep(umidi->endpoints[i].in); return 0; } -EXPORT_SYMBOL(snd_usb_create_midi_interface); +EXPORT_SYMBOL(snd_usbmidi_create); EXPORT_SYMBOL(snd_usbmidi_input_stop); EXPORT_SYMBOL(snd_usbmidi_input_start); EXPORT_SYMBOL(snd_usbmidi_disconnect); diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 00cd54c236b4..0ad061e5728b 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -62,8 +62,8 @@ static int us122l_create_usbmidi(struct snd_card *card) struct usb_device *dev = US122L(card)->chip.dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 1); - return snd_usb_create_midi_interface(&US122L(card)->chip, - iface, &quirk); + return snd_usbmidi_create(card, iface, + &US122L(card)->midi_list, &quirk); } static int us144_create_usbmidi(struct snd_card *card) @@ -84,8 +84,8 @@ static int us144_create_usbmidi(struct snd_card *card) struct usb_device *dev = US122L(card)->chip.dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 0); - return snd_usb_create_midi_interface(&US122L(card)->chip, - iface, &quirk); + return snd_usbmidi_create(card, iface, + &US122L(card)->midi_list, &quirk); } /* @@ -297,7 +297,7 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, static void us122l_stop(struct us122l *us122l) { struct list_head *p; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_stop(p); usb_stream_stop(&us122l->sk); @@ -363,7 +363,7 @@ static bool us122l_start(struct us122l *us122l, snd_printk(KERN_ERR "us122l_start error %i \n", err); goto out; } - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_start(p); success = true; out: @@ -508,7 +508,7 @@ static bool us122l_create_card(struct snd_card *card) if (err < 0) { /* release the midi resources */ struct list_head *p; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_disconnect(p); us122l_stop(us122l); @@ -546,7 +546,7 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) US122L(card)->chip.card = card; mutex_init(&US122L(card)->mutex); init_waitqueue_head(&US122L(card)->sk.sleep); - INIT_LIST_HEAD(&US122L(card)->chip.midi_list); + INIT_LIST_HEAD(&US122L(card)->midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)", @@ -638,7 +638,7 @@ static void snd_us122l_disconnect(struct usb_interface *intf) us122l->chip.shutdown = 1; /* release the midi resources */ - list_for_each(p, &us122l->chip.midi_list) { + list_for_each(p, &us122l->midi_list) { snd_usbmidi_disconnect(p); } @@ -667,7 +667,7 @@ static int snd_us122l_suspend(struct usb_interface *intf, pm_message_t message) if (!us122l) return 0; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_stop(p); mutex_lock(&us122l->mutex); @@ -720,7 +720,7 @@ static int snd_us122l_resume(struct usb_interface *intf) if (err) goto unlock; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_start(p); unlock: mutex_unlock(&us122l->mutex); diff --git a/sound/usb/usx2y/us122l.h b/sound/usb/usx2y/us122l.h index 3d10c4b2a0f5..61ce5d7de0b9 100644 --- a/sound/usb/usx2y/us122l.h +++ b/sound/usb/usx2y/us122l.h @@ -12,6 +12,7 @@ struct us122l { unsigned second_periods_polled; struct file *master; struct file *slave; + struct list_head midi_list; atomic_t mmap_count; }; diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 52e04b2f35d3..f96ab86259d0 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -171,7 +171,7 @@ static int usX2Y_create_usbmidi(struct snd_card *card) &quirk_2 : &quirk_1; snd_printdd("usX2Y_create_usbmidi \n"); - return snd_usb_create_midi_interface(&usX2Y(card)->chip, iface, quirk); + return snd_usbmidi_create(card, iface, &usX2Y(card)->midi_list, quirk); } static int usX2Y_create_alsa_devices(struct snd_card *card) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index cb4bb8373ca2..181337090e48 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -354,7 +354,7 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) usX2Y(card)->chip.card = card; init_waitqueue_head(&usX2Y(card)->prepare_wait_queue); mutex_init(&usX2Y(card)->prepare_mutex); - INIT_LIST_HEAD(&usX2Y(card)->chip.midi_list); + INIT_LIST_HEAD(&usX2Y(card)->midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)", @@ -451,7 +451,7 @@ static void usX2Y_usb_disconnect(struct usb_device *device, void* ptr) usb_kill_urb(usX2Y->In04urb); snd_card_disconnect(card); /* release the midi resources */ - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_disconnect(p); } if (usX2Y->us428ctls_sharedmem) diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 456b5fdbc339..231866ea3491 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -42,6 +42,7 @@ struct usX2Ydev { struct snd_usX2Y_substream *subs[4]; struct snd_usX2Y_substream * volatile prepare_subs; wait_queue_head_t prepare_wait_queue; + struct list_head midi_list; }; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 9efd27f6b52f..b8e2f4691493 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -740,7 +740,7 @@ static int usX2Y_format_set(struct usX2Ydev *usX2Y, snd_pcm_format_t format) alternate = 1; usX2Y->stride = 4; } - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_input_stop(p); } usb_kill_urb(usX2Y->In04urb); @@ -750,7 +750,7 @@ static int usX2Y_format_set(struct usX2Ydev *usX2Y, snd_pcm_format_t format) } usX2Y->In04urb->dev = usX2Y->chip.dev; err = usb_submit_urb(usX2Y->In04urb, GFP_KERNEL); - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_input_start(p); } usX2Y->format = format; -- cgit v1.2.2 From a014bbadb53121e243cac254593e79e3ca89742d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 16 Nov 2009 12:26:30 +0100 Subject: sound: usxxx: cleanup chip field The chip field is no longer needed. Move those of its fields that are actually used to the device structure itself. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 68 ++++++++++++++++++++--------------------- sound/usb/usx2y/us122l.h | 3 +- sound/usb/usx2y/usX2Yhwdep.c | 6 ++-- sound/usb/usx2y/usbusx2y.c | 24 +++++++-------- sound/usb/usx2y/usbusx2y.h | 5 ++- sound/usb/usx2y/usbusx2yaudio.c | 30 +++++++++--------- sound/usb/usx2y/usx2yhwdeppcm.c | 8 ++--- 7 files changed, 72 insertions(+), 72 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 0ad061e5728b..f71cd28eca6b 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -59,7 +59,7 @@ static int us122l_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_US122L, .data = &quirk_data }; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 1); return snd_usbmidi_create(card, iface, @@ -81,7 +81,7 @@ static int us144_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_US122L, .data = &quirk_data }; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 0); return snd_usbmidi_create(card, iface, @@ -194,11 +194,11 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) if (!us122l->first) us122l->first = file; - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - iface = usb_ifnum_to_if(us122l->chip.dev, 0); + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_get_interface(iface); } - iface = usb_ifnum_to_if(us122l->chip.dev, 1); + iface = usb_ifnum_to_if(us122l->dev, 1); usb_autopm_get_interface(iface); return 0; } @@ -209,11 +209,11 @@ static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file) struct usb_interface *iface; snd_printdd(KERN_DEBUG "%p %p\n", hw, file); - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - iface = usb_ifnum_to_if(us122l->chip.dev, 0); + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_put_interface(iface); } - iface = usb_ifnum_to_if(us122l->chip.dev, 1); + iface = usb_ifnum_to_if(us122l->dev, 1); usb_autopm_put_interface(iface); if (us122l->first == file) us122l->first = NULL; @@ -330,7 +330,7 @@ static bool us122l_start(struct us122l *us122l, unsigned use_packsize = 0; bool success = false; - if (us122l->chip.dev->speed == USB_SPEED_HIGH) { + if (us122l->dev->speed == USB_SPEED_HIGH) { /* The us-122l's descriptor defaults to iso max_packsize 78, which isn't needed for samplerates <= 48000. Lets save some memory: @@ -347,11 +347,11 @@ static bool us122l_start(struct us122l *us122l, break; } } - if (!usb_stream_new(&us122l->sk, us122l->chip.dev, 1, 2, + if (!usb_stream_new(&us122l->sk, us122l->dev, 1, 2, rate, use_packsize, period_frames, 6)) goto out; - err = us122l_set_sample_rate(us122l->chip.dev, rate); + err = us122l_set_sample_rate(us122l->dev, rate); if (err < 0) { us122l_stop(us122l); snd_printk(KERN_ERR "us122l_set_sample_rate error \n"); @@ -390,7 +390,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, err = -ENXIO; goto free; } - high_speed = us122l->chip.dev->speed == USB_SPEED_HIGH; + high_speed = us122l->dev->speed == USB_SPEED_HIGH; if ((cfg->sample_rate != 44100 && cfg->sample_rate != 48000 && (!high_speed || (cfg->sample_rate != 88200 && cfg->sample_rate != 96000))) || @@ -450,7 +450,7 @@ static int usb_stream_hwdep_new(struct snd_card *card) { int err; struct snd_hwdep *hw; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; err = snd_hwdep_new(card, SND_USB_STREAM_ID, 0, &hw); if (err < 0) @@ -476,26 +476,26 @@ static bool us122l_create_card(struct snd_card *card) int err; struct us122l *us122l = US122L(card); - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - err = usb_set_interface(us122l->chip.dev, 0, 1); + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); return false; } } - err = usb_set_interface(us122l->chip.dev, 1, 1); + err = usb_set_interface(us122l->dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); return false; } - pt_info_set(us122l->chip.dev, 0x11); - pt_info_set(us122l->chip.dev, 0x10); + pt_info_set(us122l->dev, 0x11); + pt_info_set(us122l->dev, 0x10); if (!us122l_start(us122l, 44100, 256)) return false; - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) + if (us122l->dev->descriptor.idProduct == USB_ID_US144) err = us144_create_usbmidi(card); else err = us122l_create_usbmidi(card); @@ -520,7 +520,7 @@ static bool us122l_create_card(struct snd_card *card) static void snd_us122l_free(struct snd_card *card) { struct us122l *us122l = US122L(card); - int index = us122l->chip.index; + int index = us122l->card_index; if (index >= 0 && index < SNDRV_CARDS) snd_us122l_card_used[index] = 0; } @@ -540,10 +540,9 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) sizeof(struct us122l), &card); if (err < 0) return err; - snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; + snd_us122l_card_used[US122L(card)->card_index = dev] = 1; card->private_free = snd_us122l_free; - US122L(card)->chip.dev = device; - US122L(card)->chip.card = card; + US122L(card)->dev = device; mutex_init(&US122L(card)->mutex); init_waitqueue_head(&US122L(card)->sk.sleep); INIT_LIST_HEAD(&US122L(card)->midi_list); @@ -554,8 +553,8 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) le16_to_cpu(device->descriptor.idVendor), le16_to_cpu(device->descriptor.idProduct), 0, - US122L(card)->chip.dev->bus->busnum, - US122L(card)->chip.dev->devnum + US122L(card)->dev->bus->busnum, + US122L(card)->dev->devnum ); *cardp = card; return 0; @@ -635,16 +634,15 @@ static void snd_us122l_disconnect(struct usb_interface *intf) mutex_lock(&us122l->mutex); us122l_stop(us122l); mutex_unlock(&us122l->mutex); - us122l->chip.shutdown = 1; /* release the midi resources */ list_for_each(p, &us122l->midi_list) { snd_usbmidi_disconnect(p); } - usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 0)); - usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 1)); - usb_put_dev(us122l->chip.dev); + usb_put_intf(usb_ifnum_to_if(us122l->dev, 0)); + usb_put_intf(usb_ifnum_to_if(us122l->dev, 1)); + usb_put_dev(us122l->dev); while (atomic_read(&us122l->mmap_count)) msleep(500); @@ -694,23 +692,23 @@ static int snd_us122l_resume(struct usb_interface *intf) mutex_lock(&us122l->mutex); /* needed, doesn't restart without: */ - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - err = usb_set_interface(us122l->chip.dev, 0, 1); + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); goto unlock; } } - err = usb_set_interface(us122l->chip.dev, 1, 1); + err = usb_set_interface(us122l->dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); goto unlock; } - pt_info_set(us122l->chip.dev, 0x11); - pt_info_set(us122l->chip.dev, 0x10); + pt_info_set(us122l->dev, 0x11); + pt_info_set(us122l->dev, 0x10); - err = us122l_set_sample_rate(us122l->chip.dev, + err = us122l_set_sample_rate(us122l->dev, us122l->sk.s->cfg.sample_rate); if (err < 0) { snd_printk(KERN_ERR "us122l_set_sample_rate error \n"); diff --git a/sound/usb/usx2y/us122l.h b/sound/usb/usx2y/us122l.h index 61ce5d7de0b9..4daf1982e821 100644 --- a/sound/usb/usx2y/us122l.h +++ b/sound/usb/usx2y/us122l.h @@ -3,7 +3,8 @@ struct us122l { - struct snd_usb_audio chip; + struct usb_device *dev; + int card_index; int stride; struct usb_stream_kernel sk; diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index f96ab86259d0..1879b72c40f8 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -114,7 +114,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, struct usX2Ydev *us428 = hw->private_data; int id = -1; - switch (le16_to_cpu(us428->chip.dev->descriptor.idProduct)) { + switch (le16_to_cpu(us428->dev->descriptor.idProduct)) { case USB_ID_US122: id = USX2Y_TYPE_122; break; @@ -164,7 +164,7 @@ static int usX2Y_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &quirk_data_2 }; - struct usb_device *dev = usX2Y(card)->chip.dev; + struct usb_device *dev = usX2Y(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 0); struct snd_usb_audio_quirk *quirk = le16_to_cpu(dev->descriptor.idProduct) == USB_ID_US428 ? @@ -202,7 +202,7 @@ static int snd_usX2Y_hwdep_dsp_load(struct snd_hwdep *hw, snd_printdd( "dsp_load %s\n", dsp->name); if (access_ok(VERIFY_READ, dsp->image, dsp->length)) { - struct usb_device* dev = priv->chip.dev; + struct usb_device* dev = priv->dev; char *buf; buf = memdup_user(dsp->image, dsp->length); diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 181337090e48..c42350eed2eb 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -239,8 +239,8 @@ static void i_usX2Y_In04Int(struct urb *urb) for (j = 0; j < URBS_AsyncSeq && !err; ++j) if (0 == usX2Y->AS04.urb[j]->status) { struct us428_p4out *p4out = us428ctls->p4out + send; // FIXME if more than 1 p4out is new, 1 gets lost. - usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->chip.dev, - usb_sndbulkpipe(usX2Y->chip.dev, 0x04), &p4out->val.vol, + usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->dev, + usb_sndbulkpipe(usX2Y->dev, 0x04), &p4out->val.vol, p4out->type == eLT_Light ? sizeof(struct us428_lights) : 5, i_usX2Y_Out04Int, usX2Y); err = usb_submit_urb(usX2Y->AS04.urb[j], GFP_ATOMIC); @@ -253,7 +253,7 @@ static void i_usX2Y_In04Int(struct urb *urb) if (err) snd_printk(KERN_ERR "In04Int() usb_submit_urb err=%i\n", err); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; usb_submit_urb(urb, GFP_ATOMIC); } @@ -273,8 +273,8 @@ int usX2Y_AsyncSeq04_init(struct usX2Ydev *usX2Y) err = -ENOMEM; break; } - usb_fill_bulk_urb( usX2Y->AS04.urb[i], usX2Y->chip.dev, - usb_sndbulkpipe(usX2Y->chip.dev, 0x04), + usb_fill_bulk_urb( usX2Y->AS04.urb[i], usX2Y->dev, + usb_sndbulkpipe(usX2Y->dev, 0x04), usX2Y->AS04.buffer + URB_DataLen_AsyncSeq*i, 0, i_usX2Y_Out04Int, usX2Y ); @@ -293,7 +293,7 @@ int usX2Y_In04_init(struct usX2Ydev *usX2Y) } init_waitqueue_head(&usX2Y->In04WaitQueue); - usb_fill_int_urb(usX2Y->In04urb, usX2Y->chip.dev, usb_rcvintpipe(usX2Y->chip.dev, 0x4), + usb_fill_int_urb(usX2Y->In04urb, usX2Y->dev, usb_rcvintpipe(usX2Y->dev, 0x4), usX2Y->In04Buf, 21, i_usX2Y_In04Int, usX2Y, 10); @@ -348,10 +348,9 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) sizeof(struct usX2Ydev), &card); if (err < 0) return err; - snd_usX2Y_card_used[usX2Y(card)->chip.index = dev] = 1; + snd_usX2Y_card_used[usX2Y(card)->card_index = dev] = 1; card->private_free = snd_usX2Y_card_private_free; - usX2Y(card)->chip.dev = device; - usX2Y(card)->chip.card = card; + usX2Y(card)->dev = device; init_waitqueue_head(&usX2Y(card)->prepare_wait_queue); mutex_init(&usX2Y(card)->prepare_mutex); INIT_LIST_HEAD(&usX2Y(card)->midi_list); @@ -362,7 +361,7 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) le16_to_cpu(device->descriptor.idVendor), le16_to_cpu(device->descriptor.idProduct), 0,//us428(card)->usbmidi.ifnum, - usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum + usX2Y(card)->dev->bus->busnum, usX2Y(card)->dev->devnum ); *cardp = card; return 0; @@ -432,8 +431,8 @@ static void snd_usX2Y_card_private_free(struct snd_card *card) usb_free_urb(usX2Y(card)->In04urb); if (usX2Y(card)->us428ctls_sharedmem) snd_free_pages(usX2Y(card)->us428ctls_sharedmem, sizeof(*usX2Y(card)->us428ctls_sharedmem)); - if (usX2Y(card)->chip.index >= 0 && usX2Y(card)->chip.index < SNDRV_CARDS) - snd_usX2Y_card_used[usX2Y(card)->chip.index] = 0; + if (usX2Y(card)->card_index >= 0 && usX2Y(card)->card_index < SNDRV_CARDS) + snd_usX2Y_card_used[usX2Y(card)->card_index] = 0; } /* @@ -445,7 +444,6 @@ static void usX2Y_usb_disconnect(struct usb_device *device, void* ptr) struct snd_card *card = ptr; struct usX2Ydev *usX2Y = usX2Y(card); struct list_head *p; - usX2Y->chip.shutdown = 1; usX2Y->chip_status = USX2Y_STAT_CHIP_HUP; usX2Y_unlinkSeq(&usX2Y->AS04); usb_kill_urb(usX2Y->In04urb); diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 231866ea3491..1d174cea352b 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -22,7 +22,8 @@ struct snd_usX2Y_urbSeq { #include "usx2yhwdeppcm.h" struct usX2Ydev { - struct snd_usb_audio chip; + struct usb_device *dev; + int card_index; int stride; struct urb *In04urb; void *In04Buf; @@ -43,6 +44,8 @@ struct usX2Ydev { struct snd_usX2Y_substream * volatile prepare_subs; wait_queue_head_t prepare_wait_queue; struct list_head midi_list; + struct list_head pcm_list; + int pcm_devs; }; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index b8e2f4691493..74a67a85aa81 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -199,7 +199,7 @@ static int usX2Y_urb_submit(struct snd_usX2Y_substream *subs, struct urb *urb, i return -ENODEV; urb->start_frame = (frame + NRURBS * nr_of_packs()); // let hcd do rollover sanity checks urb->hcpriv = NULL; - urb->dev = subs->usX2Y->chip.dev; /* we need to set this at each time */ + urb->dev = subs->usX2Y->dev; /* we need to set this at each time */ if ((err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { snd_printk(KERN_ERR "usb_submit_urb() returned %i\n", err); return err; @@ -300,7 +300,7 @@ static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" "Most propably some urb of usb-frame %i is still missing.\n" "Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); usX2Y_clients_stop(usX2Y); @@ -313,7 +313,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) if (unlikely(atomic_read(&subs->state) < state_PREPARED)) { snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", urb->status, urb->start_frame); return; @@ -424,7 +424,7 @@ static int usX2Y_urbs_allocate(struct snd_usX2Y_substream *subs) int i; unsigned int pipe; int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; - struct usb_device *dev = subs->usX2Y->chip.dev; + struct usb_device *dev = subs->usX2Y->dev; pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) : usb_rcvisocpipe(dev, subs->endpoint); @@ -500,7 +500,7 @@ static int usX2Y_urbs_start(struct snd_usX2Y_substream *subs) unsigned long pack; if (0 == i) atomic_set(&subs->state, state_STARTING3); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * pack; @@ -692,7 +692,7 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) } ((char*)(usbdata + i))[0] = ra[i].c1; ((char*)(usbdata + i))[1] = ra[i].c2; - usb_fill_bulk_urb(us->urb[i], usX2Y->chip.dev, usb_sndbulkpipe(usX2Y->chip.dev, 4), + usb_fill_bulk_urb(us->urb[i], usX2Y->dev, usb_sndbulkpipe(usX2Y->dev, 4), usbdata + i, 2, i_usX2Y_04Int, usX2Y); #ifdef OLD_USB us->urb[i]->transfer_flags = USB_QUEUE_BULK; @@ -744,11 +744,11 @@ static int usX2Y_format_set(struct usX2Ydev *usX2Y, snd_pcm_format_t format) snd_usbmidi_input_stop(p); } usb_kill_urb(usX2Y->In04urb); - if ((err = usb_set_interface(usX2Y->chip.dev, 0, alternate))) { + if ((err = usb_set_interface(usX2Y->dev, 0, alternate))) { snd_printk(KERN_ERR "usb_set_interface error \n"); return err; } - usX2Y->In04urb->dev = usX2Y->chip.dev; + usX2Y->In04urb->dev = usX2Y->dev; err = usb_submit_urb(usX2Y->In04urb, GFP_KERNEL); list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_input_start(p); @@ -955,7 +955,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, struct snd_pcm *pcm; int err, i; struct snd_usX2Y_substream **usX2Y_substream = - usX2Y(card)->subs + 2 * usX2Y(card)->chip.pcm_devs; + usX2Y(card)->subs + 2 * usX2Y(card)->pcm_devs; for (i = playback_endpoint ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; i <= SNDRV_PCM_STREAM_CAPTURE; ++i) { @@ -971,7 +971,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]->endpoint = playback_endpoint; usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]->endpoint = capture_endpoint; - err = snd_pcm_new(card, NAME_ALLCAPS" Audio", usX2Y(card)->chip.pcm_devs, + err = snd_pcm_new(card, NAME_ALLCAPS" Audio", usX2Y(card)->pcm_devs, playback_endpoint ? 1 : 0, 1, &pcm); if (err < 0) { @@ -987,7 +987,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, pcm->private_free = snd_usX2Y_pcm_private_free; pcm->info_flags = 0; - sprintf(pcm->name, NAME_ALLCAPS" Audio #%d", usX2Y(card)->chip.pcm_devs); + sprintf(pcm->name, NAME_ALLCAPS" Audio #%d", usX2Y(card)->pcm_devs); if ((playback_endpoint && 0 > (err = snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, @@ -1001,7 +1001,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, snd_usX2Y_pcm_private_free(pcm); return err; } - usX2Y(card)->chip.pcm_devs++; + usX2Y(card)->pcm_devs++; return 0; } @@ -1013,14 +1013,14 @@ int usX2Y_audio_create(struct snd_card *card) { int err = 0; - INIT_LIST_HEAD(&usX2Y(card)->chip.pcm_list); + INIT_LIST_HEAD(&usX2Y(card)->pcm_list); if (0 > (err = usX2Y_audio_stream_new(card, 0xA, 0x8))) return err; - if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) == USB_ID_US428) + if (le16_to_cpu(usX2Y(card)->dev->descriptor.idProduct) == USB_ID_US428) if (0 > (err = usX2Y_audio_stream_new(card, 0, 0xA))) return err; - if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) != USB_ID_US122) + if (le16_to_cpu(usX2Y(card)->dev->descriptor.idProduct) != USB_ID_US122) err = usX2Y_rate_set(usX2Y(card), 44100); // Lets us428 recognize output-volume settings, disturbs us122. return err; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 4b2304c2e02d..9ed6c3956ca7 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -234,7 +234,7 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) if (unlikely(atomic_read(&subs->state) < state_PREPARED)) { snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", urb->status, urb->start_frame); return; @@ -318,7 +318,7 @@ static int usX2Y_usbpcm_urbs_allocate(struct snd_usX2Y_substream *subs) int i; unsigned int pipe; int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; - struct usb_device *dev = subs->usX2Y->chip.dev; + struct usb_device *dev = subs->usX2Y->dev; pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) : usb_rcvisocpipe(dev, subs->endpoint); @@ -441,7 +441,7 @@ static int usX2Y_usbpcm_urbs_start(struct snd_usX2Y_substream *subs) unsigned long pack; if (0 == u) atomic_set(&subs->state, state_STARTING3); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * (pack + u * nr_of_packs()); @@ -741,7 +741,7 @@ int usX2Y_hwdep_pcm_new(struct snd_card *card) int err; struct snd_hwdep *hw; struct snd_pcm *pcm; - struct usb_device *dev = usX2Y(card)->chip.dev; + struct usb_device *dev = usX2Y(card)->dev; if (1 != nr_of_packs()) return 0; -- cgit v1.2.2 From bbb3c644bd9967753ce8c214c5e64b27c361d2a4 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Tue, 24 Nov 2009 22:51:05 -0500 Subject: ALSA: intel8x0: Mute External Amplifier by default for Gateway 4525GZ BugLink: https://bugs.launchpad.net/bugs/487884 This Gateway model needs External Amplifier muted for audible playback, so set the inv_eapd quirk for it. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index aac20fb4aad2..b990143636f1 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2062,6 +2062,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "MSI P4 ATX 645 Ultra", .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x161f, + .subdevice = 0x203a, + .name = "Gateway 4525GZ", /* AD1981B */ + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x1734, .subdevice = 0x0088, -- cgit v1.2.2 From c0fa59df7214e546f8a37bc677867ac7b67b5c93 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Nov 2009 11:36:10 +0000 Subject: ASoC: Add BCLK calculation utility for TDM mode too Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index b16aaaeb0aab..1d07b931f3d8 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -54,6 +54,12 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) } EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size); +int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots) +{ + return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots); +} +EXPORT_SYMBOL_GPL(snd_soc_calc_bclk); + int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) { int ret; -- cgit v1.2.2 From 0b587fc4d35afb1bc0fc3d890084bb14c78372dc Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 25 Nov 2009 18:27:20 -0500 Subject: ALSA: hda: Fix max PCM level to 0 dB for Fujitsu-Siemens laptops using CX20549 (Venice) BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4792 Cristian reported that these models have really bad sound above 6 dB and proposed the original patch. I've updated the comment to reflect this change. Signed-off-by: Daniel T Chen Reported-by: Cristian Klein Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 36dd5a6bf874..60810ba899d1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1171,9 +1171,10 @@ static int patch_cxt5045(struct hda_codec *codec) switch (codec->subsystem_id >> 16) { case 0x103c: - /* HP laptop has a really bad sound over 0dB on NID 0x17. - * Fix max PCM level to 0 dB - * (originall it has 0x2b steps with 0dB offset 0x14) + case 0x1734: + /* HP & Fujitsu-Siemens laptops have really bad sound over 0dB + * on NID 0x17. Fix max PCM level to 0 dB + * (originally it has 0x2b steps with 0dB offset 0x14) */ snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, (0x14 << AC_AMPCAP_OFFSET_SHIFT) | -- cgit v1.2.2 From 657b1989dacf58e83e7a76bca6d4a91a9f294cf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 12:40:21 +0100 Subject: ALSA: pcm - Use dma_mmap_coherent() if available Use dma_mmap_coherent() for mmapping the buffers allocated via dma_alloc_coherent() if available. Currently, only ARM has this function, so we do temporarily have an ifdef pcm_native.c. This should be handled better globally in future. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 49 +++++++++++++++++++++++++++++++++---------------- 1 file changed, 33 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ab73edf2c89a..f067c5b906e4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -3094,23 +3095,42 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, return 0; } -static const struct vm_operations_struct snd_pcm_vm_ops_data = -{ +static const struct vm_operations_struct snd_pcm_vm_ops_data = { + .open = snd_pcm_mmap_data_open, + .close = snd_pcm_mmap_data_close, +}; + +static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, .fault = snd_pcm_mmap_data_fault, }; +#ifndef ARCH_HAS_DMA_MMAP_COHERENT +/* This should be defined / handled globally! */ +#ifdef CONFIG_ARM +#define ARCH_HAS_DMA_MMAP_COHERENT +#endif +#endif + /* * mmap the DMA buffer on RAM */ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - area->vm_ops = &snd_pcm_vm_ops_data; - area->vm_private_data = substream; area->vm_flags |= VM_RESERVED; - atomic_inc(&substream->mmap_count); +#ifdef ARCH_HAS_DMA_MMAP_COHERENT + if (!substream->ops->page && + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return dma_mmap_coherent(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); +#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ + /* mmap with fault handler */ + area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; } @@ -3118,12 +3138,6 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, * mmap the DMA buffer on I/O memory area */ #if SNDRV_PCM_INFO_MMAP_IOMEM -static const struct vm_operations_struct snd_pcm_vm_ops_data_mmio = -{ - .open = snd_pcm_mmap_data_open, - .close = snd_pcm_mmap_data_close, -}; - int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { @@ -3133,8 +3147,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, #ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); #endif - area->vm_ops = &snd_pcm_vm_ops_data_mmio; - area->vm_private_data = substream; area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; @@ -3142,7 +3154,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, size, area->vm_page_prot)) return -EAGAIN; - atomic_inc(&substream->mmap_count); return 0; } @@ -3159,6 +3170,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, long size; unsigned long offset; size_t dma_bytes; + int err; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (!(area->vm_flags & (VM_WRITE|VM_READ))) @@ -3183,10 +3195,15 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, if (offset > dma_bytes - size) return -EINVAL; + area->vm_ops = &snd_pcm_vm_ops_data; + area->vm_private_data = substream; if (substream->ops->mmap) - return substream->ops->mmap(substream, area); + err = substream->ops->mmap(substream, area); else - return snd_pcm_default_mmap(substream, area); + err = snd_pcm_default_mmap(substream, area); + if (!err) + atomic_inc(&substream->mmap_count); + return err; } EXPORT_SYMBOL(snd_pcm_mmap_data); -- cgit v1.2.2 From 9eb4a06788a598573c751af1a7e46639afc89513 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 12:43:39 +0100 Subject: ALSA: pcm - define snd_pcm_default_page_ops() Add a helper (inline) function as the default page ops. Any hacks wrt the page address conversion will be applied in this function. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f067c5b906e4..c906be26c312 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3062,6 +3062,13 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file } #endif /* coherent mmap */ +static inline struct page * +snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) +{ + void *vaddr = substream->runtime->dma_area + ofs; + return virt_to_page(vaddr); +} + /* * fault callback for mmapping a RAM page */ @@ -3072,7 +3079,6 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, struct snd_pcm_runtime *runtime; unsigned long offset; struct page * page; - void *vaddr; size_t dma_bytes; if (substream == NULL) @@ -3082,14 +3088,12 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) return VM_FAULT_SIGBUS; - if (substream->ops->page) { + if (substream->ops->page) page = substream->ops->page(substream, offset); - if (!page) - return VM_FAULT_SIGBUS; - } else { - vaddr = runtime->dma_area + offset; - page = virt_to_page(vaddr); - } + else + page = snd_pcm_default_page_ops(substream, offset); + if (!page) + return VM_FAULT_SIGBUS; get_page(page); vmf->page = page; return 0; -- cgit v1.2.2 From 74ea23aa6c9a8bece71b35ddeeb7ad6ae6782cd9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Nov 2009 13:55:11 +0200 Subject: ASoC: tlv320dac33: Change RT wq to singlethread wq RT workqueue is going away in the near future, replace it with singlethread wq for now, which is still supported. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2a013e46ae14..9c8903dbe647 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1118,7 +1118,8 @@ static int dac33_i2c_probe(struct i2c_client *client, } if (dac33->irq != -1) { /* Setup work queue */ - dac33->dac33_wq = create_rt_workqueue("tlv320dac33"); + dac33->dac33_wq = + create_singlethread_workqueue("tlv320dac33"); if (dac33->dac33_wq == NULL) { free_irq(dac33->irq, &dac33->codec); ret = -ENOMEM; -- cgit v1.2.2 From 66b6cfacfc5aa2fda37b0d40cd54931ca5ef8cd7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 12:50:01 +0100 Subject: ALSA: pcm - fix page conversion on non-coherent MIPS arch The non-coherent MIPS arch doesn't give the correct address by a simple virt_to_page() for pages allocated via dma_alloc_coherent(). Original patch by Wu Zhangjin . [Ralf mentioned: "The origins of this patch go back far further. The oldest patch I could find which is a superset of this was written by Atsushi Nemoto and various incarnations of it have been sumitted to and reject by me a number of times through the years."] A proper check of the buffer allocation type was added to avoid the wrong conversion. Note that this doesn't fix perfectly: the pages should be marked with proper pgprot value. This will be done in a future implementation like the conversion to dma_mmap_coherent(). Acked-by: Ralf Baechle Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c906be26c312..e48c5f618578 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3066,6 +3066,10 @@ static inline struct page * snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) { void *vaddr = substream->runtime->dma_area + ofs; +#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return virt_to_page(CAC_ADDR(vaddr)); +#endif return virt_to_page(vaddr); } -- cgit v1.2.2 From 6985c8877a711c7c307af05203858cb7c3c89d0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 15:04:24 +0100 Subject: ALSA: pcm - fix page conversion on non-coherent PPC arch The non-cohernet PPC arch doesn't give the correct address by a simple virt_to_page() for pages allocated via dma_alloc_coherent(). This patch adds a hack to fix the conversion similarly like MIPS. Note that this doesn't fix perfectly: the pages should be marked with proper pgprot value. This will be done in a future implementation like the conversion to dma_mmap_coherent(). Acked-by: Benjamin Herrenschmidt Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e48c5f618578..29ab46a12e11 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3069,6 +3069,16 @@ snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) #if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return virt_to_page(CAC_ADDR(vaddr)); +#endif +#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { + dma_addr_t addr = substream->runtime->dma_addr + ofs; + addr -= get_dma_offset(substream->dma_buffer.dev.dev); + /* assume dma_handle set via pfn_to_phys() in + * mm/dma-noncoherent.c + */ + return pfn_to_page(addr >> PAGE_SHIFT); + } #endif return virt_to_page(vaddr); } -- cgit v1.2.2 From d6797322231af98b9bb4afb175dd614cf511e5f7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 15:08:54 +0100 Subject: ALSA: Remove old DMA-mmap code from arm/devdma.c The call of dma_mmap_coherent() is done in the PCM core now. Signed-off-by: Takashi Iwai --- sound/arm/Makefile | 2 +- sound/arm/aaci.c | 16 ++++------- sound/arm/devdma.c | 80 ------------------------------------------------------ sound/arm/devdma.h | 3 -- 4 files changed, 6 insertions(+), 95 deletions(-) delete mode 100644 sound/arm/devdma.c delete mode 100644 sound/arm/devdma.h (limited to 'sound') diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 5a549ed6c8aa..8c0c851d4641 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -3,7 +3,7 @@ # obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o -snd-aaci-objs := aaci.o devdma.o +snd-aaci-objs := aaci.o obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o snd-pxa2xx-pcm-objs := pxa2xx-pcm.o diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f8213e2d5..e59372887f36 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -30,7 +30,6 @@ #include #include "aaci.h" -#include "devdma.h" #define DRIVER_NAME "aaci-pl041" @@ -492,7 +491,7 @@ static int aaci_pcm_hw_free(struct snd_pcm_substream *substream) /* * Clear out the DMA and any allocated buffers. */ - devdma_hw_free(NULL, substream); + snd_pcm_lib_free_pages(substream); return 0; } @@ -505,8 +504,8 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aaci_pcm_hw_free(substream); - err = devdma_hw_alloc(NULL, substream, - params_buffer_bytes(params)); + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); if (err < 0) goto out; @@ -551,11 +550,6 @@ static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(runtime, bytes); } -static int aaci_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - return devdma_mmap(NULL, substream, vma); -} - /* * Playback specific ALSA stuff @@ -722,7 +716,6 @@ static struct snd_pcm_ops aaci_playback_ops = { .prepare = aaci_pcm_prepare, .trigger = aaci_pcm_playback_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, @@ -850,7 +843,6 @@ static struct snd_pcm_ops aaci_capture_ops = { .prepare = aaci_pcm_capture_prepare, .trigger = aaci_pcm_capture_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; /* @@ -1040,6 +1032,8 @@ static int __devinit aaci_init_pcm(struct aaci *aaci) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, 0, 64 * 104); } return ret; diff --git a/sound/arm/devdma.c b/sound/arm/devdma.c deleted file mode 100644 index 9d1e6665b546..000000000000 --- a/sound/arm/devdma.c +++ /dev/null @@ -1,80 +0,0 @@ -/* - * linux/sound/arm/devdma.c - * - * Copyright (C) 2003-2004 Russell King, All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * ARM DMA shim for ALSA. - */ -#include -#include - -#include -#include - -#include "devdma.h" - -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - - if (runtime->dma_area == NULL) - return; - - if (buf != &substream->dma_buffer) { - dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, buf->addr); - kfree(runtime->dma_buffer_p); - } - - snd_pcm_set_runtime_buffer(substream, NULL); -} - -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - int ret = 0; - - if (buf) { - if (buf->bytes >= size) - goto out; - devdma_hw_free(dev, substream); - } - - if (substream->dma_buffer.area != NULL && substream->dma_buffer.bytes >= size) { - buf = &substream->dma_buffer; - } else { - buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); - if (!buf) - goto nomem; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = dev; - buf->area = dma_alloc_coherent(dev, size, &buf->addr, GFP_KERNEL); - buf->bytes = size; - buf->private_data = NULL; - - if (!buf->area) - goto free; - } - snd_pcm_set_runtime_buffer(substream, buf); - ret = 1; - out: - runtime->dma_bytes = size; - return ret; - - free: - kfree(buf); - nomem: - return -ENOMEM; -} - -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - return dma_mmap_coherent(dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); -} diff --git a/sound/arm/devdma.h b/sound/arm/devdma.h deleted file mode 100644 index d025329c8a0f..000000000000 --- a/sound/arm/devdma.h +++ /dev/null @@ -1,3 +0,0 @@ -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream); -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size); -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma); -- cgit v1.2.2 From 8700055e0a30b3f67c1474b09200b59c32dd3796 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 27 Nov 2009 11:20:56 +0100 Subject: ALSA: opti-miro: fix OOPS if hardware is not detected If a hardware is not detected there is a kernel crash due to not initialized snd_miro->aci pointer. This pointer is initialized after detection of the opti (miro) chip. This bug was introduced by patches to expose ACI mikser outside the snd-miro driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 40b64cd54c85..e374869e3e21 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1229,7 +1229,8 @@ static void snd_card_miro_free(struct snd_card *card) struct snd_miro *miro = card->private_data; release_and_free_resource(miro->res_aci_port); - miro->aci->aci_port = 0; + if (miro->aci) + miro->aci->aci_port = 0; release_and_free_resource(miro->res_mc_base); } -- cgit v1.2.2 From bfc9902599549736b9c6445e1e2235b8542f64a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Nov 2009 12:22:44 +0100 Subject: ALSA: hda - Don't trigger pin-sense for STAC/IDT codecs STAC/IDT codecs seem to behave weird when SET_PIN_SENSE verb is issued before reading the jack-detection although the TRIG_REQ pin capability is given by the hardware. Since snd_hda_jack_detect() issues the SET_PIN_SENSE verb simply judging from the pincap, we have to revert the change in the commit d56757abc11a21996d9839c0d4e3b2c3666cd318 ALSA: hda - Replace the rest of jack-detections with snd_hda_jack_detect() to plain GET_PIN_SENSE verb without triggering. Reported-by: Jiri Slaby Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2a45375d79f8..6b0bc040c3b1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4440,7 +4440,14 @@ static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) { if (!nid) return 0; - return snd_hda_jack_detect(codec, nid); + /* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT + * codecs behave wrongly when SET_PIN_SENSE is triggered, although + * the pincap gives TRIG_REQ bit. + */ + if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE) + return 1; + return 0; } static void stac92xx_line_out_detect(struct hda_codec *codec, -- cgit v1.2.2 From a22eaf4ce106404f6c5283da30b4d514ede964c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Nov 2009 15:14:09 +0100 Subject: ASoC: Revert missing reset_err in wm97*.c The commit fe3e78e073d25308756f38019956061153267769 ASoC: Factor out snd_soc_init_card() removed the error paths that are still valid for wm97* codecs, causing the compile errors like sound/soc/codecs/wm9705.c:399: error: label 'reset_err' used but not defined sound/soc/codecs/wm9712.c:687: error: label 'reset_err' used but not defined sound/soc/codecs/wm9713.c:1237: error: label 'reset_err' used but not defined Revert the removed error path codes. Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9705.c | 2 ++ sound/soc/codecs/wm9712.c | 2 ++ sound/soc/codecs/wm9713.c | 2 ++ 3 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dfffc6c778c0..ec54c6da9856 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -404,6 +404,8 @@ static int wm9705_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2a0872273007..0ac1215dcd9b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -697,6 +697,8 @@ static int wm9712_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 00bac315fb3b..4d74ecb0e56b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1249,6 +1249,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); -- cgit v1.2.2 From 49af574b60669a58a2e96960ac694ce953119083 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 27 Nov 2009 13:47:10 +0100 Subject: ALSA: ARM: add Raumfeld audio support Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/raumfeld.c | 335 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 346 insertions(+) create mode 100644 sound/soc/pxa/raumfeld.c (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index d4f4031afa33..376e14a9c273 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -118,6 +118,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b480335..f3e08fd40ca2 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 000000000000..f272269c05d1 --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,335 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood + * Richard Purdie + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +#define GPIO_SPDIF_RESET (38) +#define GPIO_MCLK_RESET (111) +#define GPIO_CODEC_RESET (120) + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +#define MAX9485_MCLK_FREQ_112896 0x22 +#define MAX9485_MCLK_FREQ_122880 0x23 + +static void set_max9485_clk(char clk) +{ + i2c_master_send(max9486_client, &clk, 1); +} + +static void raumfeld_enable_audio(bool en) +{ + if (en) { + gpio_set_value(GPIO_MCLK_RESET, 1); + + /* wait some time to let the clocks become stable */ + msleep(100); + + gpio_set_value(GPIO_SPDIF_RESET, 1); + gpio_set_value(GPIO_CODEC_RESET, 1); + } else { + gpio_set_value(GPIO_MCLK_RESET, 0); + gpio_set_value(GPIO_SPDIF_RESET, 0); + gpio_set_value(GPIO_CODEC_RESET, 0); + } +} + +/* CS4270 */ +static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, +}; + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .hw_params = raumfeld_ak4104_hw_params, +}; + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + /* no S/PDIF on Speakers */ + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + raumfeld_enable_audio(true); + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + raumfeld_enable_audio(false); + + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); + + gpio_free(GPIO_MCLK_RESET); + gpio_free(GPIO_CODEC_RESET); + gpio_free(GPIO_SPDIF_RESET); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack "); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); -- cgit v1.2.2 From dd2e5a156525f11754d9b1e0583f6bb49c253d62 Mon Sep 17 00:00:00 2001 From: Dominik Brodowski Date: Tue, 3 Nov 2009 10:27:34 +0100 Subject: pcmcia: remove deprecated handle_to_dev() macro Update remaining users and remove deprecated handle_to_dev() macro CC: Harald Welte CC: netdev@vger.kernel.org CC: linux-wireless@vger.kernel.org CC: linux-serial@vger.kernel.org Signed-off-by: Dominik Brodowski --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 2 +- sound/pcmcia/vx/vxpocket.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 64b859925c0b..447aaaee3be6 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -131,7 +131,7 @@ static int snd_pdacf_probe(struct pcmcia_device *link) return err; } - snd_card_set_dev(card, &handle_to_dev(link)); + snd_card_set_dev(card, &link->dev); pdacf->index = i; card_list[i] = card; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 1492744ad67f..5a5db48a91a9 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -244,7 +244,7 @@ static int vxpocket_config(struct pcmcia_device *link) if (ret) goto failed; - chip->dev = &handle_to_dev(link); + chip->dev = &link->dev; snd_card_set_dev(chip->card, chip->dev); if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq.AssignedIRQ) < 0) -- cgit v1.2.2 From 5fa9167a1bf5f5a4b7282f5e7ac56a4a5a1fa044 Mon Sep 17 00:00:00 2001 From: Dominik Brodowski Date: Sun, 8 Nov 2009 17:24:46 +0100 Subject: pcmcia: rework the irq_req_t typedef Most of the irq_req_t typedef'd struct can be re-worked quite easily: (1) IRQInfo2 was unused in any case, so drop it. (2) IRQInfo1 was used write-only, so drop it. (3) Instance (private data to be passed to the IRQ handler): Most PCMCIA drivers using pcmcia_request_irq() to actually register an IRQ handler set the "dev_id" to the same pointer as the "priv" pointer in struct pcmcia_device. Modify the two exceptions (ipwireless, ibmtr_cs) to also work this waym and set the IRQ handler's "dev_id" to p_dev->priv unconditionally. (4) Handler is to be of type irq_handler_t. (5) Handler != NULL already tells whether an IRQ handler is present. Therefore, we do not need the IRQ_HANDLER_PRESENT flag in irq_req_t.Attributes. CC: netdev@vger.kernel.org CC: linux-bluetooth@vger.kernel.org CC: linux-ide@vger.kernel.org CC: linux-wireless@vger.kernel.org CC: linux-scsi@vger.kernel.org CC: alsa-devel@alsa-project.org CC: Jaroslav Kysela CC: Jiri Kosina CC: Karsten Keil for the Bluetooth parts: Acked-by: Marcel Holtmann Signed-off-by: Dominik Brodowski --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 4 +--- sound/pcmcia/vx/vxpocket.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 447aaaee3be6..7717e01fc071 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -142,12 +142,10 @@ static int snd_pdacf_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT | IRQ_FORCED_PULSE; + link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_FORCED_PULSE; // link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED; - link->irq.IRQInfo1 = 0 /* | IRQ_LEVEL_ID */; link->irq.Handler = pdacf_interrupt; - link->irq.Instance = pdacf; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 5a5db48a91a9..7be3b3357045 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -161,11 +161,9 @@ static int snd_vxpocket_new(struct snd_card *card, int ibl, link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT; + link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - link->irq.IRQInfo1 = IRQ_LEVEL_ID; link->irq.Handler = &snd_vx_irq_handler; - link->irq.Instance = chip; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; -- cgit v1.2.2 From 70a5f1187bcb3fac93a7d5c5fcfc5fc76b9c3f55 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 30 Nov 2009 07:45:47 +0100 Subject: ALSA: opti-miro: separate comon probing code Separate common probing code in order to use it for PnP probing. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 273 +++++++++++++++++++++++++---------------------- 1 file changed, 147 insertions(+), 126 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index e374869e3e21..c67bc3cd2c65 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1142,28 +1142,39 @@ __skip_mpu: return 0; } +static int __devinit snd_miro_opti_check(struct snd_miro *chip) +{ + unsigned char value; + + chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -ENOMEM; + + value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); + if (value != 0xff && value != inb(chip->mc_base + OPTi9XX_MC_REG(1))) + if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) + return 0; + + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; + + return -ENODEV; +} + static int __devinit snd_card_miro_detect(struct snd_card *card, struct snd_miro *chip) { int i, err; - unsigned char value; for (i = OPTi9XX_HW_82C929; i <= OPTi9XX_HW_82C924; i++) { if ((err = snd_miro_init(chip, i)) < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; - - value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); - if ((value != 0xff) && (value != inb(chip->mc_base + 1))) - if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) - return 1; - - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; - + err = snd_miro_opti_check(chip); + if (err == 0) + return 1; } return -ENODEV; @@ -1234,151 +1245,69 @@ static void snd_card_miro_free(struct snd_card *card) release_and_free_resource(miro->res_mc_base); } -static int __devinit snd_miro_match(struct device *devptr, unsigned int n) -{ - return 1; -} - -static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) +static int __devinit snd_miro_probe(struct snd_card *card) { - static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; - static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; - static int possible_irqs[] = {11, 9, 10, 7, -1}; - static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; - static int possible_dma1s[] = {3, 1, 0, -1}; - static int possible_dma2s[][2] = {{1,-1}, {0,-1}, {-1,-1}, {0,-1}}; - int error; - struct snd_miro *miro; + struct snd_miro *miro = card->private_data; struct snd_wss *codec; struct snd_timer *timer; - struct snd_card *card; struct snd_pcm *pcm; struct snd_rawmidi *rmidi; - error = snd_card_create(index, id, THIS_MODULE, - sizeof(struct snd_miro), &card); - if (error < 0) - return error; - - card->private_free = snd_card_miro_free; - miro = card->private_data; - - error = snd_card_miro_detect(card, miro); - if (error < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); - return -ENODEV; + if (!miro->res_mc_base) { + miro->res_mc_base = request_region(miro->mc_base, + miro->mc_base_size, + "miro (OPTi9xx MC)"); + if (miro->res_mc_base == NULL) { + snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); + return -ENOMEM; + } } - if ((error = snd_card_miro_aci_detect(card, miro)) < 0) { + error = snd_card_miro_aci_detect(card, miro); + if (error < 0) { snd_card_free(card); snd_printk(KERN_ERR "unable to detect aci chip\n"); return -ENODEV; } - /* init proc interface */ - snd_miro_proc_init(card, miro); - - - if (! miro->res_mc_base && - (miro->res_mc_base = request_region(miro->mc_base, miro->mc_base_size, - "miro (OPTi9xx MC)")) == NULL) { - snd_card_free(card); - snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); - return -ENOMEM; - } - miro->wss_base = port; + miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; miro->dma1 = dma1; miro->dma2 = dma2; - if (miro->wss_base == SNDRV_AUTO_PORT) { - if ((miro->wss_base = snd_legacy_find_free_ioport(possible_ports, 4)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free WSS port\n"); - return -EBUSY; - } - } - - if (mpu_port == SNDRV_AUTO_PORT) { - mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); - if (mpu_port < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY; - } - } - miro->mpu_port = mpu_port; - - if (miro->irq == SNDRV_AUTO_IRQ) { - if ((miro->irq = snd_legacy_find_free_irq(possible_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free IRQ\n"); - return -EBUSY; - } - } - if (miro->mpu_irq == SNDRV_AUTO_IRQ) { - if ((miro->mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); - return -EBUSY; - } - } - if (miro->dma1 == SNDRV_AUTO_DMA) { - if ((miro->dma1 = snd_legacy_find_free_dma(possible_dma1s)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA1\n"); - return -EBUSY; - } - } - if (miro->dma2 == SNDRV_AUTO_DMA) { - if ((miro->dma2 = snd_legacy_find_free_dma(possible_dma2s[miro->dma1 % 4])) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA2\n"); - return -EBUSY; - } - } + /* init proc interface */ + snd_miro_proc_init(card, miro); error = snd_miro_configure(miro); - if (error) { - snd_card_free(card); + if (error) return error; - } error = snd_wss_create(card, miro->wss_base + 4, -1, - miro->irq, miro->dma1, miro->dma2, - WSS_HW_AD1845, 0, &codec); - if (error < 0) { - snd_card_free(card); + miro->irq, miro->dma1, miro->dma2, + WSS_HW_DETECT, 0, &codec); + if (error < 0) return error; - } error = snd_wss_pcm(codec, 0, &pcm); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_mixer(codec); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_timer(codec, 0, &timer); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } miro->pcm = pcm; error = snd_miro_mixer(card, miro); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } if (miro->aci->aci_vendor == 'm') { /* It looks like a miro sound card. */ @@ -1425,20 +1354,111 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; struct snd_opl4 *opl4; + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", fm_port); } - if ((error = snd_set_aci_init_values(miro)) < 0) { - snd_card_free(card); + error = snd_set_aci_init_values(miro); + if (error < 0) return error; + + return snd_card_register(card); +} + +static int __devinit snd_miro_isa_match(struct device *devptr, unsigned int n) +{ + return 1; +} + +static int __devinit snd_miro_isa_probe(struct device *devptr, unsigned int n) +{ + static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; + static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; + static int possible_irqs[] = {11, 9, 10, 7, -1}; + static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; + static int possible_dma1s[] = {3, 1, 0, -1}; + static int possible_dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, + {0, -1} }; + + int error; + struct snd_miro *miro; + struct snd_card *card; + + error = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (error < 0) + return error; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + error = snd_card_miro_detect(card, miro); + if (error < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + return -ENODEV; + } + + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free WSS port\n"); + return -EBUSY; + } + } + + if (mpu_port == SNDRV_AUTO_PORT) { + mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); + if (mpu_port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 port\n"); + return -EBUSY; + } + } + + if (irq == SNDRV_AUTO_IRQ) { + irq = snd_legacy_find_free_irq(possible_irqs); + if (irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (mpu_irq == SNDRV_AUTO_IRQ) { + mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs); + if (mpu_irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 IRQ\n"); + return -EBUSY; + } + } + if (dma1 == SNDRV_AUTO_DMA) { + dma1 = snd_legacy_find_free_dma(possible_dma1s); + if (dma1 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2 == SNDRV_AUTO_DMA) { + dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4]); + if (dma2 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA2\n"); + return -EBUSY; + } } snd_card_set_dev(card, devptr); - if ((error = snd_card_register(card))) { + error = snd_miro_probe(card); + if (error < 0) { snd_card_free(card); return error; } @@ -1447,7 +1467,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return 0; } -static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) +static int __devexit snd_miro_isa_remove(struct device *devptr, + unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); dev_set_drvdata(devptr, NULL); @@ -1457,9 +1478,9 @@ static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) #define DEV_NAME "miro" static struct isa_driver snd_miro_driver = { - .match = snd_miro_match, - .probe = snd_miro_probe, - .remove = __devexit_p(snd_miro_remove), + .match = snd_miro_isa_match, + .probe = snd_miro_isa_probe, + .remove = __devexit_p(snd_miro_isa_remove), /* FIXME: suspend/resume */ .driver = { .name = DEV_NAME -- cgit v1.2.2 From 306ecee926cf79f1b3b5f6035be09ef3d83f1b76 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 30 Nov 2009 07:46:56 +0100 Subject: ALSA: opti-miro: add PnP detection The PCM12 and PCM20 can be set into the ISA PnP mode. The PCM12 PnP was sold as the PnP device. Add code to handle detection of these cards using ISA PnP framework. Tested on the PCM20 in PnP mode. The PCM12 PnP has the same MS Windows INF file except for a card name displayed for user. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 203 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 192 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c67bc3cd2c65..6123c7531110 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -60,6 +61,9 @@ static int dma1 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int dma2 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int wss; static int ide; +#ifdef CONFIG_PNP +static int isapnp = 1; /* Enable ISA PnP detection */ +#endif module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for miro soundcard."); @@ -83,6 +87,10 @@ module_param(wss, int, 0444); MODULE_PARM_DESC(wss, "wss mode"); module_param(ide, int, 0444); MODULE_PARM_DESC(ide, "enable ide port"); +#ifdef CONFIG_PNP +module_param(isapnp, bool, 0444); +MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif #define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 @@ -131,6 +139,21 @@ static char * snd_opti9xx_names[] = { "82C930", "82C931", "82C933" }; +static int snd_miro_pnp_is_probed; + +#ifdef CONFIG_PNP + +static struct pnp_card_device_id snd_miro_pnpids[] = { + /* PCM20 and PCM12 in PnP mode */ + { .id = "MIR0924", + .devs = { { "MIR0000" }, { "MIR0002" }, { "MIR0005" } }, }, + { .id = "" } +}; + +MODULE_DEVICE_TABLE(pnp_card, snd_miro_pnpids); + +#endif /* CONFIG_PNP */ + /* * ACI control */ @@ -781,17 +804,23 @@ static int __devinit snd_miro_init(struct snd_miro *chip, chip->mpu_port = -1; chip->mpu_irq = -1; + chip->pwd_reg = 3; + +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; + else +#endif + chip->mc_base = 0xf8c; + switch (hardware) { case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = 0xe3; - chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: - chip->mc_base = 0xf8c; chip->password = 0xe5; - chip->pwd_reg = 3; break; default: @@ -1014,17 +1043,22 @@ static int __devinit snd_miro_configure(struct snd_miro *chip) return -EINVAL; } - switch (chip->wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (chip->wss_base & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: @@ -1238,7 +1272,7 @@ static int __devinit snd_card_miro_aci_detect(struct snd_card *card, static void snd_card_miro_free(struct snd_card *card) { struct snd_miro *miro = card->private_data; - + release_and_free_resource(miro->res_aci_port); if (miro->aci) miro->aci->aci_port = 0; @@ -1370,6 +1404,12 @@ static int __devinit snd_miro_probe(struct snd_card *card) static int __devinit snd_miro_isa_match(struct device *devptr, unsigned int n) { +#ifdef CONFIG_PNP + if (snd_miro_pnp_is_probed) + return 0; + if (isapnp) + return 0; +#endif return 1; } @@ -1487,14 +1527,155 @@ static struct isa_driver snd_miro_driver = { }, }; +#ifdef CONFIG_PNP + +static int __devinit snd_card_miro_pnp(struct snd_miro *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *pid) +{ + struct pnp_dev *pdev; + int err; + struct pnp_dev *devmpu; + struct pnp_dev *devmc; + + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) + return -EBUSY; + + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + if (devmpu == NULL) + return -EBUSY; + + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); + return err; + } + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + err); + return err; + } + + port = pnp_port_start(pdev, 1); + fm_port = pnp_port_start(pdev, 2) + 8; + + /* + * The MC(0) is never accessed and the miroSOUND PCM20 card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; + + irq = pnp_irq(pdev, 0); + dma1 = pnp_dma(pdev, 0); + dma2 = pnp_dma(pdev, 1); + + if (mpu_port > 0) { + err = pnp_activate_dev(devmpu); + if (err < 0) { + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); + mpu_port = -1; + return err; + } + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); + } + return 0; +} + +static int __devinit snd_miro_pnp_probe(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct snd_card *card; + int err; + struct snd_miro *miro; + + if (snd_miro_pnp_is_probed) + return -EBUSY; + if (!isapnp) + return -ENODEV; + err = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (err < 0) + return err; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + err = snd_card_miro_pnp(miro, pcard, pid); + if (err) { + snd_card_free(card); + return err; + } + + /* only miroSOUND PCM20 and PCM12 == OPTi924 */ + err = snd_miro_init(miro, OPTi9XX_HW_82C924); + if (err) { + snd_card_free(card); + return err; + } + + err = snd_miro_opti_check(miro); + if (err) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return err; + } + + snd_card_set_dev(card, &pcard->card->dev); + err = snd_miro_probe(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + snd_miro_pnp_is_probed = 1; + return 0; +} + +static void __devexit snd_miro_pnp_remove(struct pnp_card_link * pcard) +{ + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); + snd_miro_pnp_is_probed = 0; +} + +static struct pnp_card_driver miro_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "miro", + .id_table = snd_miro_pnpids, + .probe = snd_miro_pnp_probe, + .remove = __devexit_p(snd_miro_pnp_remove), +}; +#endif + static int __init alsa_card_miro_init(void) { +#ifdef CONFIG_PNP + pnp_register_card_driver(&miro_pnpc_driver); + if (snd_miro_pnp_is_probed) + return 0; + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif return isa_register_driver(&snd_miro_driver, 1); } static void __exit alsa_card_miro_exit(void) { - isa_unregister_driver(&snd_miro_driver); + if (!snd_miro_pnp_is_probed) { + isa_unregister_driver(&snd_miro_driver); + return; + } +#ifdef CONFIG_PNP + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif } module_init(alsa_card_miro_init) -- cgit v1.2.2 From 45d4ebf1a6255f2234a041685789cbecac3453f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Nov 2009 11:58:30 +0100 Subject: ALSA: hda - Add a position_fix quirk for MSI Wind U115 MSI Wind U115 seems to require position_fix=1 explicitly. Otherwise it screws up PulseAudio. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91bcbdad5af5..238651bab3f5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2234,6 +2234,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), {} }; -- cgit v1.2.2 From 785d1c45ce11820d5838eb6399caa0ac98c836cf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Nov 2009 20:24:48 +0900 Subject: ASoC: sh: fsi: Add runtime PM support This patch add support runtime PM. Driver callbacks for Runtime PM are empty because the device registers are always re-initialized after pm_runtime_get_sync(). The Runtime PM functions replaces the clock framework module stop bit handling in this driver. Signed-off-by: Kuninori Morimoto Acked-by: Paul Mundt Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 39 +++++++++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index e1a3d1a2b4c8..9c49c11c43ce 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -17,7 +17,7 @@ #include #include #include -#include +#include #include #include #include @@ -105,7 +105,6 @@ struct fsi_priv { struct fsi_master { void __iomem *base; int irq; - struct clk *clk; struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; @@ -559,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, int is_master; int ret = 0; - clk_enable(master->clk); + pm_runtime_get_sync(dai->dev); /* CKG1 */ data = is_play ? (1 << 0) : (1 << 4); @@ -674,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - clk_disable(master->clk); + pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -872,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct resource *res; - char clk_name[8]; unsigned int irq; int ret; @@ -903,14 +901,8 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - /* FSI is based on SPU mstp */ - snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); - master->clk = clk_get(NULL, clk_name); - if (IS_ERR(master->clk)) { - dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); - ret = -EIO; - goto exit_iounmap; - } + pm_runtime_enable(&pdev->dev); + pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; fsi_soc_dai[1].dev = &pdev->dev; @@ -935,6 +927,7 @@ exit_free_irq: free_irq(irq, master); exit_iounmap: iounmap(master->base); + pm_runtime_disable(&pdev->dev); exit_kfree: kfree(master); master = NULL; @@ -947,7 +940,7 @@ static int fsi_remove(struct platform_device *pdev) snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); - clk_put(master->clk); + pm_runtime_disable(&pdev->dev); free_irq(master->irq, master); @@ -957,9 +950,27 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static int fsi_runtime_nop(struct device *dev) +{ + /* Runtime PM callback shared between ->runtime_suspend() + * and ->runtime_resume(). Simply returns success. + * + * This driver re-initializes all registers after + * pm_runtime_get_sync() anyway so there is no need + * to save and restore registers here. + */ + return 0; +} + +static struct dev_pm_ops fsi_pm_ops = { + .runtime_suspend = fsi_runtime_nop, + .runtime_resume = fsi_runtime_nop, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", + .pm = &fsi_pm_ops, }, .probe = fsi_probe, .remove = fsi_remove, -- cgit v1.2.2 From a649d1fcc9bd2299cb06b6594fabb429fa50f174 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 30 Nov 2009 14:06:37 +0100 Subject: ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API ALSA's for-2.6.33 branch has a new source argument to snd_soc_dai_set_pll(). Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/raumfeld.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index f272269c05d1..acfce1c0f1c9 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -116,7 +116,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; @@ -205,7 +205,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; -- cgit v1.2.2 From 854206b074581957e7b5c955001c329f94986b4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Nov 2009 18:22:04 +0100 Subject: ALSA: hda - Fix Cxt5047 test mode The NID 0x1a of Conexant 5047 chip is a mic boost volume only with the output amp unlike 5045 chip. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 60810ba899d1..a09c03c3f62b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1410,16 +1410,7 @@ static struct snd_kcontrol_new cxt5047_test_mixer[] = { .get = conexant_mux_enum_get, .put = conexant_mux_enum_put, }, - HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), { } /* end */ }; -- cgit v1.2.2 From cfc9b06f0befe50ef02253f72b76946363549031 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Dec 2009 12:19:37 +0100 Subject: ALSA: hda - Add a pin-fix for FSC Amilo Pi1505 FSC Amilo Pi 1505 has a buggy BIOS and doesn't set up the HP and speaker pins properly. Add the pinfix entry for that. Reference: Novell bnc#557403 https://bugzilla.novell.com/show_bug.cgi?id=557403 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7e8b17a1769a..a38a81e53863 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14543,6 +14543,27 @@ static struct alc_config_preset alc861_presets[] = { }, }; +/* Pin config fixes */ +enum { + PINFIX_FSC_AMILO_PI1505, +}; + +static struct alc_pincfg alc861_fsc_amilo_pi1505_pinfix[] = { + { 0x0b, 0x0221101f }, /* HP */ + { 0x0f, 0x90170310 }, /* speaker */ + { } +}; + +static const struct alc_fixup alc861_fixups[] = { + [PINFIX_FSC_AMILO_PI1505] = { + .pins = alc861_fsc_amilo_pi1505_pinfix + }, +}; + +static struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), + {} +}; static int patch_alc861(struct hda_codec *codec) { @@ -14566,6 +14587,8 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups); + if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ err = alc861_parse_auto_config(codec); -- cgit v1.2.2 From 2f703e7a2ea5f6d5ea14a7b2cd0d31be07839ac6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Dec 2009 14:17:37 +0100 Subject: ALSA: hda - Add position_fix quirk for HP dv3 HP dv3 requires position_fix=1. Reference: Novell bnc#555935 https://bugzilla.novell.com/show_bug.cgi?id=555935 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 238651bab3f5..d822bfc6cad6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2233,6 +2233,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), + SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), {} -- cgit v1.2.2 From e0feefc70c1bb3f51aa9bb42acfd22cd7472a5d9 Mon Sep 17 00:00:00 2001 From: Alexey Fisher Date: Tue, 1 Dec 2009 13:40:53 +0100 Subject: ALSA: usb - Fix mixer map for Hercules Gamesurround Muse Pocket LT Muse Pocket use brocken mixer names, so alsamixer and PA can't use it correctly This patch add quirk to overwirte default mixers. Signed-off-by: Alexey Fisher Signed-off-by: Takashi Iwai --- sound/usb/usbmixer_maps.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 3e5d66cf1f5a..77c35885e21c 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -277,6 +277,22 @@ static struct usbmix_name_map scratch_live_map[] = { { 0 } /* terminator */ }; +/* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+" + * most importand difference is SU[8], it should be set to "Capture Source" + * to make alsamixer and PA working properly. + * FIXME: or mp3plus_map should use "Capture Source" too, + * so this maps can be merget + */ +static struct usbmix_name_map hercules_usb51_map[] = { + { 8, "Capture Source" }, /* SU, default "PCM Capture Source" */ + { 9, "Master Playback" }, /* FU, default "Speaker Playback" */ + { 10, "Mic Boost", 7 }, /* FU, default "Auto Gain Input" */ + { 11, "Line Capture" }, /* FU, default "PCM Capture" */ + { 13, "Mic Bypass Playback" }, /* FU, default "Mic Playback" */ + { 14, "Line Bypass Playback" }, /* FU, default "Line Playback" */ + { 0 } /* terminator */ +}; + /* * Control map entries */ @@ -315,6 +331,13 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x06f8, 0xd002), .ignore_ctl_error = 1, }, + { + /* Hercules Gamesurround Muse Pocket LT + * (USB 5.1 Channel Audio Adapter) + */ + .id = USB_ID(0x06f8, 0xc000), + .map = hercules_usb51_map, + }, { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, -- cgit v1.2.2 From cf5bd652c384cf58544f43bea097bbc9cf14e4f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Dec 2009 16:36:56 +0100 Subject: ALSA: aaci - Clean up duplicate code Now snd_ac97_pcm_open() is called with the exactly same arguments for both playback and capture directions. Remove the unneeded check. Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index eb715e732106..83b0328d389e 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -511,15 +511,9 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) goto out; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[0].slots); - else - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[0].slots); - + err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), + params_channels(params), + aacirun->pcm->r[0].slots); if (err) goto out; -- cgit v1.2.2 From d8ea23931ce83b56801976e6f1fa893462c1c477 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 2 Dec 2009 23:27:12 +0100 Subject: ALSA: opti9xx: remove snd_opti9xx fields Remove snd_opti9xx fields which are indirect arguments to the snd_opti9xx_configure(). Pass these values as function arguments. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 110 +++++++++++++++---------------------- 1 file changed, 43 insertions(+), 67 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd555325b9d..d08c38906449 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -141,15 +141,7 @@ struct snd_opti9xx { spinlock_t lock; - long wss_base; int irq; - int dma1; - int dma2; - - long fm_port; - - long mpu_port; - int mpu_irq; #ifdef CONFIG_PNP struct pnp_dev *dev; @@ -216,13 +208,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, spin_lock_init(&chip->lock); - chip->wss_base = -1; chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->fm_port = -1; - chip->mpu_port = -1; - chip->mpu_irq = -1; switch (hardware) { #ifndef OPTi93X @@ -348,7 +334,10 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, + long wss_base, + int irq, int dma1, int dma2, + long mpu_port, int mpu_irq) { unsigned char wss_base_bits; unsigned char irq_bits; @@ -416,7 +405,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) return -EINVAL; } - switch (chip->wss_base) { + switch (wss_base) { case 0x530: wss_base_bits = 0x00; break; @@ -430,14 +419,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", - chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); __skip_base: - switch (chip->irq) { + switch (irq) { //#ifdef OPTi93X case 5: irq_bits = 0x05; @@ -456,11 +444,11 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", irq); goto __skip_resources; } - switch (chip->dma1) { + switch (dma1) { case 0: dma_bits = 0x01; break; @@ -471,38 +459,36 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", - chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) - if (chip->dma1 == chip->dma2) { + if (dma1 == dma2) { snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } - switch (chip->dma2) { + switch (dma2) { case 0: case 1: break; default: - snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", - chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", dma2); goto __skip_resources; } dma_bits |= 0x04; #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, chip->wss_base); + outb(irq_bits << 3 | dma_bits, wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ __skip_resources: if (chip->hardware > OPTi9XX_HW_82C928) { - switch (chip->mpu_port) { + switch (mpu_port) { case 0: case -1: break; @@ -520,12 +506,11 @@ __skip_resources: break; default: snd_printk(KERN_WARNING - "MPU-401 port 0x%lx not valid\n", - chip->mpu_port); + "MPU-401 port 0x%lx not valid\n", mpu_port); goto __skip_mpu; } - switch (chip->mpu_irq) { + switch (mpu_irq) { case 5: mpu_irq_bits = 0x02; break; @@ -540,12 +525,12 @@ __skip_resources: break; default: snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", - chip->mpu_irq); + mpu_irq); goto __skip_mpu; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), - (chip->mpu_port <= 0) ? 0x00 : + (mpu_port <= 0) ? 0x00 : 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3, 0xf8); } @@ -701,6 +686,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) { static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; + int xdma2; struct snd_opti9xx *chip = card->private_data; struct snd_wss *codec; #ifdef CS4231 @@ -715,31 +701,25 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) "OPTi9xx MC")) == NULL) return -ENOMEM; - chip->wss_base = port; - chip->fm_port = fm_port; - chip->mpu_port = mpu_port; - chip->irq = irq; - chip->mpu_irq = mpu_irq; - chip->dma1 = dma1; #if defined(CS4231) || defined(OPTi93X) - chip->dma2 = dma2; + xdma2 = dma2; #else - chip->dma2 = -1; + xdma2 = -1; #endif - if (chip->wss_base == SNDRV_AUTO_PORT) { - chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); - if (chip->wss_base < 0) { + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } - error = snd_opti9xx_configure(chip); + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); if (error) return error; - error = snd_wss_create(card, chip->wss_base + 4, -1, - chip->irq, chip->dma1, chip->dma2, + error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -763,35 +743,35 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) return error; #endif #ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, + error = request_irq(irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec); if (error < 0) { snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); return error; } #endif + chip->irq = irq; strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1, chip->dma2); + card->shortname, pcm->name, port + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1); + card->shortname, pcm->name, port + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ - if (chip->mpu_port <= 0 || chip->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 0, chip->mpu_irq, IRQF_DISABLED, - &rmidi))) + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi); + if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - chip->mpu_port); + mpu_port); + } - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; #ifndef OPTi93X if (chip->hardware == OPTi9XX_HW_82C928 || @@ -801,9 +781,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) /* assume we have an OPL4 */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); - if (snd_opl4_create(card, - chip->fm_port, - chip->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) { /* no luck, use OPL3 instead */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), @@ -811,12 +789,10 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) } } #endif /* !OPTi93X */ - if (!opl3 && snd_opl3_create(card, - chip->fm_port, - chip->fm_port + 2, + if (!opl3 && snd_opl3_create(card, fm_port, fm_port + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - chip->fm_port, chip->fm_port + 4 - 1); + fm_port, fm_port + 4 - 1); } if (opl3) { error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); -- cgit v1.2.2 From 274693f37090ada2cadd09944ab883f05ea6ebe6 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 3 Dec 2009 10:07:50 +0100 Subject: ALSA: hda - Add ALC661/259, ALC892/888VD support Fixed List: 1. Add alc_read_coef_idx function 2. Add ALC661 ALC259 3. Add ALC892 ALC888VD Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 44 +++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 42 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a38a81e53863..98e117bac90a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1394,6 +1394,17 @@ static void alc_pick_fixup(struct hda_codec *codec, add_verb(codec->spec, fix->verbs); } +static int alc_read_coef_idx(struct hda_codec *codec, + unsigned int coef_idx) +{ + unsigned int val; + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, + coef_idx); + val = snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_GET_PROC_COEF, 0); + return val; +} + /* * ALC888 */ @@ -3472,7 +3483,7 @@ static int alc_build_pcms(struct hda_codec *codec) snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog), "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - + if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) return -EINVAL; @@ -13445,6 +13456,13 @@ static int patch_alc269(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x04, 15); + if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){ + kfree(codec->chip_name); + codec->chip_name = kstrdup("ALC259", GFP_KERNEL); + if (!codec->chip_name) + return -ENOMEM; + } + board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, alc269_cfg_tbl); @@ -17444,6 +17462,13 @@ static int patch_alc662(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x04, 15); + if (alc_read_coef_idx(codec, 0)==0x8020){ + kfree(codec->chip_name); + codec->chip_name = kstrdup("ALC661", GFP_KERNEL); + if (!codec->chip_name) + return -ENOMEM; + } + board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, alc662_models, alc662_cfg_tbl); @@ -17510,6 +17535,20 @@ static int patch_alc662(struct hda_codec *codec) return 0; } +static int patch_alc888(struct hda_codec *codec) +{ + if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){ + kfree(codec->chip_name); + codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL); + if (!codec->chip_name) + return -ENOMEM; + patch_alc662(codec); + } else { + patch_alc882(codec); + } + return 0; +} + /* * patch entries */ @@ -17541,8 +17580,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc882 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, + { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, {} /* terminator */ }; -- cgit v1.2.2 From ac2c92e0cd06387ecee8115f5fa385fba6413c42 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 3 Dec 2009 10:14:10 +0100 Subject: ALSA: hda - Fix memory leaks in the previous patch The previous hack for replacing the codec name give memory leaks at error paths. This patch fixes them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 98e117bac90a..d967836f36bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13459,8 +13459,10 @@ static int patch_alc269(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){ kfree(codec->chip_name); codec->chip_name = kstrdup("ALC259", GFP_KERNEL); - if (!codec->chip_name) + if (!codec->chip_name) { + alc_free(codec); return -ENOMEM; + } } board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, @@ -17465,8 +17467,10 @@ static int patch_alc662(struct hda_codec *codec) if (alc_read_coef_idx(codec, 0)==0x8020){ kfree(codec->chip_name); codec->chip_name = kstrdup("ALC661", GFP_KERNEL); - if (!codec->chip_name) + if (!codec->chip_name) { + alc_free(codec); return -ENOMEM; + } } board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, @@ -17540,13 +17544,13 @@ static int patch_alc888(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){ kfree(codec->chip_name); codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL); - if (!codec->chip_name) + if (!codec->chip_name) { + alc_free(codec); return -ENOMEM; - patch_alc662(codec); - } else { - patch_alc882(codec); + } + return patch_alc662(codec); } - return 0; + return patch_alc882(codec); } /* -- cgit v1.2.2 From 1bc8079879e8edfff451b62b7550bdd18523f963 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Tue, 1 Dec 2009 18:10:34 +0100 Subject: ASoC: au1x: dbdma2: fix oops on soc device removal. platform_device_unregister() frees resources for us, no need to do it explicitly. Fixes an oops when machine code removes the soc-audio device. Signed-off-by: Manuel Lauss Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index fe9f4657c959..2ca33b09a867 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -488,11 +488,8 @@ EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); void au1xpsc_pcm_destroy(struct platform_device *dmapd) { - if (dmapd) { - kfree(dmapd->resource); - dmapd->resource = NULL; + if (dmapd) platform_device_unregister(dmapd); - } } EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy); -- cgit v1.2.2 From efd9eb96d5604c2c133e500f7b8c7b3f3fbdece8 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Tue, 1 Dec 2009 18:10:35 +0100 Subject: ASoC: au1x: dbdma2: plug memleak in pcm device creation error path free the allocated pcm platform device in the error path. Signed-off-by: Manuel Lauss Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 2ca33b09a867..19e4d37eba1c 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -480,6 +480,7 @@ struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) if (!ret) return pd; + platform_device_put(pd); out: kfree(res); return NULL; -- cgit v1.2.2 From 71f6e0645be42f93c0f90dfcc93b9d2d277c2ee6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 2 Dec 2009 15:11:08 +0900 Subject: ASoC: sh_fsi: avoid using global variable Current FSI driver use global variable to access device data. But this style will be broken if SuperH come with multiple FSI blocks in future. To solve this problem, this patch use cpu_dai->private_data. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 115 +++++++++++++++++++++++++++++------------------------ 1 file changed, 62 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9c49c11c43ce..7506ef6d287a 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -92,6 +92,7 @@ struct fsi_priv { void __iomem *base; struct snd_pcm_substream *substream; + struct fsi_master *master; int fifo_max; int chan; @@ -110,8 +111,6 @@ struct fsi_master { struct sh_fsi_platform_info *info; }; -static struct fsi_master *master; - /************************************************************************ @@ -166,7 +165,7 @@ static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } -static int fsi_master_write(u32 reg, u32 data) +static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -175,7 +174,7 @@ static int fsi_master_write(u32 reg, u32 data) return __fsi_reg_write((u32)(master->base + reg), data); } -static u32 fsi_master_read(u32 reg) +static u32 fsi_master_read(struct fsi_master *master, u32 reg) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -184,7 +183,8 @@ static u32 fsi_master_read(u32 reg) return __fsi_reg_read((u32)(master->base + reg)); } -static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +static int fsi_master_mask_set(struct fsi_master *master, + u32 reg, u32 mask, u32 data) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -200,43 +200,29 @@ static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) ************************************************************************/ -static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +static struct fsi_master *fsi_get_master(struct fsi_priv *fsi) { - struct snd_soc_pcm_runtime *rtd; - struct fsi_priv *fsi = NULL; - - if (!substream || !master) - return NULL; - - rtd = substream->private_data; - switch (rtd->dai->cpu_dai->id) { - case 0: - fsi = &master->fsia; - break; - case 1: - fsi = &master->fsib; - break; - } - - return fsi; + return fsi->master; } static int fsi_is_port_a(struct fsi_priv *fsi) { - /* return - * 1 : port a - * 0 : port b - */ + return fsi->master->base == fsi->base; +} - if (fsi == &master->fsia) - return 1; +static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *machine = rtd->dai; + struct snd_soc_dai *dai = machine->cpu_dai; - return 0; + return dai->private_data; } static u32 fsi_get_info_flags(struct fsi_priv *fsi) { int is_porta = fsi_is_port_a(fsi); + struct fsi_master *master = fsi_get_master(fsi); return is_porta ? master->info->porta_flags : master->info->portb_flags; @@ -314,27 +300,30 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) { u32 data = fsi_port_ab_io_bit(fsi, is_play); + struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(IMSK, data, data); - fsi_master_mask_set(IEMSK, data, data); + fsi_master_mask_set(master, IMSK, data, data); + fsi_master_mask_set(master, IEMSK, data, data); } static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) { u32 data = fsi_port_ab_io_bit(fsi, is_play); + struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(IMSK, data, 0); - fsi_master_mask_set(IEMSK, data, 0); + fsi_master_mask_set(master, IMSK, data, 0); + fsi_master_mask_set(master, IEMSK, data, 0); } static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) { u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + struct fsi_master *master = fsi_get_master(fsi); if (enable) - fsi_master_mask_set(CLK_RST, val, val); + fsi_master_mask_set(master, CLK_RST, val, val); else - fsi_master_mask_set(CLK_RST, val, 0); + fsi_master_mask_set(master, CLK_RST, val, 0); } static void fsi_irq_init(struct fsi_priv *fsi, int is_play) @@ -355,23 +344,23 @@ static void fsi_irq_init(struct fsi_priv *fsi, int is_play) fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); /* clear interrupt factor */ - fsi_master_mask_set(INT_ST, data, 0); + fsi_master_mask_set(fsi_get_master(fsi), INT_ST, data, 0); } -static void fsi_soft_all_reset(void) +static void fsi_soft_all_reset(struct fsi_master *master) { - u32 status = fsi_master_read(SOFT_RST); + u32 status = fsi_master_read(master, SOFT_RST); /* port AB reset */ status &= 0x000000ff; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); mdelay(10); /* soft reset */ status &= 0x000000f0; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); status |= 0x00000001; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); mdelay(10); } @@ -517,12 +506,13 @@ static int fsi_data_pop(struct fsi_priv *fsi) static irqreturn_t fsi_interrupt(int irq, void *data) { - u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; - u32 int_st = fsi_master_read(INT_ST); + struct fsi_master *master = data; + u32 status = fsi_master_read(master, SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(master, INT_ST); /* clear irq status */ - fsi_master_write(SOFT_RST, status); - fsi_master_write(SOFT_RST, status | 0x00000010); + fsi_master_write(master, SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status | 0x00000010); if (int_st & INT_A_OUT) fsi_data_push(&master->fsia); @@ -533,7 +523,7 @@ static irqreturn_t fsi_interrupt(int irq, void *data) if (int_st & INT_B_IN) fsi_data_pop(&master->fsib); - fsi_master_write(INT_ST, 0x0000000); + fsi_master_write(master, INT_ST, 0x0000000); return IRQ_HANDLED; } @@ -548,7 +538,7 @@ static irqreturn_t fsi_interrupt(int irq, void *data) static int fsi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); const char *msg; u32 flags = fsi_get_info_flags(fsi); u32 fmt; @@ -667,7 +657,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; fsi_irq_disable(fsi, is_play); @@ -679,7 +669,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); struct snd_pcm_runtime *runtime = substream->runtime; int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; @@ -760,7 +750,7 @@ static int fsi_hw_free(struct snd_pcm_substream *substream) static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); long location; location = (fsi->byte_offset - 1); @@ -870,10 +860,16 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); ************************************************************************/ static int fsi_probe(struct platform_device *pdev) { + struct fsi_master *master; struct resource *res; unsigned int irq; int ret; + if (0 != pdev->id) { + dev_err(&pdev->dev, "current fsi support id 0 only now\n"); + return -ENODEV; + } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); irq = platform_get_irq(pdev, 0); if (!res || !irq) { @@ -899,15 +895,19 @@ static int fsi_probe(struct platform_device *pdev) master->irq = irq; master->info = pdev->dev.platform_data; master->fsia.base = master->base; + master->fsia.master = master; master->fsib.base = master->base + 0x40; + master->fsib.master = master; pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; + fsi_soc_dai[0].private_data = &master->fsia; fsi_soc_dai[1].dev = &pdev->dev; + fsi_soc_dai[1].private_data = &master->fsib; - fsi_soft_all_reset(); + fsi_soft_all_reset(master); ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); if (ret) { @@ -937,6 +937,10 @@ exit: static int fsi_remove(struct platform_device *pdev) { + struct fsi_master *master; + + master = fsi_get_master(fsi_soc_dai[0].private_data); + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); @@ -946,7 +950,12 @@ static int fsi_remove(struct platform_device *pdev) iounmap(master->base); kfree(master); - master = NULL; + + fsi_soc_dai[0].dev = NULL; + fsi_soc_dai[0].private_data = NULL; + fsi_soc_dai[1].dev = NULL; + fsi_soc_dai[1].private_data = NULL; + return 0; } -- cgit v1.2.2 From 1233faa891451dee9eaddd7f8a616ba1ddd77919 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Fri, 27 Nov 2009 18:19:28 +0100 Subject: ALSA: tea575x-tuner: fix mute Fix mute state reporting in tea575x-tuner. This fixes mute function in kradio on SF64-PCR radio card. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/i2c/other/tea575x-tuner.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index d31c373e076d..c4c6ef73f9bf 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -225,7 +225,7 @@ static int vidioc_s_ctrl(struct file *file, void *priv, case V4L2_CID_AUDIO_MUTE: if (tea->ops->mute) { tea->ops->mute(tea, ctrl->value); - tea->mute = 1; + tea->mute = ctrl->value; return 0; } } -- cgit v1.2.2 From fb716c0b7bed36064cd41d800c8f339f41adf084 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Fri, 27 Nov 2009 18:18:33 +0100 Subject: snd-fm801: autodetect SF64-PCR (tuner-only) card MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When primary AC97 is not found, don't fail with tons of AC97 errors. Assume that the card is SF64-PCR (tuner-only). This makes the SF64-PCR radio card work "out of the box". Also fixes a bug that can cause an oops here:         if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { when tea575x_tuner == 16, it passes this check and causes problems a couple lines below:         chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; Tested with SF64-PCR, but I don't have any of those sound or sound+radio cards to test if I didn't break anything. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 40 +++++++++++++++++++++++++++------------- 1 file changed, 27 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 60cdb9e0b68d..83508b3964fb 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -55,7 +55,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * * 1 = MediaForte 256-PCS * 2 = MediaForte 256-PCPR * 3 = MediaForte 64-PCR - * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card + * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 */ static int tea575x_tuner[SNDRV_CARDS]; @@ -67,7 +67,10 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); + +#define TUNER_ONLY (1<<4) +#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) /* * Direct registers @@ -160,7 +163,7 @@ struct fm801 { unsigned int multichannel: 1, /* multichannel support */ secondary: 1; /* secondary codec */ unsigned char secondary_addr; /* address of the secondary codec */ - unsigned int tea575x_tuner; /* tuner flags */ + unsigned int tea575x_tuner; /* tuner access method & flags */ unsigned short ply_ctrl; /* playback control */ unsigned short cap_ctrl; /* capture control */ @@ -1287,7 +1290,7 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) { unsigned short cmdw; - if (chip->tea575x_tuner & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __ac97_ok; /* codec cold reset + AC'97 warm reset */ @@ -1296,11 +1299,13 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) udelay(100); outw(0, FM801_REG(chip, CODEC_CTRL)); - if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) { - snd_printk(KERN_ERR "Primary AC'97 codec not found\n"); - if (! resume) - return -EIO; - } + if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) + if (!resume) { + snd_printk(KERN_INFO "Primary AC'97 codec not found, " + "assume SF64-PCR (tuner-only)\n"); + chip->tea575x_tuner = 3 | TUNER_ONLY; + goto __ac97_ok; + } if (chip->multichannel) { if (chip->secondary_addr) { @@ -1414,7 +1419,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, return err; } chip->port = pci_resource_start(pci, 0); - if ((tea575x_tuner & 0x0010) == 0) { + if ((tea575x_tuner & TUNER_ONLY) == 0) { if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, "FM801", chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); @@ -1429,6 +1434,14 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->multichannel = 1; snd_fm801_chip_init(chip, 0); + /* init might set tuner access method */ + tea575x_tuner = chip->tea575x_tuner; + + if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) { + pci_clear_master(pci); + free_irq(chip->irq, chip); + chip->irq = -1; + } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); @@ -1438,12 +1451,13 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { + if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && + (tea575x_tuner & TUNER_TYPE_MASK) < 4) { chip->tea.dev_nr = tea575x_tuner >> 16; chip->tea.card = card; chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; + chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1]; snd_tea575x_init(&chip->tea); } #endif @@ -1483,7 +1497,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->port, chip->irq); - if (tea575x_tuner[dev] & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __fm801_tuner_only; if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) { -- cgit v1.2.2 From 3482594802d80a595ca50b16d3a25bcc1eb480c8 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Fri, 4 Dec 2009 15:12:10 +0900 Subject: ASoC: Rename controls with a / in wm_hubs This renames from a character / to : of controls. A / occurs below error messages. ASoC: Failed to create IN2RP/VXRP debugfs file ASoC: Failed to create IN2LP/VXRN debugfs file Signed-off-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 810a563d0ebf..d73c30536a2c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -438,11 +438,11 @@ static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1LN"), SND_SOC_DAPM_INPUT("IN1LP"), SND_SOC_DAPM_INPUT("IN2LN"), -SND_SOC_DAPM_INPUT("IN2LP/VXRN"), +SND_SOC_DAPM_INPUT("IN2LP:VXRN"), SND_SOC_DAPM_INPUT("IN1RN"), SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), -SND_SOC_DAPM_INPUT("IN2RP/VXRP"), +SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), @@ -537,14 +537,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, - { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" }, + { "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" }, { "IN2L PGA", "IN2LN Switch", "IN2LN" }, - { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" }, + { "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" }, { "IN2R PGA", "IN2RN Switch", "IN2RN" }, - { "Direct Voice", NULL, "IN2LP/VXRN" }, - { "Direct Voice", NULL, "IN2RP/VXRP" }, + { "Direct Voice", NULL, "IN2LP:VXRN" }, + { "Direct Voice", NULL, "IN2RP:VXRP" }, { "MIXINL", "IN1L Switch", "IN1L PGA" }, { "MIXINL", "IN2L Switch", "IN2L PGA" }, @@ -565,7 +565,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Left Output Mixer", "Right Input Switch", "MIXINR" }, { "Left Output Mixer", "IN2RN Switch", "IN2RN" }, { "Left Output Mixer", "IN2LN Switch", "IN2LN" }, - { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" }, + { "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" }, { "Left Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Left Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -573,7 +573,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Right Output Mixer", "Right Input Switch", "MIXINR" }, { "Right Output Mixer", "IN2LN Switch", "IN2LN" }, { "Right Output Mixer", "IN2RN Switch", "IN2RN" }, - { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" }, + { "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" }, { "Right Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Right Output Mixer", "IN1R Switch", "IN1R PGA" }, -- cgit v1.2.2 From a47979b5aa2117848b742828c98abe7eea42a9ff Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Thu, 3 Dec 2009 18:56:56 +0530 Subject: ASoC: DaVinci: Update suspend/resume support for McASP driver Add clock enable and disable calls to resume and suspend respectively. Also add a member to the audio device data structure which tracks the clock status. Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied. [1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/ 2009-November/016958.html Signed-off-by: Chaithrika U S Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 18 ++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- 3 files changed, 18 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0a302e1080d9..a613bbb0bc91 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -767,14 +767,27 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int ret = 0; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + if (!dev->clk_active) { + clk_enable(dev->clk); + dev->clk_active = 1; + } + /* Fall through */ + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + davinci_mcasp_stop(dev, substream->stream); + if (dev->clk_active) { + clk_disable(dev->clk); + dev->clk_active = 0; + } + + break; + + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(dev, substream->stream); break; @@ -866,6 +879,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } clk_enable(dev->clk); + dev->clk_active = 1; dev->base = (void __iomem *)IO_ADDRESS(mem->start); dev->op_mode = pdata->op_mode; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 582c9249ef09..e755b5121ec7 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -44,6 +44,7 @@ struct davinci_audio_dev { int sample_rate; struct clk *clk; unsigned int codec_fmt; + u8 clk_active; /* McASP specific data */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ad4d7f47a86b..80c7fdf2f521 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -49,7 +49,7 @@ static void print_buf_info(int slot, char *name) static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | -- cgit v1.2.2 From 3a7aaed714bbe3c071000d720f0cce186d1897a4 Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Fri, 4 Dec 2009 13:49:10 +0200 Subject: ASoC: tlv320dac33: Add support for regulator framework Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 92 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 79 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 9c8903dbe647..5037454974b6 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -469,6 +499,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -476,12 +508,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -959,6 +998,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + return 0; pcm_err: @@ -1039,7 +1081,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1130,6 +1172,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1149,6 +1209,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1177,6 +1241,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- cgit v1.2.2 From fbfecd3712f917ca210a55c157233d88b785896b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 28 Oct 2009 20:11:04 +0100 Subject: tree-wide: fix typos "couter" -> "counter" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch was generated by git grep -E -i -l 'couter' | xargs -r perl -p -i -e 's/couter/counter/' Signed-off-by: Uwe Kleine-König Signed-off-by: Jiri Kosina --- sound/synth/emux/soundfont.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 63c8f45c0c22..67c91230c197 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -374,7 +374,7 @@ sf_zone_new(struct snd_sf_list *sflist, struct snd_soundfont *sf) /* - * increment sample couter + * increment sample counter */ static void set_sample_counter(struct snd_sf_list *sflist, struct snd_soundfont *sf, -- cgit v1.2.2 From af901ca181d92aac3a7dc265144a9081a86d8f39 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Andr=C3=A9=20Goddard=20Rosa?= Date: Sat, 14 Nov 2009 13:09:05 -0200 Subject: tree-wide: fix assorted typos all over the place MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit That is "success", "unknown", "through", "performance", "[re|un]mapping" , "access", "default", "reasonable", "[con]currently", "temperature" , "channel", "[un]used", "application", "example","hierarchy", "therefore" , "[over|under]flow", "contiguous", "threshold", "enough" and others. Signed-off-by: André Goddard Rosa Signed-off-by: Jiri Kosina --- sound/Kconfig | 2 +- sound/isa/cs423x/cs4236.c | 2 +- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/oss/dmasound/dmasound_paula.c | 2 +- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/cs46xx/imgs/cwcdma.asp | 9 +++++---- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/rme9652/hdspm.c | 4 ++-- sound/soc/codecs/uda134x.c | 4 ++-- sound/soc/codecs/wm8903.c | 6 +++--- sound/soc/codecs/wm8993.c | 4 ++-- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s6000/s6000-pcm.c | 2 +- sound/sound_core.c | 2 +- 17 files changed, 26 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/Kconfig b/sound/Kconfig index 4b5365ad6b46..fcad760f5691 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -55,7 +55,7 @@ config SOUND_OSS_CORE_PRECLAIM Please read Documentation/feature-removal-schedule.txt for details. - If unusre, say Y. + If unsure, say Y. source "sound/oss/dmasound/Kconfig" diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index a076a6ce8071..a828baaab636 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -177,7 +177,7 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { { .id = "CSC0437", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Digital PC 5000 Onboard - CS4236B */ { .id = "CSC0735", .devs = { { "CSC0000" }, { "CSC0010" } } }, - /* some uknown CS4236B */ + /* some unknown CS4236B */ { .id = "CSC0b35", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Intel PR440FX Onboard sound */ { .id = "CSC0b36", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 02e30d7c6a93..ddad60ef3f37 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -137,7 +137,7 @@ struct snd_miro { static void snd_miro_proc_init(struct snd_miro * miro); static char * snd_opti9xx_names[] = { - "unkown", + "unknown", "82C928", "82C929", "82C924", "82C925", "82C930", "82C931", "82C933" diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd555325b9d..848007508ffd 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -185,7 +185,7 @@ MODULE_DEVICE_TABLE(pnp_card, snd_opti9xx_pnpids); #endif static char * snd_opti9xx_names[] = { - "unkown", + "unknown", "82C928", "82C929", "82C924", "82C925", "82C930", "82C931", "82C933" diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 06e9e88e4c05..bb14e4c67e89 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -657,7 +657,7 @@ static int AmiStateInfo(char *buffer, size_t space) len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n", dmasound.volume_right); if (len >= space) { - printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ; + printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ; len = space ; } return len; diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d10ec61..8d13092300da 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -233,7 +233,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val snd_iprintf(buffer, "user-defined\n"); break; default: - snd_iprintf(buffer, "unkown\n"); + snd_iprintf(buffer, "unknown\n"); break; } snd_iprintf(buffer, "Sample Bits: "); diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp index 09d24c76f034..a65e1193c89a 100644 --- a/sound/pci/cs46xx/imgs/cwcdma.asp +++ b/sound/pci/cs46xx/imgs/cwcdma.asp @@ -26,10 +26,11 @@ // // // The purpose of this code is very simple: make it possible to tranfser -// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host) -// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters) -// task always alters the samples in some how, however it's from 48khz -> 48khz. The -// alterations are not audible, but AC3 wont work. +// the samples 'as they are' with no alteration from a PCMreader +// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF. +// SRC (source rate converters) task always alters the samples in somehow, +// however it's from 48khz -> 48khz. +// The alterations are not audible, but AC3 wont work. // // ... // | diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 36e08bd2b3cc..360e3809a60b 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -184,7 +184,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard."); * The hardware has 3 channels for playback and 1 for capture. * - channel 0 is the front channel * - channel 1 is the rear channel - * - channel 2 is the center/lfe chanel + * - channel 2 is the center/lfe channel * Volume is controlled by the AC97 for the front and rear channels by * the PCM Playback Volume, Sigmatel Surround Playback Volume and * Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 780e1a72114a..8917071d5b6a 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -66,7 +66,7 @@ struct cmi_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ - /* pin deafault configuration */ + /* pin default configuration */ hda_nid_t pin_nid[NUM_PINS]; unsigned int def_conf[NUM_PINS]; unsigned int pin_def_confs; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ff20048504b6..872731eb49e8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6619,7 +6619,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = { /* Front Mic (0x01) unused */ { "Line", 0x2 }, /* Line 2 (0x03) unused */ - /* CD (0x04) unsused? */ + /* CD (0x04) unused? */ }, }; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0dce331a2a3b..a1b10d1a384d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3017,7 +3017,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, insel = "Coaxial"; break; default: - insel = "Unkown"; + insel = "Unknown"; } switch (hdspm->control_register & HDSPM_SyncRefMask) { @@ -3028,7 +3028,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, syncref = "MADI"; break; default: - syncref = "Unkown"; + syncref = "Unknown"; } snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel, syncref); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c33b92edbded..8ce1c9b2e5b8 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); if (reg >= UDA134X_REGS_NUM) { - printk(KERN_ERR "%s unkown register: reg: %u", + printk(KERN_ERR "%s unknown register: reg: %u", __func__, reg); return -EINVAL; } @@ -552,7 +552,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) ARRAY_SIZE(uda1341_snd_controls)); break; default: - printk(KERN_ERR "%s unkown codec type: %d", + printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); return -EINVAL; } diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index fe1307b500cf..d72347d90b70 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -607,7 +607,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0), SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1), -SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1, +SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8903_DRC_3, 5, 124, 1, drc_tlv_thresh), SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp), SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min), @@ -617,11 +617,11 @@ SOC_ENUM("DRC Decay Rate", drc_decay), SOC_ENUM("DRC FF Delay", drc_ff_delay), SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0), SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0), -SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max), +SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max), SOC_ENUM("DRC QR Decay Rate", drc_qr_decay), SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0), SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0), -SOC_ENUM("DRC Smoothing Threashold", drc_smoothing), +SOC_ENUM("DRC Smoothing Threshold", drc_smoothing), SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup), SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d9987999e92c..bc033687b220 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -689,7 +689,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE, SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0), SOC_ENUM("DRC Path", drc_path), -SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2, +SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8993_DRC_CONTROL_2, 2, 60, 1, drc_comp_threash), SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3, 11, 30, 1, drc_comp_amp), @@ -709,7 +709,7 @@ SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0, SOC_ENUM("DRC Quick Release Rate", drc_qr_rate), SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0), SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0), -SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth), +SOC_ENUM("DRC Smoothing Hysteresis Threshold", drc_smooth), SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0, drc_startup_tlv), diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 1966e0d5652d..3c7ccb78b6ab 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -270,7 +270,7 @@ static int attach_gpio_amp(struct device *dev, gpio_direction_output(pd->amp_gain[1], 0); } - /* note, curently we assume GPA0 isn't valid amp */ + /* note, currently we assume GPA0 isn't valid amp */ if (pdata->amp_gpio > 0) { ret = gpio_request(pd->amp_gpio, "gpio-amp"); if (ret) { diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 83b8028e209d..81d6f983f51e 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -196,7 +196,7 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream) 0 /* destination skip after chunk (impossible) */, 4 /* 16 byte burst size */, -1 /* don't conserve bandwidth */, - 0 /* low watermark irq descriptor theshold */, + 0 /* low watermark irq descriptor threshold */, 0 /* disable hardware timestamps */, 1 /* enable channel */); diff --git a/sound/sound_core.c b/sound/sound_core.c index 49c998186592..dbca7c909a31 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -353,7 +353,7 @@ static struct sound_unit *chains[SOUND_STEP]; * @dev: device pointer * * Allocate a special sound device by minor number from the sound - * subsystem. The allocated number is returned on succes. On failure + * subsystem. The allocated number is returned on success. On failure * a negative error code is returned. */ -- cgit v1.2.2 From dd1b3d53c2e5b9cccec9001fc0b63f6b686a4ac9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Dec 2009 14:22:03 +0000 Subject: ASoC: Export snd_soc_update_bits_unlocked() Allows custom controls to use it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ef8f28284cb9..8b900a842677 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1427,9 +1427,9 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * * Returns 1 for change else 0. */ -static int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, - unsigned int value) +int snd_soc_update_bits_locked(struct snd_soc_codec *codec, + unsigned short reg, unsigned int mask, + unsigned int value) { int change; @@ -1439,6 +1439,7 @@ static int snd_soc_update_bits_locked(struct snd_soc_codec *codec, return change; } +EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked); /** * snd_soc_test_bits - test register for change -- cgit v1.2.2 From d033c36ae5cec22c893c710cd026fb732c4086b9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Dec 2009 15:25:56 +0000 Subject: ASoC: Display the power register in DAPM widget debugfs Make it a bit easier to tie DAPM widgets in with the register map without referring to the source by including the register location controlled by the widget. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d294ef72590..846678aa3d35 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1147,9 +1147,16 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w); dapm_clear_walk(w->codec); - ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n", + ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", w->name, w->power ? "On" : "Off", in, out); + if (w->reg >= 0) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + " - R%d(0x%x) bit %d", + w->reg, w->reg, w->shift); + + ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + if (w->sname) ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, -- cgit v1.2.2 From a91eb199e4dc8a2ab3fb7a53f1a23ce82b29fc04 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Nov 2009 11:56:07 +0000 Subject: ASoC: Initial WM8904 CODEC driver The WM8904 is a high performance ultra-low power stereo CODEC optimised for portable audio applications, with features including a class W amplifier, FLL with free running mode, Mobile ReTune and ground referenced headphone and line outputs. Support for some features, most particularly the digital microphone interface, is not yet present. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8904.c | 2538 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8904.h | 1681 ++++++++++++++++++++++++++++++ 4 files changed, 4225 insertions(+) create mode 100644 sound/soc/codecs/wm8904.c create mode 100644 sound/soc/codecs/wm8904.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f8fed4..011d3ab7e64a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C + select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8960 if I2C select SND_SOC_WM8961 if I2C @@ -203,6 +204,9 @@ config SND_SOC_WM8900 config SND_SOC_WM8903 tristate +config SND_SOC_WM8904 + tristate + config SND_SOC_WM8940 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb133ac7..0471d9044205 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -36,6 +36,7 @@ snd-soc-wm8753-objs := wm8753.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o +snd-soc-wm8904-objs := wm8904.o snd-soc-wm8940-objs := wm8940.o snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o @@ -92,6 +93,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o +obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c new file mode 100644 index 000000000000..8310e5d14b83 --- /dev/null +++ b/sound/soc/codecs/wm8904.c @@ -0,0 +1,2538 @@ +/* + * wm8904.c -- WM8904 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8904.h" + +static struct snd_soc_codec *wm8904_codec; +struct snd_soc_codec_device soc_codec_dev_wm8904; + +#define WM8904_NUM_DCS_CHANNELS 4 + +#define WM8904_NUM_SUPPLIES 5 +static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD", + "AVDD", + "CPVDD", + "MICVDD", +}; + +/* codec private data */ +struct wm8904_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8904_MAX_REGISTER + 1]; + + struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES]; + + struct wm8904_pdata *pdata; + + int deemph; + + /* Platform provided DRC configuration */ + const char **drc_texts; + int drc_cfg; + struct soc_enum drc_enum; + + /* Platform provided ReTune mobile configuration */ + int num_retune_mobile_texts; + const char **retune_mobile_texts; + int retune_mobile_cfg; + struct soc_enum retune_mobile_enum; + + /* FLL setup */ + int fll_src; + int fll_fref; + int fll_fout; + + /* Clocking configuration */ + unsigned int mclk_rate; + int sysclk_src; + unsigned int sysclk_rate; + + int tdm_width; + int tdm_slots; + int bclk; + int fs; + + /* DC servo configuration - cached offset values */ + int dcs_state[WM8904_NUM_DCS_CHANNELS]; +}; + +static const u16 wm8904_reg[WM8904_MAX_REGISTER + 1] = { + 0x8904, /* R0 - SW Reset and ID */ + 0x0000, /* R1 - Revision */ + 0x0000, /* R2 */ + 0x0000, /* R3 */ + 0x0018, /* R4 - Bias Control 0 */ + 0x0000, /* R5 - VMID Control 0 */ + 0x0000, /* R6 - Mic Bias Control 0 */ + 0x0000, /* R7 - Mic Bias Control 1 */ + 0x0001, /* R8 - Analogue DAC 0 */ + 0x9696, /* R9 - mic Filter Control */ + 0x0001, /* R10 - Analogue ADC 0 */ + 0x0000, /* R11 */ + 0x0000, /* R12 - Power Management 0 */ + 0x0000, /* R13 */ + 0x0000, /* R14 - Power Management 2 */ + 0x0000, /* R15 - Power Management 3 */ + 0x0000, /* R16 */ + 0x0000, /* R17 */ + 0x0000, /* R18 - Power Management 6 */ + 0x0000, /* R19 */ + 0x945E, /* R20 - Clock Rates 0 */ + 0x0C05, /* R21 - Clock Rates 1 */ + 0x0006, /* R22 - Clock Rates 2 */ + 0x0000, /* R23 */ + 0x0050, /* R24 - Audio Interface 0 */ + 0x000A, /* R25 - Audio Interface 1 */ + 0x00E4, /* R26 - Audio Interface 2 */ + 0x0040, /* R27 - Audio Interface 3 */ + 0x0000, /* R28 */ + 0x0000, /* R29 */ + 0x00C0, /* R30 - DAC Digital Volume Left */ + 0x00C0, /* R31 - DAC Digital Volume Right */ + 0x0000, /* R32 - DAC Digital 0 */ + 0x0008, /* R33 - DAC Digital 1 */ + 0x0000, /* R34 */ + 0x0000, /* R35 */ + 0x00C0, /* R36 - ADC Digital Volume Left */ + 0x00C0, /* R37 - ADC Digital Volume Right */ + 0x0010, /* R38 - ADC Digital 0 */ + 0x0000, /* R39 - Digital Microphone 0 */ + 0x01AF, /* R40 - DRC 0 */ + 0x3248, /* R41 - DRC 1 */ + 0x0000, /* R42 - DRC 2 */ + 0x0000, /* R43 - DRC 3 */ + 0x0085, /* R44 - Analogue Left Input 0 */ + 0x0085, /* R45 - Analogue Right Input 0 */ + 0x0044, /* R46 - Analogue Left Input 1 */ + 0x0044, /* R47 - Analogue Right Input 1 */ + 0x0000, /* R48 */ + 0x0000, /* R49 */ + 0x0000, /* R50 */ + 0x0000, /* R51 */ + 0x0000, /* R52 */ + 0x0000, /* R53 */ + 0x0000, /* R54 */ + 0x0000, /* R55 */ + 0x0000, /* R56 */ + 0x002D, /* R57 - Analogue OUT1 Left */ + 0x002D, /* R58 - Analogue OUT1 Right */ + 0x0039, /* R59 - Analogue OUT2 Left */ + 0x0039, /* R60 - Analogue OUT2 Right */ + 0x0000, /* R61 - Analogue OUT12 ZC */ + 0x0000, /* R62 */ + 0x0000, /* R63 */ + 0x0000, /* R64 */ + 0x0000, /* R65 */ + 0x0000, /* R66 */ + 0x0000, /* R67 - DC Servo 0 */ + 0x0000, /* R68 - DC Servo 1 */ + 0xAAAA, /* R69 - DC Servo 2 */ + 0x0000, /* R70 */ + 0xAAAA, /* R71 - DC Servo 4 */ + 0xAAAA, /* R72 - DC Servo 5 */ + 0x0000, /* R73 - DC Servo 6 */ + 0x0000, /* R74 - DC Servo 7 */ + 0x0000, /* R75 - DC Servo 8 */ + 0x0000, /* R76 - DC Servo 9 */ + 0x0000, /* R77 - DC Servo Readback 0 */ + 0x0000, /* R78 */ + 0x0000, /* R79 */ + 0x0000, /* R80 */ + 0x0000, /* R81 */ + 0x0000, /* R82 */ + 0x0000, /* R83 */ + 0x0000, /* R84 */ + 0x0000, /* R85 */ + 0x0000, /* R86 */ + 0x0000, /* R87 */ + 0x0000, /* R88 */ + 0x0000, /* R89 */ + 0x0000, /* R90 - Analogue HP 0 */ + 0x0000, /* R91 */ + 0x0000, /* R92 */ + 0x0000, /* R93 */ + 0x0000, /* R94 - Analogue Lineout 0 */ + 0x0000, /* R95 */ + 0x0000, /* R96 */ + 0x0000, /* R97 */ + 0x0000, /* R98 - Charge Pump 0 */ + 0x0000, /* R99 */ + 0x0000, /* R100 */ + 0x0000, /* R101 */ + 0x0000, /* R102 */ + 0x0000, /* R103 */ + 0x0004, /* R104 - Class W 0 */ + 0x0000, /* R105 */ + 0x0000, /* R106 */ + 0x0000, /* R107 */ + 0x0000, /* R108 - Write Sequencer 0 */ + 0x0000, /* R109 - Write Sequencer 1 */ + 0x0000, /* R110 - Write Sequencer 2 */ + 0x0000, /* R111 - Write Sequencer 3 */ + 0x0000, /* R112 - Write Sequencer 4 */ + 0x0000, /* R113 */ + 0x0000, /* R114 */ + 0x0000, /* R115 */ + 0x0000, /* R116 - FLL Control 1 */ + 0x0007, /* R117 - FLL Control 2 */ + 0x0000, /* R118 - FLL Control 3 */ + 0x2EE0, /* R119 - FLL Control 4 */ + 0x0004, /* R120 - FLL Control 5 */ + 0x0014, /* R121 - GPIO Control 1 */ + 0x0010, /* R122 - GPIO Control 2 */ + 0x0010, /* R123 - GPIO Control 3 */ + 0x0000, /* R124 - GPIO Control 4 */ + 0x0000, /* R125 */ + 0x0000, /* R126 - Digital Pulls */ + 0x0000, /* R127 - Interrupt Status */ + 0xFFFF, /* R128 - Interrupt Status Mask */ + 0x0000, /* R129 - Interrupt Polarity */ + 0x0000, /* R130 - Interrupt Debounce */ + 0x0000, /* R131 */ + 0x0000, /* R132 */ + 0x0000, /* R133 */ + 0x0000, /* R134 - EQ1 */ + 0x000C, /* R135 - EQ2 */ + 0x000C, /* R136 - EQ3 */ + 0x000C, /* R137 - EQ4 */ + 0x000C, /* R138 - EQ5 */ + 0x000C, /* R139 - EQ6 */ + 0x0FCA, /* R140 - EQ7 */ + 0x0400, /* R141 - EQ8 */ + 0x00D8, /* R142 - EQ9 */ + 0x1EB5, /* R143 - EQ10 */ + 0xF145, /* R144 - EQ11 */ + 0x0B75, /* R145 - EQ12 */ + 0x01C5, /* R146 - EQ13 */ + 0x1C58, /* R147 - EQ14 */ + 0xF373, /* R148 - EQ15 */ + 0x0A54, /* R149 - EQ16 */ + 0x0558, /* R150 - EQ17 */ + 0x168E, /* R151 - EQ18 */ + 0xF829, /* R152 - EQ19 */ + 0x07AD, /* R153 - EQ20 */ + 0x1103, /* R154 - EQ21 */ + 0x0564, /* R155 - EQ22 */ + 0x0559, /* R156 - EQ23 */ + 0x4000, /* R157 - EQ24 */ + 0x0000, /* R158 */ + 0x0000, /* R159 */ + 0x0000, /* R160 */ + 0x0000, /* R161 - Control Interface Test 1 */ + 0x0000, /* R162 */ + 0x0000, /* R163 */ + 0x0000, /* R164 */ + 0x0000, /* R165 */ + 0x0000, /* R166 */ + 0x0000, /* R167 */ + 0x0000, /* R168 */ + 0x0000, /* R169 */ + 0x0000, /* R170 */ + 0x0000, /* R171 */ + 0x0000, /* R172 */ + 0x0000, /* R173 */ + 0x0000, /* R174 */ + 0x0000, /* R175 */ + 0x0000, /* R176 */ + 0x0000, /* R177 */ + 0x0000, /* R178 */ + 0x0000, /* R179 */ + 0x0000, /* R180 */ + 0x0000, /* R181 */ + 0x0000, /* R182 */ + 0x0000, /* R183 */ + 0x0000, /* R184 */ + 0x0000, /* R185 */ + 0x0000, /* R186 */ + 0x0000, /* R187 */ + 0x0000, /* R188 */ + 0x0000, /* R189 */ + 0x0000, /* R190 */ + 0x0000, /* R191 */ + 0x0000, /* R192 */ + 0x0000, /* R193 */ + 0x0000, /* R194 */ + 0x0000, /* R195 */ + 0x0000, /* R196 */ + 0x0000, /* R197 */ + 0x0000, /* R198 */ + 0x0000, /* R199 */ + 0x0000, /* R200 */ + 0x0000, /* R201 */ + 0x0000, /* R202 */ + 0x0000, /* R203 */ + 0x0000, /* R204 - Analogue Output Bias 0 */ + 0x0000, /* R205 */ + 0x0000, /* R206 */ + 0x0000, /* R207 */ + 0x0000, /* R208 */ + 0x0000, /* R209 */ + 0x0000, /* R210 */ + 0x0000, /* R211 */ + 0x0000, /* R212 */ + 0x0000, /* R213 */ + 0x0000, /* R214 */ + 0x0000, /* R215 */ + 0x0000, /* R216 */ + 0x0000, /* R217 */ + 0x0000, /* R218 */ + 0x0000, /* R219 */ + 0x0000, /* R220 */ + 0x0000, /* R221 */ + 0x0000, /* R222 */ + 0x0000, /* R223 */ + 0x0000, /* R224 */ + 0x0000, /* R225 */ + 0x0000, /* R226 */ + 0x0000, /* R227 */ + 0x0000, /* R228 */ + 0x0000, /* R229 */ + 0x0000, /* R230 */ + 0x0000, /* R231 */ + 0x0000, /* R232 */ + 0x0000, /* R233 */ + 0x0000, /* R234 */ + 0x0000, /* R235 */ + 0x0000, /* R236 */ + 0x0000, /* R237 */ + 0x0000, /* R238 */ + 0x0000, /* R239 */ + 0x0000, /* R240 */ + 0x0000, /* R241 */ + 0x0000, /* R242 */ + 0x0000, /* R243 */ + 0x0000, /* R244 */ + 0x0000, /* R245 */ + 0x0000, /* R246 */ + 0x0000, /* R247 - FLL NCO Test 0 */ + 0x0019, /* R248 - FLL NCO Test 1 */ +}; + +static struct { + int readable; + int writable; + int vol; +} wm8904_access[] = { + { 0xFFFF, 0xFFFF, 1 }, /* R0 - SW Reset and ID */ + { 0x0000, 0x0000, 0 }, /* R1 - Revision */ + { 0x0000, 0x0000, 0 }, /* R2 */ + { 0x0000, 0x0000, 0 }, /* R3 */ + { 0x001F, 0x001F, 0 }, /* R4 - Bias Control 0 */ + { 0x0047, 0x0047, 0 }, /* R5 - VMID Control 0 */ + { 0x007F, 0x007F, 0 }, /* R6 - Mic Bias Control 0 */ + { 0xC007, 0xC007, 0 }, /* R7 - Mic Bias Control 1 */ + { 0x001E, 0x001E, 0 }, /* R8 - Analogue DAC 0 */ + { 0xFFFF, 0xFFFF, 0 }, /* R9 - mic Filter Control */ + { 0x0001, 0x0001, 0 }, /* R10 - Analogue ADC 0 */ + { 0x0000, 0x0000, 0 }, /* R11 */ + { 0x0003, 0x0003, 0 }, /* R12 - Power Management 0 */ + { 0x0000, 0x0000, 0 }, /* R13 */ + { 0x0003, 0x0003, 0 }, /* R14 - Power Management 2 */ + { 0x0003, 0x0003, 0 }, /* R15 - Power Management 3 */ + { 0x0000, 0x0000, 0 }, /* R16 */ + { 0x0000, 0x0000, 0 }, /* R17 */ + { 0x000F, 0x000F, 0 }, /* R18 - Power Management 6 */ + { 0x0000, 0x0000, 0 }, /* R19 */ + { 0x7001, 0x7001, 0 }, /* R20 - Clock Rates 0 */ + { 0x3C07, 0x3C07, 0 }, /* R21 - Clock Rates 1 */ + { 0xD00F, 0xD00F, 0 }, /* R22 - Clock Rates 2 */ + { 0x0000, 0x0000, 0 }, /* R23 */ + { 0x1FFF, 0x1FFF, 0 }, /* R24 - Audio Interface 0 */ + { 0x3DDF, 0x3DDF, 0 }, /* R25 - Audio Interface 1 */ + { 0x0F1F, 0x0F1F, 0 }, /* R26 - Audio Interface 2 */ + { 0x0FFF, 0x0FFF, 0 }, /* R27 - Audio Interface 3 */ + { 0x0000, 0x0000, 0 }, /* R28 */ + { 0x0000, 0x0000, 0 }, /* R29 */ + { 0x00FF, 0x01FF, 0 }, /* R30 - DAC Digital Volume Left */ + { 0x00FF, 0x01FF, 0 }, /* R31 - DAC Digital Volume Right */ + { 0x0FFF, 0x0FFF, 0 }, /* R32 - DAC Digital 0 */ + { 0x1E4E, 0x1E4E, 0 }, /* R33 - DAC Digital 1 */ + { 0x0000, 0x0000, 0 }, /* R34 */ + { 0x0000, 0x0000, 0 }, /* R35 */ + { 0x00FF, 0x01FF, 0 }, /* R36 - ADC Digital Volume Left */ + { 0x00FF, 0x01FF, 0 }, /* R37 - ADC Digital Volume Right */ + { 0x0073, 0x0073, 0 }, /* R38 - ADC Digital 0 */ + { 0x1800, 0x1800, 0 }, /* R39 - Digital Microphone 0 */ + { 0xDFEF, 0xDFEF, 0 }, /* R40 - DRC 0 */ + { 0xFFFF, 0xFFFF, 0 }, /* R41 - DRC 1 */ + { 0x003F, 0x003F, 0 }, /* R42 - DRC 2 */ + { 0x07FF, 0x07FF, 0 }, /* R43 - DRC 3 */ + { 0x009F, 0x009F, 0 }, /* R44 - Analogue Left Input 0 */ + { 0x009F, 0x009F, 0 }, /* R45 - Analogue Right Input 0 */ + { 0x007F, 0x007F, 0 }, /* R46 - Analogue Left Input 1 */ + { 0x007F, 0x007F, 0 }, /* R47 - Analogue Right Input 1 */ + { 0x0000, 0x0000, 0 }, /* R48 */ + { 0x0000, 0x0000, 0 }, /* R49 */ + { 0x0000, 0x0000, 0 }, /* R50 */ + { 0x0000, 0x0000, 0 }, /* R51 */ + { 0x0000, 0x0000, 0 }, /* R52 */ + { 0x0000, 0x0000, 0 }, /* R53 */ + { 0x0000, 0x0000, 0 }, /* R54 */ + { 0x0000, 0x0000, 0 }, /* R55 */ + { 0x0000, 0x0000, 0 }, /* R56 */ + { 0x017F, 0x01FF, 0 }, /* R57 - Analogue OUT1 Left */ + { 0x017F, 0x01FF, 0 }, /* R58 - Analogue OUT1 Right */ + { 0x017F, 0x01FF, 0 }, /* R59 - Analogue OUT2 Left */ + { 0x017F, 0x01FF, 0 }, /* R60 - Analogue OUT2 Right */ + { 0x000F, 0x000F, 0 }, /* R61 - Analogue OUT12 ZC */ + { 0x0000, 0x0000, 0 }, /* R62 */ + { 0x0000, 0x0000, 0 }, /* R63 */ + { 0x0000, 0x0000, 0 }, /* R64 */ + { 0x0000, 0x0000, 0 }, /* R65 */ + { 0x0000, 0x0000, 0 }, /* R66 */ + { 0x000F, 0x000F, 0 }, /* R67 - DC Servo 0 */ + { 0xFFFF, 0xFFFF, 1 }, /* R68 - DC Servo 1 */ + { 0x0F0F, 0x0F0F, 0 }, /* R69 - DC Servo 2 */ + { 0x0000, 0x0000, 0 }, /* R70 */ + { 0x007F, 0x007F, 0 }, /* R71 - DC Servo 4 */ + { 0x007F, 0x007F, 0 }, /* R72 - DC Servo 5 */ + { 0x00FF, 0x00FF, 1 }, /* R73 - DC Servo 6 */ + { 0x00FF, 0x00FF, 1 }, /* R74 - DC Servo 7 */ + { 0x00FF, 0x00FF, 1 }, /* R75 - DC Servo 8 */ + { 0x00FF, 0x00FF, 1 }, /* R76 - DC Servo 9 */ + { 0x0FFF, 0x0000, 1 }, /* R77 - DC Servo Readback 0 */ + { 0x0000, 0x0000, 0 }, /* R78 */ + { 0x0000, 0x0000, 0 }, /* R79 */ + { 0x0000, 0x0000, 0 }, /* R80 */ + { 0x0000, 0x0000, 0 }, /* R81 */ + { 0x0000, 0x0000, 0 }, /* R82 */ + { 0x0000, 0x0000, 0 }, /* R83 */ + { 0x0000, 0x0000, 0 }, /* R84 */ + { 0x0000, 0x0000, 0 }, /* R85 */ + { 0x0000, 0x0000, 0 }, /* R86 */ + { 0x0000, 0x0000, 0 }, /* R87 */ + { 0x0000, 0x0000, 0 }, /* R88 */ + { 0x0000, 0x0000, 0 }, /* R89 */ + { 0x00FF, 0x00FF, 0 }, /* R90 - Analogue HP 0 */ + { 0x0000, 0x0000, 0 }, /* R91 */ + { 0x0000, 0x0000, 0 }, /* R92 */ + { 0x0000, 0x0000, 0 }, /* R93 */ + { 0x00FF, 0x00FF, 0 }, /* R94 - Analogue Lineout 0 */ + { 0x0000, 0x0000, 0 }, /* R95 */ + { 0x0000, 0x0000, 0 }, /* R96 */ + { 0x0000, 0x0000, 0 }, /* R97 */ + { 0x0001, 0x0001, 0 }, /* R98 - Charge Pump 0 */ + { 0x0000, 0x0000, 0 }, /* R99 */ + { 0x0000, 0x0000, 0 }, /* R100 */ + { 0x0000, 0x0000, 0 }, /* R101 */ + { 0x0000, 0x0000, 0 }, /* R102 */ + { 0x0000, 0x0000, 0 }, /* R103 */ + { 0x0001, 0x0001, 0 }, /* R104 - Class W 0 */ + { 0x0000, 0x0000, 0 }, /* R105 */ + { 0x0000, 0x0000, 0 }, /* R106 */ + { 0x0000, 0x0000, 0 }, /* R107 */ + { 0x011F, 0x011F, 0 }, /* R108 - Write Sequencer 0 */ + { 0x7FFF, 0x7FFF, 0 }, /* R109 - Write Sequencer 1 */ + { 0x4FFF, 0x4FFF, 0 }, /* R110 - Write Sequencer 2 */ + { 0x003F, 0x033F, 0 }, /* R111 - Write Sequencer 3 */ + { 0x03F1, 0x0000, 0 }, /* R112 - Write Sequencer 4 */ + { 0x0000, 0x0000, 0 }, /* R113 */ + { 0x0000, 0x0000, 0 }, /* R114 */ + { 0x0000, 0x0000, 0 }, /* R115 */ + { 0x0007, 0x0007, 0 }, /* R116 - FLL Control 1 */ + { 0x3F77, 0x3F77, 0 }, /* R117 - FLL Control 2 */ + { 0xFFFF, 0xFFFF, 0 }, /* R118 - FLL Control 3 */ + { 0x7FEF, 0x7FEF, 0 }, /* R119 - FLL Control 4 */ + { 0x001B, 0x001B, 0 }, /* R120 - FLL Control 5 */ + { 0x003F, 0x003F, 0 }, /* R121 - GPIO Control 1 */ + { 0x003F, 0x003F, 0 }, /* R122 - GPIO Control 2 */ + { 0x003F, 0x003F, 0 }, /* R123 - GPIO Control 3 */ + { 0x038F, 0x038F, 0 }, /* R124 - GPIO Control 4 */ + { 0x0000, 0x0000, 0 }, /* R125 */ + { 0x00FF, 0x00FF, 0 }, /* R126 - Digital Pulls */ + { 0x07FF, 0x03FF, 1 }, /* R127 - Interrupt Status */ + { 0x03FF, 0x03FF, 0 }, /* R128 - Interrupt Status Mask */ + { 0x03FF, 0x03FF, 0 }, /* R129 - Interrupt Polarity */ + { 0x03FF, 0x03FF, 0 }, /* R130 - Interrupt Debounce */ + { 0x0000, 0x0000, 0 }, /* R131 */ + { 0x0000, 0x0000, 0 }, /* R132 */ + { 0x0000, 0x0000, 0 }, /* R133 */ + { 0x0001, 0x0001, 0 }, /* R134 - EQ1 */ + { 0x001F, 0x001F, 0 }, /* R135 - EQ2 */ + { 0x001F, 0x001F, 0 }, /* R136 - EQ3 */ + { 0x001F, 0x001F, 0 }, /* R137 - EQ4 */ + { 0x001F, 0x001F, 0 }, /* R138 - EQ5 */ + { 0x001F, 0x001F, 0 }, /* R139 - EQ6 */ + { 0xFFFF, 0xFFFF, 0 }, /* R140 - EQ7 */ + { 0xFFFF, 0xFFFF, 0 }, /* R141 - EQ8 */ + { 0xFFFF, 0xFFFF, 0 }, /* R142 - EQ9 */ + { 0xFFFF, 0xFFFF, 0 }, /* R143 - EQ10 */ + { 0xFFFF, 0xFFFF, 0 }, /* R144 - EQ11 */ + { 0xFFFF, 0xFFFF, 0 }, /* R145 - EQ12 */ + { 0xFFFF, 0xFFFF, 0 }, /* R146 - EQ13 */ + { 0xFFFF, 0xFFFF, 0 }, /* R147 - EQ14 */ + { 0xFFFF, 0xFFFF, 0 }, /* R148 - EQ15 */ + { 0xFFFF, 0xFFFF, 0 }, /* R149 - EQ16 */ + { 0xFFFF, 0xFFFF, 0 }, /* R150 - EQ17 */ + { 0xFFFF, 0xFFFF, 0 }, /* R151wm8523_dai - EQ18 */ + { 0xFFFF, 0xFFFF, 0 }, /* R152 - EQ19 */ + { 0xFFFF, 0xFFFF, 0 }, /* R153 - EQ20 */ + { 0xFFFF, 0xFFFF, 0 }, /* R154 - EQ21 */ + { 0xFFFF, 0xFFFF, 0 }, /* R155 - EQ22 */ + { 0xFFFF, 0xFFFF, 0 }, /* R156 - EQ23 */ + { 0xFFFF, 0xFFFF, 0 }, /* R157 - EQ24 */ + { 0x0000, 0x0000, 0 }, /* R158 */ + { 0x0000, 0x0000, 0 }, /* R159 */ + { 0x0000, 0x0000, 0 }, /* R160 */ + { 0x0002, 0x0002, 0 }, /* R161 - Control Interface Test 1 */ + { 0x0000, 0x0000, 0 }, /* R162 */ + { 0x0000, 0x0000, 0 }, /* R163 */ + { 0x0000, 0x0000, 0 }, /* R164 */ + { 0x0000, 0x0000, 0 }, /* R165 */ + { 0x0000, 0x0000, 0 }, /* R166 */ + { 0x0000, 0x0000, 0 }, /* R167 */ + { 0x0000, 0x0000, 0 }, /* R168 */ + { 0x0000, 0x0000, 0 }, /* R169 */ + { 0x0000, 0x0000, 0 }, /* R170 */ + { 0x0000, 0x0000, 0 }, /* R171 */ + { 0x0000, 0x0000, 0 }, /* R172 */ + { 0x0000, 0x0000, 0 }, /* R173 */ + { 0x0000, 0x0000, 0 }, /* R174 */ + { 0x0000, 0x0000, 0 }, /* R175 */ + { 0x0000, 0x0000, 0 }, /* R176 */ + { 0x0000, 0x0000, 0 }, /* R177 */ + { 0x0000, 0x0000, 0 }, /* R178 */ + { 0x0000, 0x0000, 0 }, /* R179 */ + { 0x0000, 0x0000, 0 }, /* R180 */ + { 0x0000, 0x0000, 0 }, /* R181 */ + { 0x0000, 0x0000, 0 }, /* R182 */ + { 0x0000, 0x0000, 0 }, /* R183 */ + { 0x0000, 0x0000, 0 }, /* R184 */ + { 0x0000, 0x0000, 0 }, /* R185 */ + { 0x0000, 0x0000, 0 }, /* R186 */ + { 0x0000, 0x0000, 0 }, /* R187 */ + { 0x0000, 0x0000, 0 }, /* R188 */ + { 0x0000, 0x0000, 0 }, /* R189 */ + { 0x0000, 0x0000, 0 }, /* R190 */ + { 0x0000, 0x0000, 0 }, /* R191 */ + { 0x0000, 0x0000, 0 }, /* R192 */ + { 0x0000, 0x0000, 0 }, /* R193 */ + { 0x0000, 0x0000, 0 }, /* R194 */ + { 0x0000, 0x0000, 0 }, /* R195 */ + { 0x0000, 0x0000, 0 }, /* R196 */ + { 0x0000, 0x0000, 0 }, /* R197 */ + { 0x0000, 0x0000, 0 }, /* R198 */ + { 0x0000, 0x0000, 0 }, /* R199 */ + { 0x0000, 0x0000, 0 }, /* R200 */ + { 0x0000, 0x0000, 0 }, /* R201 */ + { 0x0000, 0x0000, 0 }, /* R202 */ + { 0x0000, 0x0000, 0 }, /* R203 */ + { 0x0070, 0x0070, 0 }, /* R204 - Analogue Output Bias 0 */ + { 0x0000, 0x0000, 0 }, /* R205 */ + { 0x0000, 0x0000, 0 }, /* R206 */ + { 0x0000, 0x0000, 0 }, /* R207 */ + { 0x0000, 0x0000, 0 }, /* R208 */ + { 0x0000, 0x0000, 0 }, /* R209 */ + { 0x0000, 0x0000, 0 }, /* R210 */ + { 0x0000, 0x0000, 0 }, /* R211 */ + { 0x0000, 0x0000, 0 }, /* R212 */ + { 0x0000, 0x0000, 0 }, /* R213 */ + { 0x0000, 0x0000, 0 }, /* R214 */ + { 0x0000, 0x0000, 0 }, /* R215 */ + { 0x0000, 0x0000, 0 }, /* R216 */ + { 0x0000, 0x0000, 0 }, /* R217 */ + { 0x0000, 0x0000, 0 }, /* R218 */ + { 0x0000, 0x0000, 0 }, /* R219 */ + { 0x0000, 0x0000, 0 }, /* R220 */ + { 0x0000, 0x0000, 0 }, /* R221 */ + { 0x0000, 0x0000, 0 }, /* R222 */ + { 0x0000, 0x0000, 0 }, /* R223 */ + { 0x0000, 0x0000, 0 }, /* R224 */ + { 0x0000, 0x0000, 0 }, /* R225 */ + { 0x0000, 0x0000, 0 }, /* R226 */ + { 0x0000, 0x0000, 0 }, /* R227 */ + { 0x0000, 0x0000, 0 }, /* R228 */ + { 0x0000, 0x0000, 0 }, /* R229 */ + { 0x0000, 0x0000, 0 }, /* R230 */ + { 0x0000, 0x0000, 0 }, /* R231 */ + { 0x0000, 0x0000, 0 }, /* R232 */ + { 0x0000, 0x0000, 0 }, /* R233 */ + { 0x0000, 0x0000, 0 }, /* R234 */ + { 0x0000, 0x0000, 0 }, /* R235 */ + { 0x0000, 0x0000, 0 }, /* R236 */ + { 0x0000, 0x0000, 0 }, /* R237 */ + { 0x0000, 0x0000, 0 }, /* R238 */ + { 0x0000, 0x0000, 0 }, /* R239 */ + { 0x0000, 0x0000, 0 }, /* R240 */ + { 0x0000, 0x0000, 0 }, /* R241 */ + { 0x0000, 0x0000, 0 }, /* R242 */ + { 0x0000, 0x0000, 0 }, /* R243 */ + { 0x0000, 0x0000, 0 }, /* R244 */ + { 0x0000, 0x0000, 0 }, /* R245 */ + { 0x0000, 0x0000, 0 }, /* R246 */ + { 0x0001, 0x0001, 0 }, /* R247 - FLL NCO Test 0 */ + { 0x003F, 0x003F, 0 }, /* R248 - FLL NCO Test 1 */ +}; + +static int wm8904_volatile_register(unsigned int reg) +{ + return wm8904_access[reg].vol; +} + +static int wm8904_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0); +} + +static int wm8904_configure_clocking(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + unsigned int clock0, clock2, rate; + + /* Gate the clock while we're updating to avoid misclocking */ + clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_SYSCLK_SRC, 0); + + /* This should be done on init() for bypass paths */ + switch (wm8904->sysclk_src) { + case WM8904_CLK_MCLK: + dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8904->mclk_rate); + + clock2 &= ~WM8904_SYSCLK_SRC; + rate = wm8904->mclk_rate; + + /* Ensure the FLL is stopped */ + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + break; + + case WM8904_CLK_FLL: + dev_dbg(codec->dev, "Using %dHz FLL clock\n", + wm8904->fll_fout); + + clock2 |= WM8904_SYSCLK_SRC; + rate = wm8904->fll_fout; + break; + + default: + dev_err(codec->dev, "System clock not configured\n"); + return -EINVAL; + } + + /* SYSCLK shouldn't be over 13.5MHz */ + if (rate > 13500000) { + clock0 = WM8904_MCLK_DIV; + wm8904->sysclk_rate = rate / 2; + } else { + clock0 = 0; + wm8904->sysclk_rate = rate; + } + + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, WM8904_MCLK_DIV, + clock0); + + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA | WM8904_SYSCLK_SRC, clock2); + + dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm8904->sysclk_rate); + + return 0; +} + +static void wm8904_set_drc(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int save, i; + + /* Save any enables; the configuration should clear them. */ + save = snd_soc_read(codec, WM8904_DRC_0); + + for (i = 0; i < WM8904_DRC_REGS; i++) + snd_soc_update_bits(codec, WM8904_DRC_0 + i, 0xffff, + pdata->drc_cfgs[wm8904->drc_cfg].regs[i]); + + /* Reenable the DRC */ + snd_soc_update_bits(codec, WM8904_DRC_0, + WM8904_DRC_ENA | WM8904_DRC_DAC_PATH, save); +} + +static int wm8904_put_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int value = ucontrol->value.integer.value[0]; + + if (value >= pdata->num_drc_cfgs) + return -EINVAL; + + wm8904->drc_cfg = value; + + wm8904_set_drc(codec); + + return 0; +} + +static int wm8904_get_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + + ucontrol->value.enumerated.item[0] = wm8904->drc_cfg; + + return 0; +} + +static void wm8904_set_retune_mobile(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int best, best_val, save, i, cfg; + + if (!pdata || !wm8904->num_retune_mobile_texts) + return; + + /* Find the version of the currently selected configuration + * with the nearest sample rate. */ + cfg = wm8904->retune_mobile_cfg; + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8904->retune_mobile_texts[cfg]) == 0 && + abs(pdata->retune_mobile_cfgs[i].rate + - wm8904->fs) < best_val) { + best = i; + best_val = abs(pdata->retune_mobile_cfgs[i].rate + - wm8904->fs); + } + } + + dev_dbg(codec->dev, "ReTune Mobile %s/%dHz for %dHz sample rate\n", + pdata->retune_mobile_cfgs[best].name, + pdata->retune_mobile_cfgs[best].rate, + wm8904->fs); + + /* The EQ will be disabled while reconfiguring it, remember the + * current configuration. + */ + save = snd_soc_read(codec, WM8904_EQ1); + + for (i = 0; i < WM8904_EQ_REGS; i++) + snd_soc_update_bits(codec, WM8904_EQ1 + i, 0xffff, + pdata->retune_mobile_cfgs[best].regs[i]); + + snd_soc_update_bits(codec, WM8904_EQ1, WM8904_EQ_ENA, save); +} + +static int wm8904_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int value = ucontrol->value.integer.value[0]; + + if (value >= pdata->num_retune_mobile_cfgs) + return -EINVAL; + + wm8904->retune_mobile_cfg = value; + + wm8904_set_retune_mobile(codec); + + return 0; +} + +static int wm8904_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + + ucontrol->value.enumerated.item[0] = wm8904->retune_mobile_cfg; + + return 0; +} + +static int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int wm8904_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + int val, i, best; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8904->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - wm8904->fs) < + abs(deemph_settings[best] - wm8904->fs)) + best = i; + } + + val = best << WM8904_DEEMPH_SHIFT; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1, + WM8904_DEEMPH_MASK, val); +} + +static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + + return wm8904->deemph; +} + +static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + wm8904->deemph = deemph; + + return wm8904_set_deemph(codec); +} + +static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); + +static const char *input_mode_text[] = { + "Single-Ended", "Differential Line", "Differential Mic" +}; + +static const struct soc_enum lin_mode = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text); + +static const struct soc_enum rin_mode = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text); + +static const char *hpf_mode_text[] = { + "Hi-fi", "Voice 1", "Voice 2", "Voice 3" +}; + +static const struct soc_enum hpf_mode = + SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text); + +static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = { +SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv), + +SOC_ENUM("Left Caputure Mode", lin_mode), +SOC_ENUM("Right Capture Mode", rin_mode), + +/* No TLV since it depends on mode */ +SOC_DOUBLE_R("Capture Volume", WM8904_ANALOGUE_LEFT_INPUT_0, + WM8904_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, + WM8904_ANALOGUE_RIGHT_INPUT_0, 7, 1, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), +SOC_ENUM("High Pass Filter Mode", hpf_mode), + +SOC_SINGLE("ADC 128x OSR Switch", WM8904_ANALOGUE_ADC_0, 0, 1, 0), +}; + +static const char *drc_path_text[] = { + "ADC", "DAC" +}; + +static const struct soc_enum drc_path = + SOC_ENUM_SINGLE(WM8904_DRC_0, 14, 2, drc_path_text); + +static const struct snd_kcontrol_new wm8904_dac_snd_controls[] = { +SOC_SINGLE_TLV("Digital Playback Boost Volume", + WM8904_AUDIO_INTERFACE_0, 9, 3, 0, dac_boost_tlv), +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv), + +SOC_DOUBLE_R_TLV("Headphone Volume", WM8904_ANALOGUE_OUT1_LEFT, + WM8904_ANALOGUE_OUT1_RIGHT, 0, 63, 0, out_tlv), +SOC_DOUBLE_R("Headphone Switch", WM8904_ANALOGUE_OUT1_LEFT, + WM8904_ANALOGUE_OUT1_RIGHT, 8, 1, 1), +SOC_DOUBLE_R("Headphone ZC Switch", WM8904_ANALOGUE_OUT1_LEFT, + WM8904_ANALOGUE_OUT1_RIGHT, 6, 1, 0), + +SOC_DOUBLE_R_TLV("Line Output Volume", WM8904_ANALOGUE_OUT2_LEFT, + WM8904_ANALOGUE_OUT2_RIGHT, 0, 63, 0, out_tlv), +SOC_DOUBLE_R("Line Output Switch", WM8904_ANALOGUE_OUT2_LEFT, + WM8904_ANALOGUE_OUT2_RIGHT, 8, 1, 1), +SOC_DOUBLE_R("Line Output ZC Switch", WM8904_ANALOGUE_OUT2_LEFT, + WM8904_ANALOGUE_OUT2_RIGHT, 6, 1, 0), + +SOC_SINGLE("EQ Switch", WM8904_EQ1, 0, 1, 0), +SOC_SINGLE("DRC Switch", WM8904_DRC_0, 15, 1, 0), +SOC_ENUM("DRC Path", drc_path), +SOC_SINGLE("DAC OSRx2 Switch", WM8904_DAC_DIGITAL_1, 6, 1, 0), +SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + wm8904_get_deemph, wm8904_put_deemph), +}; + +static const struct snd_kcontrol_new wm8904_snd_controls[] = { +SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8904_DAC_DIGITAL_0, 4, 8, 15, 0, + sidetone_tlv), +}; + +static const struct snd_kcontrol_new wm8904_eq_controls[] = { +SOC_SINGLE_TLV("EQ1 Volume", WM8904_EQ2, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Volume", WM8904_EQ3, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Volume", WM8904_EQ4, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Volume", WM8904_EQ5, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ5 Volume", WM8904_EQ6, 0, 24, 0, eq_tlv), +}; + +static int cp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + BUG_ON(event != SND_SOC_DAPM_POST_PMU); + + /* Maximum startup time */ + udelay(500); + + return 0; +} + +static int sysclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8904_priv *wm8904 = codec->private_data; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* If we're using the FLL then we only start it when + * required; we assume that the configuration has been + * done previously and all we need to do is kick it + * off. + */ + switch (wm8904->sysclk_src) { + case WM8904_CLK_FLL: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA, + WM8904_FLL_OSC_ENA); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_ENA, + WM8904_FLL_ENA); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + break; + } + + return 0; +} + +static int out_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8904_priv *wm8904 = codec->private_data; + int reg, val; + int dcs_mask; + int dcs_l, dcs_r; + int dcs_l_reg, dcs_r_reg; + int timeout; + + /* This code is shared between HP and LINEOUT; we do all our + * power management in stereo pairs to avoid latency issues so + * we reuse shift to identify which rather than strcmp() the + * name. */ + reg = w->shift; + + switch (reg) { + case WM8904_ANALOGUE_HP_0: + dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1; + dcs_r_reg = WM8904_DC_SERVO_8; + dcs_l_reg = WM8904_DC_SERVO_9; + dcs_l = 0; + dcs_r = 1; + break; + case WM8904_ANALOGUE_LINEOUT_0: + dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3; + dcs_r_reg = WM8904_DC_SERVO_6; + dcs_l_reg = WM8904_DC_SERVO_7; + dcs_l = 2; + dcs_r = 3; + break; + default: + BUG(); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Power on the amplifier */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA | WM8904_HPR_ENA, + WM8904_HPL_ENA | WM8904_HPR_ENA); + + /* Enable the first stage */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY, + WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY); + + /* Power up the DC servo */ + snd_soc_update_bits(codec, WM8904_DC_SERVO_0, + dcs_mask, dcs_mask); + + /* Either calibrate the DC servo or restore cached state + * if we have that. + */ + if (wm8904->dcs_state[dcs_l] || wm8904->dcs_state[dcs_r]) { + dev_dbg(codec->dev, "Restoring DC servo state\n"); + + snd_soc_write(codec, dcs_l_reg, + wm8904->dcs_state[dcs_l]); + snd_soc_write(codec, dcs_r_reg, + wm8904->dcs_state[dcs_r]); + + snd_soc_write(codec, WM8904_DC_SERVO_1, dcs_mask); + + timeout = 20; + } else { + dev_dbg(codec->dev, "Calibrating DC servo\n"); + + snd_soc_write(codec, WM8904_DC_SERVO_1, + dcs_mask << WM8904_DCS_TRIG_STARTUP_0_SHIFT); + + timeout = 500; + } + + /* Wait for DC servo to complete */ + dcs_mask <<= WM8904_DCS_CAL_COMPLETE_SHIFT; + do { + val = snd_soc_read(codec, WM8904_DC_SERVO_READBACK_0); + if ((val & dcs_mask) == dcs_mask) + break; + + msleep(1); + } while (--timeout); + + if ((val & dcs_mask) != dcs_mask) + dev_warn(codec->dev, "DC servo timed out\n"); + else + dev_dbg(codec->dev, "DC servo ready\n"); + + /* Enable the output stage */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, + WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP); + + /* Unshort the output itself */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_RMV_SHORT | + WM8904_HPR_RMV_SHORT, + WM8904_HPL_RMV_SHORT | + WM8904_HPR_RMV_SHORT); + + break; + + case SND_SOC_DAPM_PRE_PMD: + /* Short the output */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_RMV_SHORT | + WM8904_HPR_RMV_SHORT, 0); + + /* Cache the DC servo configuration; this will be + * invalidated if we change the configuration. */ + wm8904->dcs_state[dcs_l] = snd_soc_read(codec, dcs_l_reg); + wm8904->dcs_state[dcs_r] = snd_soc_read(codec, dcs_r_reg); + + snd_soc_update_bits(codec, WM8904_DC_SERVO_0, + dcs_mask, 0); + + /* Disable the amplifier input and output stages */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA | WM8904_HPR_ENA | + WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY | + WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, + 0); + break; + } + + return 0; +} + +static const char *lin_text[] = { + "IN1L", "IN2L", "IN3L" +}; + +static const struct soc_enum lin_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 2, 3, lin_text); + +static const struct snd_kcontrol_new lin_mux = + SOC_DAPM_ENUM("Left Capture Mux", lin_enum); + +static const struct soc_enum lin_inv_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 4, 3, lin_text); + +static const struct snd_kcontrol_new lin_inv_mux = + SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + +static const char *rin_text[] = { + "IN1R", "IN2R", "IN3R" +}; + +static const struct soc_enum rin_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 2, 3, rin_text); + +static const struct snd_kcontrol_new rin_mux = + SOC_DAPM_ENUM("Right Capture Mux", rin_enum); + +static const struct soc_enum rin_inv_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 4, 3, rin_text); + +static const struct snd_kcontrol_new rin_inv_mux = + SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + +static const char *aif_text[] = { + "Left", "Right" +}; + +static const struct soc_enum aifoutl_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 7, 2, aif_text); + +static const struct snd_kcontrol_new aifoutl_mux = + SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum); + +static const struct soc_enum aifoutr_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 6, 2, aif_text); + +static const struct snd_kcontrol_new aifoutr_mux = + SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum); + +static const struct soc_enum aifinl_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 5, 2, aif_text); + +static const struct snd_kcontrol_new aifinl_mux = + SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum); + +static const struct soc_enum aifinr_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 4, 2, aif_text); + +static const struct snd_kcontrol_new aifinr_mux = + SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum); + +static const struct snd_soc_dapm_widget wm8904_core_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", WM8904_CLOCK_RATES_2, 2, 0, sysclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8904_CLOCK_RATES_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TOCLK", WM8904_CLOCK_RATES_2, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget wm8904_adc_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), + +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8904_MIC_BIAS_CONTROL_0, 0, 0), + +SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lin_mux), +SND_SOC_DAPM_MUX("Left Capture Inverting Mux", SND_SOC_NOPM, 0, 0, + &lin_inv_mux), +SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rin_mux), +SND_SOC_DAPM_MUX("Right Capture Inverting Mux", SND_SOC_NOPM, 0, 0, + &rin_inv_mux), + +SND_SOC_DAPM_PGA("Left Capture PGA", WM8904_POWER_MANAGEMENT_0, 1, 0, + NULL, 0), +SND_SOC_DAPM_PGA("Right Capture PGA", WM8904_POWER_MANAGEMENT_0, 0, 0, + NULL, 0), + +SND_SOC_DAPM_ADC("ADCL", NULL, WM8904_POWER_MANAGEMENT_6, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, WM8904_POWER_MANAGEMENT_6, 0, 0), + +SND_SOC_DAPM_MUX("AIFOUTL Mux", SND_SOC_NOPM, 0, 0, &aifoutl_mux), +SND_SOC_DAPM_MUX("AIFOUTR Mux", SND_SOC_NOPM, 0, 0, &aifoutr_mux), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_widget wm8904_dac_dapm_widgets[] = { +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &aifinl_mux), +SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &aifinr_mux), + +SND_SOC_DAPM_DAC("DACL", NULL, WM8904_POWER_MANAGEMENT_6, 3, 0), +SND_SOC_DAPM_DAC("DACR", NULL, WM8904_POWER_MANAGEMENT_6, 2, 0), + +SND_SOC_DAPM_SUPPLY("Charge pump", WM8904_CHARGE_PUMP_0, 0, 0, cp_event, + SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("HPL PGA", WM8904_POWER_MANAGEMENT_2, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("HPR PGA", WM8904_POWER_MANAGEMENT_2, 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA("LINEL PGA", WM8904_POWER_MANAGEMENT_3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINER PGA", WM8904_POWER_MANAGEMENT_3, 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM, WM8904_ANALOGUE_HP_0, + 0, NULL, 0, out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_E("Line Output", SND_SOC_NOPM, WM8904_ANALOGUE_LINEOUT_0, + 0, NULL, 0, out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +SND_SOC_DAPM_OUTPUT("LINEOUTL"), +SND_SOC_DAPM_OUTPUT("LINEOUTR"), +}; + +static const char *out_mux_text[] = { + "DAC", "Bypass" +}; + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 3, 2, out_mux_text); + +static const struct snd_kcontrol_new hpl_mux = + SOC_DAPM_ENUM("HPL Mux", hpl_enum); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 2, 2, out_mux_text); + +static const struct snd_kcontrol_new hpr_mux = + SOC_DAPM_ENUM("HPR Mux", hpr_enum); + +static const struct soc_enum linel_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 1, 2, out_mux_text); + +static const struct snd_kcontrol_new linel_mux = + SOC_DAPM_ENUM("LINEL Mux", linel_enum); + +static const struct soc_enum liner_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); + +static const struct snd_kcontrol_new liner_mux = + SOC_DAPM_ENUM("LINEL Mux", liner_enum); + +static const char *sidetone_text[] = { + "None", "Left", "Right" +}; + +static const struct soc_enum dacl_sidetone_enum = + SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 2, 3, sidetone_text); + +static const struct snd_kcontrol_new dacl_sidetone_mux = + SOC_DAPM_ENUM("Left Sidetone Mux", dacl_sidetone_enum); + +static const struct soc_enum dacr_sidetone_enum = + SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 0, 3, sidetone_text); + +static const struct snd_kcontrol_new dacr_sidetone_mux = + SOC_DAPM_ENUM("Right Sidetone Mux", dacr_sidetone_enum); + +static const struct snd_soc_dapm_widget wm8904_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("Class G", WM8904_CLASS_W_0, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Bypass", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Bypass", SND_SOC_NOPM, 0, 0, NULL, 0), + +SND_SOC_DAPM_MUX("Left Sidetone", SND_SOC_NOPM, 0, 0, &dacl_sidetone_mux), +SND_SOC_DAPM_MUX("Right Sidetone", SND_SOC_NOPM, 0, 0, &dacr_sidetone_mux), + +SND_SOC_DAPM_MUX("HPL Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("HPR Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("LINEL Mux", SND_SOC_NOPM, 0, 0, &linel_mux), +SND_SOC_DAPM_MUX("LINER Mux", SND_SOC_NOPM, 0, 0, &liner_mux), +}; + +static const struct snd_soc_dapm_route core_intercon[] = { + { "CLK_DSP", NULL, "SYSCLK" }, + { "TOCLK", NULL, "SYSCLK" }, +}; + +static const struct snd_soc_dapm_route adc_intercon[] = { + { "Left Capture Mux", "IN1L", "IN1L" }, + { "Left Capture Mux", "IN2L", "IN2L" }, + { "Left Capture Mux", "IN3L", "IN3L" }, + + { "Left Capture Inverting Mux", "IN1L", "IN1L" }, + { "Left Capture Inverting Mux", "IN2L", "IN2L" }, + { "Left Capture Inverting Mux", "IN3L", "IN3L" }, + + { "Right Capture Mux", "IN1R", "IN1R" }, + { "Right Capture Mux", "IN2R", "IN2R" }, + { "Right Capture Mux", "IN3R", "IN3R" }, + + { "Right Capture Inverting Mux", "IN1R", "IN1R" }, + { "Right Capture Inverting Mux", "IN2R", "IN2R" }, + { "Right Capture Inverting Mux", "IN3R", "IN3R" }, + + { "Left Capture PGA", NULL, "Left Capture Mux" }, + { "Left Capture PGA", NULL, "Left Capture Inverting Mux" }, + + { "Right Capture PGA", NULL, "Right Capture Mux" }, + { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, + + { "AIFOUTL", "Left", "ADCL" }, + { "AIFOUTL", "Right", "ADCR" }, + { "AIFOUTR", "Left", "ADCL" }, + { "AIFOUTR", "Right", "ADCR" }, + + { "ADCL", NULL, "CLK_DSP" }, + { "ADCL", NULL, "Left Capture PGA" }, + + { "ADCR", NULL, "CLK_DSP" }, + { "ADCR", NULL, "Right Capture PGA" }, +}; + +static const struct snd_soc_dapm_route dac_intercon[] = { + { "DACL", "Right", "AIFINR" }, + { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "CLK_DSP" }, + + { "DACR", "Right", "AIFINR" }, + { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "CLK_DSP" }, + + { "Charge pump", NULL, "SYSCLK" }, + + { "Headphone Output", NULL, "HPL PGA" }, + { "Headphone Output", NULL, "HPR PGA" }, + { "Headphone Output", NULL, "Charge pump" }, + { "Headphone Output", NULL, "TOCLK" }, + + { "Line Output", NULL, "LINEL PGA" }, + { "Line Output", NULL, "LINER PGA" }, + { "Line Output", NULL, "Charge pump" }, + { "Line Output", NULL, "TOCLK" }, + + { "HPOUTL", NULL, "Headphone Output" }, + { "HPOUTR", NULL, "Headphone Output" }, + + { "LINEOUTL", NULL, "Line Output" }, + { "LINEOUTR", NULL, "Line Output" }, +}; + +static const struct snd_soc_dapm_route wm8904_intercon[] = { + { "Left Sidetone", "Left", "ADCL" }, + { "Left Sidetone", "Right", "ADCR" }, + { "DACL", NULL, "Left Sidetone" }, + + { "Right Sidetone", "Left", "ADCL" }, + { "Right Sidetone", "Right", "ADCR" }, + { "DACR", NULL, "Right Sidetone" }, + + { "Left Bypass", NULL, "Class G" }, + { "Left Bypass", NULL, "Left Capture PGA" }, + + { "Right Bypass", NULL, "Class G" }, + { "Right Bypass", NULL, "Right Capture PGA" }, + + { "HPL Mux", "DAC", "DACL" }, + { "HPL Mux", "Bypass", "Left Bypass" }, + + { "HPR Mux", "DAC", "DACR" }, + { "HPR Mux", "Bypass", "Right Bypass" }, + + { "LINEL Mux", "DAC", "DACL" }, + { "LINEL Mux", "Bypass", "Left Bypass" }, + + { "LINER Mux", "DAC", "DACR" }, + { "LINER Mux", "Bypass", "Right Bypass" }, + + { "HPL PGA", NULL, "HPL Mux" }, + { "HPR PGA", NULL, "HPR Mux" }, + + { "LINEL PGA", NULL, "LINEL Mux" }, + { "LINER PGA", NULL, "LINER Mux" }, +}; + +static int wm8904_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_add_controls(codec, wm8904_adc_snd_controls, + ARRAY_SIZE(wm8904_adc_snd_controls)); + snd_soc_add_controls(codec, wm8904_dac_snd_controls, + ARRAY_SIZE(wm8904_dac_snd_controls)); + snd_soc_add_controls(codec, wm8904_snd_controls, + ARRAY_SIZE(wm8904_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets, + ARRAY_SIZE(wm8904_core_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets, + ARRAY_SIZE(wm8904_adc_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + ARRAY_SIZE(wm8904_dac_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets, + ARRAY_SIZE(wm8904_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, core_intercon, + ARRAY_SIZE(core_intercon)); + snd_soc_dapm_add_routes(codec, adc_intercon, ARRAY_SIZE(adc_intercon)); + snd_soc_dapm_add_routes(codec, dac_intercon, ARRAY_SIZE(dac_intercon)); + snd_soc_dapm_add_routes(codec, wm8904_intercon, + ARRAY_SIZE(wm8904_intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static struct { + int ratio; + unsigned int clk_sys_rate; +} clk_sys_rates[] = { + { 64, 0 }, + { 128, 1 }, + { 192, 2 }, + { 256, 3 }, + { 384, 4 }, + { 512, 5 }, + { 786, 6 }, + { 1024, 7 }, + { 1408, 8 }, + { 1536, 9 }, +}; + +static struct { + int rate; + int sample_rate; +} sample_rates[] = { + { 8000, 0 }, + { 11025, 1 }, + { 12000, 1 }, + { 16000, 2 }, + { 22050, 3 }, + { 24000, 3 }, + { 32000, 4 }, + { 44100, 5 }, + { 48000, 5 }, +}; + +static struct { + int div; /* *10 due to .5s */ + int bclk_div; +} bclk_divs[] = { + { 10, 0 }, + { 15, 1 }, + { 20, 2 }, + { 30, 3 }, + { 40, 4 }, + { 50, 5 }, + { 55, 6 }, + { 60, 7 }, + { 80, 8 }, + { 100, 9 }, + { 110, 10 }, + { 120, 11 }, + { 160, 12 }, + { 200, 13 }, + { 220, 14 }, + { 240, 16 }, + { 200, 17 }, + { 320, 18 }, + { 440, 19 }, + { 480, 20 }, +}; + + +static int wm8904_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *wm8904 = codec->private_data; + int ret, i, best, best_val, cur_val; + unsigned int aif1 = 0; + unsigned int aif2 = 0; + unsigned int aif3 = 0; + unsigned int clock1 = 0; + unsigned int dac_digital1 = 0; + + /* What BCLK do we need? */ + wm8904->fs = params_rate(params); + if (wm8904->tdm_slots) { + dev_dbg(codec->dev, "Configuring for %d %d bit TDM slots\n", + wm8904->tdm_slots, wm8904->tdm_width); + wm8904->bclk = snd_soc_calc_bclk(wm8904->fs, + wm8904->tdm_width, 2, + wm8904->tdm_slots); + } else { + wm8904->bclk = snd_soc_params_to_bclk(params); + } + + dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8904->bclk); + + ret = wm8904_configure_clocking(codec); + if (ret != 0) + return ret; + + /* Select nearest CLK_SYS_RATE */ + best = 0; + best_val = abs((wm8904->sysclk_rate / clk_sys_rates[0].ratio) + - wm8904->fs); + for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) { + cur_val = abs((wm8904->sysclk_rate / + clk_sys_rates[i].ratio) - wm8904->fs);; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n", + clk_sys_rates[best].ratio); + clock1 |= (clk_sys_rates[best].clk_sys_rate + << WM8904_CLK_SYS_RATE_SHIFT); + + /* SAMPLE_RATE */ + best = 0; + best_val = abs(wm8904->fs - sample_rates[0].rate); + for (i = 1; i < ARRAY_SIZE(sample_rates); i++) { + /* Closest match */ + cur_val = abs(wm8904->fs - sample_rates[i].rate); + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n", + sample_rates[best].rate); + clock1 |= (sample_rates[best].sample_rate + << WM8904_SAMPLE_RATE_SHIFT); + + /* Enable sloping stopband filter for low sample rates */ + if (wm8904->fs <= 24000) + dac_digital1 |= WM8904_DAC_SB_FILT; + + /* BCLK_DIV */ + best = 0; + best_val = INT_MAX; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = ((wm8904->sysclk_rate * 10) / bclk_divs[i].div) + - wm8904->bclk; + if (cur_val < 0) /* Table is sorted */ + break; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + wm8904->bclk = (wm8904->sysclk_rate * 10) / bclk_divs[best].div; + dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n", + bclk_divs[best].div, wm8904->bclk); + aif2 |= bclk_divs[best].bclk_div; + + /* LRCLK is a simple fraction of BCLK */ + dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm8904->bclk / wm8904->fs); + aif3 |= wm8904->bclk / wm8904->fs; + + /* Apply the settings */ + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1, + WM8904_DAC_SB_FILT, dac_digital1); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1, + WM8904_AIF_WL_MASK, aif1); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_2, + WM8904_BCLK_DIV_MASK, aif2); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_3, + WM8904_LRCLK_RATE_MASK, aif3); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_1, + WM8904_SAMPLE_RATE_MASK | + WM8904_CLK_SYS_RATE_MASK, clock1); + + /* Update filters for the new settings */ + wm8904_set_retune_mobile(codec); + wm8904_set_deemph(codec); + + return 0; +} + + +static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *priv = codec->private_data; + + switch (clk_id) { + case WM8904_CLK_MCLK: + priv->sysclk_src = clk_id; + priv->mclk_rate = freq; + break; + + case WM8904_CLK_FLL: + priv->sysclk_src = clk_id; + break; + + default: + return -EINVAL; + } + + dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); + + wm8904_configure_clocking(codec); + + return 0; +} + +static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int aif1 = 0; + unsigned int aif3 = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif3 |= WM8904_LRCLK_DIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif1 |= WM8904_BCLK_DIR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif1 |= WM8904_BCLK_DIR; + aif3 |= WM8904_LRCLK_DIR; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif1 |= WM8904_AIF_LRCLK_INV; + case SND_SOC_DAIFMT_DSP_A: + aif1 |= 0x3; + break; + case SND_SOC_DAIFMT_I2S: + aif1 |= 0x2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif1 |= 0x1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8904_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif1 |= WM8904_AIF_BCLK_INV | WM8904_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8904_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif1 |= WM8904_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1, + WM8904_AIF_BCLK_INV | WM8904_AIF_LRCLK_INV | + WM8904_AIF_FMT_MASK | WM8904_BCLK_DIR, aif1); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_3, + WM8904_LRCLK_DIR, aif3); + + return 0; +} + + +static int wm8904_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *wm8904 = codec->private_data; + int aif1 = 0; + + /* Don't need to validate anything if we're turning off TDM */ + if (slots == 0) + goto out; + + /* Note that we allow configurations we can't handle ourselves - + * for example, we can generate clocks for slots 2 and up even if + * we can't use those slots ourselves. + */ + aif1 |= WM8904_AIFADC_TDM | WM8904_AIFDAC_TDM; + + switch (rx_mask) { + case 3: + break; + case 0xc: + aif1 |= WM8904_AIFADC_TDM_CHAN; + break; + default: + return -EINVAL; + } + + + switch (tx_mask) { + case 3: + break; + case 0xc: + aif1 |= WM8904_AIFDAC_TDM_CHAN; + break; + default: + return -EINVAL; + } + +out: + wm8904->tdm_width = slot_width; + wm8904->tdm_slots = slots / 2; + + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1, + WM8904_AIFADC_TDM | WM8904_AIFADC_TDM_CHAN | + WM8904_AIFDAC_TDM | WM8904_AIFDAC_TDM_CHAN, aif1); + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_clk_ref_div; + u16 n; + u16 k; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + unsigned int div; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + fll_div->fll_clk_ref_div = 0; + while ((Fref / div) > 13500000) { + div *= 2; + fll_div->fll_clk_ref_div++; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + pr_debug("Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 4; + while (Fout * div < 90000000) { + div++; + if (div > 64) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + fll_div->fll_outdiv = div - 1; + + pr_debug("Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + target /= fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + /* Now, calculate N.K */ + Ndiv = target / Fref; + + fll_div->n = Ndiv; + Nmod = target % Fref; + pr_debug("Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll_div->k = K / 10; + + pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n", + fll_div->n, fll_div->k, + fll_div->fll_fratio, fll_div->fll_outdiv, + fll_div->fll_clk_ref_div); + + return 0; +} + +static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *wm8904 = codec->private_data; + struct _fll_div fll_div; + int ret, val; + int clock2, fll1; + + /* Any change? */ + if (source == wm8904->fll_src && Fref == wm8904->fll_fref && + Fout == wm8904->fll_fout) + return 0; + + if (Fout == 0) { + dev_dbg(codec->dev, "FLL disabled\n"); + + wm8904->fll_fref = 0; + wm8904->fll_fout = 0; + + /* Gate SYSCLK to avoid glitches */ + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA, 0); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + + goto out; + } + + /* Validate the FLL ID */ + switch (source) { + case WM8904_FLL_MCLK: + case WM8904_FLL_LRCLK: + case WM8904_FLL_BCLK: + ret = fll_factors(&fll_div, Fref, Fout); + if (ret != 0) + return ret; + break; + + case WM8904_FLL_FREE_RUNNING: + dev_dbg(codec->dev, "Using free running FLL\n"); + /* Force 12MHz and output/4 for now */ + Fout = 12000000; + Fref = 12000000; + + memset(&fll_div, 0, sizeof(fll_div)); + fll_div.fll_outdiv = 3; + break; + + default: + dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id); + return -EINVAL; + } + + /* Save current state then disable the FLL and SYSCLK to avoid + * misclocking */ + clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); + fll1 = snd_soc_read(codec, WM8904_FLL_CONTROL_1); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA, 0); + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + + /* Unlock forced oscilator control to switch it on/off */ + snd_soc_update_bits(codec, WM8904_CONTROL_INTERFACE_TEST_1, + WM8904_USER_KEY, WM8904_USER_KEY); + + if (fll_id == WM8904_FLL_FREE_RUNNING) { + val = WM8904_FLL_FRC_NCO; + } else { + val = 0; + } + + snd_soc_update_bits(codec, WM8904_FLL_NCO_TEST_1, WM8904_FLL_FRC_NCO, + val); + snd_soc_update_bits(codec, WM8904_CONTROL_INTERFACE_TEST_1, + WM8904_USER_KEY, 0); + + switch (fll_id) { + case WM8904_FLL_MCLK: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_SRC_MASK, 0); + break; + + case WM8904_FLL_LRCLK: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_SRC_MASK, 1); + break; + + case WM8904_FLL_BCLK: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_SRC_MASK, 2); + break; + } + + if (fll_div.k) + val = WM8904_FLL_FRACN_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_FRACN_ENA, val); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_2, + WM8904_FLL_OUTDIV_MASK | WM8904_FLL_FRATIO_MASK, + (fll_div.fll_outdiv << WM8904_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio << WM8904_FLL_FRATIO_SHIFT)); + + snd_soc_write(codec, WM8904_FLL_CONTROL_3, fll_div.k); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_4, WM8904_FLL_N_MASK, + fll_div.n << WM8904_FLL_N_SHIFT); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_DIV_MASK, + fll_div.fll_clk_ref_div + << WM8904_FLL_CLK_REF_DIV_SHIFT); + + dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); + + wm8904->fll_fref = Fref; + wm8904->fll_fout = Fout; + wm8904->fll_src = source; + + /* Enable the FLL if it was previously active */ + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA, fll1); + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_ENA, fll1); + +out: + /* Reenable SYSCLK if it was previously active */ + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA, clock2); + + return 0; +} + +static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val; + + if (mute) + val = WM8904_DAC_MUTE; + else + val = 0; + + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1, WM8904_DAC_MUTE, val); + + return 0; +} + +static int wm8904_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8904_priv *wm8904 = codec->private_data; + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID resistance 2*50k */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_RES_MASK, + 0x1 << WM8904_VMID_RES_SHIFT); + + /* Normal bias current */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_ISEL_MASK, 2 << WM8904_ISEL_SHIFT); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + /* Sync back cached values if they're + * different from the hardware default. + */ + for (i = 1; i < ARRAY_SIZE(wm8904->reg_cache); i++) { + if (!wm8904_access[i].writable) + continue; + + if (wm8904->reg_cache[i] == wm8904_reg[i]) + continue; + + snd_soc_write(codec, i, wm8904->reg_cache[i]); + } + + /* Enable bias */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_BIAS_ENA, WM8904_BIAS_ENA); + + /* Enable VMID, VMID buffering, 2*5k resistance */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_ENA | + WM8904_VMID_RES_MASK, + WM8904_VMID_ENA | + 0x3 << WM8904_VMID_RES_SHIFT); + + /* Let VMID ramp */ + msleep(1); + } + + /* Maintain VMID with 2*250k */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_RES_MASK, + 0x2 << WM8904_VMID_RES_SHIFT); + + /* Bias current *0.5 */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_ISEL_MASK, 0); + break; + + case SND_SOC_BIAS_OFF: + /* Turn off VMID */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_RES_MASK | WM8904_VMID_ENA, 0); + + /* Stop bias generation */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_BIAS_ENA, 0); + + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8904_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8904_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8904_dai_ops = { + .set_sysclk = wm8904_set_sysclk, + .set_fmt = wm8904_set_fmt, + .set_tdm_slot = wm8904_set_tdm_slot, + .set_pll = wm8904_set_fll, + .hw_params = wm8904_hw_params, + .digital_mute = wm8904_digital_mute, +}; + +struct snd_soc_dai wm8904_dai = { + .name = "WM8904", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8904_RATES, + .formats = WM8904_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8904_RATES, + .formats = WM8904_FORMATS, + }, + .ops = &wm8904_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8904_dai); + +#ifdef CONFIG_PM +static int wm8904_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8904_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8904_suspend NULL +#define wm8904_resume NULL +#endif + +static void wm8904_handle_retune_mobile_pdata(struct wm8904_priv *wm8904) +{ + struct snd_soc_codec *codec = &wm8904->codec; + struct wm8904_pdata *pdata = wm8904->pdata; + struct snd_kcontrol_new control = + SOC_ENUM_EXT("EQ Mode", + wm8904->retune_mobile_enum, + wm8904_get_retune_mobile_enum, + wm8904_put_retune_mobile_enum); + int ret, i, j; + const char **t; + + /* We need an array of texts for the enum API but the number + * of texts is likely to be less than the number of + * configurations due to the sample rate dependency of the + * configurations. */ + wm8904->num_retune_mobile_texts = 0; + wm8904->retune_mobile_texts = NULL; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + for (j = 0; j < wm8904->num_retune_mobile_texts; j++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8904->retune_mobile_texts[j]) == 0) + break; + } + + if (j != wm8904->num_retune_mobile_texts) + continue; + + /* Expand the array... */ + t = krealloc(wm8904->retune_mobile_texts, + sizeof(char *) * + (wm8904->num_retune_mobile_texts + 1), + GFP_KERNEL); + if (t == NULL) + continue; + + /* ...store the new entry... */ + t[wm8904->num_retune_mobile_texts] = + pdata->retune_mobile_cfgs[i].name; + + /* ...and remember the new version. */ + wm8904->num_retune_mobile_texts++; + wm8904->retune_mobile_texts = t; + } + + dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", + wm8904->num_retune_mobile_texts); + + wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts; + wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts; + + ret = snd_soc_add_controls(&wm8904->codec, &control, 1); + if (ret != 0) + dev_err(wm8904->codec.dev, + "Failed to add ReTune Mobile control: %d\n", ret); +} + +static void wm8904_handle_pdata(struct wm8904_priv *wm8904) +{ + struct snd_soc_codec *codec = &wm8904->codec; + struct wm8904_pdata *pdata = wm8904->pdata; + int ret, i; + + if (!pdata) { + snd_soc_add_controls(&wm8904->codec, wm8904_eq_controls, + ARRAY_SIZE(wm8904_eq_controls)); + return; + } + + dev_dbg(codec->dev, "%d DRC configurations\n", pdata->num_drc_cfgs); + + if (pdata->num_drc_cfgs) { + struct snd_kcontrol_new control = + SOC_ENUM_EXT("DRC Mode", wm8904->drc_enum, + wm8904_get_drc_enum, wm8904_put_drc_enum); + + /* We need an array of texts for the enum API */ + wm8904->drc_texts = kmalloc(sizeof(char *) + * pdata->num_drc_cfgs, GFP_KERNEL); + if (!wm8904->drc_texts) { + dev_err(wm8904->codec.dev, + "Failed to allocate %d DRC config texts\n", + pdata->num_drc_cfgs); + return; + } + + for (i = 0; i < pdata->num_drc_cfgs; i++) + wm8904->drc_texts[i] = pdata->drc_cfgs[i].name; + + wm8904->drc_enum.max = pdata->num_drc_cfgs; + wm8904->drc_enum.texts = wm8904->drc_texts; + + ret = snd_soc_add_controls(&wm8904->codec, &control, 1); + if (ret != 0) + dev_err(wm8904->codec.dev, + "Failed to add DRC mode control: %d\n", ret); + + wm8904_set_drc(codec); + } + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", + pdata->num_retune_mobile_cfgs); + + if (pdata->num_retune_mobile_cfgs) + wm8904_handle_retune_mobile_pdata(wm8904); + else + snd_soc_add_controls(&wm8904->codec, wm8904_eq_controls, + ARRAY_SIZE(wm8904_eq_controls)); +} + +static int wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8904_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8904_codec; + codec = wm8904_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + wm8904_handle_pdata(codec->private_data); + + wm8904_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +static int wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8904 = { + .probe = wm8904_probe, + .remove = wm8904_remove, + .suspend = wm8904_suspend, + .resume = wm8904_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8904); + +static int wm8904_register(struct wm8904_priv *wm8904, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8904->codec; + int i; + + if (wm8904_codec) { + dev_err(codec->dev, "Another WM8904 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8904; + codec->name = "WM8904"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8904_set_bias_level; + codec->dai = &wm8904_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8904_MAX_REGISTER; + codec->reg_cache = &wm8904->reg_cache; + codec->volatile_register = wm8904_volatile_register; + + memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) + wm8904->supplies[i].supply = wm8904_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read ID register\n"); + goto err_enable; + } + if (ret != wm8904_reg[WM8904_SW_RESET_AND_ID]) { + dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret); + ret = -EINVAL; + goto err_enable; + } + + ret = snd_soc_read(codec, WM8904_REVISION); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_enable; + } + dev_info(codec->dev, "revision %c\n", ret + 'A'); + + ret = wm8904_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_enable; + } + + wm8904_dai.dev = codec->dev; + + /* Change some default settings - latch VU and enable ZC */ + wm8904->reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU; + wm8904->reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU; + wm8904->reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU; + wm8904->reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU; + wm8904->reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU | + WM8904_HPOUTLZC; + wm8904->reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU | + WM8904_HPOUTRZC; + wm8904->reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU | + WM8904_LINEOUTLZC; + wm8904->reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU | + WM8904_LINEOUTRZC; + wm8904->reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + + /* Set Class W by default - this will be managed by the Class + * G widget at runtime where bypass paths are available. + */ + wm8904->reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR; + + /* Use normal bias source */ + wm8904->reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL; + + wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + + wm8904_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8904_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); +err: + kfree(wm8904); + return ret; +} + +static void wm8904_unregister(struct wm8904_priv *wm8904) +{ + wm8904_set_bias_level(&wm8904->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + snd_soc_unregister_dai(&wm8904_dai); + snd_soc_unregister_codec(&wm8904->codec); + kfree(wm8904); + wm8904_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8904_priv *wm8904; + struct snd_soc_codec *codec; + + wm8904 = kzalloc(sizeof(struct wm8904_priv), GFP_KERNEL); + if (wm8904 == NULL) + return -ENOMEM; + + codec = &wm8904->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8904); + codec->control_data = i2c; + wm8904->pdata = i2c->dev.platform_data; + + codec->dev = &i2c->dev; + + return wm8904_register(wm8904, SND_SOC_I2C); +} + +static __devexit int wm8904_i2c_remove(struct i2c_client *client) +{ + struct wm8904_priv *wm8904 = i2c_get_clientdata(client); + wm8904_unregister(wm8904); + return 0; +} + +static const struct i2c_device_id wm8904_i2c_id[] = { + { "wm8904", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); + +static struct i2c_driver wm8904_i2c_driver = { + .driver = { + .name = "WM8904", + .owner = THIS_MODULE, + }, + .probe = wm8904_i2c_probe, + .remove = __devexit_p(wm8904_i2c_remove), + .id_table = wm8904_i2c_id, +}; +#endif + +static int __init wm8904_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8904_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8904 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8904_modinit); + +static void __exit wm8904_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8904_i2c_driver); +#endif +} +module_exit(wm8904_exit); + +MODULE_DESCRIPTION("ASoC WM8904 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h new file mode 100644 index 000000000000..b68886df34e4 --- /dev/null +++ b/sound/soc/codecs/wm8904.h @@ -0,0 +1,1681 @@ +/* + * wm8904.h -- WM8904 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics, plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8904_H +#define _WM8904_H + +#define WM8904_CLK_MCLK 1 +#define WM8904_CLK_FLL 2 + +#define WM8904_FLL_MCLK 1 +#define WM8904_FLL_BCLK 2 +#define WM8904_FLL_LRCLK 3 +#define WM8904_FLL_FREE_RUNNING 4 + +extern struct snd_soc_dai wm8904_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8904; + +/* + * Register values. + */ +#define WM8904_SW_RESET_AND_ID 0x00 +#define WM8904_REVISION 0x01 +#define WM8904_BIAS_CONTROL_0 0x04 +#define WM8904_VMID_CONTROL_0 0x05 +#define WM8904_MIC_BIAS_CONTROL_0 0x06 +#define WM8904_MIC_BIAS_CONTROL_1 0x07 +#define WM8904_ANALOGUE_DAC_0 0x08 +#define WM8904_MIC_FILTER_CONTROL 0x09 +#define WM8904_ANALOGUE_ADC_0 0x0A +#define WM8904_POWER_MANAGEMENT_0 0x0C +#define WM8904_POWER_MANAGEMENT_2 0x0E +#define WM8904_POWER_MANAGEMENT_3 0x0F +#define WM8904_POWER_MANAGEMENT_6 0x12 +#define WM8904_CLOCK_RATES_0 0x14 +#define WM8904_CLOCK_RATES_1 0x15 +#define WM8904_CLOCK_RATES_2 0x16 +#define WM8904_AUDIO_INTERFACE_0 0x18 +#define WM8904_AUDIO_INTERFACE_1 0x19 +#define WM8904_AUDIO_INTERFACE_2 0x1A +#define WM8904_AUDIO_INTERFACE_3 0x1B +#define WM8904_DAC_DIGITAL_VOLUME_LEFT 0x1E +#define WM8904_DAC_DIGITAL_VOLUME_RIGHT 0x1F +#define WM8904_DAC_DIGITAL_0 0x20 +#define WM8904_DAC_DIGITAL_1 0x21 +#define WM8904_ADC_DIGITAL_VOLUME_LEFT 0x24 +#define WM8904_ADC_DIGITAL_VOLUME_RIGHT 0x25 +#define WM8904_ADC_DIGITAL_0 0x26 +#define WM8904_DIGITAL_MICROPHONE_0 0x27 +#define WM8904_DRC_0 0x28 +#define WM8904_DRC_1 0x29 +#define WM8904_DRC_2 0x2A +#define WM8904_DRC_3 0x2B +#define WM8904_ANALOGUE_LEFT_INPUT_0 0x2C +#define WM8904_ANALOGUE_RIGHT_INPUT_0 0x2D +#define WM8904_ANALOGUE_LEFT_INPUT_1 0x2E +#define WM8904_ANALOGUE_RIGHT_INPUT_1 0x2F +#define WM8904_ANALOGUE_OUT1_LEFT 0x39 +#define WM8904_ANALOGUE_OUT1_RIGHT 0x3A +#define WM8904_ANALOGUE_OUT2_LEFT 0x3B +#define WM8904_ANALOGUE_OUT2_RIGHT 0x3C +#define WM8904_ANALOGUE_OUT12_ZC 0x3D +#define WM8904_DC_SERVO_0 0x43 +#define WM8904_DC_SERVO_1 0x44 +#define WM8904_DC_SERVO_2 0x45 +#define WM8904_DC_SERVO_4 0x47 +#define WM8904_DC_SERVO_5 0x48 +#define WM8904_DC_SERVO_6 0x49 +#define WM8904_DC_SERVO_7 0x4A +#define WM8904_DC_SERVO_8 0x4B +#define WM8904_DC_SERVO_9 0x4C +#define WM8904_DC_SERVO_READBACK_0 0x4D +#define WM8904_ANALOGUE_HP_0 0x5A +#define WM8904_ANALOGUE_LINEOUT_0 0x5E +#define WM8904_CHARGE_PUMP_0 0x62 +#define WM8904_CLASS_W_0 0x68 +#define WM8904_WRITE_SEQUENCER_0 0x6C +#define WM8904_WRITE_SEQUENCER_1 0x6D +#define WM8904_WRITE_SEQUENCER_2 0x6E +#define WM8904_WRITE_SEQUENCER_3 0x6F +#define WM8904_WRITE_SEQUENCER_4 0x70 +#define WM8904_FLL_CONTROL_1 0x74 +#define WM8904_FLL_CONTROL_2 0x75 +#define WM8904_FLL_CONTROL_3 0x76 +#define WM8904_FLL_CONTROL_4 0x77 +#define WM8904_FLL_CONTROL_5 0x78 +#define WM8904_GPIO_CONTROL_1 0x79 +#define WM8904_GPIO_CONTROL_2 0x7A +#define WM8904_GPIO_CONTROL_3 0x7B +#define WM8904_GPIO_CONTROL_4 0x7C +#define WM8904_DIGITAL_PULLS 0x7E +#define WM8904_INTERRUPT_STATUS 0x7F +#define WM8904_INTERRUPT_STATUS_MASK 0x80 +#define WM8904_INTERRUPT_POLARITY 0x81 +#define WM8904_INTERRUPT_DEBOUNCE 0x82 +#define WM8904_EQ1 0x86 +#define WM8904_EQ2 0x87 +#define WM8904_EQ3 0x88 +#define WM8904_EQ4 0x89 +#define WM8904_EQ5 0x8A +#define WM8904_EQ6 0x8B +#define WM8904_EQ7 0x8C +#define WM8904_EQ8 0x8D +#define WM8904_EQ9 0x8E +#define WM8904_EQ10 0x8F +#define WM8904_EQ11 0x90 +#define WM8904_EQ12 0x91 +#define WM8904_EQ13 0x92 +#define WM8904_EQ14 0x93 +#define WM8904_EQ15 0x94 +#define WM8904_EQ16 0x95 +#define WM8904_EQ17 0x96 +#define WM8904_EQ18 0x97 +#define WM8904_EQ19 0x98 +#define WM8904_EQ20 0x99 +#define WM8904_EQ21 0x9A +#define WM8904_EQ22 0x9B +#define WM8904_EQ23 0x9C +#define WM8904_EQ24 0x9D +#define WM8904_CONTROL_INTERFACE_TEST_1 0xA1 +#define WM8904_ANALOGUE_OUTPUT_BIAS_0 0xCC +#define WM8904_FLL_NCO_TEST_0 0xF7 +#define WM8904_FLL_NCO_TEST_1 0xF8 + +#define WM8904_REGISTER_COUNT 101 +#define WM8904_MAX_REGISTER 0xF8 + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - SW Reset and ID + */ +#define WM8904_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8904_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8904_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R1 (0x01) - Revision + */ +#define WM8904_REVISION_MASK 0x000F /* REVISION - [3:0] */ +#define WM8904_REVISION_SHIFT 0 /* REVISION - [3:0] */ +#define WM8904_REVISION_WIDTH 16 /* REVISION - [3:0] */ + +/* + * R4 (0x04) - Bias Control 0 + */ +#define WM8904_POBCTRL 0x0010 /* POBCTRL */ +#define WM8904_POBCTRL_MASK 0x0010 /* POBCTRL */ +#define WM8904_POBCTRL_SHIFT 4 /* POBCTRL */ +#define WM8904_POBCTRL_WIDTH 1 /* POBCTRL */ +#define WM8904_ISEL_MASK 0x000C /* ISEL - [3:2] */ +#define WM8904_ISEL_SHIFT 2 /* ISEL - [3:2] */ +#define WM8904_ISEL_WIDTH 2 /* ISEL - [3:2] */ +#define WM8904_STARTUP_BIAS_ENA 0x0002 /* STARTUP_BIAS_ENA */ +#define WM8904_STARTUP_BIAS_ENA_MASK 0x0002 /* STARTUP_BIAS_ENA */ +#define WM8904_STARTUP_BIAS_ENA_SHIFT 1 /* STARTUP_BIAS_ENA */ +#define WM8904_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */ +#define WM8904_BIAS_ENA 0x0001 /* BIAS_ENA */ +#define WM8904_BIAS_ENA_MASK 0x0001 /* BIAS_ENA */ +#define WM8904_BIAS_ENA_SHIFT 0 /* BIAS_ENA */ +#define WM8904_BIAS_ENA_WIDTH 1 /* BIAS_ENA */ + +/* + * R5 (0x05) - VMID Control 0 + */ +#define WM8904_VMID_BUF_ENA 0x0040 /* VMID_BUF_ENA */ +#define WM8904_VMID_BUF_ENA_MASK 0x0040 /* VMID_BUF_ENA */ +#define WM8904_VMID_BUF_ENA_SHIFT 6 /* VMID_BUF_ENA */ +#define WM8904_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */ +#define WM8904_VMID_RES_MASK 0x0006 /* VMID_RES - [2:1] */ +#define WM8904_VMID_RES_SHIFT 1 /* VMID_RES - [2:1] */ +#define WM8904_VMID_RES_WIDTH 2 /* VMID_RES - [2:1] */ +#define WM8904_VMID_ENA 0x0001 /* VMID_ENA */ +#define WM8904_VMID_ENA_MASK 0x0001 /* VMID_ENA */ +#define WM8904_VMID_ENA_SHIFT 0 /* VMID_ENA */ +#define WM8904_VMID_ENA_WIDTH 1 /* VMID_ENA */ + +/* + * R6 (0x06) - Mic Bias Control 0 + */ +#define WM8904_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */ +#define WM8904_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */ +#define WM8904_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */ +#define WM8904_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */ +#define WM8904_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */ +#define WM8904_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */ +#define WM8904_MICDET_ENA 0x0002 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_SHIFT 1 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_WIDTH 1 /* MICDET_ENA */ +#define WM8904_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ + +/* + * R7 (0x07) - Mic Bias Control 1 + */ +#define WM8904_MIC_DET_FILTER_ENA 0x8000 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_MASK 0x8000 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_SHIFT 15 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_WIDTH 1 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA 0x4000 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_MASK 0x4000 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_SHIFT 14 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_WIDTH 1 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MICBIAS_SEL_MASK 0x0007 /* MICBIAS_SEL - [2:0] */ +#define WM8904_MICBIAS_SEL_SHIFT 0 /* MICBIAS_SEL - [2:0] */ +#define WM8904_MICBIAS_SEL_WIDTH 3 /* MICBIAS_SEL - [2:0] */ + +/* + * R8 (0x08) - Analogue DAC 0 + */ +#define WM8904_DAC_BIAS_SEL_MASK 0x0018 /* DAC_BIAS_SEL - [4:3] */ +#define WM8904_DAC_BIAS_SEL_SHIFT 3 /* DAC_BIAS_SEL - [4:3] */ +#define WM8904_DAC_BIAS_SEL_WIDTH 2 /* DAC_BIAS_SEL - [4:3] */ +#define WM8904_DAC_VMID_BIAS_SEL_MASK 0x0006 /* DAC_VMID_BIAS_SEL - [2:1] */ +#define WM8904_DAC_VMID_BIAS_SEL_SHIFT 1 /* DAC_VMID_BIAS_SEL - [2:1] */ +#define WM8904_DAC_VMID_BIAS_SEL_WIDTH 2 /* DAC_VMID_BIAS_SEL - [2:1] */ + +/* + * R9 (0x09) - mic Filter Control + */ +#define WM8904_MIC_DET_SET_THRESHOLD_MASK 0xF000 /* MIC_DET_SET_THRESHOLD - [15:12] */ +#define WM8904_MIC_DET_SET_THRESHOLD_SHIFT 12 /* MIC_DET_SET_THRESHOLD - [15:12] */ +#define WM8904_MIC_DET_SET_THRESHOLD_WIDTH 4 /* MIC_DET_SET_THRESHOLD - [15:12] */ +#define WM8904_MIC_DET_RESET_THRESHOLD_MASK 0x0F00 /* MIC_DET_RESET_THRESHOLD - [11:8] */ +#define WM8904_MIC_DET_RESET_THRESHOLD_SHIFT 8 /* MIC_DET_RESET_THRESHOLD - [11:8] */ +#define WM8904_MIC_DET_RESET_THRESHOLD_WIDTH 4 /* MIC_DET_RESET_THRESHOLD - [11:8] */ +#define WM8904_MIC_SHORT_SET_THRESHOLD_MASK 0x00F0 /* MIC_SHORT_SET_THRESHOLD - [7:4] */ +#define WM8904_MIC_SHORT_SET_THRESHOLD_SHIFT 4 /* MIC_SHORT_SET_THRESHOLD - [7:4] */ +#define WM8904_MIC_SHORT_SET_THRESHOLD_WIDTH 4 /* MIC_SHORT_SET_THRESHOLD - [7:4] */ +#define WM8904_MIC_SHORT_RESET_THRESHOLD_MASK 0x000F /* MIC_SHORT_RESET_THRESHOLD - [3:0] */ +#define WM8904_MIC_SHORT_RESET_THRESHOLD_SHIFT 0 /* MIC_SHORT_RESET_THRESHOLD - [3:0] */ +#define WM8904_MIC_SHORT_RESET_THRESHOLD_WIDTH 4 /* MIC_SHORT_RESET_THRESHOLD - [3:0] */ + +/* + * R10 (0x0A) - Analogue ADC 0 + */ +#define WM8904_ADC_OSR128 0x0001 /* ADC_OSR128 */ +#define WM8904_ADC_OSR128_MASK 0x0001 /* ADC_OSR128 */ +#define WM8904_ADC_OSR128_SHIFT 0 /* ADC_OSR128 */ +#define WM8904_ADC_OSR128_WIDTH 1 /* ADC_OSR128 */ + +/* + * R12 (0x0C) - Power Management 0 + */ +#define WM8904_INL_ENA 0x0002 /* INL_ENA */ +#define WM8904_INL_ENA_MASK 0x0002 /* INL_ENA */ +#define WM8904_INL_ENA_SHIFT 1 /* INL_ENA */ +#define WM8904_INL_ENA_WIDTH 1 /* INL_ENA */ +#define WM8904_INR_ENA 0x0001 /* INR_ENA */ +#define WM8904_INR_ENA_MASK 0x0001 /* INR_ENA */ +#define WM8904_INR_ENA_SHIFT 0 /* INR_ENA */ +#define WM8904_INR_ENA_WIDTH 1 /* INR_ENA */ + +/* + * R14 (0x0E) - Power Management 2 + */ +#define WM8904_HPL_PGA_ENA 0x0002 /* HPL_PGA_ENA */ +#define WM8904_HPL_PGA_ENA_MASK 0x0002 /* HPL_PGA_ENA */ +#define WM8904_HPL_PGA_ENA_SHIFT 1 /* HPL_PGA_ENA */ +#define WM8904_HPL_PGA_ENA_WIDTH 1 /* HPL_PGA_ENA */ +#define WM8904_HPR_PGA_ENA 0x0001 /* HPR_PGA_ENA */ +#define WM8904_HPR_PGA_ENA_MASK 0x0001 /* HPR_PGA_ENA */ +#define WM8904_HPR_PGA_ENA_SHIFT 0 /* HPR_PGA_ENA */ +#define WM8904_HPR_PGA_ENA_WIDTH 1 /* HPR_PGA_ENA */ + +/* + * R15 (0x0F) - Power Management 3 + */ +#define WM8904_LINEOUTL_PGA_ENA 0x0002 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTL_PGA_ENA_MASK 0x0002 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTL_PGA_ENA_SHIFT 1 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTL_PGA_ENA_WIDTH 1 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA 0x0001 /* LINEOUTR_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA_MASK 0x0001 /* LINEOUTR_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA_SHIFT 0 /* LINEOUTR_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA_WIDTH 1 /* LINEOUTR_PGA_ENA */ + +/* + * R18 (0x12) - Power Management 6 + */ +#define WM8904_DACL_ENA 0x0008 /* DACL_ENA */ +#define WM8904_DACL_ENA_MASK 0x0008 /* DACL_ENA */ +#define WM8904_DACL_ENA_SHIFT 3 /* DACL_ENA */ +#define WM8904_DACL_ENA_WIDTH 1 /* DACL_ENA */ +#define WM8904_DACR_ENA 0x0004 /* DACR_ENA */ +#define WM8904_DACR_ENA_MASK 0x0004 /* DACR_ENA */ +#define WM8904_DACR_ENA_SHIFT 2 /* DACR_ENA */ +#define WM8904_DACR_ENA_WIDTH 1 /* DACR_ENA */ +#define WM8904_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8904_ADCL_ENA_MASK 0x0002 /* ADCL_ENA */ +#define WM8904_ADCL_ENA_SHIFT 1 /* ADCL_ENA */ +#define WM8904_ADCL_ENA_WIDTH 1 /* ADCL_ENA */ +#define WM8904_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8904_ADCR_ENA_MASK 0x0001 /* ADCR_ENA */ +#define WM8904_ADCR_ENA_SHIFT 0 /* ADCR_ENA */ +#define WM8904_ADCR_ENA_WIDTH 1 /* ADCR_ENA */ + +/* + * R20 (0x14) - Clock Rates 0 + */ +#define WM8904_TOCLK_RATE_DIV16 0x4000 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_DIV16_MASK 0x4000 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_DIV16_SHIFT 14 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_DIV16_WIDTH 1 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_X4 0x2000 /* TOCLK_RATE_X4 */ +#define WM8904_TOCLK_RATE_X4_MASK 0x2000 /* TOCLK_RATE_X4 */ +#define WM8904_TOCLK_RATE_X4_SHIFT 13 /* TOCLK_RATE_X4 */ +#define WM8904_TOCLK_RATE_X4_WIDTH 1 /* TOCLK_RATE_X4 */ +#define WM8904_SR_MODE 0x1000 /* SR_MODE */ +#define WM8904_SR_MODE_MASK 0x1000 /* SR_MODE */ +#define WM8904_SR_MODE_SHIFT 12 /* SR_MODE */ +#define WM8904_SR_MODE_WIDTH 1 /* SR_MODE */ +#define WM8904_MCLK_DIV 0x0001 /* MCLK_DIV */ +#define WM8904_MCLK_DIV_MASK 0x0001 /* MCLK_DIV */ +#define WM8904_MCLK_DIV_SHIFT 0 /* MCLK_DIV */ +#define WM8904_MCLK_DIV_WIDTH 1 /* MCLK_DIV */ + +/* + * R21 (0x15) - Clock Rates 1 + */ +#define WM8904_CLK_SYS_RATE_MASK 0x3C00 /* CLK_SYS_RATE - [13:10] */ +#define WM8904_CLK_SYS_RATE_SHIFT 10 /* CLK_SYS_RATE - [13:10] */ +#define WM8904_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [13:10] */ +#define WM8904_SAMPLE_RATE_MASK 0x0007 /* SAMPLE_RATE - [2:0] */ +#define WM8904_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [2:0] */ +#define WM8904_SAMPLE_RATE_WIDTH 3 /* SAMPLE_RATE - [2:0] */ + +/* + * R22 (0x16) - Clock Rates 2 + */ +#define WM8904_MCLK_INV 0x8000 /* MCLK_INV */ +#define WM8904_MCLK_INV_MASK 0x8000 /* MCLK_INV */ +#define WM8904_MCLK_INV_SHIFT 15 /* MCLK_INV */ +#define WM8904_MCLK_INV_WIDTH 1 /* MCLK_INV */ +#define WM8904_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8904_SYSCLK_SRC_MASK 0x4000 /* SYSCLK_SRC */ +#define WM8904_SYSCLK_SRC_SHIFT 14 /* SYSCLK_SRC */ +#define WM8904_SYSCLK_SRC_WIDTH 1 /* SYSCLK_SRC */ +#define WM8904_TOCLK_RATE 0x1000 /* TOCLK_RATE */ +#define WM8904_TOCLK_RATE_MASK 0x1000 /* TOCLK_RATE */ +#define WM8904_TOCLK_RATE_SHIFT 12 /* TOCLK_RATE */ +#define WM8904_TOCLK_RATE_WIDTH 1 /* TOCLK_RATE */ +#define WM8904_OPCLK_ENA 0x0008 /* OPCLK_ENA */ +#define WM8904_OPCLK_ENA_MASK 0x0008 /* OPCLK_ENA */ +#define WM8904_OPCLK_ENA_SHIFT 3 /* OPCLK_ENA */ +#define WM8904_OPCLK_ENA_WIDTH 1 /* OPCLK_ENA */ +#define WM8904_CLK_SYS_ENA 0x0004 /* CLK_SYS_ENA */ +#define WM8904_CLK_SYS_ENA_MASK 0x0004 /* CLK_SYS_ENA */ +#define WM8904_CLK_SYS_ENA_SHIFT 2 /* CLK_SYS_ENA */ +#define WM8904_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */ +#define WM8904_CLK_DSP_ENA 0x0002 /* CLK_DSP_ENA */ +#define WM8904_CLK_DSP_ENA_MASK 0x0002 /* CLK_DSP_ENA */ +#define WM8904_CLK_DSP_ENA_SHIFT 1 /* CLK_DSP_ENA */ +#define WM8904_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */ +#define WM8904_TOCLK_ENA 0x0001 /* TOCLK_ENA */ +#define WM8904_TOCLK_ENA_MASK 0x0001 /* TOCLK_ENA */ +#define WM8904_TOCLK_ENA_SHIFT 0 /* TOCLK_ENA */ +#define WM8904_TOCLK_ENA_WIDTH 1 /* TOCLK_ENA */ + +/* + * R24 (0x18) - Audio Interface 0 + */ +#define WM8904_DACL_DATINV 0x1000 /* DACL_DATINV */ +#define WM8904_DACL_DATINV_MASK 0x1000 /* DACL_DATINV */ +#define WM8904_DACL_DATINV_SHIFT 12 /* DACL_DATINV */ +#define WM8904_DACL_DATINV_WIDTH 1 /* DACL_DATINV */ +#define WM8904_DACR_DATINV 0x0800 /* DACR_DATINV */ +#define WM8904_DACR_DATINV_MASK 0x0800 /* DACR_DATINV */ +#define WM8904_DACR_DATINV_SHIFT 11 /* DACR_DATINV */ +#define WM8904_DACR_DATINV_WIDTH 1 /* DACR_DATINV */ +#define WM8904_DAC_BOOST_MASK 0x0600 /* DAC_BOOST - [10:9] */ +#define WM8904_DAC_BOOST_SHIFT 9 /* DAC_BOOST - [10:9] */ +#define WM8904_DAC_BOOST_WIDTH 2 /* DAC_BOOST - [10:9] */ +#define WM8904_LOOPBACK 0x0100 /* LOOPBACK */ +#define WM8904_LOOPBACK_MASK 0x0100 /* LOOPBACK */ +#define WM8904_LOOPBACK_SHIFT 8 /* LOOPBACK */ +#define WM8904_LOOPBACK_WIDTH 1 /* LOOPBACK */ +#define WM8904_AIFADCL_SRC 0x0080 /* AIFADCL_SRC */ +#define WM8904_AIFADCL_SRC_MASK 0x0080 /* AIFADCL_SRC */ +#define WM8904_AIFADCL_SRC_SHIFT 7 /* AIFADCL_SRC */ +#define WM8904_AIFADCL_SRC_WIDTH 1 /* AIFADCL_SRC */ +#define WM8904_AIFADCR_SRC 0x0040 /* AIFADCR_SRC */ +#define WM8904_AIFADCR_SRC_MASK 0x0040 /* AIFADCR_SRC */ +#define WM8904_AIFADCR_SRC_SHIFT 6 /* AIFADCR_SRC */ +#define WM8904_AIFADCR_SRC_WIDTH 1 /* AIFADCR_SRC */ +#define WM8904_AIFDACL_SRC 0x0020 /* AIFDACL_SRC */ +#define WM8904_AIFDACL_SRC_MASK 0x0020 /* AIFDACL_SRC */ +#define WM8904_AIFDACL_SRC_SHIFT 5 /* AIFDACL_SRC */ +#define WM8904_AIFDACL_SRC_WIDTH 1 /* AIFDACL_SRC */ +#define WM8904_AIFDACR_SRC 0x0010 /* AIFDACR_SRC */ +#define WM8904_AIFDACR_SRC_MASK 0x0010 /* AIFDACR_SRC */ +#define WM8904_AIFDACR_SRC_SHIFT 4 /* AIFDACR_SRC */ +#define WM8904_AIFDACR_SRC_WIDTH 1 /* AIFDACR_SRC */ +#define WM8904_ADC_COMP 0x0008 /* ADC_COMP */ +#define WM8904_ADC_COMP_MASK 0x0008 /* ADC_COMP */ +#define WM8904_ADC_COMP_SHIFT 3 /* ADC_COMP */ +#define WM8904_ADC_COMP_WIDTH 1 /* ADC_COMP */ +#define WM8904_ADC_COMPMODE 0x0004 /* ADC_COMPMODE */ +#define WM8904_ADC_COMPMODE_MASK 0x0004 /* ADC_COMPMODE */ +#define WM8904_ADC_COMPMODE_SHIFT 2 /* ADC_COMPMODE */ +#define WM8904_ADC_COMPMODE_WIDTH 1 /* ADC_COMPMODE */ +#define WM8904_DAC_COMP 0x0002 /* DAC_COMP */ +#define WM8904_DAC_COMP_MASK 0x0002 /* DAC_COMP */ +#define WM8904_DAC_COMP_SHIFT 1 /* DAC_COMP */ +#define WM8904_DAC_COMP_WIDTH 1 /* DAC_COMP */ +#define WM8904_DAC_COMPMODE 0x0001 /* DAC_COMPMODE */ +#define WM8904_DAC_COMPMODE_MASK 0x0001 /* DAC_COMPMODE */ +#define WM8904_DAC_COMPMODE_SHIFT 0 /* DAC_COMPMODE */ +#define WM8904_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */ + +/* + * R25 (0x19) - Audio Interface 1 + */ +#define WM8904_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_MASK 0x2000 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_SHIFT 13 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_WIDTH 1 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFDAC_TDM_CHAN_MASK 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFDAC_TDM_CHAN_SHIFT 12 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFDAC_TDM_CHAN_WIDTH 1 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFADC_TDM 0x0800 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_MASK 0x0800 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_SHIFT 11 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_WIDTH 1 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_CHAN 0x0400 /* AIFADC_TDM_CHAN */ +#define WM8904_AIFADC_TDM_CHAN_MASK 0x0400 /* AIFADC_TDM_CHAN */ +#define WM8904_AIFADC_TDM_CHAN_SHIFT 10 /* AIFADC_TDM_CHAN */ +#define WM8904_AIFADC_TDM_CHAN_WIDTH 1 /* AIFADC_TDM_CHAN */ +#define WM8904_AIF_TRIS 0x0100 /* AIF_TRIS */ +#define WM8904_AIF_TRIS_MASK 0x0100 /* AIF_TRIS */ +#define WM8904_AIF_TRIS_SHIFT 8 /* AIF_TRIS */ +#define WM8904_AIF_TRIS_WIDTH 1 /* AIF_TRIS */ +#define WM8904_AIF_BCLK_INV 0x0080 /* AIF_BCLK_INV */ +#define WM8904_AIF_BCLK_INV_MASK 0x0080 /* AIF_BCLK_INV */ +#define WM8904_AIF_BCLK_INV_SHIFT 7 /* AIF_BCLK_INV */ +#define WM8904_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */ +#define WM8904_BCLK_DIR 0x0040 /* BCLK_DIR */ +#define WM8904_BCLK_DIR_MASK 0x0040 /* BCLK_DIR */ +#define WM8904_BCLK_DIR_SHIFT 6 /* BCLK_DIR */ +#define WM8904_BCLK_DIR_WIDTH 1 /* BCLK_DIR */ +#define WM8904_AIF_LRCLK_INV 0x0010 /* AIF_LRCLK_INV */ +#define WM8904_AIF_LRCLK_INV_MASK 0x0010 /* AIF_LRCLK_INV */ +#define WM8904_AIF_LRCLK_INV_SHIFT 4 /* AIF_LRCLK_INV */ +#define WM8904_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */ +#define WM8904_AIF_WL_MASK 0x000C /* AIF_WL - [3:2] */ +#define WM8904_AIF_WL_SHIFT 2 /* AIF_WL - [3:2] */ +#define WM8904_AIF_WL_WIDTH 2 /* AIF_WL - [3:2] */ +#define WM8904_AIF_FMT_MASK 0x0003 /* AIF_FMT - [1:0] */ +#define WM8904_AIF_FMT_SHIFT 0 /* AIF_FMT - [1:0] */ +#define WM8904_AIF_FMT_WIDTH 2 /* AIF_FMT - [1:0] */ + +/* + * R26 (0x1A) - Audio Interface 2 + */ +#define WM8904_OPCLK_DIV_MASK 0x0F00 /* OPCLK_DIV - [11:8] */ +#define WM8904_OPCLK_DIV_SHIFT 8 /* OPCLK_DIV - [11:8] */ +#define WM8904_OPCLK_DIV_WIDTH 4 /* OPCLK_DIV - [11:8] */ +#define WM8904_BCLK_DIV_MASK 0x001F /* BCLK_DIV - [4:0] */ +#define WM8904_BCLK_DIV_SHIFT 0 /* BCLK_DIV - [4:0] */ +#define WM8904_BCLK_DIV_WIDTH 5 /* BCLK_DIV - [4:0] */ + +/* + * R27 (0x1B) - Audio Interface 3 + */ +#define WM8904_LRCLK_DIR 0x0800 /* LRCLK_DIR */ +#define WM8904_LRCLK_DIR_MASK 0x0800 /* LRCLK_DIR */ +#define WM8904_LRCLK_DIR_SHIFT 11 /* LRCLK_DIR */ +#define WM8904_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */ +#define WM8904_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */ +#define WM8904_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */ +#define WM8904_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */ + +/* + * R30 (0x1E) - DAC Digital Volume Left + */ +#define WM8904_DAC_VU 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_MASK 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_SHIFT 8 /* DAC_VU */ +#define WM8904_DAC_VU_WIDTH 1 /* DAC_VU */ +#define WM8904_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8904_DACL_VOL_SHIFT 0 /* DACL_VOL - [7:0] */ +#define WM8904_DACL_VOL_WIDTH 8 /* DACL_VOL - [7:0] */ + +/* + * R31 (0x1F) - DAC Digital Volume Right + */ +#define WM8904_DAC_VU 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_MASK 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_SHIFT 8 /* DAC_VU */ +#define WM8904_DAC_VU_WIDTH 1 /* DAC_VU */ +#define WM8904_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8904_DACR_VOL_SHIFT 0 /* DACR_VOL - [7:0] */ +#define WM8904_DACR_VOL_WIDTH 8 /* DACR_VOL - [7:0] */ + +/* + * R32 (0x20) - DAC Digital 0 + */ +#define WM8904_ADCL_DAC_SVOL_MASK 0x0F00 /* ADCL_DAC_SVOL - [11:8] */ +#define WM8904_ADCL_DAC_SVOL_SHIFT 8 /* ADCL_DAC_SVOL - [11:8] */ +#define WM8904_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [11:8] */ +#define WM8904_ADCR_DAC_SVOL_MASK 0x00F0 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8904_ADCR_DAC_SVOL_SHIFT 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8904_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8904_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */ +#define WM8904_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */ +#define WM8904_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */ +#define WM8904_ADC_TO_DACR_MASK 0x0003 /* ADC_TO_DACR - [1:0] */ +#define WM8904_ADC_TO_DACR_SHIFT 0 /* ADC_TO_DACR - [1:0] */ +#define WM8904_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [1:0] */ + +/* + * R33 (0x21) - DAC Digital 1 + */ +#define WM8904_DAC_MONO 0x1000 /* DAC_MONO */ +#define WM8904_DAC_MONO_MASK 0x1000 /* DAC_MONO */ +#define WM8904_DAC_MONO_SHIFT 12 /* DAC_MONO */ +#define WM8904_DAC_MONO_WIDTH 1 /* DAC_MONO */ +#define WM8904_DAC_SB_FILT 0x0800 /* DAC_SB_FILT */ +#define WM8904_DAC_SB_FILT_MASK 0x0800 /* DAC_SB_FILT */ +#define WM8904_DAC_SB_FILT_SHIFT 11 /* DAC_SB_FILT */ +#define WM8904_DAC_SB_FILT_WIDTH 1 /* DAC_SB_FILT */ +#define WM8904_DAC_MUTERATE 0x0400 /* DAC_MUTERATE */ +#define WM8904_DAC_MUTERATE_MASK 0x0400 /* DAC_MUTERATE */ +#define WM8904_DAC_MUTERATE_SHIFT 10 /* DAC_MUTERATE */ +#define WM8904_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */ +#define WM8904_DAC_UNMUTE_RAMP 0x0200 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_UNMUTE_RAMP_MASK 0x0200 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_UNMUTE_RAMP_SHIFT 9 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_UNMUTE_RAMP_WIDTH 1 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_OSR128 0x0040 /* DAC_OSR128 */ +#define WM8904_DAC_OSR128_MASK 0x0040 /* DAC_OSR128 */ +#define WM8904_DAC_OSR128_SHIFT 6 /* DAC_OSR128 */ +#define WM8904_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */ +#define WM8904_DAC_MUTE 0x0008 /* DAC_MUTE */ +#define WM8904_DAC_MUTE_MASK 0x0008 /* DAC_MUTE */ +#define WM8904_DAC_MUTE_SHIFT 3 /* DAC_MUTE */ +#define WM8904_DAC_MUTE_WIDTH 1 /* DAC_MUTE */ +#define WM8904_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM8904_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM8904_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ + +/* + * R36 (0x24) - ADC Digital Volume Left + */ +#define WM8904_ADC_VU 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_MASK 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_SHIFT 8 /* ADC_VU */ +#define WM8904_ADC_VU_WIDTH 1 /* ADC_VU */ +#define WM8904_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8904_ADCL_VOL_SHIFT 0 /* ADCL_VOL - [7:0] */ +#define WM8904_ADCL_VOL_WIDTH 8 /* ADCL_VOL - [7:0] */ + +/* + * R37 (0x25) - ADC Digital Volume Right + */ +#define WM8904_ADC_VU 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_MASK 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_SHIFT 8 /* ADC_VU */ +#define WM8904_ADC_VU_WIDTH 1 /* ADC_VU */ +#define WM8904_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8904_ADCR_VOL_SHIFT 0 /* ADCR_VOL - [7:0] */ +#define WM8904_ADCR_VOL_WIDTH 8 /* ADCR_VOL - [7:0] */ + +/* + * R38 (0x26) - ADC Digital 0 + */ +#define WM8904_ADC_HPF_CUT_MASK 0x0060 /* ADC_HPF_CUT - [6:5] */ +#define WM8904_ADC_HPF_CUT_SHIFT 5 /* ADC_HPF_CUT - [6:5] */ +#define WM8904_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [6:5] */ +#define WM8904_ADC_HPF 0x0010 /* ADC_HPF */ +#define WM8904_ADC_HPF_MASK 0x0010 /* ADC_HPF */ +#define WM8904_ADC_HPF_SHIFT 4 /* ADC_HPF */ +#define WM8904_ADC_HPF_WIDTH 1 /* ADC_HPF */ +#define WM8904_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8904_ADCL_DATINV_MASK 0x0002 /* ADCL_DATINV */ +#define WM8904_ADCL_DATINV_SHIFT 1 /* ADCL_DATINV */ +#define WM8904_ADCL_DATINV_WIDTH 1 /* ADCL_DATINV */ +#define WM8904_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8904_ADCR_DATINV_MASK 0x0001 /* ADCR_DATINV */ +#define WM8904_ADCR_DATINV_SHIFT 0 /* ADCR_DATINV */ +#define WM8904_ADCR_DATINV_WIDTH 1 /* ADCR_DATINV */ + +/* + * R39 (0x27) - Digital Microphone 0 + */ +#define WM8904_DMIC_ENA 0x1000 /* DMIC_ENA */ +#define WM8904_DMIC_ENA_MASK 0x1000 /* DMIC_ENA */ +#define WM8904_DMIC_ENA_SHIFT 12 /* DMIC_ENA */ +#define WM8904_DMIC_ENA_WIDTH 1 /* DMIC_ENA */ +#define WM8904_DMIC_SRC 0x0800 /* DMIC_SRC */ +#define WM8904_DMIC_SRC_MASK 0x0800 /* DMIC_SRC */ +#define WM8904_DMIC_SRC_SHIFT 11 /* DMIC_SRC */ +#define WM8904_DMIC_SRC_WIDTH 1 /* DMIC_SRC */ + +/* + * R40 (0x28) - DRC 0 + */ +#define WM8904_DRC_ENA 0x8000 /* DRC_ENA */ +#define WM8904_DRC_ENA_MASK 0x8000 /* DRC_ENA */ +#define WM8904_DRC_ENA_SHIFT 15 /* DRC_ENA */ +#define WM8904_DRC_ENA_WIDTH 1 /* DRC_ENA */ +#define WM8904_DRC_DAC_PATH 0x4000 /* DRC_DAC_PATH */ +#define WM8904_DRC_DAC_PATH_MASK 0x4000 /* DRC_DAC_PATH */ +#define WM8904_DRC_DAC_PATH_SHIFT 14 /* DRC_DAC_PATH */ +#define WM8904_DRC_DAC_PATH_WIDTH 1 /* DRC_DAC_PATH */ +#define WM8904_DRC_GS_HYST_LVL_MASK 0x1800 /* DRC_GS_HYST_LVL - [12:11] */ +#define WM8904_DRC_GS_HYST_LVL_SHIFT 11 /* DRC_GS_HYST_LVL - [12:11] */ +#define WM8904_DRC_GS_HYST_LVL_WIDTH 2 /* DRC_GS_HYST_LVL - [12:11] */ +#define WM8904_DRC_STARTUP_GAIN_MASK 0x07C0 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM8904_DRC_STARTUP_GAIN_SHIFT 6 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM8904_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM8904_DRC_FF_DELAY 0x0020 /* DRC_FF_DELAY */ +#define WM8904_DRC_FF_DELAY_MASK 0x0020 /* DRC_FF_DELAY */ +#define WM8904_DRC_FF_DELAY_SHIFT 5 /* DRC_FF_DELAY */ +#define WM8904_DRC_FF_DELAY_WIDTH 1 /* DRC_FF_DELAY */ +#define WM8904_DRC_GS_ENA 0x0008 /* DRC_GS_ENA */ +#define WM8904_DRC_GS_ENA_MASK 0x0008 /* DRC_GS_ENA */ +#define WM8904_DRC_GS_ENA_SHIFT 3 /* DRC_GS_ENA */ +#define WM8904_DRC_GS_ENA_WIDTH 1 /* DRC_GS_ENA */ +#define WM8904_DRC_QR 0x0004 /* DRC_QR */ +#define WM8904_DRC_QR_MASK 0x0004 /* DRC_QR */ +#define WM8904_DRC_QR_SHIFT 2 /* DRC_QR */ +#define WM8904_DRC_QR_WIDTH 1 /* DRC_QR */ +#define WM8904_DRC_ANTICLIP 0x0002 /* DRC_ANTICLIP */ +#define WM8904_DRC_ANTICLIP_MASK 0x0002 /* DRC_ANTICLIP */ +#define WM8904_DRC_ANTICLIP_SHIFT 1 /* DRC_ANTICLIP */ +#define WM8904_DRC_ANTICLIP_WIDTH 1 /* DRC_ANTICLIP */ +#define WM8904_DRC_GS_HYST 0x0001 /* DRC_GS_HYST */ +#define WM8904_DRC_GS_HYST_MASK 0x0001 /* DRC_GS_HYST */ +#define WM8904_DRC_GS_HYST_SHIFT 0 /* DRC_GS_HYST */ +#define WM8904_DRC_GS_HYST_WIDTH 1 /* DRC_GS_HYST */ + +/* + * R41 (0x29) - DRC 1 + */ +#define WM8904_DRC_ATK_MASK 0xF000 /* DRC_ATK - [15:12] */ +#define WM8904_DRC_ATK_SHIFT 12 /* DRC_ATK - [15:12] */ +#define WM8904_DRC_ATK_WIDTH 4 /* DRC_ATK - [15:12] */ +#define WM8904_DRC_DCY_MASK 0x0F00 /* DRC_DCY - [11:8] */ +#define WM8904_DRC_DCY_SHIFT 8 /* DRC_DCY - [11:8] */ +#define WM8904_DRC_DCY_WIDTH 4 /* DRC_DCY - [11:8] */ +#define WM8904_DRC_QR_THR_MASK 0x00C0 /* DRC_QR_THR - [7:6] */ +#define WM8904_DRC_QR_THR_SHIFT 6 /* DRC_QR_THR - [7:6] */ +#define WM8904_DRC_QR_THR_WIDTH 2 /* DRC_QR_THR - [7:6] */ +#define WM8904_DRC_QR_DCY_MASK 0x0030 /* DRC_QR_DCY - [5:4] */ +#define WM8904_DRC_QR_DCY_SHIFT 4 /* DRC_QR_DCY - [5:4] */ +#define WM8904_DRC_QR_DCY_WIDTH 2 /* DRC_QR_DCY - [5:4] */ +#define WM8904_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */ +#define WM8904_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */ +#define WM8904_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */ +#define WM8904_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */ +#define WM8904_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */ +#define WM8904_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */ + +/* + * R42 (0x2A) - DRC 2 + */ +#define WM8904_DRC_HI_COMP_MASK 0x0038 /* DRC_HI_COMP - [5:3] */ +#define WM8904_DRC_HI_COMP_SHIFT 3 /* DRC_HI_COMP - [5:3] */ +#define WM8904_DRC_HI_COMP_WIDTH 3 /* DRC_HI_COMP - [5:3] */ +#define WM8904_DRC_LO_COMP_MASK 0x0007 /* DRC_LO_COMP - [2:0] */ +#define WM8904_DRC_LO_COMP_SHIFT 0 /* DRC_LO_COMP - [2:0] */ +#define WM8904_DRC_LO_COMP_WIDTH 3 /* DRC_LO_COMP - [2:0] */ + +/* + * R43 (0x2B) - DRC 3 + */ +#define WM8904_DRC_KNEE_IP_MASK 0x07E0 /* DRC_KNEE_IP - [10:5] */ +#define WM8904_DRC_KNEE_IP_SHIFT 5 /* DRC_KNEE_IP - [10:5] */ +#define WM8904_DRC_KNEE_IP_WIDTH 6 /* DRC_KNEE_IP - [10:5] */ +#define WM8904_DRC_KNEE_OP_MASK 0x001F /* DRC_KNEE_OP - [4:0] */ +#define WM8904_DRC_KNEE_OP_SHIFT 0 /* DRC_KNEE_OP - [4:0] */ +#define WM8904_DRC_KNEE_OP_WIDTH 5 /* DRC_KNEE_OP - [4:0] */ + +/* + * R44 (0x2C) - Analogue Left Input 0 + */ +#define WM8904_LINMUTE 0x0080 /* LINMUTE */ +#define WM8904_LINMUTE_MASK 0x0080 /* LINMUTE */ +#define WM8904_LINMUTE_SHIFT 7 /* LINMUTE */ +#define WM8904_LINMUTE_WIDTH 1 /* LINMUTE */ +#define WM8904_LIN_VOL_MASK 0x001F /* LIN_VOL - [4:0] */ +#define WM8904_LIN_VOL_SHIFT 0 /* LIN_VOL - [4:0] */ +#define WM8904_LIN_VOL_WIDTH 5 /* LIN_VOL - [4:0] */ + +/* + * R45 (0x2D) - Analogue Right Input 0 + */ +#define WM8904_RINMUTE 0x0080 /* RINMUTE */ +#define WM8904_RINMUTE_MASK 0x0080 /* RINMUTE */ +#define WM8904_RINMUTE_SHIFT 7 /* RINMUTE */ +#define WM8904_RINMUTE_WIDTH 1 /* RINMUTE */ +#define WM8904_RIN_VOL_MASK 0x001F /* RIN_VOL - [4:0] */ +#define WM8904_RIN_VOL_SHIFT 0 /* RIN_VOL - [4:0] */ +#define WM8904_RIN_VOL_WIDTH 5 /* RIN_VOL - [4:0] */ + +/* + * R46 (0x2E) - Analogue Left Input 1 + */ +#define WM8904_INL_CM_ENA 0x0040 /* INL_CM_ENA */ +#define WM8904_INL_CM_ENA_MASK 0x0040 /* INL_CM_ENA */ +#define WM8904_INL_CM_ENA_SHIFT 6 /* INL_CM_ENA */ +#define WM8904_INL_CM_ENA_WIDTH 1 /* INL_CM_ENA */ +#define WM8904_L_IP_SEL_N_MASK 0x0030 /* L_IP_SEL_N - [5:4] */ +#define WM8904_L_IP_SEL_N_SHIFT 4 /* L_IP_SEL_N - [5:4] */ +#define WM8904_L_IP_SEL_N_WIDTH 2 /* L_IP_SEL_N - [5:4] */ +#define WM8904_L_IP_SEL_P_MASK 0x000C /* L_IP_SEL_P - [3:2] */ +#define WM8904_L_IP_SEL_P_SHIFT 2 /* L_IP_SEL_P - [3:2] */ +#define WM8904_L_IP_SEL_P_WIDTH 2 /* L_IP_SEL_P - [3:2] */ +#define WM8904_L_MODE_MASK 0x0003 /* L_MODE - [1:0] */ +#define WM8904_L_MODE_SHIFT 0 /* L_MODE - [1:0] */ +#define WM8904_L_MODE_WIDTH 2 /* L_MODE - [1:0] */ + +/* + * R47 (0x2F) - Analogue Right Input 1 + */ +#define WM8904_INR_CM_ENA 0x0040 /* INR_CM_ENA */ +#define WM8904_INR_CM_ENA_MASK 0x0040 /* INR_CM_ENA */ +#define WM8904_INR_CM_ENA_SHIFT 6 /* INR_CM_ENA */ +#define WM8904_INR_CM_ENA_WIDTH 1 /* INR_CM_ENA */ +#define WM8904_R_IP_SEL_N_MASK 0x0030 /* R_IP_SEL_N - [5:4] */ +#define WM8904_R_IP_SEL_N_SHIFT 4 /* R_IP_SEL_N - [5:4] */ +#define WM8904_R_IP_SEL_N_WIDTH 2 /* R_IP_SEL_N - [5:4] */ +#define WM8904_R_IP_SEL_P_MASK 0x000C /* R_IP_SEL_P - [3:2] */ +#define WM8904_R_IP_SEL_P_SHIFT 2 /* R_IP_SEL_P - [3:2] */ +#define WM8904_R_IP_SEL_P_WIDTH 2 /* R_IP_SEL_P - [3:2] */ +#define WM8904_R_MODE_MASK 0x0003 /* R_MODE - [1:0] */ +#define WM8904_R_MODE_SHIFT 0 /* R_MODE - [1:0] */ +#define WM8904_R_MODE_WIDTH 2 /* R_MODE - [1:0] */ + +/* + * R57 (0x39) - Analogue OUT1 Left + */ +#define WM8904_HPOUTL_MUTE 0x0100 /* HPOUTL_MUTE */ +#define WM8904_HPOUTL_MUTE_MASK 0x0100 /* HPOUTL_MUTE */ +#define WM8904_HPOUTL_MUTE_SHIFT 8 /* HPOUTL_MUTE */ +#define WM8904_HPOUTL_MUTE_WIDTH 1 /* HPOUTL_MUTE */ +#define WM8904_HPOUT_VU 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_MASK 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_SHIFT 7 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_WIDTH 1 /* HPOUT_VU */ +#define WM8904_HPOUTLZC 0x0040 /* HPOUTLZC */ +#define WM8904_HPOUTLZC_MASK 0x0040 /* HPOUTLZC */ +#define WM8904_HPOUTLZC_SHIFT 6 /* HPOUTLZC */ +#define WM8904_HPOUTLZC_WIDTH 1 /* HPOUTLZC */ +#define WM8904_HPOUTL_VOL_MASK 0x003F /* HPOUTL_VOL - [5:0] */ +#define WM8904_HPOUTL_VOL_SHIFT 0 /* HPOUTL_VOL - [5:0] */ +#define WM8904_HPOUTL_VOL_WIDTH 6 /* HPOUTL_VOL - [5:0] */ + +/* + * R58 (0x3A) - Analogue OUT1 Right + */ +#define WM8904_HPOUTR_MUTE 0x0100 /* HPOUTR_MUTE */ +#define WM8904_HPOUTR_MUTE_MASK 0x0100 /* HPOUTR_MUTE */ +#define WM8904_HPOUTR_MUTE_SHIFT 8 /* HPOUTR_MUTE */ +#define WM8904_HPOUTR_MUTE_WIDTH 1 /* HPOUTR_MUTE */ +#define WM8904_HPOUT_VU 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_MASK 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_SHIFT 7 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_WIDTH 1 /* HPOUT_VU */ +#define WM8904_HPOUTRZC 0x0040 /* HPOUTRZC */ +#define WM8904_HPOUTRZC_MASK 0x0040 /* HPOUTRZC */ +#define WM8904_HPOUTRZC_SHIFT 6 /* HPOUTRZC */ +#define WM8904_HPOUTRZC_WIDTH 1 /* HPOUTRZC */ +#define WM8904_HPOUTR_VOL_MASK 0x003F /* HPOUTR_VOL - [5:0] */ +#define WM8904_HPOUTR_VOL_SHIFT 0 /* HPOUTR_VOL - [5:0] */ +#define WM8904_HPOUTR_VOL_WIDTH 6 /* HPOUTR_VOL - [5:0] */ + +/* + * R59 (0x3B) - Analogue OUT2 Left + */ +#define WM8904_LINEOUTL_MUTE 0x0100 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUTL_MUTE_MASK 0x0100 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUTL_MUTE_SHIFT 8 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUTL_MUTE_WIDTH 1 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUT_VU 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_MASK 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_SHIFT 7 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_WIDTH 1 /* LINEOUT_VU */ +#define WM8904_LINEOUTLZC 0x0040 /* LINEOUTLZC */ +#define WM8904_LINEOUTLZC_MASK 0x0040 /* LINEOUTLZC */ +#define WM8904_LINEOUTLZC_SHIFT 6 /* LINEOUTLZC */ +#define WM8904_LINEOUTLZC_WIDTH 1 /* LINEOUTLZC */ +#define WM8904_LINEOUTL_VOL_MASK 0x003F /* LINEOUTL_VOL - [5:0] */ +#define WM8904_LINEOUTL_VOL_SHIFT 0 /* LINEOUTL_VOL - [5:0] */ +#define WM8904_LINEOUTL_VOL_WIDTH 6 /* LINEOUTL_VOL - [5:0] */ + +/* + * R60 (0x3C) - Analogue OUT2 Right + */ +#define WM8904_LINEOUTR_MUTE 0x0100 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUTR_MUTE_MASK 0x0100 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUTR_MUTE_SHIFT 8 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUTR_MUTE_WIDTH 1 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUT_VU 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_MASK 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_SHIFT 7 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_WIDTH 1 /* LINEOUT_VU */ +#define WM8904_LINEOUTRZC 0x0040 /* LINEOUTRZC */ +#define WM8904_LINEOUTRZC_MASK 0x0040 /* LINEOUTRZC */ +#define WM8904_LINEOUTRZC_SHIFT 6 /* LINEOUTRZC */ +#define WM8904_LINEOUTRZC_WIDTH 1 /* LINEOUTRZC */ +#define WM8904_LINEOUTR_VOL_MASK 0x003F /* LINEOUTR_VOL - [5:0] */ +#define WM8904_LINEOUTR_VOL_SHIFT 0 /* LINEOUTR_VOL - [5:0] */ +#define WM8904_LINEOUTR_VOL_WIDTH 6 /* LINEOUTR_VOL - [5:0] */ + +/* + * R61 (0x3D) - Analogue OUT12 ZC + */ +#define WM8904_HPL_BYP_ENA 0x0008 /* HPL_BYP_ENA */ +#define WM8904_HPL_BYP_ENA_MASK 0x0008 /* HPL_BYP_ENA */ +#define WM8904_HPL_BYP_ENA_SHIFT 3 /* HPL_BYP_ENA */ +#define WM8904_HPL_BYP_ENA_WIDTH 1 /* HPL_BYP_ENA */ +#define WM8904_HPR_BYP_ENA 0x0004 /* HPR_BYP_ENA */ +#define WM8904_HPR_BYP_ENA_MASK 0x0004 /* HPR_BYP_ENA */ +#define WM8904_HPR_BYP_ENA_SHIFT 2 /* HPR_BYP_ENA */ +#define WM8904_HPR_BYP_ENA_WIDTH 1 /* HPR_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA 0x0002 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA_MASK 0x0002 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA_SHIFT 1 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA_WIDTH 1 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA 0x0001 /* LINEOUTR_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA_MASK 0x0001 /* LINEOUTR_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA_SHIFT 0 /* LINEOUTR_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA_WIDTH 1 /* LINEOUTR_BYP_ENA */ + +/* + * R67 (0x43) - DC Servo 0 + */ +#define WM8904_DCS_ENA_CHAN_3 0x0008 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_3_MASK 0x0008 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_3_SHIFT 3 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_3_WIDTH 1 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_2 0x0004 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_2_MASK 0x0004 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_2_SHIFT 2 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_2_WIDTH 1 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_1 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_1_MASK 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_1_SHIFT 1 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_1_WIDTH 1 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_0 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8904_DCS_ENA_CHAN_0_MASK 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8904_DCS_ENA_CHAN_0_SHIFT 0 /* DCS_ENA_CHAN_0 */ +#define WM8904_DCS_ENA_CHAN_0_WIDTH 1 /* DCS_ENA_CHAN_0 */ + +/* + * R68 (0x44) - DC Servo 1 + */ +#define WM8904_DCS_TRIG_SINGLE_3 0x8000 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_3_MASK 0x8000 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_3_SHIFT 15 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_3_WIDTH 1 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_2 0x4000 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_2_MASK 0x4000 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_2_SHIFT 14 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_2_WIDTH 1 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_1 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_1_MASK 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_1_SHIFT 13 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_1_WIDTH 1 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_0 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SINGLE_0_MASK 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SINGLE_0_SHIFT 12 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SINGLE_0_WIDTH 1 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SERIES_3 0x0800 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_3_MASK 0x0800 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_3_SHIFT 11 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_3_WIDTH 1 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_2 0x0400 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_2_MASK 0x0400 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_2_SHIFT 10 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_2_WIDTH 1 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_1 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_1_MASK 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_1_SHIFT 9 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_1_WIDTH 1 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_0 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_SERIES_0_MASK 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_SERIES_0_SHIFT 8 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_SERIES_0_WIDTH 1 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_STARTUP_3 0x0080 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_3_MASK 0x0080 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_3_SHIFT 7 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_3_WIDTH 1 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_2 0x0040 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_2_MASK 0x0040 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_2_SHIFT 6 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_2_WIDTH 1 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_1 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_1_MASK 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_1_SHIFT 5 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_1_WIDTH 1 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_0 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_STARTUP_0_MASK 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_STARTUP_0_SHIFT 4 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_STARTUP_0_WIDTH 1 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_DAC_WR_3 0x0008 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_3_MASK 0x0008 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_3_SHIFT 3 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_3_WIDTH 1 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_2 0x0004 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_2_MASK 0x0004 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_2_SHIFT 2 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_2_WIDTH 1 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_1 0x0002 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_1_MASK 0x0002 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_1_SHIFT 1 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_1_WIDTH 1 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_0 0x0001 /* DCS_TRIG_DAC_WR_0 */ +#define WM8904_DCS_TRIG_DAC_WR_0_MASK 0x0001 /* DCS_TRIG_DAC_WR_0 */ +#define WM8904_DCS_TRIG_DAC_WR_0_SHIFT 0 /* DCS_TRIG_DAC_WR_0 */ +#define WM8904_DCS_TRIG_DAC_WR_0_WIDTH 1 /* DCS_TRIG_DAC_WR_0 */ + +/* + * R69 (0x45) - DC Servo 2 + */ +#define WM8904_DCS_TIMER_PERIOD_23_MASK 0x0F00 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8904_DCS_TIMER_PERIOD_23_SHIFT 8 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8904_DCS_TIMER_PERIOD_23_WIDTH 4 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8904_DCS_TIMER_PERIOD_01_MASK 0x000F /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8904_DCS_TIMER_PERIOD_01_SHIFT 0 /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8904_DCS_TIMER_PERIOD_01_WIDTH 4 /* DCS_TIMER_PERIOD_01 - [3:0] */ + +/* + * R71 (0x47) - DC Servo 4 + */ +#define WM8904_DCS_SERIES_NO_23_MASK 0x007F /* DCS_SERIES_NO_23 - [6:0] */ +#define WM8904_DCS_SERIES_NO_23_SHIFT 0 /* DCS_SERIES_NO_23 - [6:0] */ +#define WM8904_DCS_SERIES_NO_23_WIDTH 7 /* DCS_SERIES_NO_23 - [6:0] */ + +/* + * R72 (0x48) - DC Servo 5 + */ +#define WM8904_DCS_SERIES_NO_01_MASK 0x007F /* DCS_SERIES_NO_01 - [6:0] */ +#define WM8904_DCS_SERIES_NO_01_SHIFT 0 /* DCS_SERIES_NO_01 - [6:0] */ +#define WM8904_DCS_SERIES_NO_01_WIDTH 7 /* DCS_SERIES_NO_01 - [6:0] */ + +/* + * R73 (0x49) - DC Servo 6 + */ +#define WM8904_DCS_DAC_WR_VAL_3_MASK 0x00FF /* DCS_DAC_WR_VAL_3 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_3_SHIFT 0 /* DCS_DAC_WR_VAL_3 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_3_WIDTH 8 /* DCS_DAC_WR_VAL_3 - [7:0] */ + +/* + * R74 (0x4A) - DC Servo 7 + */ +#define WM8904_DCS_DAC_WR_VAL_2_MASK 0x00FF /* DCS_DAC_WR_VAL_2 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_2_SHIFT 0 /* DCS_DAC_WR_VAL_2 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_2_WIDTH 8 /* DCS_DAC_WR_VAL_2 - [7:0] */ + +/* + * R75 (0x4B) - DC Servo 8 + */ +#define WM8904_DCS_DAC_WR_VAL_1_MASK 0x00FF /* DCS_DAC_WR_VAL_1 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_1_SHIFT 0 /* DCS_DAC_WR_VAL_1 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_1_WIDTH 8 /* DCS_DAC_WR_VAL_1 - [7:0] */ + +/* + * R76 (0x4C) - DC Servo 9 + */ +#define WM8904_DCS_DAC_WR_VAL_0_MASK 0x00FF /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_0_SHIFT 0 /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_0_WIDTH 8 /* DCS_DAC_WR_VAL_0 - [7:0] */ + +/* + * R77 (0x4D) - DC Servo Readback 0 + */ +#define WM8904_DCS_CAL_COMPLETE_MASK 0x0F00 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8904_DCS_CAL_COMPLETE_SHIFT 8 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8904_DCS_CAL_COMPLETE_WIDTH 4 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8904_DCS_DAC_WR_COMPLETE_MASK 0x00F0 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8904_DCS_DAC_WR_COMPLETE_SHIFT 4 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8904_DCS_DAC_WR_COMPLETE_WIDTH 4 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8904_DCS_STARTUP_COMPLETE_MASK 0x000F /* DCS_STARTUP_COMPLETE - [3:0] */ +#define WM8904_DCS_STARTUP_COMPLETE_SHIFT 0 /* DCS_STARTUP_COMPLETE - [3:0] */ +#define WM8904_DCS_STARTUP_COMPLETE_WIDTH 4 /* DCS_STARTUP_COMPLETE - [3:0] */ + +/* + * R90 (0x5A) - Analogue HP 0 + */ +#define WM8904_HPL_RMV_SHORT 0x0080 /* HPL_RMV_SHORT */ +#define WM8904_HPL_RMV_SHORT_MASK 0x0080 /* HPL_RMV_SHORT */ +#define WM8904_HPL_RMV_SHORT_SHIFT 7 /* HPL_RMV_SHORT */ +#define WM8904_HPL_RMV_SHORT_WIDTH 1 /* HPL_RMV_SHORT */ +#define WM8904_HPL_ENA_OUTP 0x0040 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_OUTP_MASK 0x0040 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_OUTP_SHIFT 6 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_OUTP_WIDTH 1 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_DLY 0x0020 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA_DLY_MASK 0x0020 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA_DLY_SHIFT 5 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA_DLY_WIDTH 1 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA 0x0010 /* HPL_ENA */ +#define WM8904_HPL_ENA_MASK 0x0010 /* HPL_ENA */ +#define WM8904_HPL_ENA_SHIFT 4 /* HPL_ENA */ +#define WM8904_HPL_ENA_WIDTH 1 /* HPL_ENA */ +#define WM8904_HPR_RMV_SHORT 0x0008 /* HPR_RMV_SHORT */ +#define WM8904_HPR_RMV_SHORT_MASK 0x0008 /* HPR_RMV_SHORT */ +#define WM8904_HPR_RMV_SHORT_SHIFT 3 /* HPR_RMV_SHORT */ +#define WM8904_HPR_RMV_SHORT_WIDTH 1 /* HPR_RMV_SHORT */ +#define WM8904_HPR_ENA_OUTP 0x0004 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_OUTP_MASK 0x0004 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_OUTP_SHIFT 2 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_OUTP_WIDTH 1 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_DLY 0x0002 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA_DLY_MASK 0x0002 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA_DLY_SHIFT 1 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA_DLY_WIDTH 1 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA 0x0001 /* HPR_ENA */ +#define WM8904_HPR_ENA_MASK 0x0001 /* HPR_ENA */ +#define WM8904_HPR_ENA_SHIFT 0 /* HPR_ENA */ +#define WM8904_HPR_ENA_WIDTH 1 /* HPR_ENA */ + +/* + * R94 (0x5E) - Analogue Lineout 0 + */ +#define WM8904_LINEOUTL_RMV_SHORT 0x0080 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_RMV_SHORT_MASK 0x0080 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_RMV_SHORT_SHIFT 7 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_RMV_SHORT_WIDTH 1 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_ENA_OUTP 0x0040 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_OUTP_MASK 0x0040 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_OUTP_SHIFT 6 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_OUTP_WIDTH 1 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_DLY 0x0020 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA_DLY_MASK 0x0020 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA_DLY_SHIFT 5 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA_DLY_WIDTH 1 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA 0x0010 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTL_ENA_MASK 0x0010 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTL_ENA_SHIFT 4 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTL_ENA_WIDTH 1 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTR_RMV_SHORT 0x0008 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_RMV_SHORT_MASK 0x0008 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_RMV_SHORT_SHIFT 3 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_RMV_SHORT_WIDTH 1 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_ENA_OUTP 0x0004 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_OUTP_MASK 0x0004 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_OUTP_SHIFT 2 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_OUTP_WIDTH 1 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_DLY 0x0002 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA_DLY_MASK 0x0002 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA_DLY_SHIFT 1 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA_DLY_WIDTH 1 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA 0x0001 /* LINEOUTR_ENA */ +#define WM8904_LINEOUTR_ENA_MASK 0x0001 /* LINEOUTR_ENA */ +#define WM8904_LINEOUTR_ENA_SHIFT 0 /* LINEOUTR_ENA */ +#define WM8904_LINEOUTR_ENA_WIDTH 1 /* LINEOUTR_ENA */ + +/* + * R98 (0x62) - Charge Pump 0 + */ +#define WM8904_CP_ENA 0x0001 /* CP_ENA */ +#define WM8904_CP_ENA_MASK 0x0001 /* CP_ENA */ +#define WM8904_CP_ENA_SHIFT 0 /* CP_ENA */ +#define WM8904_CP_ENA_WIDTH 1 /* CP_ENA */ + +/* + * R104 (0x68) - Class W 0 + */ +#define WM8904_CP_DYN_PWR 0x0001 /* CP_DYN_PWR */ +#define WM8904_CP_DYN_PWR_MASK 0x0001 /* CP_DYN_PWR */ +#define WM8904_CP_DYN_PWR_SHIFT 0 /* CP_DYN_PWR */ +#define WM8904_CP_DYN_PWR_WIDTH 1 /* CP_DYN_PWR */ + +/* + * R108 (0x6C) - Write Sequencer 0 + */ +#define WM8904_WSEQ_ENA 0x0100 /* WSEQ_ENA */ +#define WM8904_WSEQ_ENA_MASK 0x0100 /* WSEQ_ENA */ +#define WM8904_WSEQ_ENA_SHIFT 8 /* WSEQ_ENA */ +#define WM8904_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM8904_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8904_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8904_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */ + +/* + * R109 (0x6D) - Write Sequencer 1 + */ +#define WM8904_WSEQ_DATA_WIDTH_MASK 0x7000 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8904_WSEQ_DATA_WIDTH_SHIFT 12 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8904_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8904_WSEQ_DATA_START_MASK 0x0F00 /* WSEQ_DATA_START - [11:8] */ +#define WM8904_WSEQ_DATA_START_SHIFT 8 /* WSEQ_DATA_START - [11:8] */ +#define WM8904_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [11:8] */ +#define WM8904_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */ +#define WM8904_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */ +#define WM8904_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */ + +/* + * R110 (0x6E) - Write Sequencer 2 + */ +#define WM8904_WSEQ_EOS 0x4000 /* WSEQ_EOS */ +#define WM8904_WSEQ_EOS_MASK 0x4000 /* WSEQ_EOS */ +#define WM8904_WSEQ_EOS_SHIFT 14 /* WSEQ_EOS */ +#define WM8904_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */ +#define WM8904_WSEQ_DELAY_MASK 0x0F00 /* WSEQ_DELAY - [11:8] */ +#define WM8904_WSEQ_DELAY_SHIFT 8 /* WSEQ_DELAY - [11:8] */ +#define WM8904_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [11:8] */ +#define WM8904_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */ +#define WM8904_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */ +#define WM8904_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */ + +/* + * R111 (0x6F) - Write Sequencer 3 + */ +#define WM8904_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */ +#define WM8904_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */ +#define WM8904_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */ +#define WM8904_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM8904_WSEQ_START 0x0100 /* WSEQ_START */ +#define WM8904_WSEQ_START_MASK 0x0100 /* WSEQ_START */ +#define WM8904_WSEQ_START_SHIFT 8 /* WSEQ_START */ +#define WM8904_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM8904_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */ +#define WM8904_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */ +#define WM8904_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */ + +/* + * R112 (0x70) - Write Sequencer 4 + */ +#define WM8904_WSEQ_CURRENT_INDEX_MASK 0x03F0 /* WSEQ_CURRENT_INDEX - [9:4] */ +#define WM8904_WSEQ_CURRENT_INDEX_SHIFT 4 /* WSEQ_CURRENT_INDEX - [9:4] */ +#define WM8904_WSEQ_CURRENT_INDEX_WIDTH 6 /* WSEQ_CURRENT_INDEX - [9:4] */ +#define WM8904_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */ +#define WM8904_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */ +#define WM8904_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */ +#define WM8904_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ + +/* + * R116 (0x74) - FLL Control 1 + */ +#define WM8904_FLL_FRACN_ENA 0x0004 /* FLL_FRACN_ENA */ +#define WM8904_FLL_FRACN_ENA_MASK 0x0004 /* FLL_FRACN_ENA */ +#define WM8904_FLL_FRACN_ENA_SHIFT 2 /* FLL_FRACN_ENA */ +#define WM8904_FLL_FRACN_ENA_WIDTH 1 /* FLL_FRACN_ENA */ +#define WM8904_FLL_OSC_ENA 0x0002 /* FLL_OSC_ENA */ +#define WM8904_FLL_OSC_ENA_MASK 0x0002 /* FLL_OSC_ENA */ +#define WM8904_FLL_OSC_ENA_SHIFT 1 /* FLL_OSC_ENA */ +#define WM8904_FLL_OSC_ENA_WIDTH 1 /* FLL_OSC_ENA */ +#define WM8904_FLL_ENA 0x0001 /* FLL_ENA */ +#define WM8904_FLL_ENA_MASK 0x0001 /* FLL_ENA */ +#define WM8904_FLL_ENA_SHIFT 0 /* FLL_ENA */ +#define WM8904_FLL_ENA_WIDTH 1 /* FLL_ENA */ + +/* + * R117 (0x75) - FLL Control 2 + */ +#define WM8904_FLL_OUTDIV_MASK 0x3F00 /* FLL_OUTDIV - [13:8] */ +#define WM8904_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [13:8] */ +#define WM8904_FLL_OUTDIV_WIDTH 6 /* FLL_OUTDIV - [13:8] */ +#define WM8904_FLL_CTRL_RATE_MASK 0x0070 /* FLL_CTRL_RATE - [6:4] */ +#define WM8904_FLL_CTRL_RATE_SHIFT 4 /* FLL_CTRL_RATE - [6:4] */ +#define WM8904_FLL_CTRL_RATE_WIDTH 3 /* FLL_CTRL_RATE - [6:4] */ +#define WM8904_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */ +#define WM8904_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */ +#define WM8904_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */ + +/* + * R118 (0x76) - FLL Control 3 + */ +#define WM8904_FLL_K_MASK 0xFFFF /* FLL_K - [15:0] */ +#define WM8904_FLL_K_SHIFT 0 /* FLL_K - [15:0] */ +#define WM8904_FLL_K_WIDTH 16 /* FLL_K - [15:0] */ + +/* + * R119 (0x77) - FLL Control 4 + */ +#define WM8904_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */ +#define WM8904_FLL_N_SHIFT 5 /* FLL_N - [14:5] */ +#define WM8904_FLL_N_WIDTH 10 /* FLL_N - [14:5] */ +#define WM8904_FLL_GAIN_MASK 0x000F /* FLL_GAIN - [3:0] */ +#define WM8904_FLL_GAIN_SHIFT 0 /* FLL_GAIN - [3:0] */ +#define WM8904_FLL_GAIN_WIDTH 4 /* FLL_GAIN - [3:0] */ + +/* + * R120 (0x78) - FLL Control 5 + */ +#define WM8904_FLL_CLK_REF_DIV_MASK 0x0018 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8904_FLL_CLK_REF_DIV_SHIFT 3 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8904_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8904_FLL_CLK_REF_SRC_MASK 0x0003 /* FLL_CLK_REF_SRC - [1:0] */ +#define WM8904_FLL_CLK_REF_SRC_SHIFT 0 /* FLL_CLK_REF_SRC - [1:0] */ +#define WM8904_FLL_CLK_REF_SRC_WIDTH 2 /* FLL_CLK_REF_SRC - [1:0] */ + +/* + * R121 (0x79) - GPIO Control 1 + */ +#define WM8904_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_MASK 0x0020 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_SHIFT 5 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_WIDTH 1 /* GPIO1_PU */ +#define WM8904_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_MASK 0x0010 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_SHIFT 4 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_WIDTH 1 /* GPIO1_PD */ +#define WM8904_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ +#define WM8904_GPIO1_SEL_SHIFT 0 /* GPIO1_SEL - [3:0] */ +#define WM8904_GPIO1_SEL_WIDTH 4 /* GPIO1_SEL - [3:0] */ + +/* + * R122 (0x7A) - GPIO Control 2 + */ +#define WM8904_GPIO2_PU 0x0020 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_MASK 0x0020 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_SHIFT 5 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_WIDTH 1 /* GPIO2_PU */ +#define WM8904_GPIO2_PD 0x0010 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_MASK 0x0010 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_SHIFT 4 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_WIDTH 1 /* GPIO2_PD */ +#define WM8904_GPIO2_SEL_MASK 0x000F /* GPIO2_SEL - [3:0] */ +#define WM8904_GPIO2_SEL_SHIFT 0 /* GPIO2_SEL - [3:0] */ +#define WM8904_GPIO2_SEL_WIDTH 4 /* GPIO2_SEL - [3:0] */ + +/* + * R123 (0x7B) - GPIO Control 3 + */ +#define WM8904_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_MASK 0x0020 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_SHIFT 5 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_WIDTH 1 /* GPIO3_PU */ +#define WM8904_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_MASK 0x0010 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_SHIFT 4 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_WIDTH 1 /* GPIO3_PD */ +#define WM8904_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ +#define WM8904_GPIO3_SEL_SHIFT 0 /* GPIO3_SEL - [3:0] */ +#define WM8904_GPIO3_SEL_WIDTH 4 /* GPIO3_SEL - [3:0] */ + +/* + * R124 (0x7C) - GPIO Control 4 + */ +#define WM8904_GPI7_ENA 0x0200 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_MASK 0x0200 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_SHIFT 9 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_WIDTH 1 /* GPI7_ENA */ +#define WM8904_GPI8_ENA 0x0100 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_MASK 0x0100 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_SHIFT 8 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_WIDTH 1 /* GPI8_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA 0x0080 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_MASK 0x0080 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_SHIFT 7 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_WIDTH 1 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_SEL_MASK 0x000F /* GPIO_BCLK_SEL - [3:0] */ +#define WM8904_GPIO_BCLK_SEL_SHIFT 0 /* GPIO_BCLK_SEL - [3:0] */ +#define WM8904_GPIO_BCLK_SEL_WIDTH 4 /* GPIO_BCLK_SEL - [3:0] */ + +/* + * R126 (0x7E) - Digital Pulls + */ +#define WM8904_MCLK_PU 0x0080 /* MCLK_PU */ +#define WM8904_MCLK_PU_MASK 0x0080 /* MCLK_PU */ +#define WM8904_MCLK_PU_SHIFT 7 /* MCLK_PU */ +#define WM8904_MCLK_PU_WIDTH 1 /* MCLK_PU */ +#define WM8904_MCLK_PD 0x0040 /* MCLK_PD */ +#define WM8904_MCLK_PD_MASK 0x0040 /* MCLK_PD */ +#define WM8904_MCLK_PD_SHIFT 6 /* MCLK_PD */ +#define WM8904_MCLK_PD_WIDTH 1 /* MCLK_PD */ +#define WM8904_DACDAT_PU 0x0020 /* DACDAT_PU */ +#define WM8904_DACDAT_PU_MASK 0x0020 /* DACDAT_PU */ +#define WM8904_DACDAT_PU_SHIFT 5 /* DACDAT_PU */ +#define WM8904_DACDAT_PU_WIDTH 1 /* DACDAT_PU */ +#define WM8904_DACDAT_PD 0x0010 /* DACDAT_PD */ +#define WM8904_DACDAT_PD_MASK 0x0010 /* DACDAT_PD */ +#define WM8904_DACDAT_PD_SHIFT 4 /* DACDAT_PD */ +#define WM8904_DACDAT_PD_WIDTH 1 /* DACDAT_PD */ +#define WM8904_LRCLK_PU 0x0008 /* LRCLK_PU */ +#define WM8904_LRCLK_PU_MASK 0x0008 /* LRCLK_PU */ +#define WM8904_LRCLK_PU_SHIFT 3 /* LRCLK_PU */ +#define WM8904_LRCLK_PU_WIDTH 1 /* LRCLK_PU */ +#define WM8904_LRCLK_PD 0x0004 /* LRCLK_PD */ +#define WM8904_LRCLK_PD_MASK 0x0004 /* LRCLK_PD */ +#define WM8904_LRCLK_PD_SHIFT 2 /* LRCLK_PD */ +#define WM8904_LRCLK_PD_WIDTH 1 /* LRCLK_PD */ +#define WM8904_BCLK_PU 0x0002 /* BCLK_PU */ +#define WM8904_BCLK_PU_MASK 0x0002 /* BCLK_PU */ +#define WM8904_BCLK_PU_SHIFT 1 /* BCLK_PU */ +#define WM8904_BCLK_PU_WIDTH 1 /* BCLK_PU */ +#define WM8904_BCLK_PD 0x0001 /* BCLK_PD */ +#define WM8904_BCLK_PD_MASK 0x0001 /* BCLK_PD */ +#define WM8904_BCLK_PD_SHIFT 0 /* BCLK_PD */ +#define WM8904_BCLK_PD_WIDTH 1 /* BCLK_PD */ + +/* + * R127 (0x7F) - Interrupt Status + */ +#define WM8904_IRQ 0x0400 /* IRQ */ +#define WM8904_IRQ_MASK 0x0400 /* IRQ */ +#define WM8904_IRQ_SHIFT 10 /* IRQ */ +#define WM8904_IRQ_WIDTH 1 /* IRQ */ +#define WM8904_GPIO_BCLK_EINT 0x0200 /* GPIO_BCLK_EINT */ +#define WM8904_GPIO_BCLK_EINT_MASK 0x0200 /* GPIO_BCLK_EINT */ +#define WM8904_GPIO_BCLK_EINT_SHIFT 9 /* GPIO_BCLK_EINT */ +#define WM8904_GPIO_BCLK_EINT_WIDTH 1 /* GPIO_BCLK_EINT */ +#define WM8904_WSEQ_EINT 0x0100 /* WSEQ_EINT */ +#define WM8904_WSEQ_EINT_MASK 0x0100 /* WSEQ_EINT */ +#define WM8904_WSEQ_EINT_SHIFT 8 /* WSEQ_EINT */ +#define WM8904_WSEQ_EINT_WIDTH 1 /* WSEQ_EINT */ +#define WM8904_GPIO3_EINT 0x0080 /* GPIO3_EINT */ +#define WM8904_GPIO3_EINT_MASK 0x0080 /* GPIO3_EINT */ +#define WM8904_GPIO3_EINT_SHIFT 7 /* GPIO3_EINT */ +#define WM8904_GPIO3_EINT_WIDTH 1 /* GPIO3_EINT */ +#define WM8904_GPIO2_EINT 0x0040 /* GPIO2_EINT */ +#define WM8904_GPIO2_EINT_MASK 0x0040 /* GPIO2_EINT */ +#define WM8904_GPIO2_EINT_SHIFT 6 /* GPIO2_EINT */ +#define WM8904_GPIO2_EINT_WIDTH 1 /* GPIO2_EINT */ +#define WM8904_GPIO1_EINT 0x0020 /* GPIO1_EINT */ +#define WM8904_GPIO1_EINT_MASK 0x0020 /* GPIO1_EINT */ +#define WM8904_GPIO1_EINT_SHIFT 5 /* GPIO1_EINT */ +#define WM8904_GPIO1_EINT_WIDTH 1 /* GPIO1_EINT */ +#define WM8904_GPI8_EINT 0x0010 /* GPI8_EINT */ +#define WM8904_GPI8_EINT_MASK 0x0010 /* GPI8_EINT */ +#define WM8904_GPI8_EINT_SHIFT 4 /* GPI8_EINT */ +#define WM8904_GPI8_EINT_WIDTH 1 /* GPI8_EINT */ +#define WM8904_GPI7_EINT 0x0008 /* GPI7_EINT */ +#define WM8904_GPI7_EINT_MASK 0x0008 /* GPI7_EINT */ +#define WM8904_GPI7_EINT_SHIFT 3 /* GPI7_EINT */ +#define WM8904_GPI7_EINT_WIDTH 1 /* GPI7_EINT */ +#define WM8904_FLL_LOCK_EINT 0x0004 /* FLL_LOCK_EINT */ +#define WM8904_FLL_LOCK_EINT_MASK 0x0004 /* FLL_LOCK_EINT */ +#define WM8904_FLL_LOCK_EINT_SHIFT 2 /* FLL_LOCK_EINT */ +#define WM8904_FLL_LOCK_EINT_WIDTH 1 /* FLL_LOCK_EINT */ +#define WM8904_MIC_SHRT_EINT 0x0002 /* MIC_SHRT_EINT */ +#define WM8904_MIC_SHRT_EINT_MASK 0x0002 /* MIC_SHRT_EINT */ +#define WM8904_MIC_SHRT_EINT_SHIFT 1 /* MIC_SHRT_EINT */ +#define WM8904_MIC_SHRT_EINT_WIDTH 1 /* MIC_SHRT_EINT */ +#define WM8904_MIC_DET_EINT 0x0001 /* MIC_DET_EINT */ +#define WM8904_MIC_DET_EINT_MASK 0x0001 /* MIC_DET_EINT */ +#define WM8904_MIC_DET_EINT_SHIFT 0 /* MIC_DET_EINT */ +#define WM8904_MIC_DET_EINT_WIDTH 1 /* MIC_DET_EINT */ + +/* + * R128 (0x80) - Interrupt Status Mask + */ +#define WM8904_IM_GPIO_BCLK_EINT 0x0200 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_GPIO_BCLK_EINT_MASK 0x0200 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_GPIO_BCLK_EINT_SHIFT 9 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_GPIO_BCLK_EINT_WIDTH 1 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_WSEQ_EINT 0x0100 /* IM_WSEQ_EINT */ +#define WM8904_IM_WSEQ_EINT_MASK 0x0100 /* IM_WSEQ_EINT */ +#define WM8904_IM_WSEQ_EINT_SHIFT 8 /* IM_WSEQ_EINT */ +#define WM8904_IM_WSEQ_EINT_WIDTH 1 /* IM_WSEQ_EINT */ +#define WM8904_IM_GPIO3_EINT 0x0080 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO3_EINT_MASK 0x0080 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO3_EINT_SHIFT 7 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO3_EINT_WIDTH 1 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO2_EINT 0x0040 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO2_EINT_MASK 0x0040 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO2_EINT_SHIFT 6 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO2_EINT_WIDTH 1 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO1_EINT 0x0020 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPIO1_EINT_MASK 0x0020 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPIO1_EINT_SHIFT 5 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPIO1_EINT_WIDTH 1 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPI8_EINT 0x0010 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI8_EINT_MASK 0x0010 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI8_EINT_SHIFT 4 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI8_EINT_WIDTH 1 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI7_EINT 0x0008 /* IM_GPI7_EINT */ +#define WM8904_IM_GPI7_EINT_MASK 0x0008 /* IM_GPI7_EINT */ +#define WM8904_IM_GPI7_EINT_SHIFT 3 /* IM_GPI7_EINT */ +#define WM8904_IM_GPI7_EINT_WIDTH 1 /* IM_GPI7_EINT */ +#define WM8904_IM_FLL_LOCK_EINT 0x0004 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_FLL_LOCK_EINT_MASK 0x0004 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_FLL_LOCK_EINT_SHIFT 2 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_FLL_LOCK_EINT_WIDTH 1 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_MIC_SHRT_EINT 0x0002 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_SHRT_EINT_MASK 0x0002 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_SHRT_EINT_SHIFT 1 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_SHRT_EINT_WIDTH 1 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_DET_EINT 0x0001 /* IM_MIC_DET_EINT */ +#define WM8904_IM_MIC_DET_EINT_MASK 0x0001 /* IM_MIC_DET_EINT */ +#define WM8904_IM_MIC_DET_EINT_SHIFT 0 /* IM_MIC_DET_EINT */ +#define WM8904_IM_MIC_DET_EINT_WIDTH 1 /* IM_MIC_DET_EINT */ + +/* + * R129 (0x81) - Interrupt Polarity + */ +#define WM8904_GPIO_BCLK_EINT_POL 0x0200 /* GPIO_BCLK_EINT_POL */ +#define WM8904_GPIO_BCLK_EINT_POL_MASK 0x0200 /* GPIO_BCLK_EINT_POL */ +#define WM8904_GPIO_BCLK_EINT_POL_SHIFT 9 /* GPIO_BCLK_EINT_POL */ +#define WM8904_GPIO_BCLK_EINT_POL_WIDTH 1 /* GPIO_BCLK_EINT_POL */ +#define WM8904_WSEQ_EINT_POL 0x0100 /* WSEQ_EINT_POL */ +#define WM8904_WSEQ_EINT_POL_MASK 0x0100 /* WSEQ_EINT_POL */ +#define WM8904_WSEQ_EINT_POL_SHIFT 8 /* WSEQ_EINT_POL */ +#define WM8904_WSEQ_EINT_POL_WIDTH 1 /* WSEQ_EINT_POL */ +#define WM8904_GPIO3_EINT_POL 0x0080 /* GPIO3_EINT_POL */ +#define WM8904_GPIO3_EINT_POL_MASK 0x0080 /* GPIO3_EINT_POL */ +#define WM8904_GPIO3_EINT_POL_SHIFT 7 /* GPIO3_EINT_POL */ +#define WM8904_GPIO3_EINT_POL_WIDTH 1 /* GPIO3_EINT_POL */ +#define WM8904_GPIO2_EINT_POL 0x0040 /* GPIO2_EINT_POL */ +#define WM8904_GPIO2_EINT_POL_MASK 0x0040 /* GPIO2_EINT_POL */ +#define WM8904_GPIO2_EINT_POL_SHIFT 6 /* GPIO2_EINT_POL */ +#define WM8904_GPIO2_EINT_POL_WIDTH 1 /* GPIO2_EINT_POL */ +#define WM8904_GPIO1_EINT_POL 0x0020 /* GPIO1_EINT_POL */ +#define WM8904_GPIO1_EINT_POL_MASK 0x0020 /* GPIO1_EINT_POL */ +#define WM8904_GPIO1_EINT_POL_SHIFT 5 /* GPIO1_EINT_POL */ +#define WM8904_GPIO1_EINT_POL_WIDTH 1 /* GPIO1_EINT_POL */ +#define WM8904_GPI8_EINT_POL 0x0010 /* GPI8_EINT_POL */ +#define WM8904_GPI8_EINT_POL_MASK 0x0010 /* GPI8_EINT_POL */ +#define WM8904_GPI8_EINT_POL_SHIFT 4 /* GPI8_EINT_POL */ +#define WM8904_GPI8_EINT_POL_WIDTH 1 /* GPI8_EINT_POL */ +#define WM8904_GPI7_EINT_POL 0x0008 /* GPI7_EINT_POL */ +#define WM8904_GPI7_EINT_POL_MASK 0x0008 /* GPI7_EINT_POL */ +#define WM8904_GPI7_EINT_POL_SHIFT 3 /* GPI7_EINT_POL */ +#define WM8904_GPI7_EINT_POL_WIDTH 1 /* GPI7_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL 0x0004 /* FLL_LOCK_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL_MASK 0x0004 /* FLL_LOCK_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL_SHIFT 2 /* FLL_LOCK_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL_WIDTH 1 /* FLL_LOCK_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL 0x0002 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL_MASK 0x0002 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL_SHIFT 1 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL_WIDTH 1 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL 0x0001 /* MIC_DET_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL_MASK 0x0001 /* MIC_DET_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL_SHIFT 0 /* MIC_DET_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL_WIDTH 1 /* MIC_DET_EINT_POL */ + +/* + * R130 (0x82) - Interrupt Debounce + */ +#define WM8904_GPIO_BCLK_EINT_DB 0x0200 /* GPIO_BCLK_EINT_DB */ +#define WM8904_GPIO_BCLK_EINT_DB_MASK 0x0200 /* GPIO_BCLK_EINT_DB */ +#define WM8904_GPIO_BCLK_EINT_DB_SHIFT 9 /* GPIO_BCLK_EINT_DB */ +#define WM8904_GPIO_BCLK_EINT_DB_WIDTH 1 /* GPIO_BCLK_EINT_DB */ +#define WM8904_WSEQ_EINT_DB 0x0100 /* WSEQ_EINT_DB */ +#define WM8904_WSEQ_EINT_DB_MASK 0x0100 /* WSEQ_EINT_DB */ +#define WM8904_WSEQ_EINT_DB_SHIFT 8 /* WSEQ_EINT_DB */ +#define WM8904_WSEQ_EINT_DB_WIDTH 1 /* WSEQ_EINT_DB */ +#define WM8904_GPIO3_EINT_DB 0x0080 /* GPIO3_EINT_DB */ +#define WM8904_GPIO3_EINT_DB_MASK 0x0080 /* GPIO3_EINT_DB */ +#define WM8904_GPIO3_EINT_DB_SHIFT 7 /* GPIO3_EINT_DB */ +#define WM8904_GPIO3_EINT_DB_WIDTH 1 /* GPIO3_EINT_DB */ +#define WM8904_GPIO2_EINT_DB 0x0040 /* GPIO2_EINT_DB */ +#define WM8904_GPIO2_EINT_DB_MASK 0x0040 /* GPIO2_EINT_DB */ +#define WM8904_GPIO2_EINT_DB_SHIFT 6 /* GPIO2_EINT_DB */ +#define WM8904_GPIO2_EINT_DB_WIDTH 1 /* GPIO2_EINT_DB */ +#define WM8904_GPIO1_EINT_DB 0x0020 /* GPIO1_EINT_DB */ +#define WM8904_GPIO1_EINT_DB_MASK 0x0020 /* GPIO1_EINT_DB */ +#define WM8904_GPIO1_EINT_DB_SHIFT 5 /* GPIO1_EINT_DB */ +#define WM8904_GPIO1_EINT_DB_WIDTH 1 /* GPIO1_EINT_DB */ +#define WM8904_GPI8_EINT_DB 0x0010 /* GPI8_EINT_DB */ +#define WM8904_GPI8_EINT_DB_MASK 0x0010 /* GPI8_EINT_DB */ +#define WM8904_GPI8_EINT_DB_SHIFT 4 /* GPI8_EINT_DB */ +#define WM8904_GPI8_EINT_DB_WIDTH 1 /* GPI8_EINT_DB */ +#define WM8904_GPI7_EINT_DB 0x0008 /* GPI7_EINT_DB */ +#define WM8904_GPI7_EINT_DB_MASK 0x0008 /* GPI7_EINT_DB */ +#define WM8904_GPI7_EINT_DB_SHIFT 3 /* GPI7_EINT_DB */ +#define WM8904_GPI7_EINT_DB_WIDTH 1 /* GPI7_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB 0x0004 /* FLL_LOCK_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB_MASK 0x0004 /* FLL_LOCK_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB_SHIFT 2 /* FLL_LOCK_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB_WIDTH 1 /* FLL_LOCK_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB 0x0002 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB_MASK 0x0002 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB_SHIFT 1 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB_WIDTH 1 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB 0x0001 /* MIC_DET_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB_MASK 0x0001 /* MIC_DET_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB_SHIFT 0 /* MIC_DET_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB_WIDTH 1 /* MIC_DET_EINT_DB */ + +/* + * R134 (0x86) - EQ1 + */ +#define WM8904_EQ_ENA 0x0001 /* EQ_ENA */ +#define WM8904_EQ_ENA_MASK 0x0001 /* EQ_ENA */ +#define WM8904_EQ_ENA_SHIFT 0 /* EQ_ENA */ +#define WM8904_EQ_ENA_WIDTH 1 /* EQ_ENA */ + +/* + * R135 (0x87) - EQ2 + */ +#define WM8904_EQ_B1_GAIN_MASK 0x001F /* EQ_B1_GAIN - [4:0] */ +#define WM8904_EQ_B1_GAIN_SHIFT 0 /* EQ_B1_GAIN - [4:0] */ +#define WM8904_EQ_B1_GAIN_WIDTH 5 /* EQ_B1_GAIN - [4:0] */ + +/* + * R136 (0x88) - EQ3 + */ +#define WM8904_EQ_B2_GAIN_MASK 0x001F /* EQ_B2_GAIN - [4:0] */ +#define WM8904_EQ_B2_GAIN_SHIFT 0 /* EQ_B2_GAIN - [4:0] */ +#define WM8904_EQ_B2_GAIN_WIDTH 5 /* EQ_B2_GAIN - [4:0] */ + +/* + * R137 (0x89) - EQ4 + */ +#define WM8904_EQ_B3_GAIN_MASK 0x001F /* EQ_B3_GAIN - [4:0] */ +#define WM8904_EQ_B3_GAIN_SHIFT 0 /* EQ_B3_GAIN - [4:0] */ +#define WM8904_EQ_B3_GAIN_WIDTH 5 /* EQ_B3_GAIN - [4:0] */ + +/* + * R138 (0x8A) - EQ5 + */ +#define WM8904_EQ_B4_GAIN_MASK 0x001F /* EQ_B4_GAIN - [4:0] */ +#define WM8904_EQ_B4_GAIN_SHIFT 0 /* EQ_B4_GAIN - [4:0] */ +#define WM8904_EQ_B4_GAIN_WIDTH 5 /* EQ_B4_GAIN - [4:0] */ + +/* + * R139 (0x8B) - EQ6 + */ +#define WM8904_EQ_B5_GAIN_MASK 0x001F /* EQ_B5_GAIN - [4:0] */ +#define WM8904_EQ_B5_GAIN_SHIFT 0 /* EQ_B5_GAIN - [4:0] */ +#define WM8904_EQ_B5_GAIN_WIDTH 5 /* EQ_B5_GAIN - [4:0] */ + +/* + * R140 (0x8C) - EQ7 + */ +#define WM8904_EQ_B1_A_MASK 0xFFFF /* EQ_B1_A - [15:0] */ +#define WM8904_EQ_B1_A_SHIFT 0 /* EQ_B1_A - [15:0] */ +#define WM8904_EQ_B1_A_WIDTH 16 /* EQ_B1_A - [15:0] */ + +/* + * R141 (0x8D) - EQ8 + */ +#define WM8904_EQ_B1_B_MASK 0xFFFF /* EQ_B1_B - [15:0] */ +#define WM8904_EQ_B1_B_SHIFT 0 /* EQ_B1_B - [15:0] */ +#define WM8904_EQ_B1_B_WIDTH 16 /* EQ_B1_B - [15:0] */ + +/* + * R142 (0x8E) - EQ9 + */ +#define WM8904_EQ_B1_PG_MASK 0xFFFF /* EQ_B1_PG - [15:0] */ +#define WM8904_EQ_B1_PG_SHIFT 0 /* EQ_B1_PG - [15:0] */ +#define WM8904_EQ_B1_PG_WIDTH 16 /* EQ_B1_PG - [15:0] */ + +/* + * R143 (0x8F) - EQ10 + */ +#define WM8904_EQ_B2_A_MASK 0xFFFF /* EQ_B2_A - [15:0] */ +#define WM8904_EQ_B2_A_SHIFT 0 /* EQ_B2_A - [15:0] */ +#define WM8904_EQ_B2_A_WIDTH 16 /* EQ_B2_A - [15:0] */ + +/* + * R144 (0x90) - EQ11 + */ +#define WM8904_EQ_B2_B_MASK 0xFFFF /* EQ_B2_B - [15:0] */ +#define WM8904_EQ_B2_B_SHIFT 0 /* EQ_B2_B - [15:0] */ +#define WM8904_EQ_B2_B_WIDTH 16 /* EQ_B2_B - [15:0] */ + +/* + * R145 (0x91) - EQ12 + */ +#define WM8904_EQ_B2_C_MASK 0xFFFF /* EQ_B2_C - [15:0] */ +#define WM8904_EQ_B2_C_SHIFT 0 /* EQ_B2_C - [15:0] */ +#define WM8904_EQ_B2_C_WIDTH 16 /* EQ_B2_C - [15:0] */ + +/* + * R146 (0x92) - EQ13 + */ +#define WM8904_EQ_B2_PG_MASK 0xFFFF /* EQ_B2_PG - [15:0] */ +#define WM8904_EQ_B2_PG_SHIFT 0 /* EQ_B2_PG - [15:0] */ +#define WM8904_EQ_B2_PG_WIDTH 16 /* EQ_B2_PG - [15:0] */ + +/* + * R147 (0x93) - EQ14 + */ +#define WM8904_EQ_B3_A_MASK 0xFFFF /* EQ_B3_A - [15:0] */ +#define WM8904_EQ_B3_A_SHIFT 0 /* EQ_B3_A - [15:0] */ +#define WM8904_EQ_B3_A_WIDTH 16 /* EQ_B3_A - [15:0] */ + +/* + * R148 (0x94) - EQ15 + */ +#define WM8904_EQ_B3_B_MASK 0xFFFF /* EQ_B3_B - [15:0] */ +#define WM8904_EQ_B3_B_SHIFT 0 /* EQ_B3_B - [15:0] */ +#define WM8904_EQ_B3_B_WIDTH 16 /* EQ_B3_B - [15:0] */ + +/* + * R149 (0x95) - EQ16 + */ +#define WM8904_EQ_B3_C_MASK 0xFFFF /* EQ_B3_C - [15:0] */ +#define WM8904_EQ_B3_C_SHIFT 0 /* EQ_B3_C - [15:0] */ +#define WM8904_EQ_B3_C_WIDTH 16 /* EQ_B3_C - [15:0] */ + +/* + * R150 (0x96) - EQ17 + */ +#define WM8904_EQ_B3_PG_MASK 0xFFFF /* EQ_B3_PG - [15:0] */ +#define WM8904_EQ_B3_PG_SHIFT 0 /* EQ_B3_PG - [15:0] */ +#define WM8904_EQ_B3_PG_WIDTH 16 /* EQ_B3_PG - [15:0] */ + +/* + * R151 (0x97) - EQ18 + */ +#define WM8904_EQ_B4_A_MASK 0xFFFF /* EQ_B4_A - [15:0] */ +#define WM8904_EQ_B4_A_SHIFT 0 /* EQ_B4_A - [15:0] */ +#define WM8904_EQ_B4_A_WIDTH 16 /* EQ_B4_A - [15:0] */ + +/* + * R152 (0x98) - EQ19 + */ +#define WM8904_EQ_B4_B_MASK 0xFFFF /* EQ_B4_B - [15:0] */ +#define WM8904_EQ_B4_B_SHIFT 0 /* EQ_B4_B - [15:0] */ +#define WM8904_EQ_B4_B_WIDTH 16 /* EQ_B4_B - [15:0] */ + +/* + * R153 (0x99) - EQ20 + */ +#define WM8904_EQ_B4_C_MASK 0xFFFF /* EQ_B4_C - [15:0] */ +#define WM8904_EQ_B4_C_SHIFT 0 /* EQ_B4_C - [15:0] */ +#define WM8904_EQ_B4_C_WIDTH 16 /* EQ_B4_C - [15:0] */ + +/* + * R154 (0x9A) - EQ21 + */ +#define WM8904_EQ_B4_PG_MASK 0xFFFF /* EQ_B4_PG - [15:0] */ +#define WM8904_EQ_B4_PG_SHIFT 0 /* EQ_B4_PG - [15:0] */ +#define WM8904_EQ_B4_PG_WIDTH 16 /* EQ_B4_PG - [15:0] */ + +/* + * R155 (0x9B) - EQ22 + */ +#define WM8904_EQ_B5_A_MASK 0xFFFF /* EQ_B5_A - [15:0] */ +#define WM8904_EQ_B5_A_SHIFT 0 /* EQ_B5_A - [15:0] */ +#define WM8904_EQ_B5_A_WIDTH 16 /* EQ_B5_A - [15:0] */ + +/* + * R156 (0x9C) - EQ23 + */ +#define WM8904_EQ_B5_B_MASK 0xFFFF /* EQ_B5_B - [15:0] */ +#define WM8904_EQ_B5_B_SHIFT 0 /* EQ_B5_B - [15:0] */ +#define WM8904_EQ_B5_B_WIDTH 16 /* EQ_B5_B - [15:0] */ + +/* + * R157 (0x9D) - EQ24 + */ +#define WM8904_EQ_B5_PG_MASK 0xFFFF /* EQ_B5_PG - [15:0] */ +#define WM8904_EQ_B5_PG_SHIFT 0 /* EQ_B5_PG - [15:0] */ +#define WM8904_EQ_B5_PG_WIDTH 16 /* EQ_B5_PG - [15:0] */ + +/* + * R161 (0xA1) - Control Interface Test 1 + */ +#define WM8904_USER_KEY 0x0002 /* USER_KEY */ +#define WM8904_USER_KEY_MASK 0x0002 /* USER_KEY */ +#define WM8904_USER_KEY_SHIFT 1 /* USER_KEY */ +#define WM8904_USER_KEY_WIDTH 1 /* USER_KEY */ + +/* + * R204 (0xCC) - Analogue Output Bias 0 + */ +#define WM8904_PGA_BIAS_MASK 0x0070 /* PGA_BIAS - [6:4] */ +#define WM8904_PGA_BIAS_SHIFT 4 /* PGA_BIAS - [6:4] */ +#define WM8904_PGA_BIAS_WIDTH 3 /* PGA_BIAS - [6:4] */ + +/* + * R247 (0xF7) - FLL NCO Test 0 + */ +#define WM8904_FLL_FRC_NCO 0x0001 /* FLL_FRC_NCO */ +#define WM8904_FLL_FRC_NCO_MASK 0x0001 /* FLL_FRC_NCO */ +#define WM8904_FLL_FRC_NCO_SHIFT 0 /* FLL_FRC_NCO */ +#define WM8904_FLL_FRC_NCO_WIDTH 1 /* FLL_FRC_NCO */ + +/* + * R248 (0xF8) - FLL NCO Test 1 + */ +#define WM8904_FLL_FRC_NCO_VAL_MASK 0x003F /* FLL_FRC_NCO_VAL - [5:0] */ +#define WM8904_FLL_FRC_NCO_VAL_SHIFT 0 /* FLL_FRC_NCO_VAL - [5:0] */ +#define WM8904_FLL_FRC_NCO_VAL_WIDTH 6 /* FLL_FRC_NCO_VAL - [5:0] */ + +#endif -- cgit v1.2.2 From ffbfd336f9eac361e1630cfcb17a70607551daf2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 30 Nov 2009 17:56:11 +0100 Subject: ASoC: Add regulator support to CS4270 codec driver Signed-off-by: Daniel Mack Acked-by: Timur Tabi Cc: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 43 ++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 40 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ffe122d1cd76..8b5457542a0e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,6 +28,7 @@ #include #include #include +#include #include "cs4270.h" @@ -106,6 +107,10 @@ #define CS4270_MUTE_DAC_A 0x01 #define CS4270_MUTE_DAC_B 0x02 +static const char *supply_names[] = { + "va", "vd", "vlc" +}; + /* Private data for the CS4270 */ struct cs4270_private { struct snd_soc_codec codec; @@ -114,6 +119,9 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; unsigned int manual_mute; + + /* power domain regulators */ + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; /** @@ -579,7 +587,8 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; - int ret; + struct cs4270_private *cs4270 = codec->private_data; + int i, ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ socdev->card->codec = codec; @@ -599,6 +608,15 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } + /* get the power supply regulators */ + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4270->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + goto error_free_pcms; + return 0; error_free_pcms: @@ -616,8 +634,11 @@ error_free_pcms: static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; }; @@ -799,17 +820,33 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { struct snd_soc_codec *codec = cs4270_codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + struct cs4270_private *cs4270 = codec->private_data; + int reg, ret; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + if (reg < 0) + return reg; + + ret = snd_soc_write(codec, CS4270_PWRCTL, reg); + if (ret < 0) + return ret; + + regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + + return 0; } static int cs4270_soc_resume(struct platform_device *pdev) { struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; struct i2c_client *i2c_client = codec->control_data; int reg; + regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ ndelay(500); -- cgit v1.2.2 From e15c1c1f3f903f679c9782b540f9d52c80c99610 Mon Sep 17 00:00:00 2001 From: Dominik Brodowski Date: Sat, 28 Nov 2009 18:12:06 +0100 Subject: pcmcia: remove unused IRQ_FIRST_SHARED Komuro pointed out that IRQ_FIRST_SHARED is not used at all in the PCMCIA subsystem, so remove it. Also, remove two bogus assignments. CC: Karsten Keil CC: netdev@vger.kernel.org CC: alsa-devel@alsa-project.org CC: Komuro Signed-off-by: Dominik Brodowski --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 7717e01fc071..edaa729126bb 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -143,7 +143,8 @@ static int snd_pdacf_probe(struct pcmcia_device *link) link->io.NumPorts1 = 16; link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_FORCED_PULSE; - // link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED; + /* FIXME: This driver should be updated to allow for dynamic IRQ sharing */ + /* link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING | IRQ_FORCED_PULSE; */ link->irq.Handler = pdacf_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; -- cgit v1.2.2 From e6960e194a7dfb8197822225e04eca95fbd61a7f Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 4 Dec 2009 18:30:18 +0100 Subject: ALSA: opti93x: set MC indirect registers base from PnP data The PnP data on the OPTI931 and OPTI933 contains io port range for the MC indirect registers. Use the PnP range instead of hardwired value 0xE0E. Also, request region of MC indirect registers so it is marked as used to other drivers (this was missing previously). Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 112 ++++++++++++++++++++++--------------- 1 file changed, 67 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d08c38906449..8c88401c79bc 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,6 +135,8 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + unsigned long mc_indir_size; + struct resource *res_mc_indir; struct snd_wss *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -231,7 +233,10 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - chip->mc_indir_index = 0xe0e; + if (!chip->mc_indir_index) { + chip->mc_indir_index = 0xe0e; + chip->mc_indir_size = 2; + } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -560,57 +565,69 @@ static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) #endif /* OPTi93X */ -static int __devinit snd_card_opti9xx_detect(struct snd_card *card, - struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip) { - int i, err; + unsigned char value; +#ifdef OPTi93X + unsigned long flags; +#endif + chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -EBUSY; #ifndef OPTi93X - for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) { - unsigned char value; + value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)); + if (value != 0xff && value != inb(chip->mc_base + OPTi9XX_MC_REG(1))) + if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) + return 0; +#else /* OPTi93X */ + chip->res_mc_indir = request_region(chip->mc_indir_index, + chip->mc_indir_size, + "OPTi93x MC"); + if (chip->res_mc_indir == NULL) + return -EBUSY; - if ((err = snd_opti9xx_init(chip, i)) < 0) - return err; + spin_lock_irqsave(&chip->lock, flags); + outb(chip->password, chip->mc_base + chip->pwd_reg); + outb(((chip->mc_indir_index & 0x1f0) >> 4), chip->mc_base); + spin_unlock_irqrestore(&chip->lock, flags); - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; + value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)); + snd_opti9xx_write(chip, OPTi9XX_MC_REG(7), 0xff - value); + if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value) + return 0; - value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)); - if ((value != 0xff) && (value != inb(chip->mc_base + 1))) - if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) - return 1; + release_and_free_resource(chip->res_mc_indir); + chip->res_mc_indir = NULL; +#endif /* OPTi93X */ + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; + return -ENODEV; +} - } -#else /* OPTi93X */ - for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) { - unsigned long flags; - unsigned char value; +static int __devinit snd_card_opti9xx_detect(struct snd_card *card, + struct snd_opti9xx *chip) +{ + int i, err; - if ((err = snd_opti9xx_init(chip, i)) < 0) +#ifndef OPTi93X + for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) { +#else + for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) { +#endif + err = snd_opti9xx_init(chip, i); + if (err < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; - - spin_lock_irqsave(&chip->lock, flags); - outb(chip->password, chip->mc_base + chip->pwd_reg); - outb(((chip->mc_indir_index & (1 << 8)) >> 4) | - ((chip->mc_indir_index & 0xf0) >> 4), chip->mc_base); - spin_unlock_irqrestore(&chip->lock, flags); - - value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)); - snd_opti9xx_write(chip, OPTi9XX_MC_REG(7), 0xff - value); - if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value) + err = snd_opti9xx_read_check(chip); + if (err == 0) return 1; - - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; +#ifdef OPTi93X + chip->mc_indir_index = 0; +#endif } -#endif /* OPTi93X */ - return -ENODEV; } @@ -639,6 +656,8 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; + chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; + chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else if (pid->driver_data != 0x0924) port = pnp_port_start(pdev, 1); @@ -669,7 +688,7 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, static void snd_card_opti9xx_free(struct snd_card *card) { struct snd_opti9xx *chip = card->private_data; - + if (chip) { #ifdef OPTi93X struct snd_wss *codec = chip->codec; @@ -677,6 +696,7 @@ static void snd_card_opti9xx_free(struct snd_card *card) disable_irq(codec->irq); free_irq(codec->irq, codec); } + release_and_free_resource(chip->res_mc_indir); #endif release_and_free_resource(chip->res_mc_base); } @@ -696,11 +716,6 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) struct snd_rawmidi *rmidi; struct snd_hwdep *synth; - if (! chip->res_mc_base && - (chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, - "OPTi9xx MC")) == NULL) - return -ENOMEM; - #if defined(CS4231) || defined(OPTi93X) xdma2 = dma2; #else @@ -954,6 +969,13 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, } if (hw <= OPTi9XX_HW_82C930) chip->mc_base -= 0x80; + + error = snd_opti9xx_read_check(chip); + if (error) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return error; + } snd_card_set_dev(card, &pcard->card->dev); if ((error = snd_opti9xx_probe(card)) < 0) { snd_card_free(card); -- cgit v1.2.2 From 4b7e180335d23296170a5fa8c1f074722f94b253 Mon Sep 17 00:00:00 2001 From: "Justin P. Mattock" Date: Mon, 7 Dec 2009 15:07:46 -0800 Subject: ALSA: hda - iMac 9,1 sound patch. This is an updated patch for the Apple iMac 9,1 model to add sound. Original patch posted here: http://article.gmane.org/gmane.linux.alsa.devel/61361/match= I have been using this patch for a while now and have to say it works vary well, except for a few minor things: With the iMac 24-inch 3.06GHz Intel Core 2 Duo everything seems to be working as it should, although I have not looked into the microphone (never really use one, nor have any apps to test, my guess is it doesn't work, or I never figured out how to get it to work). With the iMac 24-inch 2.66GHz Intel Core 2 Duo everything is the same as with the above machine except I'm hearing a light scratchy/distortion noise come out of the speakers when using headphones(above machine does not do this). Other than that the sound level is great(especially with good Dj headphones). Signed-off-by: Justin P. Mattock Tested-by: Justin P. Mattock Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 111 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 111 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d967836f36bb..d0d14ed7ce81 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -208,6 +208,7 @@ enum { ALC885_MBP3, ALC885_MB5, ALC885_IMAC24, + ALC885_IMAC91, ALC883_3ST_2ch_DIG, ALC883_3ST_6ch_DIG, ALC883_3ST_6ch, @@ -7050,6 +7051,20 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_imac91_mixer[] = { + HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), + { } /* end */ +}; + + static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7505,6 +7520,66 @@ static struct hda_verb alc885_mbp3_init_verbs[] = { { } }; +/* iMac 9,1 */ +static struct hda_verb alc885_imac91_init_verbs[] = { + /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP Pin: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Internal Speakers: output 0 (0x0d) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Mic (rear) pin: input vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front Mic pin: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line In pin: use output 1 when in LineOut mode */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* ADC1: mute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* ADC2: mute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* ADC3: mute amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + + { } +}; + /* iMac 24 mixer. */ static struct snd_kcontrol_new alc885_imac24_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), @@ -7551,6 +7626,26 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_imac91_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_imac91_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_imac91_automute(codec); +} static struct hda_verb alc882_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -8718,6 +8813,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC885_MB5] = "mb5", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", + [ALC885_IMAC91] = "imac91", [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig", [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig", [ALC883_3ST_6ch] = "3stack-6ch", @@ -8891,6 +8987,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), + SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91), SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, * so apparently no perfect solution yet @@ -9002,6 +9099,20 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc885_imac24_setup, .init_hook = alc885_imac24_init_hook, }, + [ALC885_IMAC91] = { + .mixers = { alc885_imac91_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_imac91_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mbp_4ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), + .input_mux = &alc882_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_imac91_unsol_event, + .init_hook = alc885_imac91_automute, + }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, -- cgit v1.2.2 From 23033b2bce4361f2859ee0331f97c9056dae7091 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Dec 2009 12:36:52 +0100 Subject: ALSA: hda - Add missing Line-Out and PCM switches as slave Realtek codecs may have "PCM" and "Line-Out" playback switches, and they can be slaves for vmaster. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0d14ed7ce81..0fbcbeef1418 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2401,6 +2401,8 @@ static const char *alc_slave_sws[] = { "Speaker Playback Switch", "Mono Playback Switch", "IEC958 Playback Switch", + "Line-Out Playback Switch", + "PCM Playback Switch", NULL, }; -- cgit v1.2.2 From d11f74c62fb4a1fefd39085570fb6dfa7b9ab2bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Dec 2009 12:52:47 +0100 Subject: ALSA: hda - Exclude unusable ADCs for ALC88x On Realtek codecs, a digital mic pin is connected often only to a single ADC. But the parser tries to set up all ADCs no matter whether the digital mic is available, and results in non-selectable input source. This patch adds a check of input-source availability of each ADC, and excludes ones that don't support all input sources. Reference: Novell bnc#561235 http://bugzilla.novell.com/show_bug.cgi?id=561235 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0fbcbeef1418..2a96bc78964d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10021,10 +10021,12 @@ static int patch_alc882(struct hda_codec *codec) spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ if (!spec->adc_nids && spec->input_mux) { - int i; + int i, j; spec->num_adc_nids = 0; for (i = 0; i < ARRAY_SIZE(alc882_adc_nids); i++) { + const struct hda_input_mux *imux = spec->input_mux; hda_nid_t cap; + hda_nid_t items[16]; hda_nid_t nid = alc882_adc_nids[i]; unsigned int wcap = get_wcaps(codec, nid); /* get type */ @@ -10035,6 +10037,15 @@ static int patch_alc882(struct hda_codec *codec) err = snd_hda_get_connections(codec, nid, &cap, 1); if (err < 0) continue; + err = snd_hda_get_connections(codec, cap, items, + ARRAY_SIZE(items)); + if (err < 0) + continue; + for (j = 0; j < imux->num_items; j++) + if (imux->items[j].index >= err) + break; + if (j < imux->num_items) + continue; spec->private_capsrc_nids[spec->num_adc_nids] = cap; spec->num_adc_nids++; } -- cgit v1.2.2 From 2b6f6c0d11fcf6244b98d2b7490164d92d3e409f Mon Sep 17 00:00:00 2001 From: Tobias Hansen Date: Mon, 7 Dec 2009 19:08:19 +0100 Subject: ALSA: snd-usb-us122l: add product IDs of US-122MKII and US-144MKII I added the product IDs of the new revisions of the devices, so owners can test whether this suffices to make them work. Patched against ALSA snapshot 20091207. Signed-off-by: Tobias Hansen Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 28 ++++++++++++++++++++++------ sound/usb/usx2y/us122l.h | 2 ++ 2 files changed, 24 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index f71cd28eca6b..91bb29666d26 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -194,7 +194,8 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) if (!us122l->first) us122l->first = file; - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_get_interface(iface); } @@ -209,7 +210,8 @@ static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file) struct usb_interface *iface; snd_printdd(KERN_DEBUG "%p %p\n", hw, file); - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_put_interface(iface); } @@ -476,7 +478,8 @@ static bool us122l_create_card(struct snd_card *card) int err; struct us122l *us122l = US122L(card); - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -495,7 +498,8 @@ static bool us122l_create_card(struct snd_card *card) if (!us122l_start(us122l, 44100, 256)) return false; - if (us122l->dev->descriptor.idProduct == USB_ID_US144) + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) err = us144_create_usbmidi(card); else err = us122l_create_usbmidi(card); @@ -597,7 +601,8 @@ static int snd_us122l_probe(struct usb_interface *intf, struct snd_card *card; int err; - if (device->descriptor.idProduct == USB_ID_US144 + if ((device->descriptor.idProduct == USB_ID_US144 || + device->descriptor.idProduct == USB_ID_US144MKII) && device->speed == USB_SPEED_HIGH) { snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); return -ENODEV; @@ -692,7 +697,8 @@ static int snd_us122l_resume(struct usb_interface *intf) mutex_lock(&us122l->mutex); /* needed, doesn't restart without: */ - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -737,6 +743,16 @@ static struct usb_device_id snd_us122l_usb_id_table[] = { .idVendor = 0x0644, .idProduct = USB_ID_US144 }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US122MKII + }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US144MKII + }, { /* terminator */ } }; diff --git a/sound/usb/usx2y/us122l.h b/sound/usb/usx2y/us122l.h index 4daf1982e821..f263b3f96c86 100644 --- a/sound/usb/usx2y/us122l.h +++ b/sound/usb/usx2y/us122l.h @@ -25,5 +25,7 @@ struct us122l { #define USB_ID_US122L 0x800E #define USB_ID_US144 0x800F +#define USB_ID_US122MKII 0x8021 +#define USB_ID_US144MKII 0x8020 #endif -- cgit v1.2.2 From 370066e2b13bafa8e742673f658e617b6ed143a4 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Tue, 8 Dec 2009 01:34:22 +0100 Subject: ASoC: Wrong variable returned on error The wrong variable was returned in the case of an error Signed-off-by: Roel Kluin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/mx1_mx2-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c index b83866529397..bffffcd5ff34 100644 --- a/sound/soc/imx/mx1_mx2-pcm.c +++ b/sound/soc/imx/mx1_mx2-pcm.c @@ -322,12 +322,12 @@ static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) pr_debug("%s: Requesting dma channel (%s)\n", __func__, prtd->dma_params->name); - prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name, - DMA_PRIO_HIGH); - if (prtd->dma_ch < 0) { + ret = imx_dma_request_by_prio(prtd->dma_params->name, DMA_PRIO_HIGH); + if (ret < 0) { printk(KERN_ERR "Error %d requesting dma channel\n", ret); return ret; } + prtd->dma_ch = ret; imx_dma_config_burstlen(prtd->dma_ch, prtd->dma_params->watermark_level); -- cgit v1.2.2 From ee6e365e30f7ee89bd214ff1215aaf90e93d4c40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Dec 2009 17:23:33 +0100 Subject: ALSA: hda - Generalize EAPD inversion check in patch_analog.c Add a flag to spec field so that the EAPD inversion can be checked outside the relevant control callbacks. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 455a0494f907..447eda1f6770 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -72,7 +72,8 @@ struct ad198x_spec { hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; unsigned int jack_present :1; - unsigned int inv_jack_detect:1; + unsigned int inv_jack_detect:1; /* inverted jack-detection */ + unsigned int inv_eapd:1; /* inverted EAPD implementation */ #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; @@ -458,7 +459,7 @@ static struct hda_codec_ops ad198x_patch_ops = { /* * EAPD control - * the private value = nid | (invert << 8) + * the private value = nid */ #define ad198x_eapd_info snd_ctl_boolean_mono_info @@ -467,8 +468,7 @@ static int ad198x_eapd_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; - if (invert) + if (spec->inv_eapd) ucontrol->value.integer.value[0] = ! spec->cur_eapd; else ucontrol->value.integer.value[0] = spec->cur_eapd; @@ -480,11 +480,10 @@ static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; hda_nid_t nid = kcontrol->private_value & 0xff; unsigned int eapd; eapd = !!ucontrol->value.integer.value[0]; - if (invert) + if (spec->inv_eapd) eapd = !eapd; if (eapd == spec->cur_eapd) return 0; @@ -705,7 +704,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, - .private_value = 0x1b | (1 << 8), /* port-D, inversed */ + .private_value = 0x1b, /* port-D */ }, { } /* end */ }; @@ -1074,6 +1073,7 @@ static int patch_ad1986a(struct hda_codec *codec) spec->loopback.amplist = ad1986a_loopbacks; #endif spec->vmaster_nid = 0x1b; + spec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */ codec->patch_ops = ad198x_patch_ops; @@ -2124,7 +2124,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, - .private_value = 0x12 | (1 << 8), /* port-D, inversed */ + .private_value = 0x12, /* port-D */ }, { } /* end */ @@ -3065,6 +3065,7 @@ static int patch_ad1988(struct hda_codec *codec) spec->input_mux = &ad1988_laptop_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1988_laptop_mixers; + spec->inv_eapd = 1; /* inverted EAPD */ spec->num_init_verbs = 1; spec->init_verbs[0] = ad1988_laptop_init_verbs; if (board_config == AD1988_LAPTOP_DIG) -- cgit v1.2.2 From 396087eaead95fcb29eb36f1e59517aeb58c545e Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 9 Dec 2009 10:44:47 +0100 Subject: ALSA: hda - Terradici HDA controllers does not support 64-bit mode Confirmed from vendor and tests in RedHat bugzilla #536782 . Signed-off-by: Jaroslav Kysela Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d822bfc6cad6..efcc4f7c57f2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2450,6 +2450,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } } + /* disable 64bit DMA address for Teradici */ + /* it does not work with device 6549:1200 subsys e4a2:040b */ + if (chip->driver_type == AZX_DRIVER_TERA) + gcap &= ~ICH6_GCAP_64OK; + /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); -- cgit v1.2.2 From 7aee67466536bbf8bb44a95712c848a61c5a0acd Mon Sep 17 00:00:00 2001 From: David Santinoli Date: Wed, 9 Dec 2009 12:34:26 +0100 Subject: ALSA: hda/realtek: quirk for D945GCLF2 mainboard Quirk for the ALC662 found on the Intel D945GCLF2 (and possibly other) mainboards. Signed-off-by: David Santinoli Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2a96bc78964d..deecdd2d5d37 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16970,6 +16970,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x8086, 0xd604, "Intel mobo", ALC662_3ST_2ch_DIG), {} }; -- cgit v1.2.2 From 482e46d4b7c9bfbb2edc047fafa85cee1b0fc1e1 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 9 Dec 2009 12:43:44 +0100 Subject: ALSA: ice1724 - aureon - modify WM8770 Master & DAC volume The volume levels in original implementation are incorrect and does not match the dB scale. The real range is linear (in the sense of the dB scale) from 0dB to -100dB. Remove logaritmic table and make all volumes from range 0dB..100dB. The tests are in RedHat's bugzilla #540817. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/ice1712/aureon.c | 31 +++++++------------------------ 1 file changed, 7 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 110d16e52733..765d7bd4c3d4 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -689,32 +689,14 @@ static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -10000, 100, 1); static const DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1); static const DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0); -/* - * Logarithmic volume values for WM8770 - * Computed as 20 * Log10(255 / x) - */ -static const unsigned char wm_vol[256] = { - 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23, - 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17, - 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13, - 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11, - 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8, - 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, - 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, - 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, - 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0 -}; - -#define WM_VOL_MAX (sizeof(wm_vol) - 1) +#define WM_VOL_MAX 100 +#define WM_VOL_CNT 101 /* 0dB .. -100dB */ #define WM_VOL_MUTE 0x8000 static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) @@ -724,7 +706,8 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) nvol = 0; else - nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX]; + nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) / + WM_VOL_MAX; wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x180 | nvol); @@ -820,7 +803,7 @@ static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info * uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = voices; uinfo->value.integer.min = 0; /* mute (-101dB) */ - uinfo->value.integer.max = 0x7F; /* 0dB */ + uinfo->value.integer.max = WM_VOL_MAX; /* 0dB */ return 0; } @@ -850,7 +833,7 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * snd_ice1712_save_gpio_status(ice); for (i = 0; i < voices; i++) { unsigned int vol = ucontrol->value.integer.value[i]; - if (vol > 0x7f) + if (vol > WM_VOL_MAX) continue; vol |= spec->vol[ofs+i]; if (vol != spec->vol[ofs+i]) { -- cgit v1.2.2 From 5f60e496083efb01893a899b6885828330db971f Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 9 Dec 2009 20:12:43 +0100 Subject: ALSA: opti93x: fix irq releasing if the irq cannot be allocated Use the chip->irq to check if the irq should be released so the irq is not released if it has not been allocated. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 8c88401c79bc..d8eac3f28947 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -548,10 +548,13 @@ __skip_mpu: static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_wss *codec = dev_id; - struct snd_opti9xx *chip = codec->card->private_data; + struct snd_opti9xx *chip = dev_id; + struct snd_wss *codec = chip->codec; unsigned char status; + if (!codec) + return IRQ_HANDLED; + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); @@ -691,10 +694,9 @@ static void snd_card_opti9xx_free(struct snd_card *card) if (chip) { #ifdef OPTi93X - struct snd_wss *codec = chip->codec; - if (codec && codec->irq > 0) { - disable_irq(codec->irq); - free_irq(codec->irq, codec); + if (chip->irq > 0) { + disable_irq(chip->irq); + free_irq(chip->irq, chip); } release_and_free_resource(chip->res_mc_indir); #endif @@ -759,9 +761,9 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) #endif #ifdef OPTi93X error = request_irq(irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", codec); + IRQF_DISABLED, DEV_NAME" - WSS", chip); if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; } #endif -- cgit v1.2.2 From 6b2f3d1f769be5779b479c37800229d9a4809fc3 Mon Sep 17 00:00:00 2001 From: Christoph Hellwig Date: Tue, 27 Oct 2009 11:05:28 +0100 Subject: vfs: Implement proper O_SYNC semantics While Linux provided an O_SYNC flag basically since day 1, it took until Linux 2.4.0-test12pre2 to actually get it implemented for filesystems, since that day we had generic_osync_around with only minor changes and the great "For now, when the user asks for O_SYNC, we'll actually give O_DSYNC" comment. This patch intends to actually give us real O_SYNC semantics in addition to the O_DSYNC semantics. After Jan's O_SYNC patches which are required before this patch it's actually surprisingly simple, we just need to figure out when to set the datasync flag to vfs_fsync_range and when not. This patch renames the existing O_SYNC flag to O_DSYNC while keeping it's numerical value to keep binary compatibility, and adds a new real O_SYNC flag. To guarantee backwards compatiblity it is defined as expanding to both the O_DSYNC and the new additional binary flag (__O_SYNC) to make sure we are backwards-compatible when compiled against the new headers. This also means that all places that don't care about the differences can just check O_DSYNC and get the right behaviour for O_SYNC, too - only places that actuall care need to check __O_SYNC in addition. Drivers and network filesystems have been updated in a fail safe way to always do the full sync magic if O_DSYNC is set. The few places setting O_SYNC for lower layers are kept that way for now to stay failsafe. We enforce that O_DSYNC is set when __O_SYNC is set early in the open path to make sure we always get these sane options. Note that parisc really screwed up their headers as they already define a O_DSYNC that has always been a no-op. We try to repair it by using it for the new O_DSYNC and redefinining O_SYNC to send both the traditional O_SYNC numerical value _and_ the O_DSYNC one. Cc: Richard Henderson Cc: Ivan Kokshaysky Cc: Grant Grundler Cc: "David S. Miller" Cc: Ingo Molnar Cc: "H. Peter Anvin" Cc: Thomas Gleixner Cc: Al Viro Cc: Andreas Dilger Acked-by: Trond Myklebust Acked-by: Kyle McMartin Acked-by: Ulrich Drepper Signed-off-by: Christoph Hellwig Signed-off-by: Andrew Morton Signed-off-by: Jan Kara --- sound/core/rawmidi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 2f766123b158..0f5a194695d9 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1257,7 +1257,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, break; count -= count1; } - if (file->f_flags & O_SYNC) { + if (file->f_flags & O_DSYNC) { spin_lock_irq(&runtime->lock); while (runtime->avail != runtime->buffer_size) { wait_queue_t wait; -- cgit v1.2.2 From 761c9d45d14e0afa3c0b8eb84b4075602e50533b Mon Sep 17 00:00:00 2001 From: Olof Johansson Date: Thu, 10 Dec 2009 11:15:55 -0600 Subject: ASoC: Fix build of OMAP sound drivers There are build errors when building for some of the omap2/3 boards without enabling sound: sound/built-in.o:(.data+0x43bc): undefined reference to `soc_codec_dev_tlv320aic23' sound/built-in.o:(.data+0x43cc): undefined reference to `tlv320aic23_dai' Confused me quite a bit since the drivers that had references to the codec weren't enabled. Turns out the Makefile was using the wrong config option to enable them. Patch below. Reported-by: Anand Gadiyar Signed-off-by: Olof Johansson Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d49458a29bb7..3db8a6c523f4 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -23,9 +23,9 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o -obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o -obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- cgit v1.2.2 From c357aab02ee8de1f833579861ebd1e5683d2e806 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Fri, 11 Dec 2009 07:51:54 +0100 Subject: ALSA: hda - Fix LED GPIO setup for HP laptops with IDT codecs This patch fixes an error in processing of the HP BIOS configuration to enable GPIO based mute LED indicator control. That error causes driver to enable such control on all HP systems with the 92HD75 IDT codecs and results in unnecessary toggling of the GPIO on mute control manipulation. It also adds support of the future HP BIOS configuration extension for the named control. New configuration string has a format HP_Mute_LED_P_G where P can be 0 or 1 and defines mute LED GPIO control state (low/high) that corresponds to the NOT muted state of the master volume and G is the index of the GPIO to use (0..9) Lastly, it adds more systems to the support of the audio implementation as found on HP B-series systems Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 95 ++++++++++++++++++++++++++++++------------ 1 file changed, 68 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6b0bc040c3b1..e66672317e57 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -209,6 +209,7 @@ struct sigmatel_spec { unsigned int gpio_data; unsigned int gpio_mute; unsigned int gpio_led; + unsigned int gpio_led_polarity; /* stream */ unsigned int stream_delay; @@ -4724,13 +4725,61 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } -static int hp_bseries_system(u32 subsystem_id) +/* + * This method searches for the mute LED GPIO configuration + * provided as OEM string in SMBIOS. The format of that string + * is HP_Mute_LED_P_G or HP_Mute_LED_P + * where P can be 0 or 1 and defines mute LED GPIO control state (low/high) + * that corresponds to the NOT muted state of the master volume + * and G is the index of the GPIO to use as the mute LED control (0..9) + * If _G portion is missing it is assigned based on the codec ID + * + * So, HP B-series like systems may have HP_Mute_LED_0 (current models) + * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + */ +static int find_mute_led_gpio(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + const struct dmi_device *dev = NULL; + + if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, + NULL, dev))) { + if (sscanf(dev->name, "HP_Mute_LED_%d_%d", + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { + spec->gpio_led = 1 << spec->gpio_led; + return 1; + } + if (sscanf(dev->name, "HP_Mute_LED_%d", + &spec->gpio_led_polarity) == 1) { + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + return 1; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + return 1; + } + } + } + } + return 0; +} + +static int hp_blike_system(u32 subsystem_id) { switch (subsystem_id) { - case 0x103c307e: - case 0x103c307f: - case 0x103c3080: - case 0x103c3081: + case 0x103c1520: + case 0x103c1521: + case 0x103c1523: + case 0x103c1524: + case 0x103c1525: case 0x103c1722: case 0x103c1723: case 0x103c1724: @@ -4739,6 +4788,14 @@ static int hp_bseries_system(u32 subsystem_id) case 0x103c1727: case 0x103c1728: case 0x103c1729: + case 0x103c172a: + case 0x103c172b: + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c7007: + case 0x103c7008: return 1; } return 0; @@ -4833,7 +4890,7 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, else spec->gpio_data |= spec->gpio_led; /* white */ - if (hp_bseries_system(codec->subsystem_id)) { + if (!spec->gpio_led_polarity) { /* LED state is inverted on these systems */ spec->gpio_data ^= spec->gpio_led; } @@ -5526,7 +5583,7 @@ again: break; } - if (hp_bseries_system(codec->subsystem_id)) { + if (hp_blike_system(codec->subsystem_id)) { pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || @@ -5544,26 +5601,10 @@ again: } } - if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { - const struct dmi_device *dev = NULL; - while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, - NULL, dev))) { - if (strcmp(dev->name, "HP_Mute_LED_1")) { - switch (codec->vendor_id) { - case 0x111d7608: - spec->gpio_led = 0x01; - break; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - spec->gpio_led = 0x08; - break; - } - break; - } - } - } + if (find_mute_led_gpio(codec)) + snd_printd("mute LED gpio %d polarity %d\n", + spec->gpio_led, + spec->gpio_led_polarity); #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { -- cgit v1.2.2 From b923528ed29dc2d12832f76b1b9e05848d9de853 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 11 Dec 2009 12:28:33 +0800 Subject: ALSA: hda - show HBR(High Bit Rate) pin cap in procfs Note that the HBR capability only applies to HDMI pin. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_proc.c | 5 ++++- 2 files changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2d627613aea3..f9a084a1378e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -258,6 +258,7 @@ enum { #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ +#define AC_PINCAP_HBR (1<<27) /* High Bit Rate */ /* Vref status (used in pin cap) */ #define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */ #define AC_PINCAP_VREF_50 (1<<1) /* 50% */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 09476fc1ab64..8d381c891001 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -240,8 +240,11 @@ static void print_pin_caps(struct snd_info_buffer *buffer, /* Realtek uses this bit as a different meaning */ if ((codec->vendor_id >> 16) == 0x10ec) snd_iprintf(buffer, " R/L"); - else + else { + if (caps & AC_PINCAP_HBR) + snd_iprintf(buffer, " HBR"); snd_iprintf(buffer, " HDMI"); + } } if (caps & AC_PINCAP_TRIG_REQ) snd_iprintf(buffer, " Trigger"); -- cgit v1.2.2 From 728765b30a052317b6cb6111d4c4e66aba5c0099 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 11 Dec 2009 12:28:34 +0800 Subject: ALSA: intelhdmi - accept DisplayPort pin HDA036 spec states: DP (Display Port) indicates whether the Pin Complex Widget supports connection to a Display Port sink. Supported if set to 1. Note that it is possible for the pin widget to support more than one digital display connection type, e.g. HDMI and DP bit are both set to 1. Also export the DP pin cap in procfs. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 2 ++ sound/pci/hda/patch_intelhdmi.c | 2 +- 3 files changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index f9a084a1378e..9000d52fccca 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -255,6 +255,9 @@ enum { * in HD-audio specification */ #define AC_PINCAP_HDMI (1<<7) /* HDMI pin */ +#define AC_PINCAP_DP (1<<24) /* DisplayPort pin, can + * coexist with AC_PINCAP_HDMI + */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 8d381c891001..c9afc04adac8 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -246,6 +246,8 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " HDMI"); } } + if (caps & AC_PINCAP_DP) + snd_iprintf(buffer, " DP"); if (caps & AC_PINCAP_TRIG_REQ) snd_iprintf(buffer, " Trigger"); if (caps & AC_PINCAP_IMP_SENSE) diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 928df59be5d8..742f15eb3331 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -344,7 +344,7 @@ static int intel_hdmi_parse_codec(struct hda_codec *codec) break; case AC_WID_PIN: caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if (!(caps & AC_PINCAP_HDMI)) + if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) continue; if (intel_hdmi_add_pin(codec, nid) < 0) return -EINVAL; -- cgit v1.2.2 From 1ffc69a6e86aa9458046d1719957e091c8e95f7a Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 11 Dec 2009 12:28:35 +0800 Subject: ALSA: intelhdmi - channel mapping applies to Pin HDA036-A specifies that the Audio Sample Packet (ASP) Channel Mapping verbs apply to Digital Display Pin Complex instead of Converter. With this fix, channel mapping is working as expected for IbexPeak. Thanks to Marcin for pointing this out! Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 742f15eb3331..0d5dd1ba8205 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -436,14 +436,15 @@ static void hdmi_set_channel_count(struct hda_codec *codec, AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); } -static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid) +static void hdmi_debug_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid) { #ifdef CONFIG_SND_DEBUG_VERBOSE int i; int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, nid, 0, + slot = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_CHAN_SLOT, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", slot >> 4, slot & 0xf); @@ -619,7 +620,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, return ai->CA; } -static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t nid, +static void hdmi_setup_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid, struct hdmi_audio_infoframe *ai) { int i; @@ -633,11 +635,11 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t nid, */ for (i = 0; i < 8; i++) - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, (i << 4) | i); - hdmi_debug_channel_mapping(codec, nid); + hdmi_debug_channel_mapping(codec, pin_nid); } static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, @@ -676,7 +678,6 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, }; hdmi_setup_channel_allocation(codec, nid, &ai); - hdmi_setup_channel_mapping(codec, nid, &ai); for (i = 0; i < spec->num_pins; i++) { if (spec->pin_cvt[i] != nid) @@ -686,6 +687,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, pin_nid = spec->pin[i]; if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + hdmi_setup_channel_mapping(codec, pin_nid, &ai); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, &ai); hdmi_start_infoframe_trans(codec, pin_nid); -- cgit v1.2.2 From b14224bb74e19072c34617c501bceab94ebf579f Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 11 Dec 2009 12:28:36 +0800 Subject: ALSA: intelhdmi - add channel mapping for typical configurations IbexPeak is the first Intel HDMI audio codec to support channel mapping. Currently the outstanding problem is, the HDMI channel order do not agree with that of ALSA. This patch presents workaround for some typical use cases. It gives priority to the typical ALSA surround configurations, and defines channel mapping for them. We may need better kernel+userspace interactive channel mapping scheme. For example, in current scheme if user plays with the surround50 device, the kernel is unaware of this and will still select the surround41 channel allocation and channel mapping.. Thanks to Marcin for offering good tips! Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_intelhdmi.c | 89 +++++++++++++++++++++++++++++++---------- 1 file changed, 67 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 0d5dd1ba8205..3990182777ee 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -145,6 +145,42 @@ struct cea_channel_speaker_allocation { int spk_mask; }; +/* + * ALSA sequence is: + * + * surround40 surround41 surround50 surround51 surround71 + * ch0 front left = = = = + * ch1 front right = = = = + * ch2 rear left = = = = + * ch3 rear right = = = = + * ch4 LFE center center center + * ch5 LFE LFE + * ch6 side left + * ch7 side right + * + * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} + */ +static int hdmi_channel_mapping[0x32][8] = { + /* stereo */ + [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* 2.1 */ + [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* Dolby Surround */ + [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* surround40 */ + [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, + /* 4ch */ + [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, + /* surround41 */ + [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround50 */ + [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround51 */ + [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, + /* 7.1 */ + [0x13] = { 0x00, 0x11, 0x26, 0x37, 0x43, 0x52, 0x64, 0x75 }, +}; + /* * This is an ordered list! * @@ -152,32 +188,36 @@ struct cea_channel_speaker_allocation { * hdmi_setup_channel_allocation(). */ static struct cea_channel_speaker_allocation channel_allocations[] = { -/* channel: 8 7 6 5 4 3 2 1 */ +/* channel: 7 6 5 4 3 2 1 0 */ { .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, /* 2.1 */ { .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, /* Dolby Surround */ { .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, + /* surround40 */ +{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, + /* surround41 */ +{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, + /* surround50 */ +{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, + /* surround51 */ +{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, + /* 6.1 */ +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, + { .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, { .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, { .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, { .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, { .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, - /* 5.1 */ -{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, - /* 7.1 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, { .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, { .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, @@ -210,7 +250,6 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, }; - /* * HDA/HDMI auto parsing */ @@ -625,19 +664,25 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { int i; + int ca = ai->CA; + int err; - if (!ai->CA) - return; - - /* - * TODO: adjust channel mapping if necessary - * ALSA sequence is front/surr/clfe/side? - */ + if (hdmi_channel_mapping[ca][1] == 0) { + for (i = 0; i < channel_allocations[ca].channels; i++) + hdmi_channel_mapping[ca][i] = i | (i << 4); + for (; i < 8; i++) + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + } - for (i = 0; i < 8; i++) - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - (i << 4) | i); + for (i = 0; i < 8; i++) { + err = snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + hdmi_channel_mapping[ca][i]); + if (err) { + snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); + break; + } + } hdmi_debug_channel_mapping(codec, pin_nid); } -- cgit v1.2.2 From fcfdebe70759c74e2e701f69aaa7f0e5e32cf5a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 11 Dec 2009 12:51:05 +0100 Subject: ALSA: hrtimer - Fix lock-up The timer stop callback can be called from snd_timer_interrupt(), which is called from the hrtimer callback. Since hrtimer_cancel() waits for the callback completion, this eventually results in a lock-up. This patch fixes the problem by just toggling a flag at stop callback and call hrtimer_cancel() later. Reported-and-tested-by: Wojtek Zabolotny Cc: Signed-off-by: Takashi Iwai --- sound/core/hrtimer.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 34c7d48f5061..7f4d744ae40a 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -37,14 +37,22 @@ static unsigned int resolution; struct snd_hrtimer { struct snd_timer *timer; struct hrtimer hrt; + atomic_t running; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; + + if (!atomic_read(&stime->running)) + return HRTIMER_NORESTART; + hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); snd_timer_interrupt(stime->timer, t->sticks); + + if (!atomic_read(&stime->running)) + return HRTIMER_NORESTART; return HRTIMER_RESTART; } @@ -58,6 +66,7 @@ static int snd_hrtimer_open(struct snd_timer *t) hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); stime->timer = t; stime->hrt.function = snd_hrtimer_callback; + atomic_set(&stime->running, 0); t->private_data = stime; return 0; } @@ -78,16 +87,18 @@ static int snd_hrtimer_start(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; + atomic_set(&stime->running, 0); + hrtimer_cancel(&stime->hrt); hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution), HRTIMER_MODE_REL); + atomic_set(&stime->running, 1); return 0; } static int snd_hrtimer_stop(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - - hrtimer_cancel(&stime->hrt); + atomic_set(&stime->running, 0); return 0; } -- cgit v1.2.2 From 0287d970652027d5e299e0215578f228660a0e4e Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 11 Dec 2009 20:15:11 +0800 Subject: intelhdmi - dont power off HDA link For codecs without EPSS support (G45/IbexPeak), the hotplug event will be lost if the HDA is powered off during the time. After that the pin presence detection verb returns inaccurate info. So always power-on HDA link for !EPSS codecs. KarL offers the fact and Takashi recommends to flag hda_bus. Thanks! Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 3 ++- sound/pci/hda/patch_intelhdmi.c | 11 +++++++++++ 3 files changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9000d52fccca..1d541b7f5547 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -639,6 +639,7 @@ struct hda_bus { unsigned int rirb_error:1; /* error in codec communication */ unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ + unsigned int power_keep_link_on:1; /* don't power off HDA link */ }; /* diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index efcc4f7c57f2..e54420e691ae 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2082,7 +2082,8 @@ static void azx_power_notify(struct hda_bus *bus) } if (power_on) azx_init_chip(chip); - else if (chip->running && power_save_controller) + else if (chip->running && power_save_controller && + !bus->power_keep_link_on) azx_stop_chip(chip); } #endif /* CONFIG_SND_HDA_POWER_SAVE */ diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 3990182777ee..918f40378d52 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -391,6 +391,17 @@ static int intel_hdmi_parse_codec(struct hda_codec *codec) } } + /* + * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event + * can be lost and presence sense verb will become inaccurate if the + * HDA link is powered off at hot plug or hw initialization time. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & + AC_PWRST_EPSS)) + codec->bus->power_keep_link_on = 1; +#endif + return 0; } -- cgit v1.2.2 From 14ff3e78304e3f7fe18f950c3aa0686e6800b3fb Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 10 Dec 2009 20:39:28 +0100 Subject: ALSA: dt019x: merge into the als100 driver The als100 driver is so similar to the dt019x/als007 driver that one driver's source can be used for both drivers with only few changes. Merge the dt019x driver into the als100. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 21 +--- sound/isa/Makefile | 2 - sound/isa/als100.c | 121 ++++++++++++++------ sound/isa/dt019x.c | 321 ----------------------------------------------------- 4 files changed, 90 insertions(+), 375 deletions(-) delete mode 100644 sound/isa/dt019x.c (limited to 'sound') diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 02fe81ca88fd..194af3b01e13 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -63,15 +63,16 @@ config SND_AD1848 will be called snd-ad1848. config SND_ALS100 - tristate "Avance Logic ALS100/ALS120" + tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx" depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP help - Say Y here to include support for soundcards based on Avance - Logic ALS100, ALS110, ALS120 and ALS200 chips. + Say Y here to include support for soundcards based on the + Diamond Technologies DT-019X or Avance Logic chips: ALS007, + ALS100, ALS110, ALS120 and ALS200 chips. To compile this driver as a module, choose M here: the module will be called snd-als100. @@ -127,20 +128,6 @@ config SND_CS4236 To compile this driver as a module, choose M here: the module will be called snd-cs4236. -config SND_DT019X - tristate "Diamond Technologies DT-019X, Avance Logic ALS-007" - depends on PNP - select ISAPNP - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_SB16_DSP - help - Say Y here to include support for soundcards based on the - Diamond Technologies DT-019X or Avance Logic ALS-007 chips. - - To compile this driver as a module, choose M here: the module - will be called snd-dt019x. - config SND_ES968 tristate "Generic ESS ES968 driver" depends on PNP diff --git a/sound/isa/Makefile b/sound/isa/Makefile index b906b9a1a81e..c73d30c4f462 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -7,7 +7,6 @@ snd-adlib-objs := adlib.o snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8330-objs := cmi8330.o -snd-dt019x-objs := dt019x.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,7 +18,6 @@ obj-$(CONFIG_SND_ADLIB) += snd-adlib.o obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o -obj-$(CONFIG_SND_DT019X) += snd-dt019x.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 5fd52e4d7079..20becc89f6f6 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -2,9 +2,13 @@ /* card-als100.c - driver for Avance Logic ALS100 based soundcards. Copyright (C) 1999-2000 by Massimo Piccioni + Copyright (C) 1999-2002 by Massimo Piccioni Thanks to Pierfrancesco 'qM2' Passerini. + Generalised for soundcards based on DT-0196 and ALS-007 chips + by Jonathan Woithe : June 2002. + This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or @@ -33,10 +37,10 @@ #define PFX "als100: " -MODULE_AUTHOR("Massimo Piccioni "); -MODULE_DESCRIPTION("Avance Logic ALS1X0"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," +MODULE_DESCRIPTION("Avance Logic ALS007/ALS1X0"); +MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," + "{Avance Logic ALS-007}}" + "{{Avance Logic,ALS100 - PRO16PNP}," "{Avance Logic,ALS110}," "{Avance Logic,ALS120}," "{Avance Logic,ALS200}," @@ -45,9 +49,12 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," "{Avance Logic,ALS120}," "{RTL,RTL3000}}"); +MODULE_AUTHOR("Massimo Piccioni "); +MODULE_LICENSE("GPL"); + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ @@ -57,14 +64,15 @@ static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for als100 based soundcard."); +MODULE_PARM_DESC(index, "Index value for Avance Logic based soundcard."); module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for als100 based soundcard."); +MODULE_PARM_DESC(id, "ID string for Avance Logic based soundcard."); module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable als100 based soundcard."); +MODULE_PARM_DESC(enable, "Enable Avance Logic based soundcard."); + +MODULE_ALIAS("snd-dt019x"); struct snd_card_als100 { - int dev_no; struct pnp_dev *dev; struct pnp_dev *devmpu; struct pnp_dev *devopl; @@ -72,25 +80,43 @@ struct snd_card_als100 { }; static struct pnp_card_device_id snd_als100_pnpids[] = { + /* DT197A30 */ + { .id = "RWB1688", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_DT019X }, + /* DT0196 / ALS-007 */ + { .id = "ALS0007", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_DT019X }, /* ALS100 - PRO16PNP */ - { .id = "ALS0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, + { .id = "ALS0001", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS110 - MF1000 - Digimate 3D Sound */ - { .id = "ALS0110", .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } } }, + { .id = "ALS0110", + .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS120 */ - { .id = "ALS0120", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } }, + { .id = "ALS0120", + .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS200 */ - { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } } }, + { .id = "ALS0200", + .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS200 OEM */ - { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } } }, + { .id = "ALS0200", + .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } }, + .driver_data = SB_HW_ALS100 }, /* RTL3000 */ - { .id = "RTL3000", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } }, - { .id = "", } /* end */ + { .id = "RTL3000", + .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } }, + .driver_data = SB_HW_ALS100 }, + { .id = "" } /* end */ }; MODULE_DEVICE_TABLE(pnp_card, snd_als100_pnpids); -#define DRIVER_NAME "snd-card-als100" - static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, struct pnp_card_link *card, const struct pnp_card_device_id *id) @@ -113,8 +139,12 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, return err; } port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 1); - dma16[dev] = pnp_dma(pdev, 0); + if (id->driver_data == SB_HW_DT019X) + dma8[dev] = pnp_dma(pdev, 0); + else { + dma8[dev] = pnp_dma(pdev, 1); + dma16[dev] = pnp_dma(pdev, 0); + } irq[dev] = pnp_irq(pdev, 0); pdev = acard->devmpu; @@ -175,22 +205,33 @@ static int __devinit snd_card_als100_probe(int dev, } snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - dma16[dev], - SB_HW_ALS100, &chip)) < 0) { + if (pid->driver_data == SB_HW_DT019X) + dma16[dev] = -1; + + error = snd_sbdsp_create(card, port[dev], irq[dev], + snd_sb16dsp_interrupt, + dma8[dev], dma16[dev], + pid->driver_data, + &chip); + if (error < 0) { snd_card_free(card); return error; } acard->chip = chip; - strcpy(card->driver, "ALS100"); - strcpy(card->shortname, "Avance Logic ALS100"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev], dma16[dev]); + if (pid->driver_data == SB_HW_DT019X) { + strcpy(card->driver, "DT-019X"); + strcpy(card->shortname, "Diamond Tech. DT-019X"); + sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", + card->shortname, chip->name, chip->port, + irq[dev], dma8[dev]); + } else { + strcpy(card->driver, "ALS100"); + strcpy(card->shortname, "Avance Logic ALS100"); + sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", + card->shortname, chip->name, chip->port, + irq[dev], dma8[dev], dma16[dev]); + } if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { snd_card_free(card); @@ -203,9 +244,19 @@ static int __devinit snd_card_als100_probe(int dev, } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (snd_mpu401_uart_new(card, 0, MPU401_HW_ALS100, + int mpu_type = MPU401_HW_ALS100; + + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + + if (pid->driver_data == SB_HW_DT019X) + mpu_type = MPU401_HW_MPU401; + + if (snd_mpu401_uart_new(card, 0, + mpu_type, mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } @@ -291,7 +342,7 @@ static int snd_als100_pnp_resume(struct pnp_card_link *pcard) static struct pnp_card_driver als100_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, - .name = "als100", + .name = "als100", .id_table = snd_als100_pnpids, .probe = snd_als100_pnp_detect, .remove = __devexit_p(snd_als100_pnp_remove), @@ -312,7 +363,7 @@ static int __init alsa_card_als100_init(void) if (!als100_devices) { pnp_unregister_card_driver(&als100_pnpc_driver); #ifdef MODULE - snd_printk(KERN_ERR "no ALS100 based soundcards found\n"); + snd_printk(KERN_ERR "no Avance Logic based soundcards found\n"); #endif return -ENODEV; } diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c deleted file mode 100644 index 80f5b1af9be8..000000000000 --- a/sound/isa/dt019x.c +++ /dev/null @@ -1,321 +0,0 @@ - -/* - dt019x.c - driver for Diamond Technologies DT-0197H based soundcards. - Copyright (C) 1999, 2002 by Massimo Piccioni - - Generalised for soundcards based on DT-0196 and ALS-007 chips - by Jonathan Woithe : June 2002. - - This program is free software; you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#define PFX "dt019x: " - -MODULE_AUTHOR("Massimo Piccioni "); -MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," - "{Avance Logic ALS-007}}"); - -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ -static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ -static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ -static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ - -module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for DT-019X based soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard."); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard."); - -struct snd_card_dt019x { - struct pnp_dev *dev; - struct pnp_dev *devmpu; - struct pnp_dev *devopl; - struct snd_sb *chip; -}; - -static struct pnp_card_device_id snd_dt019x_pnpids[] = { - /* DT197A30 */ - { .id = "RWB1688", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - /* DT0196 / ALS-007 */ - { .id = "ALS0007", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - { .id = "", } -}; - -MODULE_DEVICE_TABLE(pnp_card, snd_dt019x_pnpids); - - -#define DRIVER_NAME "snd-card-dt019x" - - -static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, - struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - struct pnp_dev *pdev; - int err; - - acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (acard->dev == NULL) - return -ENODEV; - - acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL); - - pdev = acard->dev; - - err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR PFX "DT-019X AUDIO pnp configure failure\n"); - return err; - } - - port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 0); - irq[dev] = pnp_irq(pdev, 0); - snd_printdd("dt019x: found audio interface: port=0x%lx, irq=0x%x, dma=0x%x\n", - port[dev],irq[dev],dma8[dev]); - - pdev = acard->devmpu; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X MPU401 pnp configure failure, skipping\n"); - goto __mpu_error; - } - mpu_port[dev] = pnp_port_start(pdev, 0); - mpu_irq[dev] = pnp_irq(pdev, 0); - snd_printdd("dt019x: found MPU-401: port=0x%lx, irq=0x%x\n", - mpu_port[dev],mpu_irq[dev]); - } else { - __mpu_error: - acard->devmpu = NULL; - mpu_port[dev] = -1; - } - - pdev = acard->devopl; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X OPL3 pnp configure failure, skipping\n"); - goto __fm_error; - } - fm_port[dev] = pnp_port_start(pdev, 0); - snd_printdd("dt019x: found OPL3 synth: port=0x%lx\n",fm_port[dev]); - } else { - __fm_error: - acard->devopl = NULL; - fm_port[dev] = -1; - } - - return 0; -} - -static int __devinit snd_card_dt019x_probe(int dev, struct pnp_card_link *pcard, const struct pnp_card_device_id *pid) -{ - int error; - struct snd_sb *chip; - struct snd_card *card; - struct snd_card_dt019x *acard; - struct snd_opl3 *opl3; - - error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_dt019x), &card); - if (error < 0) - return error; - acard = card->private_data; - - snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_card_dt019x_pnp(dev, acard, pcard, pid))) { - snd_card_free(card); - return error; - } - - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - -1, - SB_HW_DT019X, - &chip)) < 0) { - snd_card_free(card); - return error; - } - acard->chip = chip; - - strcpy(card->driver, "DT-019X"); - strcpy(card->shortname, "Diamond Tech. DT-019X"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev]); - - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_sbmixer_new(chip)) < 0) { - snd_card_free(card); - return error; - } - - if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) - mpu_irq[dev] = -1; - if (snd_mpu401_uart_new(card, 0, -/* MPU401_HW_SB,*/ - MPU401_HW_MPU401, - mpu_port[dev], 0, - mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, - NULL) < 0) - snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]); - } - - if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, - fm_port[dev], - fm_port[dev] + 2, - OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx ?\n", - fm_port[dev], fm_port[dev] + 2); - } else { - if ((error = snd_opl3_timer_new(opl3, 0, 1)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { - snd_card_free(card); - return error; - } - } - } - - if ((error = snd_card_register(card)) < 0) { - snd_card_free(card); - return error; - } - pnp_set_card_drvdata(pcard, card); - return 0; -} - -static unsigned int __devinitdata dt019x_devices; - -static int __devinit snd_dt019x_pnp_probe(struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - static int dev; - int res; - - for ( ; dev < SNDRV_CARDS; dev++) { - if (!enable[dev]) - continue; - res = snd_card_dt019x_probe(dev, card, pid); - if (res < 0) - return res; - dev++; - dt019x_devices++; - return 0; - } - return -ENODEV; -} - -static void __devexit snd_dt019x_pnp_remove(struct pnp_card_link * pcard) -{ - snd_card_free(pnp_get_card_drvdata(pcard)); - pnp_set_card_drvdata(pcard, NULL); -} - -#ifdef CONFIG_PM -static int snd_dt019x_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); - snd_sbmixer_suspend(chip); - return 0; -} - -static int snd_dt019x_pnp_resume(struct pnp_card_link *pcard) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_sbdsp_reset(chip); - snd_sbmixer_resume(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif - -static struct pnp_card_driver dt019x_pnpc_driver = { - .flags = PNP_DRIVER_RES_DISABLE, - .name = "dt019x", - .id_table = snd_dt019x_pnpids, - .probe = snd_dt019x_pnp_probe, - .remove = __devexit_p(snd_dt019x_pnp_remove), -#ifdef CONFIG_PM - .suspend = snd_dt019x_pnp_suspend, - .resume = snd_dt019x_pnp_resume, -#endif -}; - -static int __init alsa_card_dt019x_init(void) -{ - int err; - - err = pnp_register_card_driver(&dt019x_pnpc_driver); - if (err) - return err; - - if (!dt019x_devices) { - pnp_unregister_card_driver(&dt019x_pnpc_driver); -#ifdef MODULE - snd_printk(KERN_ERR "no DT-019X / ALS-007 based soundcards found\n"); -#endif - return -ENODEV; - } - return 0; -} - -static void __exit alsa_card_dt019x_exit(void) -{ - pnp_unregister_card_driver(&dt019x_pnpc_driver); -} - -module_init(alsa_card_dt019x_init) -module_exit(alsa_card_dt019x_exit) -- cgit v1.2.2 From b2e8d7dab9d82be3851b8cbcc1ab64b1b2575844 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 10 Dec 2009 20:40:18 +0100 Subject: ALSA: opti93x: move controls definitions to opti93x driver Move OPTi93x controls definitions to the opti93x driver from the common wss-lib library module. These controls are used only by the opti93x driver. Also, fix capture source names. They are the same as opl3sa2 names. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 85 ++++++++++++++++++++++++++++++++++++++ sound/isa/wss/wss_lib.c | 80 +++++++---------------------------- 2 files changed, 100 insertions(+), 65 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 106be6e471f7..ea4a67120468 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include #include @@ -546,6 +547,85 @@ __skip_mpu: #ifdef OPTi93X +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); + +static struct snd_kcontrol_new snd_opti93x_controls[] = { +WSS_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), +WSS_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +WSS_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +WSS_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +}; + +static int __devinit snd_opti93x_mixer(struct snd_wss *chip) +{ + struct snd_card *card; + unsigned int idx; + struct snd_ctl_elem_id id1, id2; + int err; + + if (snd_BUG_ON(!chip || !chip->pcm)) + return -EINVAL; + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 switch to CD */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + return err; + } + /* reassign AUX1 switch to FM */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + return err; + } + /* remove AUX1 volume */ + strcpy(id1.name, "Aux Playback Volume"); id1.index = 1; + snd_ctl_remove_id(card, &id1); + + /* Replace WSS volume controls with OPTi93x volume controls */ + id1.index = 0; + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + strcpy(id1.name, snd_opti93x_controls[idx].name); + snd_ctl_remove_id(card, &id1); + + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], chip)); + if (err < 0) + return err; + } + return 0; +} + static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { struct snd_wss *codec = dev_id; @@ -752,6 +832,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) error = snd_wss_mixer(codec); if (error < 0) return error; +#ifdef OPTi93X + error = snd_opti93x_mixer(codec); + if (error < 0) + return error; +#endif #ifdef CS4231 error = snd_wss_timer(codec, 0, &timer); if (error < 0) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5b9d6c18bc45..9191b32d9130 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2014,6 +2014,7 @@ static int snd_wss_info_mux(struct snd_kcontrol *kcontrol, case WSS_HW_INTERWAVE: ptexts = gusmax_texts; break; + case WSS_HW_OPTI93X: case WSS_HW_OPL3SA2: ptexts = opl3sa_texts; break; @@ -2246,54 +2247,12 @@ WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), }; -static struct snd_kcontrol_new snd_opti93x_controls[] = { -WSS_DOUBLE("Master Playback Switch", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, - db_scale_6bit), -WSS_DOUBLE("PCM Playback Switch", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -WSS_DOUBLE("FM Playback Switch", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Line Playback Switch", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), -WSS_DOUBLE("Mic Playback Switch", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -WSS_DOUBLE("CD Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Aux Playback Switch", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_wss_info_mux, - .get = snd_wss_get_mux, - .put = snd_wss_put_mux, -} -}; - int snd_wss_mixer(struct snd_wss *chip) { struct snd_card *card; unsigned int idx; int err; + int count = ARRAY_SIZE(snd_wss_controls); if (snd_BUG_ON(!chip || !chip->pcm)) return -EINVAL; @@ -2302,28 +2261,19 @@ int snd_wss_mixer(struct snd_wss *chip) strcpy(card->mixername, chip->pcm->name); - if (chip->hardware == WSS_HW_OPTI93X) - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_opti93x_controls[idx], - chip)); - if (err < 0) - return err; - } - else { - int count = ARRAY_SIZE(snd_wss_controls); - - /* Use only the first 11 entries on AD1848 */ - if (chip->hardware & WSS_HW_AD1848_MASK) - count = 11; - - for (idx = 0; idx < count; idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_wss_controls[idx], - chip)); - if (err < 0) - return err; - } + /* Use only the first 11 entries on AD1848 */ + if (chip->hardware & WSS_HW_AD1848_MASK) + count = 11; + /* There is no loopback on OPTI93X */ + else if (chip->hardware == WSS_HW_OPTI93X) + count = 9; + + for (idx = 0; idx < count; idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_wss_controls[idx], + chip)); + if (err < 0) + return err; } return 0; } -- cgit v1.2.2 From 52dc438606d1ef78b96f56cc04dbea9242005730 Mon Sep 17 00:00:00 2001 From: Alexey Fisher Date: Sat, 12 Dec 2009 11:16:41 +0200 Subject: ALSA: hda - Overwrite pin config on intel DG45ID board. The pin config provided by BIOS have some problems: 0x0221401f: [Jack] HP Out at Ext Front <-- other association and sequence 0x02a19020: [Jack] Mic at Ext Front <-- other association 0x01113014: [Jack] Speaker at Ext Rear <-- line out (not speaker) 0x01114010: [Jack] Speaker at Ext Rear <-- line out 0x01a19030: [Jack] Mic at Ext Rear <-- other association 0x01111012: [Jack] Speaker at Ext Rear <-- line out 0x01116011: [Jack] Speaker at Ext Rear <-- line out 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x01451140: [Jack] SPDIF Out at Ext Rear 0x40f000f0: [N/A] Other at Ext N/A just overwrite it. Signed-off-by: Alexey Fisher Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e66672317e57..3d59f8325848 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1539,6 +1539,13 @@ static unsigned int alienware_m17x_pin_configs[13] = { 0x904601b0, }; +static unsigned int intel_dg45id_pin_configs[14] = { + 0x02214230, 0x02A19240, 0x01013214, 0x01014210, + 0x01A19250, 0x01011212, 0x01016211, 0x40f000f0, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x014510A0, + 0x074510B0, 0x40f000f0 +}; + static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, [STAC_DELL_M6_AMIC] = dell_m6_pin_configs, @@ -1546,6 +1553,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_DELL_M6_BOTH] = dell_m6_pin_configs, [STAC_DELL_EQ] = dell_m6_pin_configs, [STAC_ALIENWARE_M17X] = alienware_m17x_pin_configs, + [STAC_92HD73XX_INTEL] = intel_dg45id_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { -- cgit v1.2.2 From e9d0a803c127e2e30afb0df780ccb3af4e2adb28 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 12 Dec 2009 09:51:03 +0100 Subject: ALSA: opti93x: use dB scale for mixer controls Add dB scale for mixer controls. Fix dB scale for Master Volume control. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 36 ++++++++++++++++++++++-------------- 1 file changed, 22 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index ea4a67120468..b0ea310c87de 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -547,32 +547,40 @@ __skip_mpu: #ifdef OPTi93X -static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_step, -9300, 300, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit_12db_max, -3300, 300, 0); static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, - db_scale_6bit), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -WSS_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), + db_scale_5bit_3db_step), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1, + db_scale_5bit), +WSS_DOUBLE_TLV("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE_TLV("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), +WSS_DOUBLE_TLV("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), }; static int __devinit snd_opti93x_mixer(struct snd_wss *chip) -- cgit v1.2.2 From 5a65edbc12b6b34ef912114f1fc8215786f85b25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Nov 2009 16:10:51 +0000 Subject: mfd: Convert wm8350 IRQ handlers to irq_handler_t This is done as simple code transformation, the semantics of the IRQ API provided by the core are are still very different to those of genirq (mainly with regard to masking). Signed-off-by: Mark Brown Signed-off-by: Samuel Ortiz --- sound/soc/codecs/wm8350.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index f82125d9e85a..17a327d67fd5 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1340,9 +1340,10 @@ static int wm8350_resume(struct platform_device *pdev) return 0; } -static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) +static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; + struct wm8350 *wm8350 = priv->codec.control_data; u16 reg; int report; int mask; @@ -1365,7 +1366,7 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) if (!jack->jack) { dev_warn(wm8350->dev, "Jack interrupt called with no jack\n"); - return; + return IRQ_NONE; } /* Debounce */ @@ -1378,6 +1379,8 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) report = 0; snd_soc_jack_report(jack->jack, report, jack->report); + + return IRQ_HANDLED; } /** @@ -1421,7 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); /* Sync status */ - wm8350_hp_jack_handler(wm8350, irq, priv); + wm8350_hp_jack_handler(irq, priv); wm8350_unmask_irq(wm8350, irq); @@ -1485,9 +1488,11 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, - wm8350_hp_jack_handler, priv); + wm8350_hp_jack_handler, 0, "Left jack detect", + priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, - wm8350_hp_jack_handler, priv); + wm8350_hp_jack_handler, 0, "Right jack detect", + priv); ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { -- cgit v1.2.2 From 6a6127462eb9096419fd4b3115ec5971d83a600f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Nov 2009 16:10:52 +0000 Subject: mfd: Mask and unmask wm8350 IRQs on request and free Bring the WM8350 IRQ API more in line with the generic IRQ API by masking and unmasking interrupts as they are requested and freed. This is mostly just a case of deleting the mask and unmask calls from the individual drivers. The RTC driver is changed to mask the periodic IRQ after requesting it rather than only unmasking the alarm IRQ. If the periodic IRQ fires in the period where it is reqested then there will be a spurious notification but there should be no serious consequences from this. The CODEC drive is changed to explicitly disable headphone jack detection prior to requesting the IRQs. This will avoid the IRQ firing with no jack set up. Signed-off-by: Mark Brown Signed-off-by: Samuel Ortiz --- sound/soc/codecs/wm8350.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 17a327d67fd5..ebbf11b653a4 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1426,8 +1426,6 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, /* Sync status */ wm8350_hp_jack_handler(irq, priv); - wm8350_unmask_irq(wm8350, irq); - return 0; } EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); @@ -1485,8 +1483,10 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, WM8350_OUT2_VU | WM8350_OUT2R_MUTE); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + /* Make sure jack detect is disabled to start off with */ + wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, + WM8350_JDL_ENA | WM8350_JDR_ENA); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, wm8350_hp_jack_handler, 0, "Left jack detect", priv); @@ -1521,8 +1521,6 @@ static int wm8350_remove(struct platform_device *pdev) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); -- cgit v1.2.2 From b07682b6056eb6701f8cb86aa5800e6f2ea7919b Mon Sep 17 00:00:00 2001 From: Santosh Shilimkar Date: Sun, 13 Dec 2009 20:05:51 +0100 Subject: mfd: Rename twl4030* driver files to enable re-use The upcoming TWL6030 is companion chip for OMAP4 like the current TWL4030 for OMAP3. The common modules like RTC, Regulator creates opportunity to re-use the most of the code from twl4030. This patch renames few common drivers twl4030* files to twl* to enable the code re-use. Signed-off-by: Rajendra Nayak Signed-off-by: Balaji T K Signed-off-by: Santosh Shilimkar Acked-by: Kevin Hilman Signed-off-by: Samuel Ortiz --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 5f1681f6ca76..c3a6ceb542cb 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -26,7 +26,7 @@ #include #include #include -#include +#include #include #include #include -- cgit v1.2.2 From fc7b92fca4e546184557f1c53f84ad57c66b7695 Mon Sep 17 00:00:00 2001 From: Balaji T K Date: Sun, 13 Dec 2009 21:23:33 +0100 Subject: mfd: Rename all twl4030_i2c* This patch renames function names like twl4030_i2c_write_u8, twl4030_i2c_read_u8 to twl_i2c_write_u8, twl_i2c_read_u8 and also common variable in twl-core.c Signed-off-by: Rajendra Nayak Signed-off-by: Balaji T K Signed-off-by: Santosh Shilimkar Acked-by: Kevin Hilman Signed-off-by: Samuel Ortiz --- sound/soc/codecs/twl4030.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c3a6ceb542cb..2a27f7b56726 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -175,7 +175,7 @@ static int twl4030_write(struct snd_soc_codec *codec, { twl4030_write_reg_cache(codec, reg, value); if (likely(reg < TWL4030_REG_SW_SHADOW)) - return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); else return 0; @@ -261,7 +261,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec) do { /* this takes a little while, so don't slam i2c */ udelay(2000); - twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, TWL4030_REG_ANAMICL); } while ((i++ < 100) && ((byte & TWL4030_CNCL_OFFSET_START) == @@ -542,7 +542,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ break; \ case SND_SOC_DAPM_POST_PMD: \ reg_val = twl4030_read_reg_cache(w->codec, reg); \ - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ reg_val & (~mask), \ reg); \ break; \ @@ -679,7 +679,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / twl4030->sysclk) + 1); /* Bypass the reg_cache to mute the headset */ - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain & (~0x0f), TWL4030_REG_HS_GAIN_SET); -- cgit v1.2.2 From 950200e2ff11daae1c5d9426703bdd494603f38b Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 13 Dec 2009 14:11:02 -0500 Subject: ALSA: hda: Use ALC260_WILL quirk for another Acer model (0x1025007f) BugLink: https://bugs.launchpad.net/bugs/418627 The original reporter states that this quirk is necessary to obtain reasonable gain for playback. Without it, sound is inaudible. Tested with playback (spkr and hp) and capture. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index deecdd2d5d37..c9e860709747 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6248,6 +6248,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), + SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), -- cgit v1.2.2 From 01f5966d2f36f08eb6604665eade69c9f38ffaed Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 13 Dec 2009 16:22:58 -0500 Subject: ALSA: hda: Fix max PCM level to 0 dB for AD1981_HP BugLink: https://bugs.launchpad.net/bugs/461062 The original reporter states that PCM maxes at +12 dB and results in very bad distortion. Cap PCM at 0 dB to resolve this symptom. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 447eda1f6770..1a36137e13ec 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1789,6 +1789,14 @@ static int patch_ad1981(struct hda_codec *codec) codec->patch_ops.init = ad1981_hp_init; codec->patch_ops.unsol_event = ad1981_hp_unsol_event; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); break; case AD1981_THINKPAD: spec->mixers[0] = ad1981_thinkpad_mixers; -- cgit v1.2.2 From 0d64b568fcd48b133721c1d322e7c51d85eb12df Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 13 Dec 2009 12:42:56 +0100 Subject: ALSA: sound/isa/gus: Correct code taking the size of a pointer sizeof(share_id) is just the size of the pointer. On the other hand, block->share_id is an array, so its size seems more appropriate. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ expression *x; expression f; type T; @@ *f(...,(T)x,...) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_mem.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 661205c4dcea..af888a022fc0 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -127,7 +127,8 @@ static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc, !share_id[2] && !share_id[3]) return NULL; for (block = alloc->first; block; block = block->next) - if (!memcmp(share_id, block->share_id, sizeof(share_id))) + if (!memcmp(share_id, block->share_id, + sizeof(block->share_id))) return block; return NULL; } -- cgit v1.2.2 From 74c2b45b714e49b427584b4bd8f44f1a24d82d9c Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 13 Dec 2009 21:13:44 +0100 Subject: ALSA: sb_mixer: convert pointer tables to mixer control tables Convert table of pointers to mixer controls into tables of the mixer controls. It saves about 20% of the snd-sb-common module size reported by lsmod. The als4000 uses part of sb16's control table. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sb/sb_mixer.c | 330 +++++++++++++++++------------------------------- 1 file changed, 115 insertions(+), 215 deletions(-) (limited to 'sound') diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 318ff0c823e7..8cfc41fbe368 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -528,20 +528,11 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty * SB 2.0 specific mixer elements */ -static struct sbmix_elem snd_sb20_ctl_master_play_vol = - SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_pcm_play_vol = - SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3); -static struct sbmix_elem snd_sb20_ctl_synth_play_vol = - SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_cd_play_vol = - SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7); - -static struct sbmix_elem *snd_sb20_controls[] = { - &snd_sb20_ctl_master_play_vol, - &snd_sb20_ctl_pcm_play_vol, - &snd_sb20_ctl_synth_play_vol, - &snd_sb20_ctl_cd_play_vol +static struct sbmix_elem snd_sb20_controls[] = { + SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7), + SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3), + SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7), + SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7) }; static unsigned char snd_sb20_init_values[][2] = { @@ -552,41 +543,24 @@ static unsigned char snd_sb20_init_values[][2] = { /* * SB Pro specific mixer elements */ -static struct sbmix_elem snd_sbpro_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_filter = - SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3); -static struct sbmix_elem snd_sbpro_ctl_capture_source = +static struct sbmix_elem snd_sbpro_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7), + SB_DOUBLE("PCM Playback Volume", + SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7), + SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7), + SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7), + SB_DOUBLE("Line Playback Volume", + SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7), + SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3), { .name = "Capture Source", .type = SB_MIX_CAPTURE_PRO - }; -static struct sbmix_elem snd_sbpro_ctl_capture_filter = - SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_capture_low_filter = - SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1); - -static struct sbmix_elem *snd_sbpro_controls[] = { - &snd_sbpro_ctl_master_play_vol, - &snd_sbpro_ctl_pcm_play_vol, - &snd_sbpro_ctl_pcm_play_filter, - &snd_sbpro_ctl_synth_play_vol, - &snd_sbpro_ctl_cd_play_vol, - &snd_sbpro_ctl_line_play_vol, - &snd_sbpro_ctl_mic_play_vol, - &snd_sbpro_ctl_capture_source, - &snd_sbpro_ctl_capture_filter, - &snd_sbpro_ctl_capture_low_filter + }, + SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1), + SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1) }; static unsigned char snd_sbpro_init_values[][2] = { @@ -598,68 +572,42 @@ static unsigned char snd_sbpro_init_values[][2] = { /* * SB16 specific mixer elements */ -static struct sbmix_elem snd_sb16_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_3d_enhance_switch = - SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1); -static struct sbmix_elem snd_sb16_ctl_tone_bass = - SB_DOUBLE("Tone Control - Bass", SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_tone_treble = - SB_DOUBLE("Tone Control - Treble", SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_synth_capture_route = - SB16_INPUT_SW("Synth Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5); -static struct sbmix_elem snd_sb16_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_cd_capture_route = - SB16_INPUT_SW("CD Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_switch = - SB_DOUBLE("CD Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_line_capture_route = - SB16_INPUT_SW("Line Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3); -static struct sbmix_elem snd_sb16_ctl_line_play_switch = - SB_DOUBLE("Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1); -static struct sbmix_elem snd_sb16_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_mic_capture_route = - SB16_INPUT_SW("Mic Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0); -static struct sbmix_elem snd_sb16_ctl_mic_play_switch = - SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1); -static struct sbmix_elem snd_sb16_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31); -static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3); -static struct sbmix_elem snd_sb16_ctl_capture_vol = - SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_play_vol = - SB_DOUBLE("Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_auto_mic_gain = - SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1); - -static struct sbmix_elem *snd_sb16_controls[] = { - &snd_sb16_ctl_master_play_vol, - &snd_sb16_ctl_3d_enhance_switch, - &snd_sb16_ctl_tone_bass, - &snd_sb16_ctl_tone_treble, - &snd_sb16_ctl_pcm_play_vol, - &snd_sb16_ctl_synth_capture_route, - &snd_sb16_ctl_synth_play_vol, - &snd_sb16_ctl_cd_capture_route, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_cd_play_vol, - &snd_sb16_ctl_line_capture_route, - &snd_sb16_ctl_line_play_switch, - &snd_sb16_ctl_line_play_vol, - &snd_sb16_ctl_mic_capture_route, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_mic_play_vol, - &snd_sb16_ctl_pc_speaker_vol, - &snd_sb16_ctl_capture_vol, - &snd_sb16_ctl_play_vol, - &snd_sb16_ctl_auto_mic_gain +static struct sbmix_elem snd_sb16_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31), + SB_DOUBLE("PCM Playback Volume", + SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Synth Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5), + SB_DOUBLE("Synth Playback Volume", + SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("CD Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Mic Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), + SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3), + SB_DOUBLE("Capture Volume", + SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3), + SB_DOUBLE("Playback Volume", + SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3), + SB16_INPUT_SW("Line Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31), + SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1), + SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1), + SB_DOUBLE("Tone Control - Bass", + SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15), + SB_DOUBLE("Tone Control - Treble", + SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15) }; static unsigned char snd_sb16_init_values[][2] = { @@ -678,46 +626,34 @@ static unsigned char snd_sb16_init_values[][2] = { /* * DT019x specific mixer elements */ -static struct sbmix_elem snd_dt019x_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7); -static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7); -static struct sbmix_elem snd_dt019x_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch = - SB_DOUBLE("PCM Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2,1, 1); -static struct sbmix_elem snd_dt019x_ctl_synth_play_switch = - SB_DOUBLE("Synth Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4,3, 1); -static struct sbmix_elem snd_dt019x_ctl_capture_source = +static struct sbmix_elem snd_dt019x_controls[] = { + /* ALS4000 below has some parts which we might be lacking, + * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ + SB_DOUBLE("Master Playback Volume", + SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4, 0, 15), + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("PCM Playback Volume", + SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4, 0, 15), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4, 0, 15), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4, 0, 15), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7), + SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4, 0, 15), { .name = "Capture Source", .type = SB_MIX_CAPTURE_DT019X - }; - -static struct sbmix_elem *snd_dt019x_controls[] = { - /* ALS4000 below has some parts which we might be lacking, - * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ - &snd_dt019x_ctl_master_play_vol, - &snd_dt019x_ctl_pcm_play_vol, - &snd_dt019x_ctl_synth_play_vol, - &snd_dt019x_ctl_cd_play_vol, - &snd_dt019x_ctl_mic_play_vol, - &snd_dt019x_ctl_pc_speaker_vol, - &snd_dt019x_ctl_line_play_vol, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_line_play_switch, - &snd_dt019x_ctl_pcm_play_switch, - &snd_dt019x_ctl_synth_play_switch, - &snd_dt019x_ctl_capture_source + } }; static unsigned char snd_dt019x_init_values[][2] = { @@ -735,82 +671,37 @@ static unsigned char snd_dt019x_init_values[][2] = { /* * ALS4000 specific mixer elements */ -static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch = - SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1); -static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = { +static struct sbmix_elem snd_als4000_controls[] = { + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03), + SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1), + { .name = "Master Mono Capture Route", .type = SB_MIX_MONO_CAPTURE_ALS4K - }; -static struct sbmix_elem snd_als4000_ctl_mono_playback_switch = - SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1); -static struct sbmix_elem snd_als4000_ctl_mic_20db_boost = - SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03); -static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback = - SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); -static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback = + }, + SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1), + SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01), + SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01), SB_SINGLE("Digital Loopback Switch", - SB_ALS4000_CR3_CONFIGURATION, 7, 0x01); -/* FIXME: functionality of 3D controls might be swapped, I didn't find - * a description of how to identify what is supposed to be what */ -static struct sbmix_elem snd_als4000_3d_control_switch = - SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01); -static struct sbmix_elem snd_als4000_3d_control_ratio = - SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07); -static struct sbmix_elem snd_als4000_3d_control_freq = + SB_ALS4000_CR3_CONFIGURATION, 7, 0x01), + /* FIXME: functionality of 3D controls might be swapped, I didn't find + * a description of how to identify what is supposed to be what */ + SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07), /* FIXME: maybe there's actually some standard 3D ctrl name for it?? */ - SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03); -static struct sbmix_elem snd_als4000_3d_control_delay = + SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03), /* FIXME: ALS4000a.pdf mentions BBD (Bucket Brigade Device) time delay, * but what ALSA 3D attribute is that actually? "Center", "Depth", * "Wide" or "Space" or even "Level"? Assuming "Wide" for now... */ - SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f); -static struct sbmix_elem snd_als4000_3d_control_poweroff_switch = - SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01); -static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch = + SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f), + SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01), SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch", - SB_ALS4000_FMDAC, 5, 0x01); + SB_ALS4000_FMDAC, 5, 0x01), #ifdef NOT_AVAILABLE -static struct sbmix_elem snd_als4000_ctl_fmdac = - SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01); -static struct sbmix_elem snd_als4000_ctl_qsound = - SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f); -#endif - -static struct sbmix_elem *snd_als4000_controls[] = { - /* ALS4000a.PDF regs page */ - &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */ - &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */ - &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */ - &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */ - &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */ - &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_line_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */ - &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */ - &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */ - &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */ - &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */ - &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */ - &snd_sb16_ctl_play_vol, /* MX41/42 15 */ - &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */ - &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */ - &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */ - &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */ - &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */ - &snd_als4000_3d_control_switch, /* MX50 17 */ - &snd_als4000_3d_control_ratio, /* MX50 17 */ - &snd_als4000_3d_control_freq, /* MX50 17 */ - &snd_als4000_3d_control_delay, /* MX51 18 */ - &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */ - &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */ -#ifdef NOT_AVAILABLE - &snd_als4000_ctl_fmdac, - &snd_als4000_ctl_qsound, + SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01), + SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f), #endif }; @@ -829,11 +720,10 @@ static unsigned char snd_als4000_init_values[][2] = { { SB_ALS4000_MIC_IN_GAIN, 0 }, }; - /* */ static int snd_sbmixer_init(struct snd_sb *chip, - struct sbmix_elem **controls, + struct sbmix_elem *controls, int controls_count, unsigned char map[][2], int map_count, @@ -856,7 +746,8 @@ static int snd_sbmixer_init(struct snd_sb *chip, } for (idx = 0; idx < controls_count; idx++) { - if ((err = snd_sbmixer_add_ctl_elem(chip, controls[idx])) < 0) + err = snd_sbmixer_add_ctl_elem(chip, &controls[idx]); + if (err < 0) return err; } snd_component_add(card, name); @@ -908,6 +799,15 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_ALS4000: + /* use only the first 16 controls from SB16 */ + err = snd_sbmixer_init(chip, + snd_sb16_controls, + 16, + snd_sb16_init_values, + ARRAY_SIZE(snd_sb16_init_values), + "ALS4000"); + if (err < 0) + return err; if ((err = snd_sbmixer_init(chip, snd_als4000_controls, ARRAY_SIZE(snd_als4000_controls), -- cgit v1.2.2 From 6dd7dc767e35cfbb38f8c63a50b1c27acad25920 Mon Sep 17 00:00:00 2001 From: Stefan Ringel Date: Mon, 14 Dec 2009 11:27:11 +0100 Subject: ALSA: hda - Add PCI IDs for Nvidia G2xx-series Signed-off-by: Stefan Ringel Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e54420e691ae..9b56f937913e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2713,6 +2713,9 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, -- cgit v1.2.2 From bc2580061e42c323d7777029f01318f395edac0d Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 13 Dec 2009 12:43:15 +0100 Subject: ASoC: Correct code taking the size of a pointer sizeof(codec->reg_cache) is just the size of the pointer. Elsewhere in the file, codec->reg_cache is used with sizeof(wm8900_reg_defaults), so the code is changed to do the same here. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ expression *x; expression f; type T; @@ *f(...,(T)x,...) // Signed-off-by: Julia Lawall Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index c9438dd62df3..dbc368c08263 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -199,7 +199,7 @@ static void wm8900_reset(struct snd_soc_codec *codec) snd_soc_write(codec, WM8900_REG_RESET, 0); memcpy(codec->reg_cache, wm8900_reg_defaults, - sizeof(codec->reg_cache)); + sizeof(wm8900_reg_defaults)); } static int wm8900_hp_event(struct snd_soc_dapm_widget *w, -- cgit v1.2.2 From 63978ab3e3e963db28093b53bb4598f2702e1ad7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 14 Dec 2009 12:48:35 +0100 Subject: sound: add Edirol UA-101 support Add experimental support for the Edirol UA-101 audio/MIDI interface. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/Kconfig | 12 + sound/usb/Makefile | 2 + sound/usb/ua101.c | 1457 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/usbaudio.c | 54 -- sound/usb/usbaudio.h | 1 - sound/usb/usbquirks.h | 31 -- 6 files changed, 1471 insertions(+), 86 deletions(-) create mode 100644 sound/usb/ua101.c (limited to 'sound') diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 73525c048e7f..8c2925814ce4 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -21,6 +21,18 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_UA101 + tristate "Edirol UA-101 driver (EXPERIMENTAL)" + depends on EXPERIMENTAL + select SND_PCM + select SND_RAWMIDI + help + Say Y here to include support for the Edirol UA-101 audio/MIDI + interface. + + To compile this driver as a module, choose M here: the module + will be called snd-ua101. + config SND_USB_USX2Y tristate "Tascam US-122, US-224 and US-428 USB driver" depends on X86 || PPC || ALPHA diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abb288bfe35d..5bf64aef9558 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -4,9 +4,11 @@ snd-usb-audio-objs := usbaudio.o usbmixer.o snd-usb-lib-objs := usbmidi.o +snd-ua101-objs := ua101.o # Toplevel Module Dependency obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usb-lib.o +obj-$(CONFIG_SND_USB_UA101) += snd-ua101.o snd-usb-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usb-lib.o diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c new file mode 100644 index 000000000000..ab9f8a2e1938 --- /dev/null +++ b/sound/usb/ua101.c @@ -0,0 +1,1457 @@ +/* + * Edirol UA-101 driver + * Copyright (c) Clemens Ladisch + * + * This driver is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver. If not, see . + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "usbaudio.h" + +MODULE_DESCRIPTION("Edirol UA-101 driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101}}"); + +/* I use my UA-1A for testing because I don't have a UA-101 ... */ +#define UA1A_HACK + +/* + * Should not be lower than the minimum scheduling delay of the host + * controller. Some Intel controllers need more than one frame; as long as + * that driver doesn't tell us about this, use 1.5 frames just to be sure. + */ +#define MIN_QUEUE_LENGTH 12 +/* Somewhat random. */ +#define MAX_QUEUE_LENGTH 30 +/* + * This magic value optimizes memory usage efficiency for the UA-101's packet + * sizes at all sample rates, taking into account the stupid cache pool sizes + * that usb_buffer_alloc() uses. + */ +#define DEFAULT_QUEUE_LENGTH 21 + +#define MAX_PACKET_SIZE 672 /* hardware specific */ +#define MAX_MEMORY_BUFFERS DIV_ROUND_UP(MAX_QUEUE_LENGTH, \ + PAGE_SIZE / MAX_PACKET_SIZE) + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static unsigned int queue_length = 21; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "card index"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "enable card"); +module_param(queue_length, uint, 0644); +MODULE_PARM_DESC(queue_length, "USB queue length in microframes, " + __stringify(MIN_QUEUE_LENGTH)"-"__stringify(MAX_QUEUE_LENGTH)); + +enum { + INTF_PLAYBACK, + INTF_CAPTURE, + INTF_MIDI, + + INTF_COUNT +}; + +/* bits in struct ua101::states */ +enum { + USB_CAPTURE_RUNNING, + USB_PLAYBACK_RUNNING, + ALSA_CAPTURE_OPEN, + ALSA_PLAYBACK_OPEN, + ALSA_CAPTURE_RUNNING, + ALSA_PLAYBACK_RUNNING, + CAPTURE_URB_COMPLETED, + PLAYBACK_URB_COMPLETED, + DISCONNECTED, +}; + +struct ua101 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf[INTF_COUNT]; + int card_index; + struct snd_pcm *pcm; + struct list_head midi_list; + u64 format_bit; + unsigned int rate; + unsigned int packets_per_second; + spinlock_t lock; + struct mutex mutex; + unsigned long states; + + /* FIFO to synchronize playback rate to capture rate */ + unsigned int rate_feedback_start; + unsigned int rate_feedback_count; + u8 rate_feedback[MAX_QUEUE_LENGTH]; + + struct list_head ready_playback_urbs; + struct tasklet_struct playback_tasklet; + wait_queue_head_t alsa_capture_wait; + wait_queue_head_t rate_feedback_wait; + wait_queue_head_t alsa_playback_wait; + struct ua101_stream { + struct snd_pcm_substream *substream; + unsigned int usb_pipe; + unsigned int channels; + unsigned int frame_bytes; + unsigned int max_packet_bytes; + unsigned int period_pos; + unsigned int buffer_pos; + unsigned int queue_length; + struct ua101_urb { + struct urb urb; + struct usb_iso_packet_descriptor iso_frame_desc[1]; + struct list_head ready_list; + } *urbs[MAX_QUEUE_LENGTH]; + struct { + unsigned int size; + void *addr; + dma_addr_t dma; + } buffers[MAX_MEMORY_BUFFERS]; + } capture, playback; + + unsigned int fps[10]; + unsigned int frame_counter; +}; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver ua101_driver; + +static void abort_alsa_playback(struct ua101 *ua); +static void abort_alsa_capture(struct ua101 *ua); + +/* allocate virtual buffer; may be called more than once */ +static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, + size_t size) +{ + struct snd_pcm_runtime *runtime = subs->runtime; + + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = vmalloc_user(size); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* free virtual buffer; may be called more than once */ +static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) +{ + struct snd_pcm_runtime *runtime = subs->runtime; + + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; +} + +/* get the physical page pointer at the given offset */ +static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, + unsigned long offset) +{ + void *pageptr = subs->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); +} + +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} + +static void abort_usb_capture(struct ua101 *ua) +{ + if (test_and_clear_bit(USB_CAPTURE_RUNNING, &ua->states)) { + wake_up(&ua->alsa_capture_wait); + wake_up(&ua->rate_feedback_wait); + } +} + +static void abort_usb_playback(struct ua101 *ua) +{ + if (test_and_clear_bit(USB_PLAYBACK_RUNNING, &ua->states)) + wake_up(&ua->alsa_playback_wait); +} + +static void playback_urb_complete(struct urb *usb_urb) +{ + struct ua101_urb *urb = (struct ua101_urb *)usb_urb; + struct ua101 *ua = urb->urb.context; + unsigned long flags; + + if (unlikely(urb->urb.status == -ENOENT || /* unlinked */ + urb->urb.status == -ENODEV || /* device removed */ + urb->urb.status == -ECONNRESET || /* unlinked */ + urb->urb.status == -ESHUTDOWN)) { /* device disabled */ + abort_usb_playback(ua); + abort_alsa_playback(ua); + return; + } + + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states)) { + /* append URB to FIFO */ + spin_lock_irqsave(&ua->lock, flags); + list_add_tail(&urb->ready_list, &ua->ready_playback_urbs); + if (ua->rate_feedback_count > 0) + tasklet_schedule(&ua->playback_tasklet); + ua->playback.substream->runtime->delay -= + urb->urb.iso_frame_desc[0].length / + ua->playback.frame_bytes; + spin_unlock_irqrestore(&ua->lock, flags); + } +} + +static void first_playback_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + + urb->complete = playback_urb_complete; + playback_urb_complete(urb); + + set_bit(PLAYBACK_URB_COMPLETED, &ua->states); + wake_up(&ua->alsa_playback_wait); +} + +/* copy data from the ALSA ring buffer into the URB buffer */ +static bool copy_playback_data(struct ua101_stream *stream, struct urb *urb, + unsigned int frames) +{ + struct snd_pcm_runtime *runtime; + unsigned int frame_bytes, frames1; + const u8 *source; + + runtime = stream->substream->runtime; + frame_bytes = stream->frame_bytes; + source = runtime->dma_area + stream->buffer_pos * frame_bytes; + if (stream->buffer_pos + frames <= runtime->buffer_size) { + memcpy(urb->transfer_buffer, source, frames * frame_bytes); + } else { + /* wrap around at end of ring buffer */ + frames1 = runtime->buffer_size - stream->buffer_pos; + memcpy(urb->transfer_buffer, source, frames1 * frame_bytes); + memcpy(urb->transfer_buffer + frames1 * frame_bytes, + runtime->dma_area, (frames - frames1) * frame_bytes); + } + + stream->buffer_pos += frames; + if (stream->buffer_pos >= runtime->buffer_size) + stream->buffer_pos -= runtime->buffer_size; + stream->period_pos += frames; + if (stream->period_pos >= runtime->period_size) { + stream->period_pos -= runtime->period_size; + return true; + } + return false; +} + +static inline void add_with_wraparound(struct ua101 *ua, + unsigned int *value, unsigned int add) +{ + *value += add; + if (*value >= ua->playback.queue_length) + *value -= ua->playback.queue_length; +} + +static void playback_tasklet(unsigned long data) +{ + struct ua101 *ua = (void *)data; + unsigned long flags; + unsigned int frames; + struct ua101_urb *urb; + bool do_period_elapsed = false; + int err; + + if (unlikely(!test_bit(USB_PLAYBACK_RUNNING, &ua->states))) + return; + + /* + * Synchronizing the playback rate to the capture rate is done by using + * the same sequence of packet sizes for both streams. + * Submitting a playback URB therefore requires both a ready URB and + * the size of the corresponding capture packet, i.e., both playback + * and capture URBs must have been completed. Since the USB core does + * not guarantee that playback and capture complete callbacks are + * called alternately, we use two FIFOs for packet sizes and read URBs; + * submitting playback URBs is possible as long as both FIFOs are + * nonempty. + */ + spin_lock_irqsave(&ua->lock, flags); + while (ua->rate_feedback_count > 0 && + !list_empty(&ua->ready_playback_urbs)) { + /* take packet size out of FIFO */ + frames = ua->rate_feedback[ua->rate_feedback_start]; + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + ua->rate_feedback_count--; + + /* take URB out of FIFO */ + urb = list_first_entry(&ua->ready_playback_urbs, + struct ua101_urb, ready_list); + list_del(&urb->ready_list); + + /* fill packet with data or silence */ + urb->urb.iso_frame_desc[0].length = + frames * ua->playback.frame_bytes; + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + do_period_elapsed |= copy_playback_data(&ua->playback, + &urb->urb, + frames); + else + memset(urb->urb.transfer_buffer, 0, + urb->urb.iso_frame_desc[0].length); + + /* and off you go ... */ + err = usb_submit_urb(&urb->urb, GFP_ATOMIC); + if (unlikely(err < 0)) { + spin_unlock_irqrestore(&ua->lock, flags); + abort_usb_playback(ua); + abort_alsa_playback(ua); + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + return; + } + ua->playback.substream->runtime->delay += frames; + } + spin_unlock_irqrestore(&ua->lock, flags); + if (do_period_elapsed) + snd_pcm_period_elapsed(ua->playback.substream); +} + +/* copy data from the URB buffer into the ALSA ring buffer */ +static bool copy_capture_data(struct ua101_stream *stream, struct urb *urb, + unsigned int frames) +{ + struct snd_pcm_runtime *runtime; + unsigned int frame_bytes, frames1; + u8 *dest; + + runtime = stream->substream->runtime; + frame_bytes = stream->frame_bytes; + dest = runtime->dma_area + stream->buffer_pos * frame_bytes; + if (stream->buffer_pos + frames <= runtime->buffer_size) { + memcpy(dest, urb->transfer_buffer, frames * frame_bytes); + } else { + /* wrap around at end of ring buffer */ + frames1 = runtime->buffer_size - stream->buffer_pos; + memcpy(dest, urb->transfer_buffer, frames1 * frame_bytes); + memcpy(runtime->dma_area, + urb->transfer_buffer + frames1 * frame_bytes, + (frames - frames1) * frame_bytes); + } + + stream->buffer_pos += frames; + if (stream->buffer_pos >= runtime->buffer_size) + stream->buffer_pos -= runtime->buffer_size; + stream->period_pos += frames; + if (stream->period_pos >= runtime->period_size) { + stream->period_pos -= runtime->period_size; + return true; + } + return false; +} + +static void capture_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + struct ua101_stream *stream = &ua->capture; + unsigned long flags; + unsigned int frames, write_ptr; + bool do_period_elapsed; + int err; + + if (unlikely(urb->status == -ENOENT || /* unlinked */ + urb->status == -ENODEV || /* device removed */ + urb->status == -ECONNRESET || /* unlinked */ + urb->status == -ESHUTDOWN)) /* device disabled */ + goto stream_stopped; + + if (urb->status >= 0 && urb->iso_frame_desc[0].status >= 0) + frames = urb->iso_frame_desc[0].actual_length / + stream->frame_bytes; + else + frames = 0; + + spin_lock_irqsave(&ua->lock, flags); + + if (frames > 0 && test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + do_period_elapsed = copy_capture_data(stream, urb, frames); + else + do_period_elapsed = false; + + if (test_bit(USB_CAPTURE_RUNNING, &ua->states)) { + err = usb_submit_urb(urb, GFP_ATOMIC); + if (unlikely(err < 0)) { + spin_unlock_irqrestore(&ua->lock, flags); + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + goto stream_stopped; + } + + /* append packet size to FIFO */ + write_ptr = ua->rate_feedback_start; + add_with_wraparound(ua, &write_ptr, ua->rate_feedback_count); + ua->rate_feedback[write_ptr] = frames; + if (ua->rate_feedback_count < ua->playback.queue_length) { + ua->rate_feedback_count++; + if (ua->rate_feedback_count == + ua->playback.queue_length) + wake_up(&ua->rate_feedback_wait); + } else { + /* + * Ring buffer overflow; this happens when the playback + * stream is not running. Throw away the oldest entry, + * so that the playback stream, when it starts, sees + * the most recent packet sizes. + */ + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + } + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states) && + !list_empty(&ua->ready_playback_urbs)) + tasklet_schedule(&ua->playback_tasklet); + } + + spin_unlock_irqrestore(&ua->lock, flags); + + if (do_period_elapsed) + snd_pcm_period_elapsed(stream->substream); + + /* for debugging: measure the sample rate relative to the USB clock */ + ua->fps[ua->frame_counter++ / ua->packets_per_second] += frames; + if (ua->frame_counter >= ARRAY_SIZE(ua->fps) * ua->packets_per_second) { + printk(KERN_DEBUG "capture rate:"); + for (frames = 0; frames < ARRAY_SIZE(ua->fps); ++frames) + printk(KERN_CONT " %u", ua->fps[frames]); + printk(KERN_CONT "\n"); + memset(ua->fps, 0, sizeof(ua->fps)); + ua->frame_counter = 0; + } + return; + +stream_stopped: + abort_usb_playback(ua); + abort_usb_capture(ua); + abort_alsa_playback(ua); + abort_alsa_capture(ua); +} + +static void first_capture_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + + urb->complete = capture_urb_complete; + capture_urb_complete(urb); + + set_bit(CAPTURE_URB_COMPLETED, &ua->states); + wake_up(&ua->alsa_capture_wait); +} + +static int submit_stream_urbs(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) { + int err = usb_submit_urb(&stream->urbs[i]->urb, GFP_KERNEL); + if (err < 0) { + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + return err; + } + } + return 0; +} + +static void kill_stream_urbs(struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) + usb_kill_urb(&stream->urbs[i]->urb); +} + +static int enable_iso_interface(struct ua101 *ua, unsigned int intf_index) +{ + struct usb_host_interface *alts; + + alts = ua->intf[intf_index]->cur_altsetting; + if (alts->desc.bAlternateSetting != 1) { + int err = usb_set_interface(ua->dev, + alts->desc.bInterfaceNumber, 1); + if (err < 0) { + dev_err(&ua->dev->dev, + "cannot initialize interface; error %d: %s\n", + err, usb_error_string(err)); + return err; + } + } + return 0; +} + +static void disable_iso_interface(struct ua101 *ua, unsigned int intf_index) +{ + struct usb_host_interface *alts; + + alts = ua->intf[intf_index]->cur_altsetting; + if (alts->desc.bAlternateSetting != 0) { + int err = usb_set_interface(ua->dev, + alts->desc.bInterfaceNumber, 0); + if (err < 0 && !test_bit(DISCONNECTED, &ua->states)) + dev_warn(&ua->dev->dev, + "interface reset failed; error %d: %s\n", + err, usb_error_string(err)); + } +} + +static void stop_usb_capture(struct ua101 *ua) +{ + clear_bit(USB_CAPTURE_RUNNING, &ua->states); + + kill_stream_urbs(&ua->capture); + + disable_iso_interface(ua, INTF_CAPTURE); +} + +static int start_usb_capture(struct ua101 *ua) +{ + int err; + + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + + if (test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return 0; + + kill_stream_urbs(&ua->capture); + + err = enable_iso_interface(ua, INTF_CAPTURE); + if (err < 0) + return err; + + clear_bit(CAPTURE_URB_COMPLETED, &ua->states); + ua->capture.urbs[0]->urb.complete = first_capture_urb_complete; + ua->rate_feedback_start = 0; + ua->rate_feedback_count = 0; + + set_bit(USB_CAPTURE_RUNNING, &ua->states); + err = submit_stream_urbs(ua, &ua->capture); + if (err < 0) + stop_usb_capture(ua); + return err; +} + +static void stop_usb_playback(struct ua101 *ua) +{ + clear_bit(USB_PLAYBACK_RUNNING, &ua->states); + + kill_stream_urbs(&ua->playback); + + tasklet_kill(&ua->playback_tasklet); + + disable_iso_interface(ua, INTF_PLAYBACK); +} + +static int start_usb_playback(struct ua101 *ua) +{ + unsigned int i, frames; + struct urb *urb; + int err = 0; + + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return 0; + + kill_stream_urbs(&ua->playback); + tasklet_kill(&ua->playback_tasklet); + + err = enable_iso_interface(ua, INTF_PLAYBACK); + if (err < 0) + return err; + + clear_bit(PLAYBACK_URB_COMPLETED, &ua->states); + ua->playback.urbs[0]->urb.complete = + first_playback_urb_complete; + spin_lock_irq(&ua->lock); + INIT_LIST_HEAD(&ua->ready_playback_urbs); + spin_unlock_irq(&ua->lock); + + /* + * We submit the initial URBs all at once, so we have to wait for the + * packet size FIFO to be full. + */ + wait_event(ua->rate_feedback_wait, + ua->rate_feedback_count >= ua->playback.queue_length || + !test_bit(USB_CAPTURE_RUNNING, &ua->states) || + test_bit(DISCONNECTED, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) { + stop_usb_playback(ua); + return -ENODEV; + } + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) { + stop_usb_playback(ua); + return -EIO; + } + + for (i = 0; i < ua->playback.queue_length; ++i) { + /* all initial URBs contain silence */ + spin_lock_irq(&ua->lock); + frames = ua->rate_feedback[ua->rate_feedback_start]; + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + ua->rate_feedback_count--; + spin_unlock_irq(&ua->lock); + urb = &ua->playback.urbs[i]->urb; + urb->iso_frame_desc[0].length = + frames * ua->playback.frame_bytes; + memset(urb->transfer_buffer, 0, + urb->iso_frame_desc[0].length); + } + + set_bit(USB_PLAYBACK_RUNNING, &ua->states); + err = submit_stream_urbs(ua, &ua->playback); + if (err < 0) + stop_usb_playback(ua); + return err; +} + +static void abort_alsa_capture(struct ua101 *ua) +{ + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); +} + +static void abort_alsa_playback(struct ua101 *ua) +{ + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); +} + +static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, + unsigned int channels) +{ + int err; + + substream->runtime->hw.info = + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_FIFO_IN_FRAMES; + substream->runtime->hw.formats = ua->format_bit; + substream->runtime->hw.rates = snd_pcm_rate_to_rate_bit(ua->rate); + substream->runtime->hw.rate_min = ua->rate; + substream->runtime->hw.rate_max = ua->rate; + substream->runtime->hw.channels_min = channels; + substream->runtime->hw.channels_max = channels; + substream->runtime->hw.buffer_bytes_max = 45000 * 1024; + substream->runtime->hw.period_bytes_min = 1; + substream->runtime->hw.period_bytes_max = UINT_MAX; + substream->runtime->hw.periods_min = 2; + substream->runtime->hw.periods_max = UINT_MAX; + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 1500000 / ua->packets_per_second, + 8192000); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + return err; +} + +static int capture_pcm_open(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + ua->capture.substream = substream; + err = set_stream_hw(ua, substream, ua->capture.channels); + if (err < 0) + return err; + substream->runtime->hw.fifo_size = + DIV_ROUND_CLOSEST(ua->rate, ua->packets_per_second); + substream->runtime->delay = substream->runtime->hw.fifo_size; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + set_bit(ALSA_CAPTURE_OPEN, &ua->states); + mutex_unlock(&ua->mutex); + return err; +} + +static int playback_pcm_open(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + ua->playback.substream = substream; + err = set_stream_hw(ua, substream, ua->playback.channels); + if (err < 0) + return err; + substream->runtime->hw.fifo_size = + DIV_ROUND_CLOSEST(ua->rate * ua->playback.queue_length, + ua->packets_per_second); + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err < 0) + goto error; + err = start_usb_playback(ua); + if (err < 0) { + if (!test_bit(ALSA_CAPTURE_OPEN, &ua->states)) + stop_usb_capture(ua); + goto error; + } + set_bit(ALSA_PLAYBACK_OPEN, &ua->states); +error: + mutex_unlock(&ua->mutex); + return err; +} + +static int capture_pcm_close(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + + mutex_lock(&ua->mutex); + clear_bit(ALSA_CAPTURE_OPEN, &ua->states); + if (!test_bit(ALSA_PLAYBACK_OPEN, &ua->states)) + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + return 0; +} + +static int playback_pcm_close(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + + mutex_lock(&ua->mutex); + stop_usb_playback(ua); + clear_bit(ALSA_PLAYBACK_OPEN, &ua->states); + if (!test_bit(ALSA_CAPTURE_OPEN, &ua->states)) + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + return 0; +} + +static int capture_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + return snd_pcm_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} + +static int playback_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + err = start_usb_playback(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + return snd_pcm_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} + +static int ua101_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_free_vmalloc_buffer(substream); + return 0; +} + +static int capture_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + /* + * The EHCI driver schedules the first packet of an iso stream at 10 ms + * in the future, i.e., no data is actually captured for that long. + * Take the wait here so that the stream is known to be actually + * running when the start trigger has been called. + */ + wait_event(ua->alsa_capture_wait, + test_bit(CAPTURE_URB_COMPLETED, &ua->states) || + !test_bit(USB_CAPTURE_RUNNING, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return -EIO; + + ua->capture.period_pos = 0; + ua->capture.buffer_pos = 0; + return 0; +} + +static int playback_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + err = start_usb_playback(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + /* see the comment in capture_pcm_prepare() */ + wait_event(ua->alsa_playback_wait, + test_bit(PLAYBACK_URB_COMPLETED, &ua->states) || + !test_bit(USB_PLAYBACK_RUNNING, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + if (!test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return -EIO; + + substream->runtime->delay = 0; + ua->playback.period_pos = 0; + ua->playback.buffer_pos = 0; + return 0; +} + +static int capture_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ua101 *ua = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return -EIO; + set_bit(ALSA_CAPTURE_RUNNING, &ua->states); + return 0; + case SNDRV_PCM_TRIGGER_STOP: + clear_bit(ALSA_CAPTURE_RUNNING, &ua->states); + return 0; + default: + return -EINVAL; + } +} + +static int playback_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ua101 *ua = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (!test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return -EIO; + set_bit(ALSA_PLAYBACK_RUNNING, &ua->states); + return 0; + case SNDRV_PCM_TRIGGER_STOP: + clear_bit(ALSA_PLAYBACK_RUNNING, &ua->states); + return 0; + default: + return -EINVAL; + } +} + +static inline snd_pcm_uframes_t ua101_pcm_pointer(struct ua101 *ua, + struct ua101_stream *stream) +{ + unsigned long flags; + unsigned int pos; + + spin_lock_irqsave(&ua->lock, flags); + pos = stream->buffer_pos; + spin_unlock_irqrestore(&ua->lock, flags); + return pos; +} + +static snd_pcm_uframes_t capture_pcm_pointer(struct snd_pcm_substream *subs) +{ + struct ua101 *ua = subs->private_data; + + return ua101_pcm_pointer(ua, &ua->capture); +} + +static snd_pcm_uframes_t playback_pcm_pointer(struct snd_pcm_substream *subs) +{ + struct ua101 *ua = subs->private_data; + + return ua101_pcm_pointer(ua, &ua->playback); +} + +static struct snd_pcm_ops capture_pcm_ops = { + .open = capture_pcm_open, + .close = capture_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = capture_pcm_hw_params, + .hw_free = ua101_pcm_hw_free, + .prepare = capture_pcm_prepare, + .trigger = capture_pcm_trigger, + .pointer = capture_pcm_pointer, + .page = snd_pcm_get_vmalloc_page, +}; + +static struct snd_pcm_ops playback_pcm_ops = { + .open = playback_pcm_open, + .close = playback_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = playback_pcm_hw_params, + .hw_free = ua101_pcm_hw_free, + .prepare = playback_pcm_prepare, + .trigger = playback_pcm_trigger, + .pointer = playback_pcm_pointer, + .page = snd_pcm_get_vmalloc_page, +}; + +static const struct uac_format_type_i_discrete_descriptor * +find_format_descriptor(struct usb_interface *interface) +{ + struct usb_host_interface *alt; + u8 *extra; + int extralen; + + if (interface->num_altsetting != 2) { + dev_err(&interface->dev, "invalid num_altsetting\n"); + return NULL; + } + + alt = &interface->altsetting[0]; + if (alt->desc.bNumEndpoints != 0) { + dev_err(&interface->dev, "invalid bNumEndpoints\n"); + return NULL; + } + + alt = &interface->altsetting[1]; + if (alt->desc.bNumEndpoints != 1) { + dev_err(&interface->dev, "invalid bNumEndpoints\n"); + return NULL; + } + + extra = alt->extra; + extralen = alt->extralen; + while (extralen >= sizeof(struct usb_descriptor_header)) { + struct uac_format_type_i_discrete_descriptor *desc; + + desc = (struct uac_format_type_i_discrete_descriptor *)extra; + if (desc->bLength > extralen) { + dev_err(&interface->dev, "descriptor overflow\n"); + return NULL; + } + if (desc->bLength == UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(1) && + desc->bDescriptorType == USB_DT_CS_INTERFACE && + desc->bDescriptorSubtype == UAC_FORMAT_TYPE) { + if (desc->bFormatType != UAC_FORMAT_TYPE_I_PCM || + desc->bSamFreqType != 1) { + dev_err(&interface->dev, + "invalid format type\n"); + return NULL; + } + return desc; + } + extralen -= desc->bLength; + extra += desc->bLength; + } + dev_err(&interface->dev, "sample format descriptor not found\n"); + return NULL; +} + +static int detect_usb_format(struct ua101 *ua) +{ + const struct uac_format_type_i_discrete_descriptor *fmt_capture; + const struct uac_format_type_i_discrete_descriptor *fmt_playback; + const struct usb_endpoint_descriptor *epd; + unsigned int rate2; + + fmt_capture = find_format_descriptor(ua->intf[INTF_CAPTURE]); + fmt_playback = find_format_descriptor(ua->intf[INTF_PLAYBACK]); + if (!fmt_capture || !fmt_playback) + return -ENXIO; + + switch (fmt_capture->bSubframeSize) { + case 3: + ua->format_bit = SNDRV_PCM_FMTBIT_S24_3LE; + break; + case 4: + ua->format_bit = SNDRV_PCM_FMTBIT_S32_LE; + break; + default: + dev_err(&ua->dev->dev, "sample width is not 24 or 32 bits\n"); + return -ENXIO; + } + if (fmt_capture->bSubframeSize != fmt_playback->bSubframeSize) { + dev_err(&ua->dev->dev, + "playback/capture sample widths do not match\n"); + return -ENXIO; + } + + if (fmt_capture->bBitResolution != 24 || + fmt_playback->bBitResolution != 24) { + dev_err(&ua->dev->dev, "sample width is not 24 bits\n"); + return -ENXIO; + } + + ua->rate = combine_triple(fmt_capture->tSamFreq[0]); + rate2 = combine_triple(fmt_playback->tSamFreq[0]); + if (ua->rate != rate2) { + dev_err(&ua->dev->dev, + "playback/capture rates do not match: %u/%u\n", + rate2, ua->rate); + return -ENXIO; + } + + switch (ua->dev->speed) { + case USB_SPEED_FULL: + ua->packets_per_second = 1000; + break; + case USB_SPEED_HIGH: + ua->packets_per_second = 8000; + break; + default: + dev_err(&ua->dev->dev, "unknown device speed\n"); + return -ENXIO; + } + + ua->capture.channels = fmt_capture->bNrChannels; + ua->playback.channels = fmt_playback->bNrChannels; + ua->capture.frame_bytes = + fmt_capture->bSubframeSize * ua->capture.channels; + ua->playback.frame_bytes = + fmt_playback->bSubframeSize * ua->playback.channels; + + epd = &ua->intf[INTF_CAPTURE]->altsetting[1].endpoint[0].desc; + if (!usb_endpoint_is_isoc_in(epd)) { + dev_err(&ua->dev->dev, "invalid capture endpoint\n"); + return -ENXIO; + } + ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, usb_endpoint_num(epd)); + ua->capture.max_packet_bytes = le16_to_cpu(epd->wMaxPacketSize); + + epd = &ua->intf[INTF_PLAYBACK]->altsetting[1].endpoint[0].desc; + if (!usb_endpoint_is_isoc_out(epd)) { + dev_err(&ua->dev->dev, "invalid playback endpoint\n"); + return -ENXIO; + } + ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, usb_endpoint_num(epd)); + ua->playback.max_packet_bytes = le16_to_cpu(epd->wMaxPacketSize); + return 0; +} + +static int alloc_stream_buffers(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int remaining_packets, packets, packets_per_page, i; + size_t size; + + stream->queue_length = queue_length; + stream->queue_length = max(stream->queue_length, + (unsigned int)MIN_QUEUE_LENGTH); + stream->queue_length = min(stream->queue_length, + (unsigned int)MAX_QUEUE_LENGTH); + + /* + * The cache pool sizes used by usb_buffer_alloc() (128, 512, 2048) are + * quite bad when used with the packet sizes of this device (e.g. 280, + * 520, 624). Therefore, we allocate and subdivide entire pages, using + * a smaller buffer only for the last chunk. + */ + remaining_packets = stream->queue_length; + packets_per_page = PAGE_SIZE / stream->max_packet_bytes; + for (i = 0; i < ARRAY_SIZE(stream->buffers); ++i) { + packets = min(remaining_packets, packets_per_page); + size = packets * stream->max_packet_bytes; + stream->buffers[i].addr = + usb_buffer_alloc(ua->dev, size, GFP_KERNEL, + &stream->buffers[i].dma); + if (!stream->buffers[i].addr) + return -ENOMEM; + stream->buffers[i].size = size; + remaining_packets -= packets; + if (!remaining_packets) + break; + } + if (remaining_packets) { + dev_err(&ua->dev->dev, "too many packets\n"); + return -ENXIO; + } + return 0; +} + +static void free_stream_buffers(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(stream->buffers); ++i) + usb_buffer_free(ua->dev, + stream->buffers[i].size, + stream->buffers[i].addr, + stream->buffers[i].dma); +} + +static int alloc_stream_urbs(struct ua101 *ua, struct ua101_stream *stream, + void (*urb_complete)(struct urb *)) +{ + unsigned max_packet_size = stream->max_packet_bytes; + struct ua101_urb *urb; + unsigned int b, u = 0; + + for (b = 0; b < ARRAY_SIZE(stream->buffers); ++b) { + unsigned int size = stream->buffers[b].size; + u8 *addr = stream->buffers[b].addr; + dma_addr_t dma = stream->buffers[b].dma; + + while (size >= max_packet_size) { + if (u >= stream->queue_length) + goto bufsize_error; + urb = kmalloc(sizeof(*urb), GFP_KERNEL); + if (!urb) + return -ENOMEM; + usb_init_urb(&urb->urb); + urb->urb.dev = ua->dev; + urb->urb.pipe = stream->usb_pipe; + urb->urb.transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + urb->urb.transfer_buffer = addr; + urb->urb.transfer_dma = dma; + urb->urb.transfer_buffer_length = max_packet_size; + urb->urb.number_of_packets = 1; + urb->urb.interval = 1; + urb->urb.context = ua; + urb->urb.complete = urb_complete; + urb->urb.iso_frame_desc[0].offset = 0; + urb->urb.iso_frame_desc[0].length = max_packet_size; + stream->urbs[u++] = urb; + size -= max_packet_size; + addr += max_packet_size; + dma += max_packet_size; + } + } + if (u == stream->queue_length) + return 0; +bufsize_error: + dev_err(&ua->dev->dev, "internal buffer size error\n"); + return -ENXIO; +} + +static void free_stream_urbs(struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) + kfree(stream->urbs[i]); +} + +static void free_usb_related_resources(struct ua101 *ua, + struct usb_interface *interface) +{ + unsigned int i; + + free_stream_urbs(&ua->capture); + free_stream_urbs(&ua->playback); + free_stream_buffers(ua, &ua->capture); + free_stream_buffers(ua, &ua->playback); + + for (i = 0; i < ARRAY_SIZE(ua->intf); ++i) + if (ua->intf[i]) { + usb_set_intfdata(ua->intf[i], NULL); + if (ua->intf[i] != interface) + usb_driver_release_interface(&ua101_driver, + ua->intf[i]); + } +} + +static void ua101_card_free(struct snd_card *card) +{ + struct ua101 *ua = card->private_data; + + mutex_destroy(&ua->mutex); +} + +static int ua101_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + static const struct snd_usb_midi_endpoint_info midi_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk midi_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &midi_ep + }; + struct snd_card *card; + struct ua101 *ua; + unsigned int card_index, i; + char usb_path[32]; + int err; + + if (interface->altsetting->desc.bInterfaceNumber != 0) + return -ENODEV; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*ua), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + card->private_free = ua101_card_free; + ua = card->private_data; + ua->dev = interface_to_usbdev(interface); + ua->card = card; + ua->card_index = card_index; + INIT_LIST_HEAD(&ua->midi_list); + spin_lock_init(&ua->lock); + mutex_init(&ua->mutex); + INIT_LIST_HEAD(&ua->ready_playback_urbs); + tasklet_init(&ua->playback_tasklet, + playback_tasklet, (unsigned long)ua); + init_waitqueue_head(&ua->alsa_capture_wait); + init_waitqueue_head(&ua->rate_feedback_wait); + init_waitqueue_head(&ua->alsa_playback_wait); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { + ua->intf[2] = interface; + ua->intf[0] = usb_ifnum_to_if(ua->dev, 1); + ua->intf[1] = usb_ifnum_to_if(ua->dev, 2); + usb_driver_claim_interface(&ua101_driver, ua->intf[0], ua); + usb_driver_claim_interface(&ua101_driver, ua->intf[1], ua); + } else { +#endif + ua->intf[0] = interface; + for (i = 1; i < ARRAY_SIZE(ua->intf); ++i) { + ua->intf[i] = usb_ifnum_to_if(ua->dev, i); + if (!ua->intf[i]) { + dev_err(&ua->dev->dev, "interface %u not found\n", i); + err = -ENXIO; + goto probe_error; + } + err = usb_driver_claim_interface(&ua101_driver, + ua->intf[i], ua); + if (err < 0) { + ua->intf[i] = NULL; + err = -EBUSY; + goto probe_error; + } + } +#ifdef UA1A_HACK + } +#endif + + snd_card_set_dev(card, &interface->dev); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { + ua->format_bit = SNDRV_PCM_FMTBIT_S16_LE; + ua->rate = 44100; + ua->packets_per_second = 1000; + ua->capture.channels = 2; + ua->playback.channels = 2; + ua->capture.frame_bytes = 4; + ua->playback.frame_bytes = 4; + ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, 2); + ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, 1); + ua->capture.max_packet_bytes = 192; + ua->playback.max_packet_bytes = 192; + } else { +#endif + err = detect_usb_format(ua); + if (err < 0) + goto probe_error; +#ifdef UA1A_HACK + } +#endif + + strcpy(card->driver, "UA-101"); + strcpy(card->shortname, "UA-101"); + usb_make_path(ua->dev, usb_path, sizeof(usb_path)); + snprintf(ua->card->longname, sizeof(ua->card->longname), + "EDIROL UA-101 (serial %s), %u Hz at %s, %s speed", + ua->dev->serial ? ua->dev->serial : "?", ua->rate, usb_path, + ua->dev->speed == USB_SPEED_HIGH ? "high" : "full"); + + err = alloc_stream_buffers(ua, &ua->capture); + if (err < 0) + goto probe_error; + err = alloc_stream_buffers(ua, &ua->playback); + if (err < 0) + goto probe_error; + + err = alloc_stream_urbs(ua, &ua->capture, capture_urb_complete); + if (err < 0) + goto probe_error; + err = alloc_stream_urbs(ua, &ua->playback, playback_urb_complete); + if (err < 0) + goto probe_error; + + err = snd_pcm_new(card, "UA-101", 0, 1, 1, &ua->pcm); + if (err < 0) + goto probe_error; + ua->pcm->private_data = ua; + strcpy(ua->pcm->name, "UA-101"); + snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); + snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct != cpu_to_le16(0x0018)) { +#endif + err = snd_usbmidi_create(card, ua->intf[INTF_MIDI], + &ua->midi_list, &midi_quirk); + if (err < 0) + goto probe_error; +#ifdef UA1A_HACK + } +#endif + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, ua); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + free_usb_related_resources(ua, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void ua101_disconnect(struct usb_interface *interface) +{ + struct ua101 *ua = usb_get_intfdata(interface); + struct list_head *midi; + + if (!ua) + return; + + mutex_lock(&devices_mutex); + + set_bit(DISCONNECTED, &ua->states); + wake_up(&ua->rate_feedback_wait); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(ua->card); + + /* make sure that there are no pending USB requests */ + __list_for_each(midi, &ua->midi_list) + snd_usbmidi_disconnect(midi); + abort_alsa_playback(ua); + abort_alsa_capture(ua); + mutex_lock(&ua->mutex); + stop_usb_playback(ua); + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + + free_usb_related_resources(ua, interface); + + devices_used &= ~(1 << ua->card_index); + + snd_card_free_when_closed(ua->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_device_id ua101_ids[] = { +#ifdef UA1A_HACK + { USB_DEVICE(0x0582, 0x0018) }, +#endif + { USB_DEVICE(0x0582, 0x007d) }, + { USB_DEVICE(0x0582, 0x008d) }, + { } +}; +MODULE_DEVICE_TABLE(usb, ua101_ids); + +static struct usb_driver ua101_driver = { + .name = "snd-ua101", + .id_table = ua101_ids, + .probe = ua101_probe, + .disconnect = ua101_disconnect, +#if 0 + .suspend = ua101_suspend, + .resume = ua101_resume, +#endif +}; + +static int __init alsa_card_ua101_init(void) +{ + return usb_register(&ua101_driver); +} + +static void __exit alsa_card_ua101_exit(void) +{ + usb_deregister(&ua101_driver); + mutex_destroy(&devices_mutex); +} + +module_init(alsa_card_ua101_init); +module_exit(alsa_card_ua101_exit); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b074a594c595..f352141cf8ee 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3142,59 +3142,6 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, return 0; } -/* - * Create a stream for an Edirol UA-101 interface. - * Copy, paste and modify from Edirol UA-1000 - */ -static int create_ua101_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - static const struct audioformat ua101_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - }; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct audioformat *fp; - int stream, err; - - if (iface->num_altsetting != 2) - return -ENXIO; - alts = &iface->altsetting[1]; - altsd = get_iface_desc(alts); - if (alts->extralen != 18 || alts->extra[1] != USB_DT_CS_INTERFACE || - altsd->bNumEndpoints != 1) - return -ENXIO; - - fp = kmemdup(&ua101_format, sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; - - fp->channels = alts->extra[11]; - fp->iface = altsd->bInterfaceNumber; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]); - - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp); - return err; - } - /* FIXME: playback must be synchronized to capture */ - usb_set_interface(chip->dev, fp->iface, 0); - return 0; -} - static int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, const struct snd_usb_audio_quirk *quirk); @@ -3406,7 +3353,6 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, - [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 40ba8115fb81..9826337c76b8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,7 +159,6 @@ enum quirk_type { QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, - QUIRK_AUDIO_EDIROL_UA101, QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index a892bda03df9..bd6706c2d534 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1266,37 +1266,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -/* Roland UA-101 in High-Speed Mode only */ -{ - USB_DEVICE(0x0582, 0x007d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "UA-101", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA101 - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA101 - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, { /* has ID 0x0081 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0080), -- cgit v1.2.2 From f74890277a196949e4004fe2955e1d4fb3930f98 Mon Sep 17 00:00:00 2001 From: Steve Soule Date: Mon, 14 Dec 2009 11:06:03 -0700 Subject: ALSA: ac97_codec - increase timeout for analog sections to 5 second I have a Soundblaster 16PCI. For many years, alsa has had a bug where not all of the card's controls are detected (many alsa versions, many kernel versions). In particular, Master Playback Volume is usually not detected, and so I get no sound or extremely faint sound. The problem has always been inconsistent: sometimes all of the controls are detected correctly, and sometimes a partial set is detected. It works correctly about 10% of the time. Finally, I got around to tracking down the problem. When the driver fails, it prints the kernel message "AC'97 0 analog subsections not ready". This message is generated from the function snd_ac97_mixer() in ac97_codec.c. The message indicates that the card failed to come back after reset within the time limit. The time limit is 120 milliseconds. I tried increasing the time limit to 1 second, and found that this made the driver work about 70% of the time. I tried increasing it to 5 seconds, and it now seems to work 100% of the time. I expect that this change would be completely harmless for existing cards that work, and would only introduce additional delay for cards that do not work. ALSA bug#4032. Signed-off-by: Steve Soule Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 20cb60afb200..c11920623009 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0); - end_time = jiffies + msecs_to_jiffies(120); + end_time = jiffies + msecs_to_jiffies(5000); do { if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f) goto __ready_ok; -- cgit v1.2.2 From 5b0cb1d850c26893b1468b3a519433a1b7a176be Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 8 Dec 2009 16:13:32 +0100 Subject: ALSA: hda - add more NID->Control mapping This set of changes add missing NID values to some static control elemenents. Also, it handles all "Capture Source" or "Input Source" controls. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.c | 64 +++++++++- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_generic.c | 3 +- sound/pci/hda/hda_local.h | 5 + sound/pci/hda/hda_proc.c | 23 ++-- sound/pci/hda/patch_analog.c | 31 +++++ sound/pci/hda/patch_cirrus.c | 4 + sound/pci/hda/patch_cmedia.c | 12 +- sound/pci/hda/patch_realtek.c | 120 ++++++++++++++++++- sound/pci/hda/patch_si3054.c | 1 + sound/pci/hda/patch_via.c | 273 +++++++++++++++++++++++++----------------- 11 files changed, 415 insertions(+), 122 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9cfdb771928c..20100b1548e1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,6 +931,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) #endif list_del(&codec->list); snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); @@ -985,7 +986,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 60); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { @@ -1706,7 +1708,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); /** - * snd_hda_ctl-add - Add a control element and assign to the codec + * snd_hda_ctl_add - Add a control element and assign to the codec * @codec: HD-audio codec * @nid: corresponding NID (optional) * @kctl: the control element to assign @@ -1746,6 +1748,35 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); +/** + * snd_hda_add_nid - Assign a NID to a control element + * @codec: HD-audio codec + * @nid: corresponding NID (optional) + * @kctl: the control element to assign + * @index: index to kctl + * + * Add the given control element to an array inside the codec instance. + * This function is used when #snd_hda_ctl_add cannot be used for 1:1 + * NID:KCTL mapping - for example "Capture Source" selector. + */ +int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t nid) +{ + struct hda_nid_item *item; + + if (nid > 0) { + item = snd_array_new(&codec->nids); + if (!item) + return -ENOMEM; + item->kctl = kctl; + item->index = index; + item->nid = nid; + return 0; + } + return -EINVAL; +} +EXPORT_SYMBOL_HDA(snd_hda_add_nid); + /** * snd_hda_ctls_clear - Clear all controls assigned to the given codec * @codec: HD-audio codec @@ -1757,6 +1788,7 @@ void snd_hda_ctls_clear(struct hda_codec *codec) for (i = 0; i < codec->mixers.used; i++) snd_ctl_remove(codec->bus->card, items[i].kctl); snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); } /* pseudo device locking @@ -3476,6 +3508,8 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) for (; knew->name; knew++) { struct snd_kcontrol *kctl; + if (knew->iface == -1) /* skip this codec private value */ + continue; kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; @@ -3496,6 +3530,32 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) } EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); +/** + * snd_hda_add_nids - assign nids to controls from the array + * @codec: the HDA codec + * @kctl: struct snd_kcontrol + * @index: index to kctl + * @nids: the array of hda_nid_t + * @size: count of hda_nid_t items + * + * This helper function assigns NIDs in the given array to a control element. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t *nids, unsigned int size) +{ + int err; + + for ( ; size > 0; size--, nids++) { + err = snd_hda_add_nid(codec, kctl, index, *nids); + if (err < 0) + return err; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_add_nids); + #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1d541b7f5547..0d08ad5bd898 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -789,6 +789,7 @@ struct hda_codec { u32 *wcaps; struct snd_array mixers; /* list of assigned mixer elements */ + struct snd_array nids; /* list of mapped mixer elements */ struct hda_cache_rec amp_cache; /* cache for amp access */ struct hda_cache_rec cmd_cache; /* cache for other commands */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 092c6a7c2ff3..5ea21285ee1f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -861,7 +861,8 @@ static int build_input_controls(struct hda_codec *codec) } /* create input MUX if multiple sources are available */ - err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cap_sel, codec)); + err = snd_hda_ctl_add(codec, spec->adc_node->nid, + snd_ctl_new1(&cap_sel, codec)); if (err < 0) return err; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 5778ae882b83..98cf3f4f3755 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -342,6 +342,8 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); +int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t *nids, unsigned int size); /* * unsolicited event handler @@ -466,11 +468,14 @@ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); struct hda_nid_item { struct snd_kcontrol *kctl; + unsigned int index; hda_nid_t nid; }; int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, struct snd_kcontrol *kctl); +int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t nid); void snd_hda_ctls_clear(struct hda_codec *codec); /* diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c9afc04adac8..2e27d6a8b446 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -61,18 +61,21 @@ static const char *get_wid_type_name(unsigned int wid_value) return "UNKNOWN Widget"; } -static void print_nid_mixers(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) +static void print_nid_array(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid, + struct snd_array *array) { int i; - struct hda_nid_item *items = codec->mixers.list; + struct hda_nid_item *items = array->list, *item; struct snd_kcontrol *kctl; - for (i = 0; i < codec->mixers.used; i++) { - if (items[i].nid == nid) { - kctl = items[i].kctl; + for (i = 0; i < array->used; i++) { + item = &items[i]; + if (item->nid == nid) { + kctl = item->kctl; snd_iprintf(buffer, " Control: name=\"%s\", index=%i, device=%i\n", - kctl->id.name, kctl->id.index, kctl->id.device); + kctl->id.name, kctl->id.index + item->index, + kctl->id.device); } } } @@ -528,7 +531,8 @@ static void print_gpio(struct snd_info_buffer *buffer, (data & (1<mixers); + print_nid_array(buffer, codec, nid, &codec->nids); } static void print_codec_info(struct snd_info_entry *entry, @@ -608,7 +612,8 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " CP"); snd_iprintf(buffer, "\n"); - print_nid_mixers(buffer, codec, nid); + print_nid_array(buffer, codec, nid, &codec->mixers); + print_nid_array(buffer, codec, nid, &codec->nids); print_nid_pcms(buffer, codec, nid); /* volume knob is a special widget that always have connection diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 447eda1f6770..d418842373fd 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -174,6 +174,7 @@ static struct snd_kcontrol_new ad_beep_mixer[] = { static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; + struct snd_kcontrol *kctl; unsigned int i; int err; @@ -239,6 +240,28 @@ static int ad198x_build_controls(struct hda_codec *codec) } ad198x_free_kctls(codec); /* no longer needed */ + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + if (!kctl) + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + + /* assign IEC958 enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, + SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source"); + if (kctl) { + err = snd_hda_add_nid(codec, kctl, 0, + spec->multiout.dig_out_nid); + if (err < 0) + return err; + } + return 0; } @@ -701,6 +724,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, @@ -808,6 +832,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1a, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -1608,6 +1633,7 @@ static struct snd_kcontrol_new ad1981_hp_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x05, .name = "Master Playback Switch", .info = ad198x_eapd_info, .get = ad198x_eapd_get, @@ -2121,6 +2147,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x12, .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, @@ -2242,6 +2269,7 @@ static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "IEC958 Playback Source", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = ad1988_spdif_playback_source_info, .get = ad1988_spdif_playback_source_get, .put = ad1988_spdif_playback_source_put, @@ -3728,6 +3756,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3756,6 +3785,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4097,6 +4127,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4b200da1bd18..d0b8c6dc7322 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -759,6 +759,10 @@ static int build_input(struct hda_codec *codec) err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; + err = snd_hda_add_nids(codec, kctl, 0, spec->adc_nid, + spec->num_inputs); + if (err < 0) + return err; } if (spec->num_inputs > 1 && !spec->mic_detect) { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index a45c1169762b..cc1c22370a60 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -315,7 +315,8 @@ static struct hda_verb cmi9880_allout_init[] = { static int cmi9880_build_controls(struct hda_codec *codec) { struct cmi_spec *spec = codec->spec; - int err; + struct snd_kcontrol *kctl; + int i, err; err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer); if (err < 0) @@ -340,6 +341,15 @@ static int cmi9880_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->adc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 888b6313eeca..6b0b8728f6b7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -627,6 +627,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, #define ALC_PIN_MODE(xname, nid, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_pin_mode_info, \ .get = alc_pin_mode_get, \ .put = alc_pin_mode_put, \ @@ -678,6 +679,7 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, } #define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_gpio_data_info, \ .get = alc_gpio_data_get, \ .put = alc_gpio_data_put, \ @@ -732,6 +734,7 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, } #define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_spdif_ctrl_info, \ .get = alc_spdif_ctrl_get, \ .put = alc_spdif_ctrl_put, \ @@ -785,6 +788,7 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, #define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_eapd_ctrl_info, \ .get = alc_eapd_ctrl_get, \ .put = alc_eapd_ctrl_put, \ @@ -2410,6 +2414,15 @@ static const char *alc_slave_sws[] = { * build control elements */ +#define NID_MAPPING (-1) + +#define SUBDEV_SPEAKER_ (0 << 6) +#define SUBDEV_HP_ (1 << 6) +#define SUBDEV_LINE_ (2 << 6) +#define SUBDEV_SPEAKER(x) (SUBDEV_SPEAKER_ | ((x) & 0x3f)) +#define SUBDEV_HP(x) (SUBDEV_HP_ | ((x) & 0x3f)) +#define SUBDEV_LINE(x) (SUBDEV_LINE_ | ((x) & 0x3f)) + static void alc_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -2424,8 +2437,11 @@ static struct snd_kcontrol_new alc_beep_mixer[] = { static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; - int i; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *knew; + int i, j, err; + unsigned int u; + hda_nid_t nid; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -2494,6 +2510,73 @@ static int alc_build_controls(struct hda_codec *codec) } alc_free_kctls(codec); /* no longer needed */ + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + if (!kctl) + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + if (spec->cap_mixer) { + const char *kname = kctl ? kctl->id.name : NULL; + for (knew = spec->cap_mixer; knew->name; knew++) { + if (kname && strcmp(knew->name, kname) == 0) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nid(codec, kctl, i, + spec->adc_nids[i]); + if (err < 0) + return err; + } + } + } + + /* other nid->control mapping */ + for (i = 0; i < spec->num_mixers; i++) { + for (knew = spec->mixers[i]; knew->name; knew++) { + if (knew->iface != NID_MAPPING) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + if (kctl == NULL) + continue; + u = knew->subdevice; + for (j = 0; j < 4; j++, u >>= 8) { + nid = u & 0x3f; + if (nid == 0) + continue; + switch (u & 0xc0) { + case SUBDEV_SPEAKER_: + nid = spec->autocfg.speaker_pins[nid]; + break; + case SUBDEV_LINE_: + nid = spec->autocfg.line_out_pins[nid]; + break; + case SUBDEV_HP_: + nid = spec->autocfg.hp_pins[nid]; + break; + default: + continue; + } + err = snd_hda_add_nid(codec, kctl, 0, nid); + if (err < 0) + return err; + } + u = knew->private_value; + for (j = 0; j < 4; j++, u >>= 8) { + nid = u & 0xff; + if (nid == 0) + continue; + err = snd_hda_add_nid(codec, kctl, 0, nid); + if (err < 0) + return err; + } + } + } return 0; } @@ -3781,6 +3864,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, #define PIN_CTL_TEST(xname,nid) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_test_pin_ctl_info, \ .get = alc_test_pin_ctl_get, \ .put = alc_test_pin_ctl_put, \ @@ -3790,6 +3874,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, #define PIN_SRC_TEST(xname,nid) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_test_pin_src_info, \ .get = alc_test_pin_src_get, \ .put = alc_test_pin_src_put, \ @@ -5080,6 +5165,7 @@ static struct snd_kcontrol_new alc260_hp_output_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, @@ -5118,6 +5204,7 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, @@ -10188,8 +10275,14 @@ static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, .info = snd_ctl_boolean_mono_info, \ .get = alc262_hp_master_sw_get, \ .put = alc262_hp_master_sw_put, \ + }, \ + { \ + .iface = NID_MAPPING, \ + .name = "Master Playback Switch", \ + .private_value = 0x15 | (0x16 << 8) | (0x1b << 16), \ } + static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -10347,6 +10440,12 @@ static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, .info = snd_ctl_boolean_mono_info, \ .get = alc262_hippo_master_sw_get, \ .put = alc262_hippo_master_sw_put, \ + }, \ + { \ + .iface = NID_MAPPING, \ + .name = "Master Playback Switch", \ + .subdevice = SUBDEV_HP(0) | (SUBDEV_LINE(0) << 8) | \ + (SUBDEV_SPEAKER(0) << 16), \ } static struct snd_kcontrol_new alc262_hippo_mixer[] = { @@ -10820,11 +10919,17 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, + { + .iface = NID_MAPPING, + .name = "Master Playback Switch", + .private_value = 0x1b, + }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), @@ -10855,6 +10960,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -11009,6 +11115,11 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { .get = alc_mux_enum_get, .put = alc262_ultra_mux_enum_put, }, + { + .iface = NID_MAPPING, + .name = "Capture Source", + .private_value = 0x15, + }, { } /* end */ }; @@ -12026,6 +12137,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12041,6 +12153,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12058,6 +12171,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13010,6 +13124,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13030,6 +13145,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 43b436c5d01b..f419ee8d75f0 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -122,6 +122,7 @@ static int si3054_switch_put(struct snd_kcontrol *kcontrol, #define SI3054_KCONTROL(kname,reg,mask) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = kname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | reg, \ .info = si3054_switch_info, \ .get = si3054_switch_get, \ .put = si3054_switch_put, \ diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b70e26ad263f..64995e8e3a72 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -54,6 +54,8 @@ #include "hda_codec.h" #include "hda_local.h" +#define NID_MAPPING (-1) + /* amp values */ #define AMP_VAL_IDX_SHIFT 19 #define AMP_VAL_IDX_MASK (0x0f<<19) @@ -157,6 +159,19 @@ struct via_spec { #endif }; +static struct via_spec * via_new_spec(struct hda_codec *codec) +{ + struct via_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return NULL; + + codec->spec = spec; + spec->codec = codec; + return spec; +} + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -448,6 +463,22 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, return 0; } +static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, + struct snd_kcontrol_new *tmpl) +{ + struct snd_kcontrol_new *knew; + + snd_array_init(&spec->kctls, sizeof(*knew), 32); + knew = snd_array_new(&spec->kctls); + if (!knew) + return NULL; + *knew = *tmpl; + knew->name = kstrdup(tmpl->name, GFP_KERNEL); + if (!knew->name) + return NULL; + return 0; +} + static void via_free_kctls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1088,24 +1119,9 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct via_spec *spec = codec->spec; - hda_nid_t nid; + hda_nid_t nid = kcontrol->private_value; unsigned int pinsel; - switch (spec->codec_type) { - case VT1718S: - nid = 0x34; - break; - case VT2002P: - nid = 0x35; - break; - case VT1812: - nid = 0x3d; - break; - default: - nid = spec->autocfg.hp_pins[0]; - break; - } /* use !! to translate conn sel 2 for VT1718S */ pinsel = !!snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, @@ -1127,29 +1143,24 @@ static void activate_ctl(struct hda_codec *codec, const char *name, int active) } } +static hda_nid_t side_mute_channel(struct via_spec *spec) +{ + switch (spec->codec_type) { + case VT1708: return 0x1b; + case VT1709_10CH: return 0x29; + case VT1708B_8CH: /* fall thru */ + case VT1708S: return 0x27; + default: return 0; + } +} + static int update_side_mute_status(struct hda_codec *codec) { /* mute side channel */ struct via_spec *spec = codec->spec; unsigned int parm = spec->hp_independent_mode ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; - hda_nid_t sw3; - - switch (spec->codec_type) { - case VT1708: - sw3 = 0x1b; - break; - case VT1709_10CH: - sw3 = 0x29; - break; - case VT1708B_8CH: - case VT1708S: - sw3 = 0x27; - break; - default: - sw3 = 0; - break; - } + hda_nid_t sw3 = side_mute_channel(spec); if (sw3) snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1162,28 +1173,11 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.hp_pins[0]; + hda_nid_t nid = kcontrol->private_value; unsigned int pinsel = ucontrol->value.enumerated.item[0]; /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - - switch (spec->codec_type) { - case VT1718S: - nid = 0x34; - pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ - spec->multiout.num_dacs = 4; - break; - case VT2002P: - nid = 0x35; - break; - case VT1812: - nid = 0x3d; - break; - default: - nid = spec->autocfg.hp_pins[0]; - break; - } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid @@ -1207,18 +1201,55 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new via_hp_mixer[] = { +static struct snd_kcontrol_new via_hp_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Independent HP", - .count = 1, .info = via_independent_hp_info, .get = via_independent_hp_get, .put = via_independent_hp_put, }, - { } /* end */ + { + .iface = NID_MAPPING, + .name = "Independent HP", + }, }; +static int via_hp_build(struct via_spec *spec) +{ + struct snd_kcontrol_new *knew; + hda_nid_t nid; + + knew = via_clone_control(spec, &via_hp_mixer[0]); + if (knew == NULL) + return -ENOMEM; + + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + break; + case VT2002P: + nid = 0x35; + break; + case VT1812: + nid = 0x3d; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; + knew->private_value = nid; + + knew = via_clone_control(spec, &via_hp_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = side_mute_channel(spec); + + return 0; +} + static void notify_aa_path_ctls(struct hda_codec *codec) { int i; @@ -1376,7 +1407,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new via_smart51_mixer[] = { +static struct snd_kcontrol_new via_smart51_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Smart 5.1", @@ -1385,9 +1416,36 @@ static struct snd_kcontrol_new via_smart51_mixer[] = { .get = via_smart51_get, .put = via_smart51_put, }, - {} /* end */ + { + .iface = NID_MAPPING, + .name = "Smart 5.1", + } }; +static int via_smart51_build(struct via_spec *spec) +{ + struct snd_kcontrol_new *knew; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + hda_nid_t nid; + int i; + + knew = via_clone_control(spec, &via_smart51_mixer[0]); + if (knew == NULL) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + knew = via_clone_control(spec, &via_smart51_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = nid; + } + } + + return 0; +} + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1819,8 +1877,9 @@ static struct hda_pcm_stream vt1708_pcm_digital_capture = { static int via_build_controls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; - int err; - int i; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *knew; + int err, i; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -1845,6 +1904,28 @@ static int via_build_controls(struct hda_codec *codec) return err; } + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->mux_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + + /* other nid->control mapping */ + for (i = 0; i < spec->num_mixers; i++) { + for (knew = spec->mixers[i]; knew->name; knew++) { + if (knew->iface != NID_MAPPING) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + if (kctl == NULL) + continue; + err = snd_hda_add_nid(codec, kctl, 0, + knew->subdevice); + } + } + /* init power states */ set_jack_power_state(codec); analog_low_current_mode(codec, 1); @@ -2481,9 +2562,9 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -2554,12 +2635,10 @@ static int patch_vt1708(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708_parse_auto_config(codec); if (err < 0) { @@ -2597,7 +2676,6 @@ static int patch_vt1708(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - spec->codec = codec; INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; } @@ -3010,9 +3088,9 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -3032,12 +3110,10 @@ static int patch_vt1709_10ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - err = vt1709_parse_auto_config(codec); if (err < 0) { via_free(codec); @@ -3126,12 +3202,10 @@ static int patch_vt1709_6ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - err = vt1709_parse_auto_config(codec); if (err < 0) { via_free(codec); @@ -3581,9 +3655,9 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -3605,12 +3679,10 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) if (get_codec_type(codec) == VT1708BCE) return patch_vt1708S(codec); /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708B_parse_auto_config(codec); if (err < 0) { @@ -3657,12 +3729,10 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708B_parse_auto_config(codec); if (err < 0) { @@ -4071,9 +4141,9 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -4103,12 +4173,10 @@ static int patch_vt1708S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708S_parse_auto_config(codec); if (err < 0) { @@ -4443,7 +4511,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -4464,12 +4532,10 @@ static int patch_vt1702(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1702_parse_auto_config(codec); if (err < 0) { @@ -4865,9 +4931,9 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -4888,12 +4954,10 @@ static int patch_vt1718S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1718S_parse_auto_config(codec); if (err < 0) { @@ -5014,6 +5078,7 @@ static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Mic Capture Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x26, .count = 1, .info = vt1716s_dmic_info, .get = vt1716s_dmic_get, @@ -5361,9 +5426,9 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -5384,12 +5449,10 @@ static int patch_vt1716S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1716S_parse_auto_config(codec); if (err < 0) { @@ -5719,7 +5782,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -5741,12 +5804,10 @@ static int patch_vt2002P(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt2002P_parse_auto_config(codec); if (err < 0) { @@ -6070,7 +6131,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -6092,12 +6153,10 @@ static int patch_vt1812(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1812_parse_auto_config(codec); if (err < 0) { -- cgit v1.2.2 From 9e3fd8719f624a43575b56a4777b1552399a8be8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 8 Dec 2009 17:45:25 +0100 Subject: ALSA: hda - introduce HDA_SUBDEV_AMP_FLAG (ControlAmp in proc) The purpose of this changeset is to show information about amplifier setting in the codec proc file. Something like: Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.c | 14 +++++++++----- sound/pci/hda/hda_local.h | 11 ++++++++--- sound/pci/hda/hda_proc.c | 8 ++++++++ sound/pci/hda/patch_analog.c | 12 +++++++----- sound/pci/hda/patch_cirrus.c | 2 ++ sound/pci/hda/patch_conexant.c | 1 + sound/pci/hda/patch_realtek.c | 18 ++++++++++-------- sound/pci/hda/patch_sigmatel.c | 3 ++- sound/pci/hda/patch_via.c | 4 +++- 9 files changed, 50 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 20100b1548e1..c9af15ed7f10 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1723,19 +1723,22 @@ EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); * * snd_hda_ctl_add() checks the control subdev id field whether * #HDA_SUBDEV_NID_FLAG bit is set. If set (and @nid is zero), the lower - * bits value is taken as the NID to assign. + * bits value is taken as the NID to assign. The #HDA_NID_ITEM_AMP bit + * specifies if kctl->private_value is a HDA amplifier value. */ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, struct snd_kcontrol *kctl) { int err; + unsigned short flags = 0; struct hda_nid_item *item; - if (kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) { - if (nid == 0) - nid = kctl->id.subdevice & 0xffff; + if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) + flags |= HDA_NID_ITEM_AMP; + if ((kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) != 0 && nid == 0) + nid = kctl->id.subdevice & 0xffff; + if (kctl->id.subdevice & 0xf0000000) kctl->id.subdevice = 0; - } err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) return err; @@ -1744,6 +1747,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, return -ENOMEM; item->kctl = kctl; item->nid = nid; + item->flags = flags; return 0; } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 98cf3f4f3755..0a256471f812 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -31,6 +31,7 @@ * in snd_hda_ctl_add(), so that this value won't appear in the outside. */ #define HDA_SUBDEV_NID_FLAG (1U << 31) +#define HDA_SUBDEV_AMP_FLAG (1U << 30) /* * for mixer controls @@ -42,7 +43,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -63,7 +64,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -81,7 +82,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ @@ -466,10 +467,14 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); +/* flags for hda_nid_item */ +#define HDA_NID_ITEM_AMP (1<<0) + struct hda_nid_item { struct snd_kcontrol *kctl; unsigned int index; hda_nid_t nid; + unsigned short flags; }; int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 2e27d6a8b446..f97d35de66c4 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -76,6 +76,14 @@ static void print_nid_array(struct snd_info_buffer *buffer, " Control: name=\"%s\", index=%i, device=%i\n", kctl->id.name, kctl->id.index + item->index, kctl->id.device); + if (item->flags & HDA_NID_ITEM_AMP) + snd_iprintf(buffer, + " ControlAmp: chs=%lu, dir=%s, " + "idx=%lu, ofs=%lu\n", + get_amp_channels(kctl), + get_amp_direction(kctl) ? "Out" : "In", + get_amp_index(kctl), + get_amp_offset(kctl)); } } } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d418842373fd..5e2bb181a149 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -832,7 +832,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1a, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1a, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -2602,7 +2602,9 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, if (! knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } @@ -3756,7 +3758,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3785,7 +3787,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4127,7 +4129,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d0b8c6dc7322..e51f6658aa2c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -500,6 +500,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); + (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } @@ -513,6 +514,7 @@ static int add_volume(struct hda_codec *codec, const char *name, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); + (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a09c03c3f62b..b68650af40a9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2178,6 +2178,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x10, .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = snd_hda_mixer_amp_volume_put, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6b0b8728f6b7..87bf7bd6292a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4414,7 +4414,9 @@ static int add_control(struct alc_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } @@ -10919,7 +10921,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, @@ -10960,7 +10962,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1b, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -12137,7 +12139,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12153,7 +12155,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12171,7 +12173,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13124,7 +13126,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13145,7 +13147,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f8325848..1ee586b65b63 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2702,7 +2702,8 @@ stac_control_new(struct sigmatel_spec *spec, return NULL; } if (nid) - knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | nid; return knew; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 64995e8e3a72..b94cdee5eb53 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -458,7 +458,9 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } -- cgit v1.2.2 From 5e26dfd0615868872cb44842f1e1428c7b414ab0 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 10 Dec 2009 13:57:01 +0100 Subject: ALSA: hda - simplify usage of HDA_SUBDEV_AMP_FLAG The HDA_SUBDEV_NID_FLAG is duplicate for amplifier control elements. Move get_amp_nid_() call to the snd_hda_ctl_add() function. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.c | 7 +++++-- sound/pci/hda/hda_local.h | 6 +++--- sound/pci/hda/patch_analog.c | 16 ++++++---------- sound/pci/hda/patch_cirrus.c | 4 ++-- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 21 +++++++++------------ sound/pci/hda/patch_sigmatel.c | 8 +++----- sound/pci/hda/patch_via.c | 4 +--- 8 files changed, 30 insertions(+), 38 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c9af15ed7f10..c848ec0f085e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1733,11 +1733,14 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, unsigned short flags = 0; struct hda_nid_item *item; - if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) + if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) { flags |= HDA_NID_ITEM_AMP; + if (nid == 0) + nid = get_amp_nid_(kctl->private_value); + } if ((kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) != 0 && nid == 0) nid = kctl->id.subdevice & 0xffff; - if (kctl->id.subdevice & 0xf0000000) + if (kctl->id.subdevice & (HDA_SUBDEV_NID_FLAG|HDA_SUBDEV_AMP_FLAG)) kctl->id.subdevice = 0; err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 0a256471f812..d505d052972e 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -43,7 +43,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -64,7 +64,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -82,7 +82,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5e2bb181a149..e75b5e5a1d55 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -209,9 +209,7 @@ static int ad198x_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, - get_amp_nid_(spec->beep_amp), - kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } @@ -832,7 +830,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1a, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -2602,9 +2600,7 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, if (! knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } @@ -3758,7 +3754,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3787,7 +3783,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4129,7 +4125,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index e51f6658aa2c..eeb91f6a06c2 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -501,7 +501,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; - return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); + return snd_hda_ctl_add(codec, 0, *kctlp); } static int add_volume(struct hda_codec *codec, const char *name, @@ -515,7 +515,7 @@ static int add_volume(struct hda_codec *codec, const char *name, snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; - return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); + return snd_hda_ctl_add(codec, 0, *kctlp); } static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b68650af40a9..1ab2958a290b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2178,7 +2178,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x10, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = snd_hda_mixer_amp_volume_put, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 87bf7bd6292a..cb7679551bdc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2482,8 +2482,7 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, - get_amp_nid_(spec->beep_amp), kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } @@ -4414,9 +4413,7 @@ static int add_control(struct alc_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } @@ -10921,7 +10918,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, @@ -10962,7 +10959,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1b, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -12139,7 +12136,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12155,7 +12152,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12173,7 +12170,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13126,7 +13123,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13147,7 +13144,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1ee586b65b63..0bafea9d5106 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2685,7 +2685,7 @@ static struct snd_kcontrol_new * stac_control_new(struct sigmatel_spec *spec, struct snd_kcontrol_new *ktemp, const char *name, - hda_nid_t nid) + unsigned int subdev) { struct snd_kcontrol_new *knew; @@ -2701,9 +2701,7 @@ stac_control_new(struct sigmatel_spec *spec, spec->kctls.alloced--; return NULL; } - if (nid) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | nid; + knew->subdevice = subdev; return knew; } @@ -2713,7 +2711,7 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec, unsigned long val) { struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, - get_amp_nid_(val)); + HDA_SUBDEV_AMP_FLAG); if (!knew) return -ENOMEM; knew->index = idx; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b94cdee5eb53..de4839e46762 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -458,9 +458,7 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } -- cgit v1.2.2 From 1cf86f6f9b000e98c1b7f866f99633ae67464e2f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 15 Dec 2009 15:54:21 +0900 Subject: ASoC: ak4642: Add default return value in ak4642_modinit If ak4642 driver was compiled without I2C configs, ak4642_modinit return value will become un-stable. This patch modify this bug Reported-by: Magnus Damm Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b69861d52161..3ef16bbc8c83 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -470,7 +470,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); static int __init ak4642_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ak4642_i2c_driver); #endif -- cgit v1.2.2 From 471452104b8520337ae2fb48c4e61cd4896e025d Mon Sep 17 00:00:00 2001 From: Alexey Dobriyan Date: Mon, 14 Dec 2009 18:00:08 -0800 Subject: const: constify remaining dev_pm_ops Signed-off-by: Alexey Dobriyan Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/arm/pxa2xx-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec.h | 2 +- sound/soc/soc-core.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index b4b48afb6de6..5d9411839cd7 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -159,7 +159,7 @@ static int pxa2xx_ac97_resume(struct device *dev) return ret; } -static struct dev_pm_ops pxa2xx_ac97_pm_ops = { +static const struct dev_pm_ops pxa2xx_ac97_pm_ops = { .suspend = pxa2xx_ac97_suspend, .resume = pxa2xx_ac97_resume, }; diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index d441c3b64631..4984754f3298 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -312,7 +312,7 @@ int simtec_audio_resume(struct device *dev) return 0; } -struct dev_pm_ops simtec_audio_pmops = { +const struct dev_pm_ops simtec_audio_pmops = { .resume = simtec_audio_resume, }; EXPORT_SYMBOL_GPL(simtec_audio_pmops); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h index 2714203af161..e18faee30cce 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.h +++ b/sound/soc/s3c24xx/s3c24xx_simtec.h @@ -15,7 +15,7 @@ extern int simtec_audio_core_probe(struct platform_device *pdev, extern int simtec_audio_remove(struct platform_device *pdev); #ifdef CONFIG_PM -extern struct dev_pm_ops simtec_audio_pmops; +extern const struct dev_pm_ops simtec_audio_pmops; #define simtec_audio_pm &simtec_audio_pmops #else #define simtec_audio_pm NULL diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ef8f28284cb9..0a6440c6f54a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1236,7 +1236,7 @@ static int soc_poweroff(struct device *dev) return 0; } -static struct dev_pm_ops soc_pm_ops = { +static const struct dev_pm_ops soc_pm_ops = { .suspend = soc_suspend, .resume = soc_resume, .poweroff = soc_poweroff, -- cgit v1.2.2 From 3c55494670745e523f69b56edb66ca0b50a470c2 Mon Sep 17 00:00:00 2001 From: Andres Salomon Date: Mon, 14 Dec 2009 18:00:36 -0800 Subject: ALSA: cs5535audio: free OLPC quirks from reliance on MGEODE_LX cpu optimization Previously, OLPC support for the mic extensions was only enabled in the ALSA driver if CONFIG_OLPC and CONFIG_MGEODE_LX were both set. This was because the old geode GPIO code was written in a manner that assumed CONFIG_MGEODE_LX. With the new cs553x-gpio driver, this is no longer the case; as such, we can drop the requirement on CONFIG_MGEODE_LX and instead include a requirement on GPIOLIB. We use the generic GPIO API rather than the cs553x-specific API. Signed-off-by: Andres Salomon Cc: Takashi Iwai Cc: Jordan Crouse Cc: David Brownell Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/pci/cs5535audio/Makefile | 2 -- sound/pci/cs5535audio/cs5535audio.c | 1 + sound/pci/cs5535audio/cs5535audio.h | 4 +++- sound/pci/cs5535audio/cs5535audio_olpc.c | 26 +++++++++++++++++++------- 4 files changed, 23 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile index fda7a94c992f..ccc642269b9e 100644 --- a/sound/pci/cs5535audio/Makefile +++ b/sound/pci/cs5535audio/Makefile @@ -4,9 +4,7 @@ snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o -ifdef CONFIG_MGEODE_LX snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.o -endif # Toplevel Module Dependency obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 05f56e04849b..91e7faf69bbb 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -389,6 +389,7 @@ probefail_out: static void __devexit snd_cs5535audio_remove(struct pci_dev *pci) { + olpc_quirks_cleanup(); snd_card_free(pci_get_drvdata(pci)); pci_set_drvdata(pci, NULL); } diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index 7a298ac662e3..51966d782a3c 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -99,10 +99,11 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state); int snd_cs5535audio_resume(struct pci_dev *pci); #endif -#if defined(CONFIG_OLPC) && defined(CONFIG_MGEODE_LX) +#ifdef CONFIG_OLPC void __devinit olpc_prequirks(struct snd_card *card, struct snd_ac97_template *ac97); int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97); +void __devexit olpc_quirks_cleanup(void); void olpc_analog_input(struct snd_ac97 *ac97, int on); void olpc_mic_bias(struct snd_ac97 *ac97, int on); @@ -128,6 +129,7 @@ static inline int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97) { return 0; } +static inline void olpc_quirks_cleanup(void) { } static inline void olpc_analog_input(struct snd_ac97 *ac97, int on) { } static inline void olpc_mic_bias(struct snd_ac97 *ac97, int on) { } static inline void olpc_capture_open(struct snd_ac97 *ac97) { } diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c index 5c6814335cd7..50da49be9ae5 100644 --- a/sound/pci/cs5535audio/cs5535audio_olpc.c +++ b/sound/pci/cs5535audio/cs5535audio_olpc.c @@ -13,10 +13,13 @@ #include #include #include +#include #include #include "cs5535audio.h" +#define DRV_NAME "cs5535audio-olpc" + /* * OLPC has an additional feature on top of the regular AD1888 codec features. * It has an Analog Input mode that is switched into (after disabling the @@ -38,10 +41,7 @@ void olpc_analog_input(struct snd_ac97 *ac97, int on) } /* set Analog Input through GPIO */ - if (on) - geode_gpio_set(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL); - else - geode_gpio_clear(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL); + gpio_set_value(OLPC_GPIO_MIC_AC, on); } /* @@ -73,8 +73,7 @@ static int olpc_dc_info(struct snd_kcontrol *kctl, static int olpc_dc_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v) { - v->value.integer.value[0] = geode_gpio_isset(OLPC_GPIO_MIC_AC, - GPIO_OUTPUT_VAL); + v->value.integer.value[0] = gpio_get_value(OLPC_GPIO_MIC_AC); return 0; } @@ -153,6 +152,12 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97) if (!machine_is_olpc()) return 0; + if (gpio_request(OLPC_GPIO_MIC_AC, DRV_NAME)) { + printk(KERN_ERR DRV_NAME ": unable to allocate MIC GPIO\n"); + return -EIO; + } + gpio_direction_output(OLPC_GPIO_MIC_AC, 0); + /* drop the original AD1888 HPF control */ memset(&elem, 0, sizeof(elem)); elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -169,11 +174,18 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97) for (i = 0; i < ARRAY_SIZE(olpc_cs5535audio_ctls); i++) { err = snd_ctl_add(card, snd_ctl_new1(&olpc_cs5535audio_ctls[i], ac97->private_data)); - if (err < 0) + if (err < 0) { + gpio_free(OLPC_GPIO_MIC_AC); return err; + } } /* turn off the mic by default */ olpc_mic_bias(ac97, 0); return 0; } + +void __devexit olpc_quirks_cleanup(void) +{ + gpio_free(OLPC_GPIO_MIC_AC); +} -- cgit v1.2.2 From e7d2860b690d4f3bed6824757c540579638e3d1e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Andr=C3=A9=20Goddard=20Rosa?= Date: Mon, 14 Dec 2009 18:01:06 -0800 Subject: tree-wide: convert open calls to remove spaces to skip_spaces() lib function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Makes use of skip_spaces() defined in lib/string.c for removing leading spaces from strings all over the tree. It decreases lib.a code size by 47 bytes and reuses the function tree-wide: text data bss dec hex filename 64688 584 592 65864 10148 (TOTALS-BEFORE) 64641 584 592 65817 10119 (TOTALS-AFTER) Also, while at it, if we see (*str && isspace(*str)), we can be sure to remove the first condition (*str) as the second one (isspace(*str)) also evaluates to 0 whenever *str == 0, making it redundant. In other words, "a char equals zero is never a space". Julia Lawall tried the semantic patch (http://coccinelle.lip6.fr) below, and found occurrences of this pattern on 3 more files: drivers/leds/led-class.c drivers/leds/ledtrig-timer.c drivers/video/output.c @@ expression str; @@ ( // ignore skip_spaces cases while (*str && isspace(*str)) { \(str++;\|++str;\) } | - *str && isspace(*str) ) Signed-off-by: André Goddard Rosa Cc: Julia Lawall Cc: Martin Schwidefsky Cc: Jeff Dike Cc: Ingo Molnar Cc: Thomas Gleixner Cc: "H. Peter Anvin" Cc: Richard Purdie Cc: Neil Brown Cc: Kyle McMartin Cc: Henrique de Moraes Holschuh Cc: David Howells Cc: Cc: Samuel Ortiz Cc: Patrick McHardy Cc: Takashi Iwai Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/pci/hda/hda_hwdep.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index d24328661c6a..40ccb419b6e9 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include "hda_codec.h" @@ -428,8 +429,7 @@ static int parse_hints(struct hda_codec *codec, const char *buf) char *key, *val; struct hda_hint *hint; - while (isspace(*buf)) - buf++; + buf = skip_spaces(buf); if (!*buf || *buf == '#' || *buf == '\n') return 0; if (*buf == '=') @@ -444,8 +444,7 @@ static int parse_hints(struct hda_codec *codec, const char *buf) return -EINVAL; } *val++ = 0; - while (isspace(*val)) - val++; + val = skip_spaces(val); remove_trail_spaces(key); remove_trail_spaces(val); hint = get_hint(codec, key); -- cgit v1.2.2 From 75b46c1321785c29cfbc4f839b6dc031428734ad Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Tue, 15 Dec 2009 20:53:44 -0500 Subject: ASoC: Fix disable of SPDIF on STAC9766 codec Change code so that switching to playing music through the analog output disables SPDIF out instead of disabling it when stream ends. Signed-off-by: Jon Smirl Acked-by: Mark Brown --- sound/soc/codecs/stac9766.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index bbc72c2ddfca..81b8c9dfe7fc 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -191,6 +191,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); vra |= 0x1; /* enable variable rate audio */ + vra &= ~0x4; /* disable SPDIF output */ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); @@ -221,22 +222,6 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, return stac9766_ac97_write(codec, reg, runtime->rate); } -static int ac97_digital_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned short vra; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_STOP: - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); - vra &= !0x04; - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); - break; - } - return 0; -} - static int stac9766_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -315,7 +300,6 @@ static struct snd_soc_dai_ops stac9766_dai_ops_analog = { static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, - .trigger = ac97_digital_trigger, }; struct snd_soc_dai stac9766_dai[] = { -- cgit v1.2.2 From 283375cefbf4f91ce51d93d010634c48d0d39044 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Dec 2009 18:09:03 +0000 Subject: ASoC: Push registers out of mixer power decision No need for the mixers to know about this, and it allows for virtual controls. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 846678aa3d35..4cf58911f3b3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1262,8 +1262,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, /* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int reg, - int val_mask, int val, int invert) + struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; int found = 0; @@ -1273,9 +1272,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, widget->id != snd_soc_dapm_switch) return -ENODEV; - if (!snd_soc_test_bits(widget->codec, reg, val_mask, val)) - return 0; - /* find dapm widget path assoc with kcontrol */ list_for_each_entry(path, &widget->codec->dapm_paths, list) { if (path->kcontrol != kcontrol) @@ -1283,12 +1279,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; - if (val) - /* new connection */ - path->connect = invert ? 0:1; - else - /* old connection must be powered down */ - path->connect = invert ? 1:0; + path->connect = connect; break; } @@ -1695,6 +1686,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val2, val_mask; + int connect; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1721,7 +1713,17 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, return 1; } - dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert); + if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { + if (val) + /* new connection */ + connect = invert ? 0:1; + else + /* old connection must be powered down */ + connect = invert ? 1:0; + + dapm_mixer_update_power(widget, kcontrol, connect); + } + if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, kcontrol, -- cgit v1.2.2 From d207c68dd92455a3d618c37b5a9f0dc598723fd6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Dec 2009 17:13:55 +0000 Subject: ASoC: Sort DAPM sequences by CODEC as well In preparation for multiple device support. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4cf58911f3b3..de22c2f1842e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -739,6 +739,8 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, int sort[]) { + if (a->codec != b->codec) + return (unsigned long)a - (unsigned long)b; if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; if (a->reg != b->reg) -- cgit v1.2.2 From cce2e9db718d823f33ac846c019763cdc84e8658 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Dec 2009 21:50:01 +0000 Subject: ASoC: Register the CODEC in WM8727 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8727.c | 66 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 49 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index d8ffbd641d71..63a254e293ca 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -44,23 +44,16 @@ struct snd_soc_dai wm8727_dai = { }; EXPORT_SYMBOL_GPL(wm8727_dai); +static struct snd_soc_codec *wm8727_codec; + static int wm8727_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec; int ret = 0; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - mutex_init(&codec->mutex); - codec->name = "WM8727"; - codec->owner = THIS_MODULE; - codec->dai = &wm8727_dai; - codec->num_dai = 1; - socdev->card->codec = codec; - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + BUG_ON(!wm8727_codec); + + socdev->card->codec = wm8727_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); @@ -80,12 +73,9 @@ pcm_err: static int wm8727_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - if (codec == NULL) - return 0; snd_soc_free_pcms(socdev); - kfree(codec); + return 0; } @@ -98,13 +88,55 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8727); static __devinit int wm8727_platform_probe(struct platform_device *pdev) { + struct snd_soc_codec *codec; + int ret; + + if (wm8727_codec) { + dev_err(&pdev->dev, "Another WM8727 is registered\n"); + return -EBUSY; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + wm8727_codec = codec; + + platform_set_drvdata(pdev, codec); + + mutex_init(&codec->mutex); + codec->dev = &pdev->dev; + codec->name = "WM8727"; + codec->owner = THIS_MODULE; + codec->dai = &wm8727_dai; + codec->num_dai = 1; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + wm8727_dai.dev = &pdev->dev; - return snd_soc_register_dai(&wm8727_dai); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to register CODEC: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8727_dai); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(codec); + return ret; } static int __devexit wm8727_platform_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&wm8727_dai); + snd_soc_unregister_codec(platform_get_drvdata(pdev)); return 0; } -- cgit v1.2.2 From 168db50d967e09133feda8247d4dcb3c73437766 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 9 Dec 2009 13:29:20 +0900 Subject: ASoC: S3C64XX: Remove unnecessary header includes Removed redundant header includes which make no difference to compilation. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index cc7edb5f792d..8feb029b99fe 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -15,16 +15,10 @@ #include #include #include -#include #include -#include #include #include -#include -#include -#include -#include #include #include -- cgit v1.2.2 From 0fe692292a26f57b6522fe859cc8b2549ec0cd97 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 9 Dec 2009 13:29:25 +0900 Subject: ASoC: S3C64XX: Compress and generalize the CPU driver The driver can be 'generalized' a bit by not hardcoding '2'(the number of I2Sv3 controllers that the driver can handle) at many places, instead we define a macro for it. That makes it easier to increase number of controllers by changing the parameter at just one place, this will be useful when there is support for newer SoCs, which have the same controller, only more in number. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 114 +++++++++++++++------------------------- 1 file changed, 41 insertions(+), 73 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 8feb029b99fe..93ed3aad1631 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -32,6 +32,11 @@ #include "s3c-dma.h" #include "s3c64xx-i2s.h" +/* The value should be set to maximum of the total number + * of I2Sv3 controllers that any supported SoC has. + */ +#define MAX_I2SV3 2 + static struct s3c2410_dma_client s3c64xx_dma_client_out = { .name = "I2S PCM Stereo out" }; @@ -40,37 +45,12 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { - [0] = { - .channel = DMACH_I2S0_OUT, - .client = &s3c64xx_dma_client_out, - .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD, - .dma_size = 4, - }, - [1] = { - .channel = DMACH_I2S1_OUT, - .client = &s3c64xx_dma_client_out, - .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD, - .dma_size = 4, - }, -}; - -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { - [0] = { - .channel = DMACH_I2S0_IN, - .client = &s3c64xx_dma_client_in, - .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD, - .dma_size = 4, - }, - [1] = { - .channel = DMACH_I2S1_IN, - .client = &s3c64xx_dma_client_in, - .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD, - .dma_size = 4, - }, -}; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; +static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; -static struct s3c_i2sv2_info s3c64xx_i2s[2]; +struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) { @@ -163,55 +143,13 @@ static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { .set_sysclk = s3c64xx_i2s_set_sysclk, }; -struct snd_soc_dai s3c64xx_i2s_dai[] = { - { - .name = "s3c64xx-i2s", - .id = 0, - .probe = s3c64xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .ops = &s3c64xx_i2s_dai_ops, - .symmetric_rates = 1, - }, - { - .name = "s3c64xx-i2s", - .id = 1, - .probe = s3c64xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .ops = &s3c64xx_i2s_dai_ops, - .symmetric_rates = 1, - }, -}; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); - static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) { struct s3c_i2sv2_info *i2s; struct snd_soc_dai *dai; int ret; - if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) { + if (pdev->id >= MAX_I2SV3) { dev_err(&pdev->dev, "id %d out of range\n", pdev->id); return -EINVAL; } @@ -219,10 +157,40 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) i2s = &s3c64xx_i2s[pdev->id]; dai = &s3c64xx_i2s_dai[pdev->id]; dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s"; + dai->id = pdev->id; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2s_probe; + dai->ops = &s3c64xx_i2s_dai_ops; i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + if (pdev->id == 0) { + i2s->dma_capture->channel = DMACH_I2S0_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S0_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; + } else { + i2s->dma_capture->channel = DMACH_I2S1_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S1_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; + } + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); if (IS_ERR(i2s->iis_cclk)) { dev_err(&pdev->dev, "failed to get audio-bus\n"); -- cgit v1.2.2 From 7c4e6492205b677a5786b85bcf72ce7c8f4adf15 Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Wed, 9 Dec 2009 12:05:50 +0200 Subject: ASoC: tpa6130a2: Add support for regulator framework Take the regulator framework in use for managing the power sources Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Acked-by: Eduardo Valentin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 87 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 70 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 6b650c1aa3d1..0eb33d49942e 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -34,10 +35,17 @@ static struct i2c_client *tpa6130a2_client; +#define TPA6130A2_NUM_SUPPLIES 2 +static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { + "CPVSS", + "Vdd", +}; + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; unsigned char regs[TPA6130A2_CACHEREGNUM]; + struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; int power_gpio; unsigned char power_state; }; @@ -106,10 +114,11 @@ static void tpa6130a2_initialize(void) tpa6130a2_i2c_write(i, data->regs[i]); } -static void tpa6130a2_power(int power) +static int tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; + int ret; BUG_ON(tpa6130a2_client == NULL); data = i2c_get_clientdata(tpa6130a2_client); @@ -117,11 +126,20 @@ static void tpa6130a2_power(int power) mutex_lock(&data->mutex); if (power) { /* Power on */ - if (data->power_gpio >= 0) { + if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - data->power_state = 1; - tpa6130a2_initialize(); + + ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(&tpa6130a2_client->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + data->power_state = 1; + tpa6130a2_initialize(); + /* Clear SWS */ val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val &= ~TPA6130A2_SWS; @@ -131,13 +149,25 @@ static void tpa6130a2_power(int power) val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val |= TPA6130A2_SWS; tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + /* Power off */ - if (data->power_gpio >= 0) { + if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - data->power_state = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(&tpa6130a2_client->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + data->power_state = 0; } + +exit: mutex_unlock(&data->mutex); + return ret; } static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, @@ -299,15 +329,17 @@ static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w, static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + int ret = 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: - tpa6130a2_power(1); + ret = tpa6130a2_power(1); break; case SND_SOC_DAPM_POST_PMD: - tpa6130a2_power(0); + ret = tpa6130a2_power(0); break; } - return 0; + return ret; } static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { @@ -352,7 +384,7 @@ static int tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; - int ret; + int i, ret; dev = &client->dev; @@ -387,15 +419,25 @@ static int tpa6130a2_probe(struct i2c_client *client, if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); - goto fail; + goto err_gpio; } gpio_direction_output(data->power_gpio, 0); - } else { - data->power_state = 1; - tpa6130a2_initialize(); } - tpa6130a2_power(1); + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + + ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + goto err_regulator; + } + + ret = tpa6130a2_power(1); + if (ret != 0) + goto err_power; + /* Read version */ ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) & @@ -404,10 +446,18 @@ static int tpa6130a2_probe(struct i2c_client *client, dev_warn(dev, "UNTESTED version detected (%d)\n", ret); /* Disable the chip */ - tpa6130a2_power(0); + ret = tpa6130a2_power(0); + if (ret != 0) + goto err_power; return 0; -fail: + +err_power: + regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); +err_regulator: + if (data->power_gpio >= 0) + gpio_free(data->power_gpio); +err_gpio: kfree(data); i2c_set_clientdata(tpa6130a2_client, NULL); tpa6130a2_client = NULL; @@ -423,6 +473,9 @@ static int tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); + + regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + kfree(data); tpa6130a2_client = NULL; -- cgit v1.2.2 From 98615454f66175e923f239ab1d1bd85cd618363e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Dec 2009 13:21:56 +0900 Subject: ASoC: Add DA7210 codec device support for ALSA This original driver was created by Dialog Semiconductor, and cleanuped by Kuninori Morimoto. Special thanks to David Chen. This became very simple ASoC codec driver, and it is tested by EcoVec24 board. Signed-off-by: David Chen Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7210.c | 586 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7210.h | 24 ++ 4 files changed, 616 insertions(+) create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 011d3ab7e64a..691abe7df087 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -23,6 +23,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_DA7210 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -113,6 +114,9 @@ config SND_SOC_AK4671 config SND_SOC_CS4270 tristate +config SND_SOC_DA7210 + tristate + # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0471d9044205..b328f293be65 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -10,6 +10,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o +snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o @@ -67,6 +68,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o +obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c new file mode 100644 index 000000000000..14f5f344b1d5 --- /dev/null +++ b/sound/soc/codecs/da7210.c @@ -0,0 +1,586 @@ +/* + * DA7210 ALSA Soc codec driver + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto + * + * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "da7210.h" + +/* DA7210 register space */ +#define DA7210_STATUS 0x02 +#define DA7210_STARTUP1 0x03 +#define DA7210_MIC_L 0x07 +#define DA7210_MIC_R 0x08 +#define DA7210_INMIX_L 0x0D +#define DA7210_INMIX_R 0x0E +#define DA7210_ADC_HPF 0x0F +#define DA7210_ADC 0x10 +#define DA7210_DAC_HPF 0x14 +#define DA7210_DAC_L 0x15 +#define DA7210_DAC_R 0x16 +#define DA7210_DAC_SEL 0x17 +#define DA7210_OUTMIX_L 0x1C +#define DA7210_OUTMIX_R 0x1D +#define DA7210_HP_L_VOL 0x21 +#define DA7210_HP_R_VOL 0x22 +#define DA7210_HP_CFG 0x23 +#define DA7210_DAI_SRC_SEL 0x25 +#define DA7210_DAI_CFG1 0x26 +#define DA7210_DAI_CFG3 0x28 +#define DA7210_PLL_DIV3 0x2B +#define DA7210_PLL 0x2C + +/* STARTUP1 bit fields */ +#define DA7210_SC_MST_EN (1 << 0) + +/* MIC_L bit fields */ +#define DA7210_MICBIAS_EN (1 << 6) +#define DA7210_MIC_L_EN (1 << 7) + +/* MIC_R bit fields */ +#define DA7210_MIC_R_EN (1 << 7) + +/* INMIX_L bit fields */ +#define DA7210_IN_L_EN (1 << 7) + +/* INMIX_R bit fields */ +#define DA7210_IN_R_EN (1 << 7) + +/* ADC_HPF bit fields */ +#define DA7210_ADC_VOICE_EN (1 << 7) + +/* ADC bit fields */ +#define DA7210_ADC_L_EN (1 << 3) +#define DA7210_ADC_R_EN (1 << 7) + +/* DAC_SEL bit fields */ +#define DA7210_DAC_L_SRC_DAI_L (4 << 0) +#define DA7210_DAC_L_EN (1 << 3) +#define DA7210_DAC_R_SRC_DAI_R (5 << 4) +#define DA7210_DAC_R_EN (1 << 7) + +/* OUTMIX_L bit fields */ +#define DA7210_OUT_L_EN (1 << 7) + +/* OUTMIX_R bit fields */ +#define DA7210_OUT_R_EN (1 << 7) + +/* HP_CFG bit fields */ +#define DA7210_HP_2CAP_MODE (1 << 1) +#define DA7210_HP_SENSE_EN (1 << 2) +#define DA7210_HP_L_EN (1 << 3) +#define DA7210_HP_MODE (1 << 6) +#define DA7210_HP_R_EN (1 << 7) + +/* DAI_SRC_SEL bit fields */ +#define DA7210_DAI_OUT_L_SRC (6 << 0) +#define DA7210_DAI_OUT_R_SRC (7 << 4) + +/* DAI_CFG1 bit fields */ +#define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_MASTER (1 << 7) + +/* DAI_CFG3 bit fields */ +#define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_OE (1 << 3) +#define DA7210_DAI_EN (1 << 7) + +/*PLL_DIV3 bit fields */ +#define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) +#define DA7210_PLL_BYP (1 << 6) + +/* PLL bit fields */ +#define DA7210_PLL_FS_48000 (11 << 0) + +#define DA7210_VERSION "0.0.1" + +/* Codec private data */ +struct da7210_priv { + struct snd_soc_codec codec; +}; + +static struct snd_soc_codec *da7210_codec; + +/* + * Register cache + */ +static const u8 da7210_reg[] = { + 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */ + 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */ + 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */ + 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */ + 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */ + 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */ + 0x00, /* R88 */ +}; + +/* + * Read da7210 register cache + */ +static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) +{ + u8 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(da7210_reg)); + return cache[reg]; +} + +/* + * Write to the da7210 register space + */ +static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg >= codec->reg_cache_size) + return -EIO; + + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + + cache[reg] = value; + return 0; +} + +/* + * Read from the da7210 register space. + */ +static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) +{ + if (DA7210_STATUS == reg) + return i2c_smbus_read_byte_data(codec->control_data, reg); + + return da7210_read_reg_cache(codec, reg); +} + +static int da7210_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* PlayBack Volume 40 */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + + /* Enable Out */ + snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); + snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); + + } else { + /* Volume 7 */ + snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); + snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); + + /* Enable Mic */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + } + + return 0; +} + +/* + * Set PCM DAI word length. + */ +static int da7210_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u32 dai_cfg1; + u32 reg, mask; + + /* set DAI source to Left and Right ADC */ + da7210_write(codec, DA7210_DAI_SRC_SEL, + DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); + + /* Enable DAI */ + da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + + dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + dai_cfg1 |= DA7210_DAI_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dai_cfg1 |= DA7210_DAI_WORD_S24_LE; + break; + default: + return -EINVAL; + } + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + + /* FIXME + * + * It support 48K only now + */ + switch (params_rate(params)) { + case 48000: + if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { + reg = DA7210_DAC_HPF; + mask = DA7210_DAC_VOICE_EN; + } else { + reg = DA7210_ADC_HPF; + mask = DA7210_ADC_VOICE_EN; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg, mask, 0); + + return 0; +} + +/* + * Set DAI mode and Format + */ +static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 dai_cfg1; + u32 dai_cfg3; + + dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_cfg1 |= DA7210_DAI_MODE_MASTER; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support I2S only now + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support 64bit data transmission only now + */ + dai_cfg1 |= DA7210_DAI_FLEN_64BIT; + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + + return 0; +} + +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +/* DAI operations */ +static struct snd_soc_dai_ops da7210_dai_ops = { + .startup = da7210_startup, + .hw_params = da7210_hw_params, + .set_fmt = da7210_set_dai_fmt, +}; + +struct snd_soc_dai da7210_dai = { + .name = "DA7210 IIS", + .id = 0, + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + .ops = &da7210_dai_ops, +}; +EXPORT_SYMBOL_GPL(da7210_dai); + +/* + * Initialize the DA7210 driver + * register the mixer and dsp interfaces with the kernel + */ +static int da7210_init(struct da7210_priv *da7210) +{ + struct snd_soc_codec *codec = &da7210->codec; + int ret = 0; + + if (da7210_codec) { + dev_err(codec->dev, "Another da7210 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = da7210; + codec->name = "DA7210"; + codec->owner = THIS_MODULE; + codec->read = da7210_read; + codec->write = da7210_write; + codec->dai = &da7210_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(da7210_reg); + codec->reg_cache = kmemdup(da7210_reg, + sizeof(da7210_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + da7210_dai.dev = codec->dev; + da7210_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register CODEC: %d\n", ret); + goto init_err; + } + + ret = snd_soc_register_dai(&da7210_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto init_err; + } + + /* FIXME + * + * This driver use fixed value here + */ + + /* + * ADC settings + */ + + /* Enable Left & Right MIC PGA and Mic Bias */ + da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + + /* Enable Left and Right input PGA */ + da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + + /* Enable Left and Right ADC */ + da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + + /* + * DAC settings + */ + + /* Enable Left and Right DAC */ + da7210_write(codec, DA7210_DAC_SEL, + DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | + DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); + + /* Enable Left and Right out PGA */ + da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + + /* Enable Left and Right HeadPhone PGA */ + da7210_write(codec, DA7210_HP_CFG, + DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | + DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + + /* Diable PLL and bypass it */ + da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + + /* Bypass PLL and set MCLK freq rang to 10-20MHz */ + da7210_write(codec, DA7210_PLL_DIV3, + DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + + /* Activate all enabled subsystem */ + da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + + return ret; + +init_err: + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return ret; + +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int da7210_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7210_priv *da7210; + struct snd_soc_codec *codec; + int ret; + + da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + codec = &da7210->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, da7210); + codec->control_data = i2c; + + ret = da7210_init(da7210); + if (ret < 0) + pr_err("Failed to initialise da7210 audio codec\n"); + + return ret; +} + +static int da7210_i2c_remove(struct i2c_client *client) +{ + struct da7210_priv *da7210 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&da7210_dai); + kfree(da7210->codec.reg_cache); + kfree(da7210); + da7210_codec = NULL; + + return 0; +} + +static const struct i2c_device_id da7210_i2c_id[] = { + { "da7210", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da7210_i2c_driver = { + .driver = { + .name = "DA7210 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = da7210_i2c_probe, + .remove = __devexit_p(da7210_i2c_remove), + .id_table = da7210_i2c_id, +}; +#endif + +static int da7210_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (!da7210_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = da7210_codec; + codec = da7210_codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); + +pcm_err: + return ret; +} + +static int da7210_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_da7210 = { + .probe = da7210_probe, + .remove = da7210_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_da7210); + +static int __init da7210_modinit(void) +{ + int ret = 0; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&da7210_i2c_driver); +#endif + return ret; +} +module_init(da7210_modinit); + +static void __exit da7210_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&da7210_i2c_driver); +#endif +} +module_exit(da7210_exit); + +MODULE_DESCRIPTION("ASoC DA7210 driver"); +MODULE_AUTHOR("David Chen, Kuninori Morimoto"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7210.h b/sound/soc/codecs/da7210.h new file mode 100644 index 000000000000..390d621eb742 --- /dev/null +++ b/sound/soc/codecs/da7210.h @@ -0,0 +1,24 @@ +/* + * da7210.h -- audio driver for da7210 + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _DA7210_H +#define _DA7210_H + +extern struct snd_soc_dai da7210_dai; +extern struct snd_soc_codec_device soc_codec_dev_da7210; + +#endif + -- cgit v1.2.2 From 038494059f795849012a96adba2ab73e65b94ba5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Dec 2009 13:22:00 +0900 Subject: ASoC: Add FSI-DA7210 sound support for SuperH Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 8 +++++ sound/soc/sh/Makefile | 2 ++ sound/soc/sh/fsi-da7210.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 93 insertions(+) create mode 100644 sound/soc/sh/fsi-da7210.c (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9e6976586554..8072a6d1c4db 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -47,4 +47,12 @@ config SND_FSI_AK4642 This option enables generic sound support for the FSI - AK4642 unit +config SND_FSI_DA7210 + bool "FSI-DA7210 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_DA7210 + help + This option enables generic sound support for the + FSI - DA7210 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a6997872f24e..1d0ec0af74b7 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -13,6 +13,8 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o +snd-soc-fsi-da7210-objs := fsi-da7210.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o +obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c new file mode 100644 index 000000000000..33b4d177f466 --- /dev/null +++ b/sound/soc/sh/fsi-da7210.c @@ -0,0 +1,83 @@ +/* + * fsi-da7210.c + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "../codecs/da7210.h" + +static int fsi_da7210_init(struct snd_soc_codec *codec) +{ + return snd_soc_dai_set_fmt(&da7210_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_dai_link fsi_da7210_dai = { + .name = "DA7210", + .stream_name = "DA7210", + .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .codec_dai = &da7210_dai, + .init = fsi_da7210_init, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_da7210_dai, + .num_links = 1, +}; + +static struct snd_soc_device fsi_da7210_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_da7210, +}; + +static struct platform_device *fsi_da7210_snd_device; + +static int __init fsi_da7210_sound_init(void) +{ + int ret; + + fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_da7210_snd_device) + return -ENOMEM; + + platform_set_drvdata(fsi_da7210_snd_device, &fsi_da7210_snd_devdata); + fsi_da7210_snd_devdata.dev = &fsi_da7210_snd_device->dev; + ret = platform_device_add(fsi_da7210_snd_device); + if (ret) + platform_device_put(fsi_da7210_snd_device); + + return ret; +} + +static void __exit fsi_da7210_sound_exit(void) +{ + platform_device_unregister(fsi_da7210_snd_device); +} + +module_init(fsi_da7210_sound_init); +module_exit(fsi_da7210_sound_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC FSI DA2710"); +MODULE_AUTHOR("Kuninori Morimoto "); +MODULE_LICENSE("GPL"); -- cgit v1.2.2 From ebeb53e1e1f11a51d8a93843a437f516e3528bfa Mon Sep 17 00:00:00 2001 From: Balaji T K Date: Tue, 15 Dec 2009 20:09:02 +0530 Subject: mfd: twl: fix twl4030 rename for remaining driver, board files Recent drivers/mfd/twl4030* renames to twl broke compile for various boards as the series was missing a patch to change the board-*.c files. This patch renames include twl4030.h to include twl.h and also renames twl4030_i2c_ routines. Signed-off-by: Balaji T K Acked-by: Mark Brown Reviewed-by: Felipe Balbi Cc: Samuel Ortiz Signed-off-by: Tony Lindgren --- sound/soc/omap/sdp3430.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index c071f9603a38..3c85c0f92823 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,7 +24,7 @@ #include #include -#include +#include #include #include #include @@ -321,11 +321,11 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, TWL4030_INTBR_PMBR1); pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); -- cgit v1.2.2 From 3497b91946a3df42830c826939424d98251a3b0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 15 Dec 2009 20:58:56 +0000 Subject: ASoC: Fix sorting of codecs Makefile entries Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b328f293be65..c0fd3c86edad 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -96,11 +96,11 @@ obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o -obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o -obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o +obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o -- cgit v1.2.2 From 255173b40db448ce063a2caa680a552fb637ad20 Mon Sep 17 00:00:00 2001 From: Peter Meerwald Date: Mon, 14 Dec 2009 14:44:56 +0100 Subject: ASoC: PLL computation in TLV320AIC3x SoC driver fix precision of PLL computation for TLV320AIC3x SoC driver, test results are at http://pmeerw.net/clk Signed-off-by: Peter Meerwald Acked-by: Vladimir Barinov Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 75 +++++++++++++++++++++++++++--------------- 1 file changed, 49 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 2b4dc2b0b017..5a8f53ce2250 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -765,9 +765,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_priv *aic3x = codec->private_data; int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; - u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; - u16 pll_d = 1; + u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 d, pll_d = 1; u8 reg; + int clk; /* select data word length */ data = @@ -833,48 +834,70 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, if (bypass_pll) return 0; - /* Use PLL - * find an apropriate setup for j, d, r and p by iterating over - * p and r - j and d are calculated for each fraction. - * Up to 128 values are probed, the closest one wins the game. + /* Use PLL, compute apropriate setup for j, d, r and p, the closest + * one wins the game. Try with d==0 first, next with d!=0. + * Constraints for j are according to the datasheet. * The sysclk is divided by 1000 to prevent integer overflows. */ + codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); for (r = 1; r <= 16; r++) for (p = 1; p <= 8; p++) { - int clk, tmp = (codec_clk * pll_r * 10) / pll_p; - u8 j = tmp / 10000; - u16 d = tmp % 10000; + for (j = 4; j <= 55; j++) { + /* This is actually 1000*((j+(d/10000))*r)/p + * The term had to be converted to get + * rid of the division by 10000; d = 0 here + */ + int clk = (1000 * j * r) / p; + + /* Check whether this values get closer than + * the best ones we had before + */ + if (abs(codec_clk - clk) < + abs(codec_clk - last_clk)) { + pll_j = j; pll_d = 0; + pll_r = r; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } + } - if (j > 63) - continue; + /* try with d != 0 */ + for (p = 1; p <= 8; p++) { + j = codec_clk * p / 1000; - if (d != 0 && aic3x->sysclk < 10000000) - continue; + if (j < 4 || j > 11) + continue; - /* This is actually 1000 * ((j + (d/10000)) * r) / p - * The term had to be converted to get rid of the - * division by 10000 */ - clk = ((10000 * j * r) + (d * r)) / (10 * p); + /* do not use codec_clk here since we'd loose precision */ + d = ((2048 * p * fsref) - j * aic3x->sysclk) + * 100 / (aic3x->sysclk/100); - /* check whether this values get closer than the best - * ones we had before */ - if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { - pll_j = j; pll_d = d; pll_r = r; pll_p = p; - last_clk = clk; - } + clk = (10000 * j + d) / (10 * p); - /* Early exit for exact matches */ - if (clk == codec_clk) - break; + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = 1; pll_p = p; + last_clk = clk; } + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } + if (last_clk == 0) { printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); return -EINVAL; } +found: data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); -- cgit v1.2.2 From c2151433847e88ba05c6bb539d9397ea90d755e6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Dec 2009 20:36:37 +0000 Subject: ASoC: Fix build of DA7210 DAC_VOICE_EN was not defined - looks to have been overly enthusiastically deleted from a previous revision of the patch, pull the value from v1. Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 14f5f344b1d5..fbf3ab482015 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -81,6 +81,9 @@ #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) +/* DAC_HPF fields */ +#define DA7210_DAC_VOICE_EN (1 << 7) + /* DAC_SEL bit fields */ #define DA7210_DAC_L_SRC_DAI_L (4 << 0) #define DA7210_DAC_L_EN (1 << 3) -- cgit v1.2.2 From 2fbe74b90bafebce615466b4c20f96b0465df1ae Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 16 Dec 2009 16:54:43 +0100 Subject: sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot() limit and jiffies are unsigned so the test did not work. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/pss.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 83f5ee236b12..e19dd5dcc2de 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -269,7 +269,7 @@ static int pss_reset_dsp(pss_confdata * devc) unsigned long i, limit = jiffies + HZ/10; outw(0x2000, REG(PSS_CONTROL)); - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) inw(REG(PSS_CONTROL)); outw(0x0000, REG(PSS_CONTROL)); return 1; @@ -369,11 +369,11 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size outw(0, REG(PSS_DATA)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit - jiffies >= 0); i++) + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) val = inw(REG(PSS_STATUS)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) { val = inw(REG(PSS_STATUS)); if (val & 0x4000) -- cgit v1.2.2 From ebb83eeb6469bedda83b4dc6f23ddf93eb32b347 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 17 Dec 2009 12:23:00 +0100 Subject: ALSA: hda - More ALC663 fixes and support of compatible chips 1. Add more ASUS NB model. 2. Fixed alc663_m51va_setup M51VA has Digital Mic that NID is 0x12. The record source index is 0x9 for ALC663. So, to modify the alc663_m51va_setup function to index 0x9 and add analog Mic aupport function alc663_mode1_setup. 3. Add ASUS mode7 and mode8 modules for ALC663 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 306 ++++++++++++++++++++++++++++++++++++++---- 1 file changed, 282 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c9e860709747..287bb6019df9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -131,8 +131,8 @@ enum { enum { ALC269_BASIC, ALC269_QUANTA_FL1, - ALC269_ASUS_EEEPC_P703, - ALC269_ASUS_EEEPC_P901, + ALC269_ASUS_AMIC, + ALC269_ASUS_DMIC, ALC269_FUJITSU, ALC269_LIFEBOOK, ALC269_AUTO, @@ -188,6 +188,8 @@ enum { ALC663_ASUS_MODE4, ALC663_ASUS_MODE5, ALC663_ASUS_MODE6, + ALC663_ASUS_MODE7, + ALC663_ASUS_MODE8, ALC272_DELL, ALC272_DELL_ZM1, ALC272_SAMSUNG_NC10, @@ -13232,10 +13234,12 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc269_speaker_automute(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + unsigned int nid = spec->autocfg.hp_pins[0]; unsigned int present; unsigned char bits; - present = snd_hda_jack_detect(codec, 0x15); + present = snd_hda_jack_detect(codec, nid); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -13460,8 +13464,8 @@ static void alc269_auto_init(struct hda_codec *codec) static const char *alc269_models[ALC269_MODEL_LAST] = { [ALC269_BASIC] = "basic", [ALC269_QUANTA_FL1] = "quanta", - [ALC269_ASUS_EEEPC_P703] = "eeepc-p703", - [ALC269_ASUS_EEEPC_P901] = "eeepc-p901", + [ALC269_ASUS_AMIC] = "asus-amic", + [ALC269_ASUS_DMIC] = "asus-dmic", [ALC269_FUJITSU] = "fujitsu", [ALC269_LIFEBOOK] = "lifebook", [ALC269_AUTO] = "auto", @@ -13470,18 +13474,41 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { static struct snd_pci_quirk alc269_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1), SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", - ALC269_ASUS_EEEPC_P703), - SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_EEEPC_P703), - SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_EEEPC_P703), - SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_EEEPC_P703), - SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_EEEPC_P703), - SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_EEEPC_P703), - SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_EEEPC_P703), + ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80JT", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_DMIC), + SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_AMIC), + SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_AMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901", - ALC269_ASUS_EEEPC_P901), + ALC269_ASUS_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", - ALC269_ASUS_EEEPC_P901), - SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_EEEPC_P901), + ALC269_ASUS_DMIC), + SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_ASUS_DMIC), + SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_ASUS_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), {} @@ -13511,7 +13538,7 @@ static struct alc_config_preset alc269_presets[] = { .setup = alc269_quanta_fl1_setup, .init_hook = alc269_quanta_fl1_init_hook, }, - [ALC269_ASUS_EEEPC_P703] = { + [ALC269_ASUS_AMIC] = { .mixers = { alc269_eeepc_mixer }, .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, @@ -13525,7 +13552,7 @@ static struct alc_config_preset alc269_presets[] = { .setup = alc269_eeepc_amic_setup, .init_hook = alc269_eeepc_inithook, }, - [ALC269_ASUS_EEEPC_P901] = { + [ALC269_ASUS_DMIC] = { .mixers = { alc269_eeepc_mixer }, .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, @@ -16160,6 +16187,52 @@ static struct snd_kcontrol_new alc663_g50v_mixer[] = { { } /* end */ }; +static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc663_mode7_mixer[] = { + HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch), + HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol), + HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch), + HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_mode8_mixer[] = { + HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch), + HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol), + HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch), + HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + + static struct snd_kcontrol_new alc662_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -16447,6 +16520,45 @@ static struct hda_verb alc272_dell_init_verbs[] = { {} }; +static struct hda_verb alc663_mode7_init_verbs[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_mode8_init_verbs[] = { + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + static struct snd_kcontrol_new alc662_auto_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), @@ -16626,6 +16738,54 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) } } +static void alc663_two_hp_m7_speaker_automute(struct hda_codec *codec) +{ + unsigned int present1, present2; + + present1 = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + present2 = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + if (present1 || present2) { + snd_hda_codec_write_cache(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } else { + snd_hda_codec_write_cache(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + } +} + +static void alc663_two_hp_m8_speaker_automute(struct hda_codec *codec) +{ + unsigned int present1, present2; + + present1 = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + present2 = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + if (present1 || present2) { + snd_hda_codec_write_cache(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } else { + snd_hda_codec_write_cache(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + } +} + static void alc663_m51va_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -16645,7 +16805,7 @@ static void alc663_m51va_setup(struct hda_codec *codec) spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 1; + spec->int_mic.mux_idx = 9; spec->auto_mic = 1; } @@ -16657,7 +16817,17 @@ static void alc663_m51va_inithook(struct hda_codec *codec) /* ***************** Mode1 ******************************/ #define alc663_mode1_unsol_event alc663_m51va_unsol_event -#define alc663_mode1_setup alc663_m51va_setup + +static void alc663_mode1_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; +} + #define alc663_mode1_inithook alc663_m51va_inithook /* ***************** Mode2 ******************************/ @@ -16674,7 +16844,7 @@ static void alc662_mode2_unsol_event(struct hda_codec *codec, } } -#define alc662_mode2_setup alc663_m51va_setup +#define alc662_mode2_setup alc663_mode1_setup static void alc662_mode2_inithook(struct hda_codec *codec) { @@ -16695,7 +16865,7 @@ static void alc663_mode3_unsol_event(struct hda_codec *codec, } } -#define alc663_mode3_setup alc663_m51va_setup +#define alc663_mode3_setup alc663_mode1_setup static void alc663_mode3_inithook(struct hda_codec *codec) { @@ -16716,7 +16886,7 @@ static void alc663_mode4_unsol_event(struct hda_codec *codec, } } -#define alc663_mode4_setup alc663_m51va_setup +#define alc663_mode4_setup alc663_mode1_setup static void alc663_mode4_inithook(struct hda_codec *codec) { @@ -16737,7 +16907,7 @@ static void alc663_mode5_unsol_event(struct hda_codec *codec, } } -#define alc663_mode5_setup alc663_m51va_setup +#define alc663_mode5_setup alc663_mode1_setup static void alc663_mode5_inithook(struct hda_codec *codec) { @@ -16758,7 +16928,7 @@ static void alc663_mode6_unsol_event(struct hda_codec *codec, } } -#define alc663_mode6_setup alc663_m51va_setup +#define alc663_mode6_setup alc663_mode1_setup static void alc663_mode6_inithook(struct hda_codec *codec) { @@ -16766,6 +16936,50 @@ static void alc663_mode6_inithook(struct hda_codec *codec) alc_mic_automute(codec); } +/* ***************** Mode7 ******************************/ +static void alc663_mode7_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_two_hp_m7_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } +} + +#define alc663_mode7_setup alc663_mode1_setup + +static void alc663_mode7_inithook(struct hda_codec *codec) +{ + alc663_two_hp_m7_speaker_automute(codec); + alc_mic_automute(codec); +} + +/* ***************** Mode8 ******************************/ +static void alc663_mode8_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_two_hp_m8_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } +} + +#define alc663_mode8_setup alc663_m51va_setup + +static void alc663_mode8_inithook(struct hda_codec *codec) +{ + alc663_two_hp_m8_speaker_automute(codec); + alc_mic_automute(codec); +} + static void alc663_g71v_hp_automute(struct hda_codec *codec) { unsigned int present; @@ -16900,6 +17114,8 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC663_ASUS_MODE4] = "asus-mode4", [ALC663_ASUS_MODE5] = "asus-mode5", [ALC663_ASUS_MODE6] = "asus-mode6", + [ALC663_ASUS_MODE7] = "asus-mode7", + [ALC663_ASUS_MODE8] = "asus-mode8", [ALC272_DELL] = "dell", [ALC272_DELL_ZM1] = "dell-zm1", [ALC272_SAMSUNG_NC10] = "samsung-nc10", @@ -16916,12 +17132,22 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7), + SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7), + SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8), + SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA), SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2), @@ -17205,6 +17431,36 @@ static struct alc_config_preset alc662_presets[] = { .setup = alc663_mode6_setup, .init_hook = alc663_mode6_inithook, }, + [ALC663_ASUS_MODE7] = { + .mixers = { alc663_mode7_mixer }, + .cap_mixer = alc662_auto_capture_mixer, + .init_verbs = { alc662_init_verbs, + alc663_mode7_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .unsol_event = alc663_mode7_unsol_event, + .setup = alc663_mode7_setup, + .init_hook = alc663_mode7_inithook, + }, + [ALC663_ASUS_MODE8] = { + .mixers = { alc663_mode8_mixer }, + .cap_mixer = alc662_auto_capture_mixer, + .init_verbs = { alc662_init_verbs, + alc663_mode8_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .unsol_event = alc663_mode8_unsol_event, + .setup = alc663_mode8_setup, + .init_hook = alc663_mode8_inithook, + }, [ALC272_DELL] = { .mixers = { alc663_m51va_mixer }, .cap_mixer = alc272_auto_capture_mixer, @@ -17688,7 +17944,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 }, + { .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 }, { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, + { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- cgit v1.2.2 From 254bba6a7e28c77d8b32d9eaeba02d4943ee844e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Einar=20R=C3=BCnkaru?= Date: Wed, 16 Dec 2009 22:16:13 +0200 Subject: ALSA: hda - Fixed internal mic initialization for Dell Vostro 1015 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixed initialization of internal mic and added internal mic boost control Renamed analog mic boost control to ext mic boost contol. Name pair analog/digital seems too confusing for a normal user. Signed-off-by: Einar Rünkaru Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 39 +++++++++++++++++++++++++++++++++------ 1 file changed, 33 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a09c03c3f62b..ca9ad9fddbf2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -111,6 +111,7 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; unsigned char ext_mic_bias; + unsigned int dell_vostro; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -2109,9 +2110,12 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int val; + hda_nid_t nid = kcontrol->private_value & 0xff; + int inout = (kcontrol->private_value & 0x100) ? + AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT; - val = snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT); + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_AMP_GAIN_MUTE, inout); ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; return 0; @@ -2123,6 +2127,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; unsigned int idx; + hda_nid_t nid = kcontrol->private_value & 0xff; + int inout = (kcontrol->private_value & 0x100) ? + AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT; if (!imux->num_items) return 0; @@ -2130,9 +2137,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, if (idx >= imux->num_items) idx = imux->num_items - 1; - snd_hda_codec_write_cache(codec, 0x17, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout | imux->items[idx].index); return 1; @@ -2201,10 +2208,11 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Mic Boost Capture Enum", + .name = "Ext Mic Boost Capture Enum", .info = cxt5066_mic_boost_mux_enum_info, .get = cxt5066_mic_boost_mux_enum_get, .put = cxt5066_mic_boost_mux_enum_put, + .private_value = 0x17, }, HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), @@ -2212,6 +2220,18 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Int Mic Boost Capture Enum", + .info = cxt5066_mic_boost_mux_enum_info, + .get = cxt5066_mic_boost_mux_enum_get, + .put = cxt5066_mic_boost_mux_enum_put, + .private_value = 0x23 | 0x100, + }, + {} +}; + static struct hda_verb cxt5066_init_verbs[] = { {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ @@ -2397,11 +2417,16 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + snd_printdd("CXT5066: init\n"); conexant_init(codec); if (codec->patch_ops.unsol_event) { cxt5066_hp_automute(codec); - cxt5066_automic(codec); + if (spec->dell_vostro) + cxt5066_vostro_automic(codec); + else + cxt5066_automic(codec); } return 0; } @@ -2500,7 +2525,9 @@ static int patch_cxt5066(struct hda_codec *codec) spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; + spec->dell_vostro = 1; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; -- cgit v1.2.2 From c0f8faf0c7cd497ec7c1d61e1e9424f4384c1f68 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Einar=20R=C3=BCnkaru?= Date: Wed, 16 Dec 2009 22:41:36 +0200 Subject: ALSA: hda - Make use of beep device found in Dell Vostro 1015n MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Conexant CX20583-10Z has digital beep device with volume control. Making use of them. Signed-off-by: Einar Rünkaru Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ca9ad9fddbf2..c578c28f368e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -29,6 +29,7 @@ #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define CXT_PIN_DIR_IN 0x00 #define CXT_PIN_DIR_OUT 0x01 @@ -477,6 +478,7 @@ static void conexant_free(struct hda_codec *codec) snd_array_free(&spec->jacks); } #endif + snd_hda_detach_beep_device(codec); kfree(codec->spec); } @@ -2229,6 +2231,7 @@ static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { .put = cxt5066_mic_boost_mux_enum_put, .private_value = 0x23 | 0x100, }, + HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), {} }; @@ -2528,6 +2531,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; spec->dell_vostro = 1; + snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; -- cgit v1.2.2 From 035eb0cff0671ada49ba9f3e5c9e7b0cb950efea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Dec 2009 15:00:26 +0100 Subject: ALSA: hda - Fix missing capsrc_nids for ALC88x Some model quirks missed the corresponding capsrc_nids. This resulted in non-working capture source selection. Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 287bb6019df9..d9a9f0c7cf5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9238,6 +9238,8 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), .adc_nids = alc889_adc_nids, + .capsrc_nids = alc889_capsrc_nids, + .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, .slave_dig_outs = alc883_slave_dig_outs, @@ -9284,6 +9286,7 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), + .capsrc_nids = alc883_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, @@ -9430,6 +9433,7 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), + .capsrc_nids = alc883_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, @@ -9491,6 +9495,7 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), + .capsrc_nids = alc883_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_lenovo_101e_capture_source, @@ -9670,6 +9675,7 @@ static struct alc_config_preset alc882_presets[] = { alc880_gpio1_init_verbs }, .adc_nids = alc883_adc_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .capsrc_nids = alc883_capsrc_nids, .dac_nids = alc883_dac_nids, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .channel_mode = alc889A_mb31_6ch_modes, -- cgit v1.2.2 From d1409ae4cecb4af260759bdfdf88fafca23a9940 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Dec 2009 15:01:31 +0100 Subject: ALSA: hda - Fix NULL dereference in kctl-NID mapping in patch_realtek.c capsrc_nids can be NULL, and adc_nids should be taken as fallback. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 36556b10357a..012435212e58 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2517,7 +2517,10 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + hda_nid_t *nids = spec->capsrc_nids; + if (!nids) + nids = spec->adc_nids; + err = snd_hda_add_nids(codec, kctl, i, nids, spec->input_mux->num_items); if (err < 0) return err; -- cgit v1.2.2 From 2fef62c825f09e29d2f52dc187ddf6f99e28c7f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Dec 2009 08:48:42 +0100 Subject: ALSA: hda - Fix quirk for Maxdata obook4-1 Works fine with the auto-parser. Reference: Novell bnc#564940 https://bugzilla.novell.com/show_bug.cgi?id=564940 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d9a9f0c7cf5b..84bc2c7c4421 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8921,7 +8921,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ - SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), -- cgit v1.2.2 From 3e85fd614c7b6bb7f33bb04a0dcb5a3bfca4c0fe Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:27:24 +0100 Subject: sound: sgio2audio/pdaudiocf/usb-audio: initialize PCM buffer When allocating the PCM buffer, use vmalloc_user() instead of vmalloc(). Otherwise, it would be possible for applications to play the previous contents of the kernel memory to the speakers, or to read it directly if the buffer is exported to userspace. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 2 +- sound/usb/usbaudio.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 8691f4cf6191..f1d9d16b5486 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -609,7 +609,7 @@ static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, /* alloc virtual 'dma' area */ if (runtime->dma_area) vfree(runtime->dma_area); - runtime->dma_area = vmalloc(size); + runtime->dma_area = vmalloc_user(size); if (runtime->dma_area == NULL) return -ENOMEM; runtime->dma_bytes = size; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index d057e6489643..5cfa608823f7 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -51,7 +51,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s return 0; /* already enough large */ vfree(runtime->dma_area); } - runtime->dma_area = vmalloc_32(size); + runtime->dma_area = vmalloc_32_user(size); if (! runtime->dma_area) return -ENOMEM; runtime->dma_bytes = size; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b074a594c595..4963defee18a 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -752,7 +752,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s return 0; /* already large enough */ vfree(runtime->dma_area); } - runtime->dma_area = vmalloc(size); + runtime->dma_area = vmalloc_user(size); if (!runtime->dma_area) return -ENOMEM; runtime->dma_bytes = size; -- cgit v1.2.2 From 681b84e17747e1c208e8e1acc54cc5e612da84d1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:00 +0100 Subject: sound: pcm: add vmalloc buffer helper functions There are now five copies of the code to allocate a PCM buffer using vmalloc(). Add a sixth in the core so that the others can be removed. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 54 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 54 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index caa7796bc2f5..d9727c74b2e1 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -434,3 +434,57 @@ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(snd_pcm_lib_free_pages); + +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = __vmalloc(size, gfp_flags, PAGE_KERNEL); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(_snd_pcm_lib_alloc_vmalloc_buffer); + +/** + * snd_pcm_lib_free_vmalloc_buffer - free vmalloc buffer + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + */ +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; +} +EXPORT_SYMBOL(snd_pcm_lib_free_vmalloc_buffer); + +/** + * snd_pcm_lib_get_vmalloc_page - map vmalloc buffer offset to page struct + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + * @offset: offset in the buffer + * + * This function is to be used as the page callback in the PCM ops. + */ +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} +EXPORT_SYMBOL(snd_pcm_lib_get_vmalloc_page); -- cgit v1.2.2 From d20fb5dc076a4cf0fdd00ab5a4e752ea3611e484 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:49 +0100 Subject: sound: pdaudiocf: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 51 +++------------------------------- 1 file changed, 4 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 5cfa608823f7..0afa683c900e 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -21,56 +21,12 @@ */ #include -#include #include #include #include #include "pdaudiocf.h" -/* - * we use a vmalloc'ed (sg-)buffer - */ - -/* get the physical page pointer on the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* - * hw_params callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already enough large */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_32_user(size); - if (! runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* - * hw_free callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - /* * clear the SRAM contents */ @@ -147,7 +103,8 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_32_buffer + (subs, params_buffer_bytes(hw_params)); } /* @@ -155,7 +112,7 @@ static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, */ static int pdacf_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_vmalloc_buffer(subs); } /* @@ -319,7 +276,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = { .prepare = pdacf_pcm_prepare, .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v1.2.2 From 6cedf8696d6a01bba391bdae06231243cfe2f48a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:30:24 +0100 Subject: sound: sgio2audio: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 31 ++++++------------------------- 1 file changed, 6 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index f1d9d16b5486..9b486beeb932 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include #include @@ -603,25 +602,14 @@ static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_pcm_runtime *runtime = substream->runtime; - int size = params_buffer_bytes(hw_params); - - /* alloc virtual 'dma' area */ - if (runtime->dma_area) - vfree(runtime->dma_area); - runtime->dma_area = vmalloc_user(size); - if (runtime->dma_area == NULL) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } /* hw_free callback */ static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) { - vfree(substream->runtime->dma_area); - substream->runtime->dma_area = NULL; - return 0; + return snd_pcm_lib_free_vmalloc_buffer(substream); } /* prepare callback */ @@ -692,13 +680,6 @@ snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) chip->channel[chan->idx].pos); } -/* get the physical page pointer on the given offset */ -static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, - unsigned long offset) -{ - return vmalloc_to_page(substream->runtime->dma_area + offset); -} - /* operators */ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .open = snd_sgio2audio_playback1_open, @@ -709,7 +690,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -721,7 +702,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -733,7 +714,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; /* -- cgit v1.2.2 From 149feef54bf543448dd4ec5820ef8ae178021c3a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:30:55 +0100 Subject: sound: vx: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_pcm.c | 59 ++++------------------------------------------- 1 file changed, 5 insertions(+), 54 deletions(-) (limited to 'sound') diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 6644d0034fba..c8385d26a16f 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -46,7 +46,6 @@ */ #include -#include #include #include #include @@ -55,55 +54,6 @@ #include "vx_cmd.h" -/* - * we use a vmalloc'ed (sg-)buffer - */ - -/* get the physical page pointer on the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* - * allocate a buffer via vmalloc_32(). - * called from hw_params - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - /* already allocated */ - if (runtime->dma_bytes >= size) - return 0; /* already enough large */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_32(size); - if (! runtime->dma_area) - return -ENOMEM; - memset(runtime->dma_area, 0, size); - runtime->dma_bytes = size; - return 1; /* changed */ -} - -/* - * free the buffer. - * called from hw_free callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - - /* * read three pending pcm bytes via inb() */ @@ -865,7 +815,8 @@ static snd_pcm_uframes_t vx_pcm_playback_pointer(struct snd_pcm_substream *subs) static int vx_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_32_buffer + (subs, params_buffer_bytes(hw_params)); } /* @@ -873,7 +824,7 @@ static int vx_pcm_hw_params(struct snd_pcm_substream *subs, */ static int vx_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_vmalloc_buffer(subs); } /* @@ -953,7 +904,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; @@ -1173,7 +1124,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v1.2.2 From c55675e348d9630c1ca69a190529bed1108c649d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:31:31 +0100 Subject: sound: usb-audio: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 46 +++++----------------------------------------- 1 file changed, 5 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index af8869a8a79e..31b63ea098b7 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -44,7 +44,6 @@ #include #include #include -#include #include #include #include @@ -735,41 +734,6 @@ static void snd_complete_sync_urb(struct urb *urb) } -/* get the physical page pointer at the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* allocate virtual buffer; may be called more than once */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already large enough */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_user(size); - if (!runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* free virtual buffer; may be called more than once */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - - /* * unlink active urbs. */ @@ -1449,8 +1413,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, unsigned int channels, rate, format; int ret, changed; - ret = snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); if (ret < 0) return ret; @@ -1507,7 +1471,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->period_bytes = 0; if (!subs->stream->chip->shutdown) release_substream_urbs(subs, 0); - return snd_pcm_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_vmalloc_buffer(substream); } /* @@ -1973,7 +1937,7 @@ static struct snd_pcm_ops snd_usb_playback_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_pcm_playback_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_usb_capture_ops = { @@ -1985,7 +1949,7 @@ static struct snd_pcm_ops snd_usb_capture_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_pcm_capture_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v1.2.2 From 5b4b2a41a1a80f5560364b7ef001486cd8fb5230 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:32:00 +0100 Subject: sound: ua101: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/ua101.c | 52 +++++++--------------------------------------------- 1 file changed, 7 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index ab9f8a2e1938..16dc7bd5e120 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include @@ -145,42 +144,6 @@ static struct usb_driver ua101_driver; static void abort_alsa_playback(struct ua101 *ua); static void abort_alsa_capture(struct ua101 *ua); -/* allocate virtual buffer; may be called more than once */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, - size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already large enough */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_user(size); - if (!runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* free virtual buffer; may be called more than once */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - -/* get the physical page pointer at the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - static const char *usb_error_string(int err) { switch (err) { @@ -791,8 +754,8 @@ static int capture_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int playback_pcm_hw_params(struct snd_pcm_substream *substream, @@ -809,14 +772,13 @@ static int playback_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int ua101_pcm_hw_free(struct snd_pcm_substream *substream) { - snd_pcm_free_vmalloc_buffer(substream); - return 0; + return snd_pcm_lib_free_vmalloc_buffer(substream); } static int capture_pcm_prepare(struct snd_pcm_substream *substream) @@ -948,7 +910,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .prepare = capture_pcm_prepare, .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops playback_pcm_ops = { @@ -960,7 +922,7 @@ static struct snd_pcm_ops playback_pcm_ops = { .prepare = playback_pcm_prepare, .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static const struct uac_format_type_i_discrete_descriptor * -- cgit v1.2.2 From 48c03ce72f2665f79a3fe54fc6d71b8cc3d30803 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 17 Dec 2009 14:51:35 +0100 Subject: ASoC: wm8974: fix a wrong bit definition The wm8974 datasheet defines BUFIOEN as bit 2. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8974.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 81c57b5c591c..a808675388fc 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -47,7 +47,7 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { }; #define WM8974_POWER1_BIASEN 0x08 -#define WM8974_POWER1_BUFIOEN 0x10 +#define WM8974_POWER1_BUFIOEN 0x04 struct wm8974_priv { struct snd_soc_codec codec; -- cgit v1.2.2 From b35a28af0a64a1e8e389bc009b76253256d8fe7b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Dec 2009 12:00:22 +0000 Subject: ASoC: Add initial WM8955 CODEC driver The WM8955 is a low power, high quality stereo DAC with integrated headphone and loudspeaker amplifiers, designed to reduce external component requirements in portable digital audio applications. This is an initial driver implementing support for the majority of the functionality in the device, currently OUT3 is not supported. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8955.c | 1151 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8955.h | 489 +++++++++++++++++++ 4 files changed, 1646 insertions(+) create mode 100644 sound/soc/codecs/wm8955.c create mode 100644 sound/soc/codecs/wm8955.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 691abe7df087..62ff26a08a2f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -52,6 +52,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C + select SND_SOC_WM8955 if I2C select SND_SOC_WM8960 if I2C select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C @@ -214,6 +215,9 @@ config SND_SOC_WM8904 config SND_SOC_WM8940 tristate +config SND_SOC_WM8955 + tristate + config SND_SOC_WM8960 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c0fd3c86edad..ea9835412e6a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o snd-soc-wm8940-objs := wm8940.o +snd-soc-wm8955-objs := wm8955.o snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o @@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o +obj-$(CONFIG_SND_SOC_WM8955) += snd-soc-wm8955.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c new file mode 100644 index 000000000000..615dab2b62ef --- /dev/null +++ b/sound/soc/codecs/wm8955.c @@ -0,0 +1,1151 @@ +/* + * wm8955.c -- WM8955 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8955.h" + +static struct snd_soc_codec *wm8955_codec; +struct snd_soc_codec_device soc_codec_dev_wm8955; + +#define WM8955_NUM_SUPPLIES 4 +static const char *wm8955_supply_names[WM8955_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD", + "HPVDD", + "AVDD", +}; + +/* codec private data */ +struct wm8955_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8955_MAX_REGISTER + 1]; + + unsigned int mclk_rate; + + int deemph; + int fs; + + struct regulator_bulk_data supplies[WM8955_NUM_SUPPLIES]; + + struct wm8955_pdata *pdata; +}; + +static const u16 wm8955_reg[WM8955_MAX_REGISTER + 1] = { + 0x0000, /* R0 */ + 0x0000, /* R1 */ + 0x0079, /* R2 - LOUT1 volume */ + 0x0079, /* R3 - ROUT1 volume */ + 0x0000, /* R4 */ + 0x0008, /* R5 - DAC Control */ + 0x0000, /* R6 */ + 0x000A, /* R7 - Audio Interface */ + 0x0000, /* R8 - Sample Rate */ + 0x0000, /* R9 */ + 0x00FF, /* R10 - Left DAC volume */ + 0x00FF, /* R11 - Right DAC volume */ + 0x000F, /* R12 - Bass control */ + 0x000F, /* R13 - Treble control */ + 0x0000, /* R14 */ + 0x0000, /* R15 - Reset */ + 0x0000, /* R16 */ + 0x0000, /* R17 */ + 0x0000, /* R18 */ + 0x0000, /* R19 */ + 0x0000, /* R20 */ + 0x0000, /* R21 */ + 0x0000, /* R22 */ + 0x00C1, /* R23 - Additional control (1) */ + 0x0000, /* R24 - Additional control (2) */ + 0x0000, /* R25 - Power Management (1) */ + 0x0000, /* R26 - Power Management (2) */ + 0x0000, /* R27 - Additional Control (3) */ + 0x0000, /* R28 */ + 0x0000, /* R29 */ + 0x0000, /* R30 */ + 0x0000, /* R31 */ + 0x0000, /* R32 */ + 0x0000, /* R33 */ + 0x0050, /* R34 - Left out Mix (1) */ + 0x0050, /* R35 - Left out Mix (2) */ + 0x0050, /* R36 - Right out Mix (1) */ + 0x0050, /* R37 - Right Out Mix (2) */ + 0x0050, /* R38 - Mono out Mix (1) */ + 0x0050, /* R39 - Mono out Mix (2) */ + 0x0079, /* R40 - LOUT2 volume */ + 0x0079, /* R41 - ROUT2 volume */ + 0x0079, /* R42 - MONOOUT volume */ + 0x0000, /* R43 - Clocking / PLL */ + 0x0103, /* R44 - PLL Control 1 */ + 0x0024, /* R45 - PLL Control 2 */ + 0x01BA, /* R46 - PLL Control 3 */ + 0x0000, /* R47 */ + 0x0000, /* R48 */ + 0x0000, /* R49 */ + 0x0000, /* R50 */ + 0x0000, /* R51 */ + 0x0000, /* R52 */ + 0x0000, /* R53 */ + 0x0000, /* R54 */ + 0x0000, /* R55 */ + 0x0000, /* R56 */ + 0x0000, /* R57 */ + 0x0000, /* R58 */ + 0x0000, /* R59 - PLL Control 4 */ +}; + +static int wm8955_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8955_RESET, 0); +} + +struct pll_factors { + int n; + int k; + int outdiv; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 22) * 10) + +static int wm8995_pll_factors(struct device *dev, + int Fref, int Fout, struct pll_factors *pll) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + + dev_dbg(dev, "Fref=%u Fout=%u\n", Fref, Fout); + + /* The oscilator should run at should be 90-100MHz, and + * there's a divide by 4 plus an optional divide by 2 in the + * output path to generate the system clock. The clock table + * is sortd so we should always generate a suitable target. */ + target = Fout * 4; + if (target < 90000000) { + pll->outdiv = 1; + target *= 2; + } else { + pll->outdiv = 0; + } + + WARN_ON(target < 90000000 || target > 100000000); + + dev_dbg(dev, "Fvco=%dHz\n", target); + + /* Now, calculate N.K */ + Ndiv = target / Fref; + + pll->n = Ndiv; + Nmod = target % Fref; + dev_dbg(dev, "Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + pll->k = K / 10; + + dev_dbg(dev, "N=%x K=%x OUTDIV=%x\n", pll->n, pll->k, pll->outdiv); + + return 0; +} + +/* Lookup table specifiying SRATE (table 25 in datasheet); some of the + * output frequencies have been rounded to the standard frequencies + * they are intended to match where the error is slight. */ +static struct { + int mclk; + int fs; + int usb; + int sr; +} clock_cfgs[] = { + { 18432000, 8000, 0, 3, }, + { 18432000, 12000, 0, 9, }, + { 18432000, 16000, 0, 11, }, + { 18432000, 24000, 0, 29, }, + { 18432000, 32000, 0, 13, }, + { 18432000, 48000, 0, 1, }, + { 18432000, 96000, 0, 15, }, + + { 16934400, 8018, 0, 19, }, + { 16934400, 11025, 0, 25, }, + { 16934400, 22050, 0, 27, }, + { 16934400, 44100, 0, 17, }, + { 16934400, 88200, 0, 31, }, + + { 12000000, 8000, 1, 2, }, + { 12000000, 11025, 1, 25, }, + { 12000000, 12000, 1, 8, }, + { 12000000, 16000, 1, 10, }, + { 12000000, 22050, 1, 27, }, + { 12000000, 24000, 1, 28, }, + { 12000000, 32000, 1, 12, }, + { 12000000, 44100, 1, 17, }, + { 12000000, 48000, 1, 0, }, + { 12000000, 88200, 1, 31, }, + { 12000000, 96000, 1, 14, }, + + { 12288000, 8000, 0, 2, }, + { 12288000, 12000, 0, 8, }, + { 12288000, 16000, 0, 10, }, + { 12288000, 24000, 0, 28, }, + { 12288000, 32000, 0, 12, }, + { 12288000, 48000, 0, 0, }, + { 12288000, 96000, 0, 14, }, + + { 12289600, 8018, 0, 18, }, + { 12289600, 11025, 0, 24, }, + { 12289600, 22050, 0, 26, }, + { 11289600, 44100, 0, 16, }, + { 11289600, 88200, 0, 31, }, +}; + +static int wm8955_configure_clocking(struct snd_soc_codec *codec) +{ + struct wm8955_priv *wm8955 = codec->private_data; + int i, ret, val; + int clocking = 0; + int srate = 0; + int sr = -1; + struct pll_factors pll; + + /* If we're not running a sample rate currently just pick one */ + if (wm8955->fs == 0) + wm8955->fs = 8000; + + /* Can we generate an exact output? */ + for (i = 0; i < ARRAY_SIZE(clock_cfgs); i++) { + if (wm8955->fs != clock_cfgs[i].fs) + continue; + sr = i; + + if (wm8955->mclk_rate == clock_cfgs[i].mclk) + break; + } + + /* We should never get here with an unsupported sample rate */ + if (sr == -1) { + dev_err(codec->dev, "Sample rate %dHz unsupported\n", + wm8955->fs); + WARN_ON(sr == -1); + return -EINVAL; + } + + if (i == ARRAY_SIZE(clock_cfgs)) { + /* If we can't generate the right clock from MCLK then + * we should configure the PLL to supply us with an + * appropriate clock. + */ + clocking |= WM8955_MCLKSEL; + + /* Use the last divider configuration we saw for the + * sample rate. */ + ret = wm8995_pll_factors(codec->dev, wm8955->mclk_rate, + clock_cfgs[sr].mclk, &pll); + if (ret != 0) { + dev_err(codec->dev, + "Unable to generate %dHz from %dHz MCLK\n", + wm8955->fs, wm8955->mclk_rate); + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_1, + WM8955_N_MASK | WM8955_K_21_18_MASK, + (pll.n << WM8955_N_SHIFT) | + pll.k >> 18); + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, + WM8955_K_17_9_MASK, + (pll.k >> 9) & WM8955_K_17_9_MASK); + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, + WM8955_K_8_0_MASK, + pll.k & WM8955_K_8_0_MASK); + if (pll.k) + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_4, + WM8955_KEN, WM8955_KEN); + else + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_4, + WM8955_KEN, 0); + + if (pll.outdiv) + val = WM8955_PLL_RB | WM8955_PLLOUTDIV2; + else + val = WM8955_PLL_RB; + + /* Now start the PLL running */ + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLL_RB | WM8955_PLLOUTDIV2, val); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLLEN, WM8955_PLLEN); + } + + srate = clock_cfgs[sr].usb | (clock_cfgs[sr].sr << WM8955_SR_SHIFT); + + snd_soc_update_bits(codec, WM8955_SAMPLE_RATE, + WM8955_USB | WM8955_SR_MASK, srate); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_MCLKSEL, clocking); + + return 0; +} + +static int wm8955_sysclk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + int ret = 0; + + /* Always disable the clocks - if we're doing reconfiguration this + * avoids misclocking. + */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_DIGENB, 0); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLL_RB | WM8955_PLLEN, 0); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + break; + case SND_SOC_DAPM_PRE_PMU: + ret = wm8955_configure_clocking(codec); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int wm8955_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8955_priv *wm8955 = codec->private_data; + int val, i, best; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8955->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - wm8955->fs) < + abs(deemph_settings[best] - wm8955->fs)) + best = i; + } + + val = best << WM8955_DEEMPH_SHIFT; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, WM8955_DAC_CONTROL, + WM8955_DEEMPH_MASK, val); +} + +static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8955_priv *wm8955 = codec->private_data; + + return wm8955->deemph; +} + +static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8955_priv *wm8955 = codec->private_data; + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + wm8955->deemph = deemph; + + return wm8955_set_deemph(codec); +} + +static const char *bass_mode_text[] = { + "Linear", "Adaptive", +}; + +static const struct soc_enum bass_mode = + SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 7, 2, bass_mode_text); + +static const char *bass_cutoff_text[] = { + "Low", "High" +}; + +static const struct soc_enum bass_cutoff = + SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 6, 2, bass_cutoff_text); + +static const char *treble_cutoff_text[] = { + "High", "Low" +}; + +static const struct soc_enum treble_cutoff = + SOC_ENUM_SINGLE(WM8955_TREBLE_CONTROL, 6, 2, treble_cutoff_text); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(atten_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -2100, 300, 0); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(treble_tlv, -1200, 150, 1); + +static const struct snd_kcontrol_new wm8955_snd_controls[] = { +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8955_LEFT_DAC_VOLUME, + WM8955_RIGHT_DAC_VOLUME, 0, 255, 0, digital_tlv), +SOC_SINGLE_TLV("Playback Attenuation Volume", WM8955_DAC_CONTROL, 7, 1, 1, + atten_tlv), +SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + wm8955_get_deemph, wm8955_put_deemph), + +SOC_ENUM("Bass Mode", bass_mode), +SOC_ENUM("Bass Cutoff", bass_cutoff), +SOC_SINGLE("Bass Volume", WM8955_BASS_CONTROL, 0, 15, 1), + +SOC_ENUM("Treble Cutoff", treble_cutoff), +SOC_SINGLE_TLV("Treble Volume", WM8955_TREBLE_CONTROL, 0, 14, 1, treble_tlv), + +SOC_SINGLE_TLV("Left Bypass Volume", WM8955_LEFT_OUT_MIX_1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Left Mono Volume", WM8955_LEFT_OUT_MIX_2, 4, 7, 1, + bypass_tlv), + +SOC_SINGLE_TLV("Right Mono Volume", WM8955_RIGHT_OUT_MIX_1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Bypass Volume", WM8955_RIGHT_OUT_MIX_2, 4, 7, 1, + bypass_tlv), + +/* Not a stereo pair so they line up with the DAPM switches */ +SOC_SINGLE_TLV("Mono Left Bypass Volume", WM8955_MONO_OUT_MIX_1, 4, 7, 1, + mono_tlv), +SOC_SINGLE_TLV("Mono Right Bypass Volume", WM8955_MONO_OUT_MIX_2, 4, 7, 1, + mono_tlv), + +SOC_DOUBLE_R_TLV("Headphone Volume", WM8955_LOUT1_VOLUME, + WM8955_ROUT1_VOLUME, 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Headphone ZC Switch", WM8955_LOUT1_VOLUME, + WM8955_ROUT1_VOLUME, 7, 1, 0), + +SOC_DOUBLE_R_TLV("Speaker Volume", WM8955_LOUT2_VOLUME, + WM8955_ROUT2_VOLUME, 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Speaker ZC Switch", WM8955_LOUT2_VOLUME, + WM8955_ROUT2_VOLUME, 7, 1, 0), + +SOC_SINGLE_TLV("Mono Volume", WM8955_MONOOUT_VOLUME, 0, 127, 0, out_tlv), +SOC_SINGLE("Mono ZC Switch", WM8955_MONOOUT_VOLUME, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lmixer[] = { +SOC_DAPM_SINGLE("Playback Switch", WM8955_LEFT_OUT_MIX_1, 8, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8955_LEFT_OUT_MIX_1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Playback Switch", WM8955_LEFT_OUT_MIX_2, 8, 1, 0), +SOC_DAPM_SINGLE("Mono Switch", WM8955_LEFT_OUT_MIX_2, 7, 1, 0), +}; + +static const struct snd_kcontrol_new rmixer[] = { +SOC_DAPM_SINGLE("Left Playback Switch", WM8955_RIGHT_OUT_MIX_1, 8, 1, 0), +SOC_DAPM_SINGLE("Mono Switch", WM8955_RIGHT_OUT_MIX_1, 7, 1, 0), +SOC_DAPM_SINGLE("Playback Switch", WM8955_RIGHT_OUT_MIX_2, 8, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8955_RIGHT_OUT_MIX_2, 7, 1, 0), +}; + +static const struct snd_kcontrol_new mmixer[] = { +SOC_DAPM_SINGLE("Left Playback Switch", WM8955_MONO_OUT_MIX_1, 8, 1, 0), +SOC_DAPM_SINGLE("Left Bypass Switch", WM8955_MONO_OUT_MIX_1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Playback Switch", WM8955_MONO_OUT_MIX_2, 8, 1, 0), +SOC_DAPM_SINGLE("Right Bypass Switch", WM8955_MONO_OUT_MIX_2, 7, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8955_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MONOIN-"), +SND_SOC_DAPM_INPUT("MONOIN+"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("LINEINL"), + +SND_SOC_DAPM_PGA("Mono Input", SND_SOC_NOPM, 0, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("SYSCLK", WM8955_POWER_MANAGEMENT_1, 0, 1, wm8955_sysclk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("TSDEN", WM8955_ADDITIONAL_CONTROL_1, 8, 0, NULL, 0), + +SND_SOC_DAPM_DAC("DACL", "Playback", WM8955_POWER_MANAGEMENT_2, 8, 0), +SND_SOC_DAPM_DAC("DACR", "Playback", WM8955_POWER_MANAGEMENT_2, 7, 0), + +SND_SOC_DAPM_PGA("LOUT1 PGA", WM8955_POWER_MANAGEMENT_2, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("ROUT1 PGA", WM8955_POWER_MANAGEMENT_2, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("LOUT2 PGA", WM8955_POWER_MANAGEMENT_2, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("ROUT2 PGA", WM8955_POWER_MANAGEMENT_2, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MOUT PGA", WM8955_POWER_MANAGEMENT_2, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("OUT3 PGA", WM8955_POWER_MANAGEMENT_2, 1, 0, NULL, 0), + +/* The names are chosen to make the control names nice */ +SND_SOC_DAPM_MIXER("Left", SND_SOC_NOPM, 0, 0, + lmixer, ARRAY_SIZE(lmixer)), +SND_SOC_DAPM_MIXER("Right", SND_SOC_NOPM, 0, 0, + rmixer, ARRAY_SIZE(rmixer)), +SND_SOC_DAPM_MIXER("Mono", SND_SOC_NOPM, 0, 0, + mmixer, ARRAY_SIZE(mmixer)), + +SND_SOC_DAPM_OUTPUT("LOUT1"), +SND_SOC_DAPM_OUTPUT("ROUT1"), +SND_SOC_DAPM_OUTPUT("LOUT2"), +SND_SOC_DAPM_OUTPUT("ROUT2"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("OUT3"), +}; + +static const struct snd_soc_dapm_route wm8955_intercon[] = { + { "DACL", NULL, "SYSCLK" }, + { "DACR", NULL, "SYSCLK" }, + + { "Mono Input", NULL, "MONOIN-" }, + { "Mono Input", NULL, "MONOIN+" }, + + { "Left", "Playback Switch", "DACL" }, + { "Left", "Right Playback Switch", "DACR" }, + { "Left", "Bypass Switch", "LINEINL" }, + { "Left", "Mono Switch", "Mono Input" }, + + { "Right", "Playback Switch", "DACR" }, + { "Right", "Left Playback Switch", "DACL" }, + { "Right", "Bypass Switch", "LINEINR" }, + { "Right", "Mono Switch", "Mono Input" }, + + { "Mono", "Left Playback Switch", "DACL" }, + { "Mono", "Right Playback Switch", "DACR" }, + { "Mono", "Left Bypass Switch", "LINEINL" }, + { "Mono", "Right Bypass Switch", "LINEINR" }, + + { "LOUT1 PGA", NULL, "Left" }, + { "LOUT1", NULL, "TSDEN" }, + { "LOUT1", NULL, "LOUT1 PGA" }, + + { "ROUT1 PGA", NULL, "Right" }, + { "ROUT1", NULL, "TSDEN" }, + { "ROUT1", NULL, "ROUT1 PGA" }, + + { "LOUT2 PGA", NULL, "Left" }, + { "LOUT2", NULL, "TSDEN" }, + { "LOUT2", NULL, "LOUT2 PGA" }, + + { "ROUT2 PGA", NULL, "Right" }, + { "ROUT2", NULL, "TSDEN" }, + { "ROUT2", NULL, "ROUT2 PGA" }, + + { "MOUT PGA", NULL, "Mono" }, + { "MONOOUT", NULL, "MOUT PGA" }, + + /* OUT3 not currently implemented */ + { "OUT3", NULL, "OUT3 PGA" }, +}; + +static int wm8955_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_add_controls(codec, wm8955_snd_controls, + ARRAY_SIZE(wm8955_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8955_dapm_widgets, + ARRAY_SIZE(wm8955_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, wm8955_intercon, + ARRAY_SIZE(wm8955_intercon)); + + return 0; +} + +static int wm8955_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8955_priv *wm8955 = codec->private_data; + int ret; + int wl; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wl = 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wl = 0x4; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wl = 0x8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wl = 0xc; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, WM8955_AUDIO_INTERFACE, + WM8955_WL_MASK, wl); + + wm8955->fs = params_rate(params); + wm8955_set_deemph(codec); + + /* If the chip is clocked then disable the clocks and force a + * reconfiguration, otherwise DAPM will power up the + * clocks for us later. */ + ret = snd_soc_read(codec, WM8955_POWER_MANAGEMENT_1); + if (ret < 0) + return ret; + if (ret & WM8955_DIGENB) { + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_DIGENB, 0); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLL_RB | WM8955_PLLEN, 0); + + wm8955_configure_clocking(codec); + } + + return 0; +} + + +static int wm8955_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8955_priv *priv = codec->private_data; + int div; + + switch (clk_id) { + case WM8955_CLK_MCLK: + if (freq > 15000000) { + priv->mclk_rate = freq /= 2; + div = WM8955_MCLKDIV2; + } else { + priv->mclk_rate = freq; + div = 0; + } + + snd_soc_update_bits(codec, WM8955_SAMPLE_RATE, + WM8955_MCLKDIV2, div); + break; + + default: + return -EINVAL; + } + + dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); + + return 0; +} + +static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u16 aif = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif |= WM8955_MS; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif |= WM8955_LRP; + case SND_SOC_DAIFMT_DSP_A: + aif |= 0x3; + break; + case SND_SOC_DAIFMT_I2S: + aif |= 0x2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif |= 0x1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif |= WM8955_BCLKINV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif |= WM8955_BCLKINV | WM8955_LRP; + break; + case SND_SOC_DAIFMT_IB_NF: + aif |= WM8955_BCLKINV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif |= WM8955_LRP; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8955_AUDIO_INTERFACE, + WM8955_MS | WM8955_FORMAT_MASK | WM8955_BCLKINV | + WM8955_LRP, aif); + + return 0; +} + + +static int wm8955_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val; + + if (mute) + val = WM8955_DACMU; + else + val = 0; + + snd_soc_update_bits(codec, WM8955_DAC_CONTROL, WM8955_DACMU, val); + + return 0; +} + +static int wm8955_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8955_priv *wm8955 = codec->private_data; + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID resistance 2*50k */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VMIDSEL_MASK, + 0x1 << WM8955_VMIDSEL_SHIFT); + + /* Default bias current */ + snd_soc_update_bits(codec, WM8955_ADDITIONAL_CONTROL_1, + WM8955_VSEL_MASK, + 0x2 << WM8955_VSEL_SHIFT); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + /* Sync back cached values if they're + * different from the hardware default. + */ + for (i = 0; i < ARRAY_SIZE(wm8955->reg_cache); i++) { + if (i == WM8955_RESET) + continue; + + if (wm8955->reg_cache[i] == wm8955_reg[i]) + continue; + + snd_soc_write(codec, i, wm8955->reg_cache[i]); + } + + /* Enable VREF and VMID */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VREF | + WM8955_VMIDSEL_MASK, + WM8955_VREF | + 0x3 << WM8955_VREF_SHIFT); + + /* Let VMID ramp */ + msleep(500); + + /* High resistance VROI to maintain outputs */ + snd_soc_update_bits(codec, + WM8955_ADDITIONAL_CONTROL_3, + WM8955_VROI, WM8955_VROI); + } + + /* Maintain VMID with 2*250k */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VMIDSEL_MASK, + 0x2 << WM8955_VMIDSEL_SHIFT); + + /* Minimum bias current */ + snd_soc_update_bits(codec, WM8955_ADDITIONAL_CONTROL_1, + WM8955_VSEL_MASK, 0); + break; + + case SND_SOC_BIAS_OFF: + /* Low resistance VROI to help discharge */ + snd_soc_update_bits(codec, + WM8955_ADDITIONAL_CONTROL_3, + WM8955_VROI, 0); + + /* Turn off VMID and VREF */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VREF | + WM8955_VMIDSEL_MASK, 0); + + regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8955_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8955_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8955_dai_ops = { + .set_sysclk = wm8955_set_sysclk, + .set_fmt = wm8955_set_fmt, + .hw_params = wm8955_hw_params, + .digital_mute = wm8955_digital_mute, +}; + +struct snd_soc_dai wm8955_dai = { + .name = "WM8955", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8955_RATES, + .formats = WM8955_FORMATS, + }, + .ops = &wm8955_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8955_dai); + +#ifdef CONFIG_PM +static int wm8955_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8955_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8955_suspend NULL +#define wm8955_resume NULL +#endif + +static int wm8955_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8955_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8955_codec; + codec = wm8955_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + wm8955_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +static int wm8955_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8955 = { + .probe = wm8955_probe, + .remove = wm8955_remove, + .suspend = wm8955_suspend, + .resume = wm8955_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8955); + +static int wm8955_register(struct wm8955_priv *wm8955, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8955->codec; + int i; + + if (wm8955_codec) { + dev_err(codec->dev, "Another WM8955 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8955; + codec->name = "WM8955"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8955_set_bias_level; + codec->dai = &wm8955_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8955_MAX_REGISTER; + codec->reg_cache = &wm8955->reg_cache; + + memcpy(codec->reg_cache, wm8955_reg, sizeof(wm8955_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8955->supplies); i++) + wm8955->supplies[i].supply = wm8955_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = wm8955_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err_enable; + } + + wm8955_dai.dev = codec->dev; + + /* Change some default settings - latch VU and enable ZC */ + wm8955->reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU; + wm8955->reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU; + wm8955->reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC; + wm8955->reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC; + wm8955->reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC; + wm8955->reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC; + wm8955->reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC; + + /* Also enable adaptive bass boost by default */ + wm8955->reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB; + + /* Set platform data values */ + if (wm8955->pdata) { + if (wm8955->pdata->out2_speaker) + wm8955->reg_cache[WM8955_ADDITIONAL_CONTROL_2] + |= WM8955_ROUT2INV; + + if (wm8955->pdata->monoin_diff) + wm8955->reg_cache[WM8955_MONO_OUT_MIX_1] + |= WM8955_DMEN; + } + + wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); + + wm8955_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8955_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); +err: + kfree(wm8955); + return ret; +} + +static void wm8955_unregister(struct wm8955_priv *wm8955) +{ + wm8955_set_bias_level(&wm8955->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); + snd_soc_unregister_dai(&wm8955_dai); + snd_soc_unregister_codec(&wm8955->codec); + kfree(wm8955); + wm8955_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8955_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8955_priv *wm8955; + struct snd_soc_codec *codec; + + wm8955 = kzalloc(sizeof(struct wm8955_priv), GFP_KERNEL); + if (wm8955 == NULL) + return -ENOMEM; + + codec = &wm8955->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8955); + codec->control_data = i2c; + wm8955->pdata = i2c->dev.platform_data; + + codec->dev = &i2c->dev; + + return wm8955_register(wm8955, SND_SOC_I2C); +} + +static __devexit int wm8955_i2c_remove(struct i2c_client *client) +{ + struct wm8955_priv *wm8955 = i2c_get_clientdata(client); + wm8955_unregister(wm8955); + return 0; +} + +static const struct i2c_device_id wm8955_i2c_id[] = { + { "wm8955", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); + +static struct i2c_driver wm8955_i2c_driver = { + .driver = { + .name = "wm8955", + .owner = THIS_MODULE, + }, + .probe = wm8955_i2c_probe, + .remove = __devexit_p(wm8955_i2c_remove), + .id_table = wm8955_i2c_id, +}; +#endif + +static int __init wm8955_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8955_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8955 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8955_modinit); + +static void __exit wm8955_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8955_i2c_driver); +#endif +} +module_exit(wm8955_exit); + +MODULE_DESCRIPTION("ASoC WM8955 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8955.h b/sound/soc/codecs/wm8955.h new file mode 100644 index 000000000000..ae349c8531f6 --- /dev/null +++ b/sound/soc/codecs/wm8955.h @@ -0,0 +1,489 @@ +/* + * wm8955.h -- WM8904 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics, plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8955_H +#define _WM8955_H + +#define WM8955_CLK_MCLK 1 + +extern struct snd_soc_dai wm8955_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8955; + +/* + * Register values. + */ +#define WM8955_LOUT1_VOLUME 0x02 +#define WM8955_ROUT1_VOLUME 0x03 +#define WM8955_DAC_CONTROL 0x05 +#define WM8955_AUDIO_INTERFACE 0x07 +#define WM8955_SAMPLE_RATE 0x08 +#define WM8955_LEFT_DAC_VOLUME 0x0A +#define WM8955_RIGHT_DAC_VOLUME 0x0B +#define WM8955_BASS_CONTROL 0x0C +#define WM8955_TREBLE_CONTROL 0x0D +#define WM8955_RESET 0x0F +#define WM8955_ADDITIONAL_CONTROL_1 0x17 +#define WM8955_ADDITIONAL_CONTROL_2 0x18 +#define WM8955_POWER_MANAGEMENT_1 0x19 +#define WM8955_POWER_MANAGEMENT_2 0x1A +#define WM8955_ADDITIONAL_CONTROL_3 0x1B +#define WM8955_LEFT_OUT_MIX_1 0x22 +#define WM8955_LEFT_OUT_MIX_2 0x23 +#define WM8955_RIGHT_OUT_MIX_1 0x24 +#define WM8955_RIGHT_OUT_MIX_2 0x25 +#define WM8955_MONO_OUT_MIX_1 0x26 +#define WM8955_MONO_OUT_MIX_2 0x27 +#define WM8955_LOUT2_VOLUME 0x28 +#define WM8955_ROUT2_VOLUME 0x29 +#define WM8955_MONOOUT_VOLUME 0x2A +#define WM8955_CLOCKING_PLL 0x2B +#define WM8955_PLL_CONTROL_1 0x2C +#define WM8955_PLL_CONTROL_2 0x2D +#define WM8955_PLL_CONTROL_3 0x2E +#define WM8955_PLL_CONTROL_4 0x3B + +#define WM8955_REGISTER_COUNT 29 +#define WM8955_MAX_REGISTER 0x3B + +/* + * Field Definitions. + */ + +/* + * R2 (0x02) - LOUT1 volume + */ +#define WM8955_LO1VU 0x0100 /* LO1VU */ +#define WM8955_LO1VU_MASK 0x0100 /* LO1VU */ +#define WM8955_LO1VU_SHIFT 8 /* LO1VU */ +#define WM8955_LO1VU_WIDTH 1 /* LO1VU */ +#define WM8955_LO1ZC 0x0080 /* LO1ZC */ +#define WM8955_LO1ZC_MASK 0x0080 /* LO1ZC */ +#define WM8955_LO1ZC_SHIFT 7 /* LO1ZC */ +#define WM8955_LO1ZC_WIDTH 1 /* LO1ZC */ +#define WM8955_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8955_LOUTVOL_SHIFT 0 /* LOUTVOL - [6:0] */ +#define WM8955_LOUTVOL_WIDTH 7 /* LOUTVOL - [6:0] */ + +/* + * R3 (0x03) - ROUT1 volume + */ +#define WM8955_RO1VU 0x0100 /* RO1VU */ +#define WM8955_RO1VU_MASK 0x0100 /* RO1VU */ +#define WM8955_RO1VU_SHIFT 8 /* RO1VU */ +#define WM8955_RO1VU_WIDTH 1 /* RO1VU */ +#define WM8955_RO1ZC 0x0080 /* RO1ZC */ +#define WM8955_RO1ZC_MASK 0x0080 /* RO1ZC */ +#define WM8955_RO1ZC_SHIFT 7 /* RO1ZC */ +#define WM8955_RO1ZC_WIDTH 1 /* RO1ZC */ +#define WM8955_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8955_ROUTVOL_SHIFT 0 /* ROUTVOL - [6:0] */ +#define WM8955_ROUTVOL_WIDTH 7 /* ROUTVOL - [6:0] */ + +/* + * R5 (0x05) - DAC Control + */ +#define WM8955_DAT 0x0080 /* DAT */ +#define WM8955_DAT_MASK 0x0080 /* DAT */ +#define WM8955_DAT_SHIFT 7 /* DAT */ +#define WM8955_DAT_WIDTH 1 /* DAT */ +#define WM8955_DACMU 0x0008 /* DACMU */ +#define WM8955_DACMU_MASK 0x0008 /* DACMU */ +#define WM8955_DACMU_SHIFT 3 /* DACMU */ +#define WM8955_DACMU_WIDTH 1 /* DACMU */ +#define WM8955_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM8955_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM8955_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ + +/* + * R7 (0x07) - Audio Interface + */ +#define WM8955_BCLKINV 0x0080 /* BCLKINV */ +#define WM8955_BCLKINV_MASK 0x0080 /* BCLKINV */ +#define WM8955_BCLKINV_SHIFT 7 /* BCLKINV */ +#define WM8955_BCLKINV_WIDTH 1 /* BCLKINV */ +#define WM8955_MS 0x0040 /* MS */ +#define WM8955_MS_MASK 0x0040 /* MS */ +#define WM8955_MS_SHIFT 6 /* MS */ +#define WM8955_MS_WIDTH 1 /* MS */ +#define WM8955_LRSWAP 0x0020 /* LRSWAP */ +#define WM8955_LRSWAP_MASK 0x0020 /* LRSWAP */ +#define WM8955_LRSWAP_SHIFT 5 /* LRSWAP */ +#define WM8955_LRSWAP_WIDTH 1 /* LRSWAP */ +#define WM8955_LRP 0x0010 /* LRP */ +#define WM8955_LRP_MASK 0x0010 /* LRP */ +#define WM8955_LRP_SHIFT 4 /* LRP */ +#define WM8955_LRP_WIDTH 1 /* LRP */ +#define WM8955_WL_MASK 0x000C /* WL - [3:2] */ +#define WM8955_WL_SHIFT 2 /* WL - [3:2] */ +#define WM8955_WL_WIDTH 2 /* WL - [3:2] */ +#define WM8955_FORMAT_MASK 0x0003 /* FORMAT - [1:0] */ +#define WM8955_FORMAT_SHIFT 0 /* FORMAT - [1:0] */ +#define WM8955_FORMAT_WIDTH 2 /* FORMAT - [1:0] */ + +/* + * R8 (0x08) - Sample Rate + */ +#define WM8955_BCLKDIV2 0x0080 /* BCLKDIV2 */ +#define WM8955_BCLKDIV2_MASK 0x0080 /* BCLKDIV2 */ +#define WM8955_BCLKDIV2_SHIFT 7 /* BCLKDIV2 */ +#define WM8955_BCLKDIV2_WIDTH 1 /* BCLKDIV2 */ +#define WM8955_MCLKDIV2 0x0040 /* MCLKDIV2 */ +#define WM8955_MCLKDIV2_MASK 0x0040 /* MCLKDIV2 */ +#define WM8955_MCLKDIV2_SHIFT 6 /* MCLKDIV2 */ +#define WM8955_MCLKDIV2_WIDTH 1 /* MCLKDIV2 */ +#define WM8955_SR_MASK 0x003E /* SR - [5:1] */ +#define WM8955_SR_SHIFT 1 /* SR - [5:1] */ +#define WM8955_SR_WIDTH 5 /* SR - [5:1] */ +#define WM8955_USB 0x0001 /* USB */ +#define WM8955_USB_MASK 0x0001 /* USB */ +#define WM8955_USB_SHIFT 0 /* USB */ +#define WM8955_USB_WIDTH 1 /* USB */ + +/* + * R10 (0x0A) - Left DAC volume + */ +#define WM8955_LDVU 0x0100 /* LDVU */ +#define WM8955_LDVU_MASK 0x0100 /* LDVU */ +#define WM8955_LDVU_SHIFT 8 /* LDVU */ +#define WM8955_LDVU_WIDTH 1 /* LDVU */ +#define WM8955_LDACVOL_MASK 0x00FF /* LDACVOL - [7:0] */ +#define WM8955_LDACVOL_SHIFT 0 /* LDACVOL - [7:0] */ +#define WM8955_LDACVOL_WIDTH 8 /* LDACVOL - [7:0] */ + +/* + * R11 (0x0B) - Right DAC volume + */ +#define WM8955_RDVU 0x0100 /* RDVU */ +#define WM8955_RDVU_MASK 0x0100 /* RDVU */ +#define WM8955_RDVU_SHIFT 8 /* RDVU */ +#define WM8955_RDVU_WIDTH 1 /* RDVU */ +#define WM8955_RDACVOL_MASK 0x00FF /* RDACVOL - [7:0] */ +#define WM8955_RDACVOL_SHIFT 0 /* RDACVOL - [7:0] */ +#define WM8955_RDACVOL_WIDTH 8 /* RDACVOL - [7:0] */ + +/* + * R12 (0x0C) - Bass control + */ +#define WM8955_BB 0x0080 /* BB */ +#define WM8955_BB_MASK 0x0080 /* BB */ +#define WM8955_BB_SHIFT 7 /* BB */ +#define WM8955_BB_WIDTH 1 /* BB */ +#define WM8955_BC 0x0040 /* BC */ +#define WM8955_BC_MASK 0x0040 /* BC */ +#define WM8955_BC_SHIFT 6 /* BC */ +#define WM8955_BC_WIDTH 1 /* BC */ +#define WM8955_BASS_MASK 0x000F /* BASS - [3:0] */ +#define WM8955_BASS_SHIFT 0 /* BASS - [3:0] */ +#define WM8955_BASS_WIDTH 4 /* BASS - [3:0] */ + +/* + * R13 (0x0D) - Treble control + */ +#define WM8955_TC 0x0040 /* TC */ +#define WM8955_TC_MASK 0x0040 /* TC */ +#define WM8955_TC_SHIFT 6 /* TC */ +#define WM8955_TC_WIDTH 1 /* TC */ +#define WM8955_TRBL_MASK 0x000F /* TRBL - [3:0] */ +#define WM8955_TRBL_SHIFT 0 /* TRBL - [3:0] */ +#define WM8955_TRBL_WIDTH 4 /* TRBL - [3:0] */ + +/* + * R15 (0x0F) - Reset + */ +#define WM8955_RESET_MASK 0x01FF /* RESET - [8:0] */ +#define WM8955_RESET_SHIFT 0 /* RESET - [8:0] */ +#define WM8955_RESET_WIDTH 9 /* RESET - [8:0] */ + +/* + * R23 (0x17) - Additional control (1) + */ +#define WM8955_TSDEN 0x0100 /* TSDEN */ +#define WM8955_TSDEN_MASK 0x0100 /* TSDEN */ +#define WM8955_TSDEN_SHIFT 8 /* TSDEN */ +#define WM8955_TSDEN_WIDTH 1 /* TSDEN */ +#define WM8955_VSEL_MASK 0x00C0 /* VSEL - [7:6] */ +#define WM8955_VSEL_SHIFT 6 /* VSEL - [7:6] */ +#define WM8955_VSEL_WIDTH 2 /* VSEL - [7:6] */ +#define WM8955_DMONOMIX_MASK 0x0030 /* DMONOMIX - [5:4] */ +#define WM8955_DMONOMIX_SHIFT 4 /* DMONOMIX - [5:4] */ +#define WM8955_DMONOMIX_WIDTH 2 /* DMONOMIX - [5:4] */ +#define WM8955_DACINV 0x0002 /* DACINV */ +#define WM8955_DACINV_MASK 0x0002 /* DACINV */ +#define WM8955_DACINV_SHIFT 1 /* DACINV */ +#define WM8955_DACINV_WIDTH 1 /* DACINV */ +#define WM8955_TOEN 0x0001 /* TOEN */ +#define WM8955_TOEN_MASK 0x0001 /* TOEN */ +#define WM8955_TOEN_SHIFT 0 /* TOEN */ +#define WM8955_TOEN_WIDTH 1 /* TOEN */ + +/* + * R24 (0x18) - Additional control (2) + */ +#define WM8955_OUT3SW_MASK 0x0180 /* OUT3SW - [8:7] */ +#define WM8955_OUT3SW_SHIFT 7 /* OUT3SW - [8:7] */ +#define WM8955_OUT3SW_WIDTH 2 /* OUT3SW - [8:7] */ +#define WM8955_ROUT2INV 0x0010 /* ROUT2INV */ +#define WM8955_ROUT2INV_MASK 0x0010 /* ROUT2INV */ +#define WM8955_ROUT2INV_SHIFT 4 /* ROUT2INV */ +#define WM8955_ROUT2INV_WIDTH 1 /* ROUT2INV */ +#define WM8955_DACOSR 0x0001 /* DACOSR */ +#define WM8955_DACOSR_MASK 0x0001 /* DACOSR */ +#define WM8955_DACOSR_SHIFT 0 /* DACOSR */ +#define WM8955_DACOSR_WIDTH 1 /* DACOSR */ + +/* + * R25 (0x19) - Power Management (1) + */ +#define WM8955_VMIDSEL_MASK 0x0180 /* VMIDSEL - [8:7] */ +#define WM8955_VMIDSEL_SHIFT 7 /* VMIDSEL - [8:7] */ +#define WM8955_VMIDSEL_WIDTH 2 /* VMIDSEL - [8:7] */ +#define WM8955_VREF 0x0040 /* VREF */ +#define WM8955_VREF_MASK 0x0040 /* VREF */ +#define WM8955_VREF_SHIFT 6 /* VREF */ +#define WM8955_VREF_WIDTH 1 /* VREF */ +#define WM8955_DIGENB 0x0001 /* DIGENB */ +#define WM8955_DIGENB_MASK 0x0001 /* DIGENB */ +#define WM8955_DIGENB_SHIFT 0 /* DIGENB */ +#define WM8955_DIGENB_WIDTH 1 /* DIGENB */ + +/* + * R26 (0x1A) - Power Management (2) + */ +#define WM8955_DACL 0x0100 /* DACL */ +#define WM8955_DACL_MASK 0x0100 /* DACL */ +#define WM8955_DACL_SHIFT 8 /* DACL */ +#define WM8955_DACL_WIDTH 1 /* DACL */ +#define WM8955_DACR 0x0080 /* DACR */ +#define WM8955_DACR_MASK 0x0080 /* DACR */ +#define WM8955_DACR_SHIFT 7 /* DACR */ +#define WM8955_DACR_WIDTH 1 /* DACR */ +#define WM8955_LOUT1 0x0040 /* LOUT1 */ +#define WM8955_LOUT1_MASK 0x0040 /* LOUT1 */ +#define WM8955_LOUT1_SHIFT 6 /* LOUT1 */ +#define WM8955_LOUT1_WIDTH 1 /* LOUT1 */ +#define WM8955_ROUT1 0x0020 /* ROUT1 */ +#define WM8955_ROUT1_MASK 0x0020 /* ROUT1 */ +#define WM8955_ROUT1_SHIFT 5 /* ROUT1 */ +#define WM8955_ROUT1_WIDTH 1 /* ROUT1 */ +#define WM8955_LOUT2 0x0010 /* LOUT2 */ +#define WM8955_LOUT2_MASK 0x0010 /* LOUT2 */ +#define WM8955_LOUT2_SHIFT 4 /* LOUT2 */ +#define WM8955_LOUT2_WIDTH 1 /* LOUT2 */ +#define WM8955_ROUT2 0x0008 /* ROUT2 */ +#define WM8955_ROUT2_MASK 0x0008 /* ROUT2 */ +#define WM8955_ROUT2_SHIFT 3 /* ROUT2 */ +#define WM8955_ROUT2_WIDTH 1 /* ROUT2 */ +#define WM8955_MONO 0x0004 /* MONO */ +#define WM8955_MONO_MASK 0x0004 /* MONO */ +#define WM8955_MONO_SHIFT 2 /* MONO */ +#define WM8955_MONO_WIDTH 1 /* MONO */ +#define WM8955_OUT3 0x0002 /* OUT3 */ +#define WM8955_OUT3_MASK 0x0002 /* OUT3 */ +#define WM8955_OUT3_SHIFT 1 /* OUT3 */ +#define WM8955_OUT3_WIDTH 1 /* OUT3 */ + +/* + * R27 (0x1B) - Additional Control (3) + */ +#define WM8955_VROI 0x0040 /* VROI */ +#define WM8955_VROI_MASK 0x0040 /* VROI */ +#define WM8955_VROI_SHIFT 6 /* VROI */ +#define WM8955_VROI_WIDTH 1 /* VROI */ + +/* + * R34 (0x22) - Left out Mix (1) + */ +#define WM8955_LD2LO 0x0100 /* LD2LO */ +#define WM8955_LD2LO_MASK 0x0100 /* LD2LO */ +#define WM8955_LD2LO_SHIFT 8 /* LD2LO */ +#define WM8955_LD2LO_WIDTH 1 /* LD2LO */ +#define WM8955_LI2LO 0x0080 /* LI2LO */ +#define WM8955_LI2LO_MASK 0x0080 /* LI2LO */ +#define WM8955_LI2LO_SHIFT 7 /* LI2LO */ +#define WM8955_LI2LO_WIDTH 1 /* LI2LO */ +#define WM8955_LI2LOVOL_MASK 0x0070 /* LI2LOVOL - [6:4] */ +#define WM8955_LI2LOVOL_SHIFT 4 /* LI2LOVOL - [6:4] */ +#define WM8955_LI2LOVOL_WIDTH 3 /* LI2LOVOL - [6:4] */ + +/* + * R35 (0x23) - Left out Mix (2) + */ +#define WM8955_RD2LO 0x0100 /* RD2LO */ +#define WM8955_RD2LO_MASK 0x0100 /* RD2LO */ +#define WM8955_RD2LO_SHIFT 8 /* RD2LO */ +#define WM8955_RD2LO_WIDTH 1 /* RD2LO */ +#define WM8955_RI2LO 0x0080 /* RI2LO */ +#define WM8955_RI2LO_MASK 0x0080 /* RI2LO */ +#define WM8955_RI2LO_SHIFT 7 /* RI2LO */ +#define WM8955_RI2LO_WIDTH 1 /* RI2LO */ +#define WM8955_RI2LOVOL_MASK 0x0070 /* RI2LOVOL - [6:4] */ +#define WM8955_RI2LOVOL_SHIFT 4 /* RI2LOVOL - [6:4] */ +#define WM8955_RI2LOVOL_WIDTH 3 /* RI2LOVOL - [6:4] */ + +/* + * R36 (0x24) - Right out Mix (1) + */ +#define WM8955_LD2RO 0x0100 /* LD2RO */ +#define WM8955_LD2RO_MASK 0x0100 /* LD2RO */ +#define WM8955_LD2RO_SHIFT 8 /* LD2RO */ +#define WM8955_LD2RO_WIDTH 1 /* LD2RO */ +#define WM8955_LI2RO 0x0080 /* LI2RO */ +#define WM8955_LI2RO_MASK 0x0080 /* LI2RO */ +#define WM8955_LI2RO_SHIFT 7 /* LI2RO */ +#define WM8955_LI2RO_WIDTH 1 /* LI2RO */ +#define WM8955_LI2ROVOL_MASK 0x0070 /* LI2ROVOL - [6:4] */ +#define WM8955_LI2ROVOL_SHIFT 4 /* LI2ROVOL - [6:4] */ +#define WM8955_LI2ROVOL_WIDTH 3 /* LI2ROVOL - [6:4] */ + +/* + * R37 (0x25) - Right Out Mix (2) + */ +#define WM8955_RD2RO 0x0100 /* RD2RO */ +#define WM8955_RD2RO_MASK 0x0100 /* RD2RO */ +#define WM8955_RD2RO_SHIFT 8 /* RD2RO */ +#define WM8955_RD2RO_WIDTH 1 /* RD2RO */ +#define WM8955_RI2RO 0x0080 /* RI2RO */ +#define WM8955_RI2RO_MASK 0x0080 /* RI2RO */ +#define WM8955_RI2RO_SHIFT 7 /* RI2RO */ +#define WM8955_RI2RO_WIDTH 1 /* RI2RO */ +#define WM8955_RI2ROVOL_MASK 0x0070 /* RI2ROVOL - [6:4] */ +#define WM8955_RI2ROVOL_SHIFT 4 /* RI2ROVOL - [6:4] */ +#define WM8955_RI2ROVOL_WIDTH 3 /* RI2ROVOL - [6:4] */ + +/* + * R38 (0x26) - Mono out Mix (1) + */ +#define WM8955_LD2MO 0x0100 /* LD2MO */ +#define WM8955_LD2MO_MASK 0x0100 /* LD2MO */ +#define WM8955_LD2MO_SHIFT 8 /* LD2MO */ +#define WM8955_LD2MO_WIDTH 1 /* LD2MO */ +#define WM8955_LI2MO 0x0080 /* LI2MO */ +#define WM8955_LI2MO_MASK 0x0080 /* LI2MO */ +#define WM8955_LI2MO_SHIFT 7 /* LI2MO */ +#define WM8955_LI2MO_WIDTH 1 /* LI2MO */ +#define WM8955_LI2MOVOL_MASK 0x0070 /* LI2MOVOL - [6:4] */ +#define WM8955_LI2MOVOL_SHIFT 4 /* LI2MOVOL - [6:4] */ +#define WM8955_LI2MOVOL_WIDTH 3 /* LI2MOVOL - [6:4] */ +#define WM8955_DMEN 0x0001 /* DMEN */ +#define WM8955_DMEN_MASK 0x0001 /* DMEN */ +#define WM8955_DMEN_SHIFT 0 /* DMEN */ +#define WM8955_DMEN_WIDTH 1 /* DMEN */ + +/* + * R39 (0x27) - Mono out Mix (2) + */ +#define WM8955_RD2MO 0x0100 /* RD2MO */ +#define WM8955_RD2MO_MASK 0x0100 /* RD2MO */ +#define WM8955_RD2MO_SHIFT 8 /* RD2MO */ +#define WM8955_RD2MO_WIDTH 1 /* RD2MO */ +#define WM8955_RI2MO 0x0080 /* RI2MO */ +#define WM8955_RI2MO_MASK 0x0080 /* RI2MO */ +#define WM8955_RI2MO_SHIFT 7 /* RI2MO */ +#define WM8955_RI2MO_WIDTH 1 /* RI2MO */ +#define WM8955_RI2MOVOL_MASK 0x0070 /* RI2MOVOL - [6:4] */ +#define WM8955_RI2MOVOL_SHIFT 4 /* RI2MOVOL - [6:4] */ +#define WM8955_RI2MOVOL_WIDTH 3 /* RI2MOVOL - [6:4] */ + +/* + * R40 (0x28) - LOUT2 volume + */ +#define WM8955_LO2VU 0x0100 /* LO2VU */ +#define WM8955_LO2VU_MASK 0x0100 /* LO2VU */ +#define WM8955_LO2VU_SHIFT 8 /* LO2VU */ +#define WM8955_LO2VU_WIDTH 1 /* LO2VU */ +#define WM8955_LO2ZC 0x0080 /* LO2ZC */ +#define WM8955_LO2ZC_MASK 0x0080 /* LO2ZC */ +#define WM8955_LO2ZC_SHIFT 7 /* LO2ZC */ +#define WM8955_LO2ZC_WIDTH 1 /* LO2ZC */ +#define WM8955_LOUT2VOL_MASK 0x007F /* LOUT2VOL - [6:0] */ +#define WM8955_LOUT2VOL_SHIFT 0 /* LOUT2VOL - [6:0] */ +#define WM8955_LOUT2VOL_WIDTH 7 /* LOUT2VOL - [6:0] */ + +/* + * R41 (0x29) - ROUT2 volume + */ +#define WM8955_RO2VU 0x0100 /* RO2VU */ +#define WM8955_RO2VU_MASK 0x0100 /* RO2VU */ +#define WM8955_RO2VU_SHIFT 8 /* RO2VU */ +#define WM8955_RO2VU_WIDTH 1 /* RO2VU */ +#define WM8955_RO2ZC 0x0080 /* RO2ZC */ +#define WM8955_RO2ZC_MASK 0x0080 /* RO2ZC */ +#define WM8955_RO2ZC_SHIFT 7 /* RO2ZC */ +#define WM8955_RO2ZC_WIDTH 1 /* RO2ZC */ +#define WM8955_ROUT2VOL_MASK 0x007F /* ROUT2VOL - [6:0] */ +#define WM8955_ROUT2VOL_SHIFT 0 /* ROUT2VOL - [6:0] */ +#define WM8955_ROUT2VOL_WIDTH 7 /* ROUT2VOL - [6:0] */ + +/* + * R42 (0x2A) - MONOOUT volume + */ +#define WM8955_MOZC 0x0080 /* MOZC */ +#define WM8955_MOZC_MASK 0x0080 /* MOZC */ +#define WM8955_MOZC_SHIFT 7 /* MOZC */ +#define WM8955_MOZC_WIDTH 1 /* MOZC */ +#define WM8955_MOUTVOL_MASK 0x007F /* MOUTVOL - [6:0] */ +#define WM8955_MOUTVOL_SHIFT 0 /* MOUTVOL - [6:0] */ +#define WM8955_MOUTVOL_WIDTH 7 /* MOUTVOL - [6:0] */ + +/* + * R43 (0x2B) - Clocking / PLL + */ +#define WM8955_MCLKSEL 0x0100 /* MCLKSEL */ +#define WM8955_MCLKSEL_MASK 0x0100 /* MCLKSEL */ +#define WM8955_MCLKSEL_SHIFT 8 /* MCLKSEL */ +#define WM8955_MCLKSEL_WIDTH 1 /* MCLKSEL */ +#define WM8955_PLLOUTDIV2 0x0020 /* PLLOUTDIV2 */ +#define WM8955_PLLOUTDIV2_MASK 0x0020 /* PLLOUTDIV2 */ +#define WM8955_PLLOUTDIV2_SHIFT 5 /* PLLOUTDIV2 */ +#define WM8955_PLLOUTDIV2_WIDTH 1 /* PLLOUTDIV2 */ +#define WM8955_PLL_RB 0x0010 /* PLL_RB */ +#define WM8955_PLL_RB_MASK 0x0010 /* PLL_RB */ +#define WM8955_PLL_RB_SHIFT 4 /* PLL_RB */ +#define WM8955_PLL_RB_WIDTH 1 /* PLL_RB */ +#define WM8955_PLLEN 0x0008 /* PLLEN */ +#define WM8955_PLLEN_MASK 0x0008 /* PLLEN */ +#define WM8955_PLLEN_SHIFT 3 /* PLLEN */ +#define WM8955_PLLEN_WIDTH 1 /* PLLEN */ + +/* + * R44 (0x2C) - PLL Control 1 + */ +#define WM8955_N_MASK 0x01E0 /* N - [8:5] */ +#define WM8955_N_SHIFT 5 /* N - [8:5] */ +#define WM8955_N_WIDTH 4 /* N - [8:5] */ +#define WM8955_K_21_18_MASK 0x000F /* K(21:18) - [3:0] */ +#define WM8955_K_21_18_SHIFT 0 /* K(21:18) - [3:0] */ +#define WM8955_K_21_18_WIDTH 4 /* K(21:18) - [3:0] */ + +/* + * R45 (0x2D) - PLL Control 2 + */ +#define WM8955_K_17_9_MASK 0x01FF /* K(17:9) - [8:0] */ +#define WM8955_K_17_9_SHIFT 0 /* K(17:9) - [8:0] */ +#define WM8955_K_17_9_WIDTH 9 /* K(17:9) - [8:0] */ + +/* + * R46 (0x2E) - PLL Control 3 + */ +#define WM8955_K_8_0_MASK 0x01FF /* K(8:0) - [8:0] */ +#define WM8955_K_8_0_SHIFT 0 /* K(8:0) - [8:0] */ +#define WM8955_K_8_0_WIDTH 9 /* K(8:0) - [8:0] */ + +/* + * R59 (0x3B) - PLL Control 4 + */ +#define WM8955_KEN 0x0080 /* KEN */ +#define WM8955_KEN_MASK 0x0080 /* KEN */ +#define WM8955_KEN_SHIFT 7 /* KEN */ +#define WM8955_KEN_WIDTH 1 /* KEN */ + +#endif -- cgit v1.2.2 From 56927eb054abd2c7371c769f359cc49a04ab488e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Dec 2009 13:11:12 +0000 Subject: ASoC: Set AIF word length for WM8904 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8904.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 8310e5d14b83..e44ee31c2184 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1503,6 +1503,23 @@ static int wm8904_hw_params(struct snd_pcm_substream *substream, wm8904->bclk = snd_soc_params_to_bclk(params); } + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif1 |= 0x40; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif1 |= 0x80; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif1 |= 0xc0; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8904->bclk); ret = wm8904_configure_clocking(codec); -- cgit v1.2.2 From 18240b67c8ca5efbbb2e8bb11942cc3db033fb16 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Dec 2009 14:20:35 +0000 Subject: ASoC: Host clock2 read up in WM8904 FLL configuration Avoids skipping over the read for disable cases. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index e44ee31c2184..992a7f23df5c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1893,6 +1893,8 @@ static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source, Fout == wm8904->fll_fout) return 0; + clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); + if (Fout == 0) { dev_dbg(codec->dev, "FLL disabled\n"); @@ -1936,7 +1938,6 @@ static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source, /* Save current state then disable the FLL and SYSCLK to avoid * misclocking */ - clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); fll1 = snd_soc_read(codec, WM8904_FLL_CONTROL_1); snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, WM8904_CLK_SYS_ENA, 0); -- cgit v1.2.2 From 0c2fd1bf4c6cb6095d8b3088d285167e66c12147 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Dec 2009 16:41:39 +0100 Subject: ALSA: hda - Check class to identify Nvidia controller chips Instead of listing all individual PCI IDs, check the matching with the PCI class together with the vendor id for Nvidia. This simplifies the pci id entries. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 30 ++++-------------------------- 1 file changed, 4 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9b56f937913e..93eaf4fc39be 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2694,32 +2694,10 @@ static struct pci_device_id azx_ids[] = { /* ULI M5461 */ { PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI }, /* NVIDIA MCP */ - { PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(PCI_VENDOR_ID_NVIDIA, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* Creative X-Fi (CA0110-IBG) */ -- cgit v1.2.2 From d49464318a7c51676c44cbd1e2480f651cc43807 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Dec 2009 20:25:30 +0100 Subject: ALSA: aaci - Fix a typo Fixed a typo of the max buffer size specified for buffer allocation changed in the commit d6797322231af98b9bb4afb175dd614cf511e5f7. Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1497dce1b04a..ae38f2c342cc 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1028,7 +1028,7 @@ static int __devinit aaci_init_pcm(struct aaci *aaci) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - NULL, 0, 64 * 104); + NULL, 0, 64 * 1024); } return ret; -- cgit v1.2.2 From 6ca867c827c84d21316e9dc4035abe9480f8347c Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:35 +0000 Subject: ALSA: AACI: simplify codec rate information There's no need for a specific rule; ALSA's generic AC'97 support calculates the necessary rate constraint information itself, and we can use this directly. Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 75 +++----------------------------------------------------- 1 file changed, 3 insertions(+), 72 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index ae38f2c342cc..ea3be874c84f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -330,63 +330,6 @@ static irqreturn_t aaci_irq(int irq, void *devid) /* * ALSA support. */ - -struct aaci_stream { - unsigned char codec_idx; - unsigned char rate_idx; -}; - -static struct aaci_stream aaci_streams[] = { - [ACSTREAM_FRONT] = { - .codec_idx = 0, - .rate_idx = AC97_RATES_FRONT_DAC, - }, - [ACSTREAM_SURROUND] = { - .codec_idx = 0, - .rate_idx = AC97_RATES_SURR_DAC, - }, - [ACSTREAM_LFE] = { - .codec_idx = 0, - .rate_idx = AC97_RATES_LFE_DAC, - }, -}; - -static inline unsigned int aaci_rate_mask(struct aaci *aaci, int streamid) -{ - struct aaci_stream *s = aaci_streams + streamid; - return aaci->ac97_bus->codec[s->codec_idx]->rates[s->rate_idx]; -} - -static unsigned int rate_list[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 -}; - -/* - * Double-rate rule: we can support double rate iff channels == 2 - * (unimplemented) - */ -static int -aaci_rule_rate_by_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule) -{ - struct aaci *aaci = rule->private; - unsigned int rate_mask = SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_5512; - struct snd_interval *c = hw_param_interval(p, SNDRV_PCM_HW_PARAM_CHANNELS); - - switch (c->max) { - case 6: - rate_mask &= aaci_rate_mask(aaci, ACSTREAM_LFE); - case 4: - rate_mask &= aaci_rate_mask(aaci, ACSTREAM_SURROUND); - case 2: - rate_mask &= aaci_rate_mask(aaci, ACSTREAM_FRONT); - } - - return snd_interval_list(hw_param_interval(p, rule->var), - ARRAY_SIZE(rate_list), rate_list, - rate_mask); -} - static struct snd_pcm_hardware aaci_hw_info = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -400,10 +343,7 @@ static struct snd_pcm_hardware aaci_hw_info = { */ .formats = SNDRV_PCM_FMTBIT_S16_LE, - /* should this be continuous or knot? */ - .rates = SNDRV_PCM_RATE_CONTINUOUS, - .rate_max = 48000, - .rate_min = 4000, + /* rates are setup from the AC'97 codec */ .channels_min = 2, .channels_max = 6, .buffer_bytes_max = 64 * 1024, @@ -423,6 +363,8 @@ static int __aaci_pcm_open(struct aaci *aaci, aacirun->substream = substream; runtime->private_data = aacirun; runtime->hw = aaci_hw_info; + runtime->hw.rates = aacirun->pcm->rates; + snd_pcm_limit_hw_rates(runtime); /* * FIXME: ALSA specifies fifo_size in bytes. If we're in normal @@ -433,17 +375,6 @@ static int __aaci_pcm_open(struct aaci *aaci, */ runtime->hw.fifo_size = aaci->fifosize * 2; - /* - * Add rule describing hardware rate dependency - * on the number of channels. - */ - ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - aaci_rule_rate_by_channels, aaci, - SNDRV_PCM_HW_PARAM_CHANNELS, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (ret) - goto out; - ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED, DRIVER_NAME, aaci); if (ret) -- cgit v1.2.2 From 4e30b69108b20eca80f1a323f969bf7629c7795f Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:37 +0000 Subject: ALSA: AACI: cleanup aaci_pcm_hw_params Since the recording and playback paths are now the same, eliminate the needless conditionals. Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index ea3be874c84f..2e28748a3d8d 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -438,18 +438,14 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (err < 0) - goto out; - - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[0].slots); - if (err) - goto out; + if (err >= 0) { + err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), + params_channels(params), + aacirun->pcm->r[0].slots); - aacirun->pcm_open = 1; + aacirun->pcm_open = err == 0; + } - out: return err; } @@ -458,7 +454,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct aaci_runtime *aacirun = runtime->private_data; - aacirun->start = (void *)runtime->dma_area; + aacirun->start = runtime->dma_area; aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; aacirun->period = -- cgit v1.2.2 From d3aee7996c30f928bbbbfd0994148e35d2e83084 Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:40 +0000 Subject: ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 2e28748a3d8d..b88bbded2f4f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -444,6 +444,11 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aacirun->pcm->r[0].slots); aacirun->pcm_open = err == 0; + aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; + aacirun->fifosz = aaci->fifosize * 4; + + if (aacirun->cr & CR_COMPACT) + aacirun->fifosz >>= 1; } return err; @@ -554,14 +559,9 @@ static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, * Enable FIFO, compact mode, 16 bits per sample. * FIXME: double rate slots? */ - if (ret >= 0) { - aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; + if (ret >= 0) aacirun->cr |= channels_to_txmask[channels]; - aacirun->fifosz = aaci->fifosize * 4; - if (aacirun->cr & CR_COMPACT) - aacirun->fifosz >>= 1; - } return ret; } @@ -648,18 +648,10 @@ static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, int ret; ret = aaci_pcm_hw_params(substream, aacirun, params); - - if (ret >= 0) { - aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; - + if (ret >= 0) /* Line in record: slot 3 and 4 */ aacirun->cr |= CR_SL3 | CR_SL4; - aacirun->fifosz = aaci->fifosize * 4; - - if (aacirun->cr & CR_COMPACT) - aacirun->fifosz >>= 1; - } return ret; } -- cgit v1.2.2 From a08d56583f6b87e2981d1b6e9ee891bdc741cc44 Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:45 +0000 Subject: ALSA: AACI: add double-rate support Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index b88bbded2f4f..b377370af2d7 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -366,6 +366,10 @@ static int __aaci_pcm_open(struct aaci *aaci, runtime->hw.rates = aacirun->pcm->rates; snd_pcm_limit_hw_rates(runtime); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + aacirun->pcm->r[1].slots) + snd_ac97_pcm_double_rate_rules(runtime); + /* * FIXME: ALSA specifies fifo_size in bytes. If we're in normal * mode, each 32-bit word contains one sample. If we're in @@ -439,9 +443,12 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (err >= 0) { - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), + unsigned int rate = params_rate(params); + int dbl = rate > 48000; + + err = snd_ac97_pcm_open(aacirun->pcm, rate, params_channels(params), - aacirun->pcm->r[0].slots); + aacirun->pcm->r[dbl].slots); aacirun->pcm_open = err == 0; aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; @@ -808,6 +815,12 @@ static struct ac97_pcm ac97_defs[] __devinitdata = { (1 << AC97_SLOT_PCM_SRIGHT) | (1 << AC97_SLOT_LFE), }, + [1] = { + .slots = (1 << AC97_SLOT_PCM_LEFT) | + (1 << AC97_SLOT_PCM_RIGHT) | + (1 << AC97_SLOT_PCM_LEFT_0) | + (1 << AC97_SLOT_PCM_RIGHT_0), + }, }, }, [1] = { /* PCM in */ -- cgit v1.2.2 From d6a89fefa50feda5516cd5210ad0008a44632b52 Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:50 +0000 Subject: ALSA: AACI: switch to per-pcm locking We can use finer-grained locking, which makes things easier when we gain DMA support. Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 49 +++++++++++++++++++++++++++++-------------------- sound/arm/aaci.h | 2 +- 2 files changed, 30 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index b377370af2d7..c5699863643b 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -172,14 +172,15 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) return v; } -static inline void aaci_chan_wait_ready(struct aaci_runtime *aacirun) +static inline void +aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask) { u32 val; int timeout = 5000; do { val = readl(aacirun->base + AACI_SR); - } while (val & (SR_TXB|SR_RXB) && timeout--); + } while (val & mask && timeout--); } @@ -208,8 +209,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) writel(0, aacirun->base + AACI_IE); return; } - ptr = aacirun->ptr; + spin_lock(&aacirun->lock); + + ptr = aacirun->ptr; do { unsigned int len = aacirun->fifosz; u32 val; @@ -217,9 +220,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (aacirun->bytes <= 0) { aacirun->bytes += aacirun->period; aacirun->ptr = ptr; - spin_unlock(&aaci->lock); + spin_unlock(&aacirun->lock); snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aaci->lock); + spin_lock(&aacirun->lock); } if (!(aacirun->cr & CR_EN)) break; @@ -245,7 +248,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) ptr = aacirun->start; } } while(1); + aacirun->ptr = ptr; + + spin_unlock(&aacirun->lock); } if (mask & ISR_URINTR) { @@ -263,6 +269,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) return; } + spin_lock(&aacirun->lock); + ptr = aacirun->ptr; do { unsigned int len = aacirun->fifosz; @@ -271,9 +279,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (aacirun->bytes <= 0) { aacirun->bytes += aacirun->period; aacirun->ptr = ptr; - spin_unlock(&aaci->lock); + spin_unlock(&aacirun->lock); snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aaci->lock); + spin_lock(&aacirun->lock); } if (!(aacirun->cr & CR_EN)) break; @@ -301,6 +309,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) } while (1); aacirun->ptr = ptr; + + spin_unlock(&aacirun->lock); } } @@ -310,7 +320,6 @@ static irqreturn_t aaci_irq(int irq, void *devid) u32 mask; int i; - spin_lock(&aaci->lock); mask = readl(aaci->base + AACI_ALLINTS); if (mask) { u32 m = mask; @@ -320,7 +329,6 @@ static irqreturn_t aaci_irq(int irq, void *devid) } } } - spin_unlock(&aaci->lock); return mask ? IRQ_HANDLED : IRQ_NONE; } @@ -580,7 +588,7 @@ static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun) ie &= ~(IE_URIE|IE_TXIE); writel(ie, aacirun->base + AACI_IE); aacirun->cr &= ~CR_EN; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_TXB); writel(aacirun->cr, aacirun->base + AACI_TXCR); } @@ -588,7 +596,7 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun) { u32 ie; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_TXB); aacirun->cr |= CR_EN; ie = readl(aacirun->base + AACI_IE); @@ -599,12 +607,12 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun) static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; unsigned long flags; int ret = 0; - spin_lock_irqsave(&aaci->lock, flags); + spin_lock_irqsave(&aacirun->lock, flags); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: aaci_pcm_playback_start(aacirun); @@ -631,7 +639,8 @@ static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cm default: ret = -EINVAL; } - spin_unlock_irqrestore(&aaci->lock, flags); + + spin_unlock_irqrestore(&aacirun->lock, flags); return ret; } @@ -666,7 +675,7 @@ static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun) { u32 ie; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_RXB); ie = readl(aacirun->base + AACI_IE); ie &= ~(IE_ORIE | IE_RXIE); @@ -681,7 +690,7 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun) { u32 ie; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_RXB); #ifdef DEBUG /* RX Timeout value: bits 28:17 in RXCR */ @@ -698,12 +707,11 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun) static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; unsigned long flags; int ret = 0; - spin_lock_irqsave(&aaci->lock, flags); + spin_lock_irqsave(&aacirun->lock, flags); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -732,7 +740,7 @@ static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd ret = -EINVAL; } - spin_unlock_irqrestore(&aaci->lock, flags); + spin_unlock_irqrestore(&aacirun->lock, flags); return ret; } @@ -933,7 +941,6 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) aaci = card->private_data; mutex_init(&aaci->ac97_sem); - spin_lock_init(&aaci->lock); aaci->card = card; aaci->dev = dev; @@ -1020,12 +1027,14 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) /* * Playback uses AACI channel 0 */ + spin_lock_init(&aaci->playback.lock); aaci->playback.base = aaci->base + AACI_CSCH1; aaci->playback.fifo = aaci->base + AACI_DR1; /* * Capture uses AACI channel 0 */ + spin_lock_init(&aaci->capture.lock); aaci->capture.base = aaci->base + AACI_CSCH1; aaci->capture.fifo = aaci->base + AACI_DR1; diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h index 924f69c1c44c..6a4a2eebdda1 100644 --- a/sound/arm/aaci.h +++ b/sound/arm/aaci.h @@ -202,6 +202,7 @@ struct aaci_runtime { void __iomem *base; void __iomem *fifo; + spinlock_t lock; struct ac97_pcm *pcm; int pcm_open; @@ -232,7 +233,6 @@ struct aaci { struct snd_ac97 *ac97; u32 maincr; - spinlock_t lock; struct aaci_runtime playback; struct aaci_runtime capture; -- cgit v1.2.2 From ef86f581f7e8b29cb58d7f4e892e1a91b3805124 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 19 Dec 2009 08:18:03 +0100 Subject: ALSA: Use kzalloc for allocating only one thing Use kzalloc rather than kcalloc(1,...) The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // @@ @@ - kcalloc(1, + kzalloc( ...) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_midi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index cb9aa4c4edd0..4be562b2cf21 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *card, int device) err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); if (err < 0) return err; - mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + mpu = kzalloc(sizeof(*mpu), GFP_KERNEL); if (mpu == NULL) { snd_device_free(card, rmidi); return -ENOMEM; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aeed4cc5aa79..20c1828e4bac 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12857,7 +12857,7 @@ static int patch_alc268(struct hda_codec *codec) int board_config; int i, has_beep, err; - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; -- cgit v1.2.2 From 440b004cf953bec2bc8cd91c64ae707fd7e25327 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 12:04:08 +0100 Subject: ALSA: hda/realtek: Remove extra .capsrc_nids initialization for ALC889_INTEL Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b375771b3ab..2d3f4f893ef3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9238,8 +9238,6 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), .adc_nids = alc889_adc_nids, - .capsrc_nids = alc889_capsrc_nids, - .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, .slave_dig_outs = alc883_slave_dig_outs, -- cgit v1.2.2 From e2595322a3a353a59cecd7f57e7aa421ecb02d12 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sat, 19 Dec 2009 18:19:02 -0500 Subject: ALSA: hda: Set Front Mic to input vref 50% for Lenovo 3000 Y410 BugLink: https://bugs.launchpad.net/bugs/479373 The OR has verified with hda-verb that the internal microphone needs VREF50 set for audible capture. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 84bc2c7c4421..1554c3a6fd2e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10686,6 +10686,13 @@ static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = { {} }; +static struct hda_verb alc262_lenovo_3000_init_verbs[] = { + /* Front Mic pin: input vref at 50% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {} +}; + static struct hda_input_mux alc262_fujitsu_capture_source = { .num_items = 3, .items = { @@ -11728,7 +11735,8 @@ static struct alc_config_preset alc262_presets[] = { [ALC262_LENOVO_3000] = { .mixers = { alc262_lenovo_3000_mixer }, .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, - alc262_lenovo_3000_unsol_verbs }, + alc262_lenovo_3000_unsol_verbs, + alc262_lenovo_3000_init_verbs }, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, .hp_nid = 0x03, -- cgit v1.2.2 From 0f86a228f4a4639b3142ce0dad208433b2db377a Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Sun, 20 Dec 2009 22:51:18 +0100 Subject: ALSA: HDA: simplify Aspire 8930G verb array This patch just simplifies the 8930G verb array a bit. Just use the common ALC889 EAPD verb array to make things more consistent. The file is already huge enough already. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1554c3a6fd2e..cb97323acc17 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1665,9 +1665,6 @@ static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { /* some bit here disables the other DACs. Init=0x4900 */ {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, -/* Enable amplifiers */ - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, /* DMIC fix * This laptop has a stereo digital microphone. The mics are only 1cm apart * which makes the stereo useless. However, either the mic or the ALC889 @@ -9386,7 +9383,8 @@ static struct alc_config_preset alc882_presets[] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc889_acer_aspire_8930g_verbs }, + alc889_acer_aspire_8930g_verbs, + alc889_eapd_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), -- cgit v1.2.2 From 556eea9a926bff8f014b4f80522b4de97ae84213 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Sun, 20 Dec 2009 22:51:23 +0100 Subject: ALSA: HDA: remove useless mixers on Aspire 8930G This patch removes some extra mixers that do nothing on the Acer Aspire 8930G. The CD mixer is useless because the SATA DVD/Blu-Ray drive has no analog audio output, and the Side mixer is useless because we max out at 6ch anyway. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cb97323acc17..faeb74f28207 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1777,6 +1777,25 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + + static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9380,7 +9399,7 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_8930G] = { - .mixers = { alc888_base_mixer, + .mixers = { alc889_acer_aspire_8930g_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, alc889_acer_aspire_8930g_verbs, -- cgit v1.2.2 From f5de24b06aa46427500d0fdbe8616b73a71d8c28 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Sun, 20 Dec 2009 22:51:31 +0100 Subject: ALSA: HDA: add powersaving hook for Realtek The current Realtek code makes no specific provision for turning stuff off. The codec chip is placed into low-power mode generically, but this doesn't turn off any external hardware connected to it, in particular external amplifiers. This patch creates a hook function that is called by the codec suspend/resume functions. It ought to disable any external hardware in a device-specific way. I've implemented a generic ALC889 function that sets the EAPD pin properly, and used it for the Acer Aspire 8930G which can benefit from this feature. On my laptop, this results in ~0.5W extra savings. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index faeb74f28207..b3abe9ca826d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -337,6 +337,9 @@ struct alc_spec { /* hooks */ void (*init_hook)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*power_hook)(struct hda_codec *codec, int power); +#endif /* for pin sensing */ unsigned int sense_updated: 1; @@ -388,6 +391,7 @@ struct alc_config_preset { void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_amp_list *loopbacks; + void (*power_hook)(struct hda_codec *codec, int power); #endif }; @@ -900,6 +904,7 @@ static void setup_preset(struct hda_codec *codec, spec->unsol_event = preset->unsol_event; spec->init_hook = preset->init_hook; #ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = preset->power_hook; spec->loopback.amplist = preset->loopbacks; #endif @@ -1826,6 +1831,16 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x1b; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static void alc889_power_eapd(struct hda_codec *codec, int power) +{ + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); +} +#endif + /* * ALC880 3-stack model * @@ -3619,12 +3634,29 @@ static void alc_free(struct hda_codec *codec) snd_hda_detach_beep_device(codec); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int alc_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct alc_spec *spec = codec->spec; + if (spec && spec->power_hook) + spec->power_hook(codec, 0); + return 0; +} +#endif + #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct alc_spec *spec = codec->spec; +#endif codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (spec && spec->power_hook) + spec->power_hook(codec, 1); +#endif return 0; } #endif @@ -3641,6 +3673,7 @@ static struct hda_codec_ops alc_patch_ops = { .resume = alc_resume, #endif #ifdef CONFIG_SND_HDA_POWER_SAVE + .suspend = alc_suspend, .check_power_status = alc_check_power_status, #endif }; @@ -9420,6 +9453,9 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc_automute_amp_unsol_event, .setup = alc889_acer_aspire_8930g_setup, .init_hook = alc_automute_amp, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .power_hook = alc889_power_eapd, +#endif }, [ALC888_ACER_ASPIRE_7730G] = { .mixers = { alc883_3ST_6ch_mixer, -- cgit v1.2.2 From 40962d7c741de1c21b6ce8516c1d9f8836fb383e Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 19 Dec 2009 18:31:04 +0100 Subject: ALSA: fix incorrect rounding direction in snd_interval_ratnum() The direction of rounding is incorrect in the snd_interval_ratnum() It was detected with following parameters (sb8 driver playing 8kHz stereo file): - num is always 1000000 - requested frequency rate is from 7999 to 7999 (single frequency) The first loop calculates div_down(num, freq->min) which is 125. Thus, a frequency range's minimum value is 1000000 / 125 = 8000 Hz. The second loop calculates div_up(num, freq->max) which is 126 The frequency range's maximum value is 1000000 / 126 = 7936 Hz. The range maximum is lower than the range minimum so the function fails due to empty result range. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f410832a25..a27545b23ee9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -758,7 +758,7 @@ int snd_interval_ratnum(struct snd_interval *i, int diff; if (q == 0) q = 1; - den = div_down(num, q); + den = div_up(num, q); if (den < rats[k].den_min) continue; if (den > rats[k].den_max) @@ -794,7 +794,7 @@ int snd_interval_ratnum(struct snd_interval *i, i->empty = 1; return -EINVAL; } - den = div_up(num, q); + den = div_down(num, q); if (den > rats[k].den_max) continue; if (den < rats[k].den_min) -- cgit v1.2.2 From db8cf334f66bdf1ba2b3d2f7128095fc9b7a6e2b Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 20 Dec 2009 20:15:19 +0100 Subject: ALSA: sbawe: fix memory detection Memory amount is increased before a successful write-read sequence is done. Thus, 512 kB of onboard memory is detected on memoryless cards like SB32. Move the increasing of memory counter after successful read is done. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 96678d5d3834..751762f1c59a 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -393,8 +393,6 @@ size_dram(struct snd_emu8000 *emu) while (size < EMU8000_MAX_DRAM) { - size += 512 * 1024; /* increment 512kbytes */ - /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -414,7 +412,9 @@ size_dram(struct snd_emu8000 *emu) /*snd_emu8000_read_wait(emu);*/ EMU8000_SMLD_READ(emu); /* discard stale data */ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) - break; /* we must have wrapped around */ + break; /* no memory at this address */ + + size += 512 * 1024; /* increment 512kbytes */ snd_emu8000_read_wait(emu); -- cgit v1.2.2 From ad8decb7f5dfd556e4a8400e37b127cd20d8e4c5 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 20 Dec 2009 19:01:50 +0100 Subject: ALSA: jazz16: Add support for Media Vision Jazz16 chipset This is one of Sound Blaster Pro compatible chipsets which is supported by Linux OSS driver and was missing native supoort for ALSA. The Jazz16 audio codec is Crystal CS4216 which is capable of playback and recording up to 48 kHz stereo. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 16 ++ sound/isa/sb/Makefile | 2 + sound/isa/sb/jazz16.c | 385 +++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/sb/sb8_main.c | 117 ++++++++++++-- sound/isa/sb/sb_common.c | 3 + sound/isa/sb/sb_mixer.c | 3 + 6 files changed, 510 insertions(+), 16 deletions(-) create mode 100644 sound/isa/sb/jazz16.c (limited to 'sound') diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 194af3b01e13..755a0a5f0e3f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -239,6 +239,22 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16 card and compatibles" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB8_DSP + help + Say Y here to include support for soundcards based on the + Media Vision Jazz16 chipset: digital chip MVD1216 (Jazz16), + codec MVA416 (CS4216) and mixer MVA514 (ICS2514). + Media Vision's Jazz16 cards were sold under names Pro Sonic 16, + Premium 3-D and Pro 3-D. There were also OEMs cards with the + Jazz16 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" select SND_OPL3_LIB diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index faeffceb01b7..af3669681788 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SB8) += snd-sb8.o obj-$(CONFIG_SND_SB16) += snd-sb16.o obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o obj-$(CONFIG_SND_ES968) += snd-es968.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 000000000000..d52966b75846 --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,385 @@ + +/* + * jazz16.c - driver for Media Vision Jazz16 based soundcards. + * Copyright (C) 2009 Krzysztof Helt + * Based on patches posted by Rask Ingemann Lambertsen and Rene Herman. + * Based on OSS Sound Blaster driver. + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive for + * more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include + +#define PFX "jazz16: " + +MODULE_DESCRIPTION("Media Vision Jazz16"); +MODULE_SUPPORTED_DEVICE("{{Media Vision ??? }," + "{RTL,RTL3000}}"); + +MODULE_AUTHOR("Krzysztof Helt "); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ +static unsigned long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static unsigned long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Media Vision Jazz16 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Media Vision Jazz16 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Media Vision Jazz16 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for jazz16 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for jazz16 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for jazz16 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for jazz16 driver."); +module_param_array(dma8, int, NULL, 0444); +MODULE_PARM_DESC(dma8, "DMA8 # for jazz16 driver."); +module_param_array(dma16, int, NULL, 0444); +MODULE_PARM_DESC(dma16, "DMA16 # for jazz16 driver."); + +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_DSP_GET_JAZZ_BRD_REV 0xfa +#define SB_JAZZ16_SET_DMAINTR 0xfb +#define SB_DSP_GET_JAZZ_MODEL 0xfe + +struct snd_card_jazz16 { + struct snd_sb *chip; +}; + +static irqreturn_t jazz16_interrupt(int irq, void *chip) +{ + return snd_sb8dsp_interrupt(chip); +} + +static int __devinit jazz16_configure_ports(unsigned long port, + unsigned long mpu_port, int idx) +{ + unsigned char val; + + if (!request_region(0x201, 1, "jazz16 config")) { + snd_printk(KERN_ERR "config port region is already in use.\n"); + return -EBUSY; + } + outb(SB_JAZZ16_WAKEUP - idx, 0x201); + udelay(100); + outb(SB_JAZZ16_SET_PORTS + idx, 0x201); + udelay(100); + val = port & 0x70; + val |= (mpu_port & 0x30) >> 4; + outb(val, 0x201); + + release_region(0x201, 1); + return 0; +} + +static int __devinit jazz16_detect_board(unsigned long port, + unsigned long mpu_port) +{ + int err; + int val; + struct snd_sb chip; + + if (!request_region(port, 0x10, "jazz16")) { + snd_printk(KERN_ERR "I/O port region is already in use.\n"); + return -EBUSY; + } + /* just to call snd_sbdsp_command/reset/get_byte() */ + chip.port = port; + + err = snd_sbdsp_reset(&chip); + if (err < 0) + for (val = 0; val < 4; val++) { + err = jazz16_configure_ports(port, mpu_port, val); + if (err < 0) + break; + + err = snd_sbdsp_reset(&chip); + if (!err) + break; + } + if (err < 0) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_BRD_REV)) { + err = -EBUSY; + goto err_unmap; + } + val = snd_sbdsp_get_byte(&chip); + if (val >= 0x30) + snd_sbdsp_get_byte(&chip); + + if ((val & 0xf0) != 0x10) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_MODEL)) { + err = -EBUSY; + goto err_unmap; + } + snd_sbdsp_get_byte(&chip); + err = snd_sbdsp_get_byte(&chip); + snd_printd("Media Vision Jazz16 board detected: rev 0x%x, model 0x%x\n", + val, err); + + err = 0; + +err_unmap: + release_region(port, 0x10); + return err; +} + +static int __devinit jazz16_configure_board(struct snd_sb *chip, int mpu_irq) +{ + static unsigned char jazz_irq_bits[] = { 0, 0, 2, 3, 0, 1, 0, 4, + 0, 2, 5, 0, 0, 0, 0, 6 }; + static unsigned char jazz_dma_bits[] = { 0, 1, 0, 2, 0, 3, 0, 4 }; + + if (jazz_dma_bits[chip->dma8] == 0 || + jazz_dma_bits[chip->dma16] == 0 || + jazz_irq_bits[chip->irq] == 0) + return -EINVAL; + + if (!snd_sbdsp_command(chip, SB_JAZZ16_SET_DMAINTR)) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_dma_bits[chip->dma8] | + (jazz_dma_bits[chip->dma16] << 4))) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_irq_bits[chip->irq] | + (jazz_irq_bits[mpu_irq] << 4))) + return -EBUSY; + + return 0; +} + +static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please specify port\n"); + return 0; + } + if (dma16[dev] != SNDRV_AUTO_DMA && + dma16[dev] != 5 && dma16[dev] != 7) { + snd_printk(KERN_ERR "dma16 must be 5 or 7"); + return 0; + } + return 1; +} + +static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) +{ + struct snd_card *card; + struct snd_card_jazz16 *jazz16; + struct snd_sb *chip; + struct snd_opl3 *opl3; + static int possible_irqs[] = {2, 3, 5, 7, 9, 10, 15, -1}; + static int possible_dmas8[] = {1, 3, -1}; + static int possible_dmas16[] = {5, 7, -1}; + int err, xirq, xdma8, xdma16, xmpu_port, xmpu_irq; + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_jazz16), &card); + if (err < 0) + return err; + + jazz16 = card->private_data; + + xirq = irq[dev]; + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto err_free; + } + } + xdma8 = dma8[dev]; + if (xdma8 == SNDRV_AUTO_DMA) { + xdma8 = snd_legacy_find_free_dma(possible_dmas8); + if (xdma8 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA8\n"); + err = -EBUSY; + goto err_free; + } + } + xdma16 = dma16[dev]; + if (xdma16 == SNDRV_AUTO_DMA) { + xdma16 = snd_legacy_find_free_dma(possible_dmas16); + if (xdma16 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA16\n"); + err = -EBUSY; + goto err_free; + } + } + + xmpu_port = mpu_port[dev]; + if (xmpu_port == SNDRV_AUTO_PORT) + xmpu_port = 0; + err = jazz16_detect_board(port[dev], xmpu_port); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 board not detected\n"); + goto err_free; + } + err = snd_sbdsp_create(card, port[dev], irq[dev], + jazz16_interrupt, + dma8[dev], dma16[dev], + SB_HW_JAZZ16, + &chip); + if (err < 0) + goto err_free; + + xmpu_irq = mpu_irq[dev]; + if (xmpu_irq == SNDRV_AUTO_IRQ || mpu_port[dev] == SNDRV_AUTO_PORT) + xmpu_irq = 0; + err = jazz16_configure_board(chip, xmpu_irq); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 configuration failed\n"); + goto err_free; + } + + jazz16->chip = chip; + + strcpy(card->driver, "jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, + "Media Vision Jazz16 at 0x%lx, irq %d, dma8 %d, dma16 %d", + port[dev], xirq, xdma8, xdma16); + + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto err_free; + err = snd_sbmixer_new(chip); + if (err < 0) + goto err_free; + + err = snd_opl3_create(card, chip->port, chip->port + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", + chip->port, chip->port + 2); + else { + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto err_free; + } + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", + mpu_port[dev]); + } + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto err_free; + + dev_set_drvdata(devptr, card); + return 0; + +err_free: + snd_card_free(card); + return err; +} + +static int __devexit snd_jazz16_remove(struct device *devptr, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(devptr); + + dev_set_drvdata(devptr, NULL); + snd_card_free(card); + return 0; +} + +#ifdef CONFIG_PM +static int snd_jazz16_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + snd_sbmixer_suspend(chip); + return 0; +} + +static int snd_jazz16_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_sbdsp_reset(chip); + snd_sbmixer_resume(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + +static struct isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), +#ifdef CONFIG_PM + .suspend = snd_jazz16_suspend, + .resume = snd_jazz16_resume, +#endif + .driver = { + .name = "jazz16" + }, +}; + +static int __init alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init(alsa_card_jazz16_init) +module_exit(alsa_card_jazz16_exit) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 658d55769c9c..3222aed5fac6 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -106,9 +106,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_CAPTURE_16) + return -EBUSY; + else + chip->mode |= SB_MODE_PLAYBACK_16; + } + chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -133,11 +145,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_PLAYBACK_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_PLAYBACK_8; + dma = chip->dma8; + } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); - if (runtime->channels > 1) { + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW); @@ -147,15 +169,14 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) /* Soundblaster hardware programming reference guide, 3-23 */ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT); runtime->dma_area[0] = 0x80; - snd_dma_program(chip->dma8, runtime->dma_addr, 1, DMA_MODE_WRITE); + snd_dma_program(dma, runtime->dma_addr, 1, DMA_MODE_WRITE); /* force interrupt */ - chip->mode = SB_MODE_HALT; snd_sbdsp_command(chip, SB_DSP_OUTPUT); snd_sbdsp_command(chip, 0); snd_sbdsp_command(chip, 0); } snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save output filter status and turn it off */ @@ -168,13 +189,15 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -212,7 +235,6 @@ static int snd_sb8_playback_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_PLAYBACK_8 : SB_MODE_HALT; return 0; } @@ -234,9 +256,21 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_PLAYBACK_16) + return -EBUSY; + else + chip->mode |= SB_MODE_CAPTURE_16; + } + chip->capture_format = SB_DSP_LO_INPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -262,14 +296,24 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_CAPTURE_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_CAPTURE_8; + dma = chip->dma8; + } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save input filter status and turn it off */ @@ -282,13 +326,15 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_INPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -328,7 +374,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_CAPTURE_8 : SB_MODE_HALT; return 0; } @@ -339,13 +384,21 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) snd_sb_ack_8bit(chip); switch (chip->mode) { - case SB_MODE_PLAYBACK_8: /* ok.. playback is active */ + case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ + case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); break; + case SB_MODE_CAPTURE_16: + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; runtime = substream->runtime; @@ -361,10 +414,15 @@ static snd_pcm_uframes_t snd_sb8_playback_pointer(struct snd_pcm_substream *subs { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_PLAYBACK_8) + if (chip->mode & SB_MODE_PLAYBACK_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_PLAYBACK_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -372,10 +430,15 @@ static snd_pcm_uframes_t snd_sb8_capture_pointer(struct snd_pcm_substream *subst { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_CAPTURE_8) + if (chip->mode & SB_MODE_CAPTURE_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_CAPTURE_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -446,6 +509,13 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; + runtime->hw.rate_min = 4000; + runtime->hw.rate_max = 50000; + runtime->hw.channels_max = 2; + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -468,6 +538,14 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3 || chip->dma16 >= 0) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -480,6 +558,10 @@ static int snd_sb8_close(struct snd_pcm_substream *substream) chip->capture_substream = NULL; spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~SB_OPEN_PCM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + chip->mode &= ~SB_MODE_PLAYBACK; + else + chip->mode &= ~SB_MODE_CAPTURE; spin_unlock_irqrestore(&chip->open_lock, flags); return 0; } @@ -515,6 +597,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc = 64 * 1024; if (rpcm) *rpcm = NULL; @@ -527,9 +610,11 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + if (chip->dma8 > 3 || chip->dma16 >= 0) + max_prealloc = 128 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 27a651502251..eae6c1c0eff9 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -170,6 +170,9 @@ static int snd_sbdsp_probe(struct snd_sb * chip) case SB_HW_CS5530: str = "16 (CS5530)"; break; + case SB_HW_JAZZ16: + str = "Pro (Jazz16)"; + break; default: return -ENODEV; } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 8cfc41fbe368..6496822c1808 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -779,6 +779,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -929,6 +930,7 @@ void snd_sbmixer_suspend(struct snd_sb *chip) save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -955,6 +957,7 @@ void snd_sbmixer_resume(struct snd_sb *chip) restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: -- cgit v1.2.2 From ee7c343c0134bf126b4235e65c407711b77174da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Dec 2009 12:41:37 +0100 Subject: ALSA: pcm - Add missing inclusion of linux/vmalloc.h Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index d9727c74b2e1..d6d49d6651f9 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include -- cgit v1.2.2 From d8d881dd2c814e1500558889d800cf78d11cf898 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 07:52:49 +0100 Subject: ALSA: hda - Fix NULL dereference with enable_beep=0 option Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f8325848..417fb22ae83c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3779,15 +3779,16 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = snd_hda_attach_beep_device(codec, nid); if (err < 0) return err; - /* IDT/STAC codecs have linear beep tone parameter */ - codec->beep->linear_tone = 1; - /* if no beep switch is available, make its own one */ - caps = query_amp_caps(codec, nid, HDA_OUTPUT); - if (codec->beep && - !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) { - err = stac92xx_beep_switch_ctl(codec); - if (err < 0) - return err; + if (codec->beep) { + /* IDT/STAC codecs have linear beep tone parameter */ + codec->beep->linear_tone = 1; + /* if no beep switch is available, make its own one */ + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + if (!(caps & AC_AMPCAP_MUTE)) { + err = stac92xx_beep_switch_ctl(codec); + if (err < 0) + return err; + } } } #endif -- cgit v1.2.2 From 8374e24c23448cabf6e78db2c83841c56c5df1e1 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 21 Dec 2009 17:07:08 +0100 Subject: ALSA: refine rate selection in snd_interval_ratnum() Refine the rate selection by choosing the rate closer to the requested one in case of selecting single frequency. Previously, the higher rate was always selected. Also, fix problem with the best_diff unsigned int value wrapping (turning negative). Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a27545b23ee9..b07cc361afb1 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -745,10 +745,13 @@ int snd_interval_ratnum(struct snd_interval *i, unsigned int rats_count, struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp) { - unsigned int best_num, best_diff, best_den; + unsigned int best_num, best_den; + int best_diff; unsigned int k; struct snd_interval t; int err; + unsigned int result_num, result_den; + int result_diff; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { @@ -770,6 +773,8 @@ int snd_interval_ratnum(struct snd_interval *i, den -= r; } diff = num - q * den; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -784,6 +789,9 @@ int snd_interval_ratnum(struct snd_interval *i, t.min = div_down(best_num, best_den); t.openmin = !!(best_num % best_den); + result_num = best_num; + result_diff = best_diff; + result_den = best_den; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { unsigned int num = rats[k].num; @@ -806,6 +814,8 @@ int snd_interval_ratnum(struct snd_interval *i, den += rats[k].den_step - r; } diff = q * den - num; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -825,10 +835,14 @@ int snd_interval_ratnum(struct snd_interval *i, return err; if (snd_interval_single(i)) { + if (best_diff * result_den < result_diff * best_den) { + result_num = best_num; + result_den = best_den; + } if (nump) - *nump = best_num; + *nump = result_num; if (denp) - *denp = best_den; + *denp = result_den; } return err; } -- cgit v1.2.2 From 1a5ba2e9fc7999b8de2a71c7e7b9f58d752c05e4 Mon Sep 17 00:00:00 2001 From: Rafael Avila de Espindola Date: Tue, 22 Dec 2009 07:59:37 +0100 Subject: ALSA: hda - Add support for the new 27 inch IMacs With the attached patch I am able to use the sound on a new IMac 27. What works: *) Internal speakers *) Internal microphone *) Headphone I don't have an external mic or a SPDIF device to test the rest. Signed-off-by: Rafael Avila de Espindola Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4b200da1bd18..fe0423c39598 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -66,6 +66,7 @@ struct cs_spec { /* available models */ enum { CS420X_MBP55, + CS420X_IMAC27, CS420X_AUTO, CS420X_MODELS }; @@ -827,7 +828,8 @@ static void cs_automute(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); } - if (spec->board_config == CS420X_MBP55) { + if (spec->board_config == CS420X_MBP55 || + spec->board_config == CS420X_IMAC27) { unsigned int gpio = hp_present ? 0x02 : 0x08; snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); @@ -1069,12 +1071,14 @@ static int cs_parse_auto_config(struct hda_codec *codec) static const char *cs420x_models[CS420X_MODELS] = { [CS420X_MBP55] = "mbp55", + [CS420X_IMAC27] = "imac27", [CS420X_AUTO] = "auto", }; static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), + SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), {} /* terminator */ }; @@ -1097,8 +1101,23 @@ static struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; +static struct cs_pincfg imac27_pincfgs[] = { + { 0x09, 0x012b4050 }, + { 0x0a, 0x90100140 }, + { 0x0b, 0x90100142 }, + { 0x0c, 0x018b3020 }, + { 0x0d, 0x90a00110 }, + { 0x0e, 0x400000f0 }, + { 0x0f, 0x01cbe030 }, + { 0x10, 0x014be060 }, + { 0x12, 0x01ab9070 }, + { 0x15, 0x400000f0 }, + {} /* terminator */ +}; + static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP55] = mbp55_pincfgs, + [CS420X_IMAC27] = imac27_pincfgs, }; static void fix_pincfg(struct hda_codec *codec, int model) @@ -1128,6 +1147,7 @@ static int patch_cs420x(struct hda_codec *codec) fix_pincfg(codec, spec->board_config); switch (spec->board_config) { + case CS420X_IMAC27: case CS420X_MBP55: /* GPIO1 = headphones */ /* GPIO3 = speakers */ -- cgit v1.2.2 From 9dc8398bab52931435fce403ce2eaf5822f28e58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 08:15:01 +0100 Subject: ALSA: hda - Add MSI blacklist A machine with AMD CPU with Nvidia board doesn't work with MSI. Reported-by: Robert J. King Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9b56f937913e..ff8ad46cc50e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2322,6 +2322,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) * white/black-list for enable_msi */ static struct snd_pci_quirk msi_black_list[] __devinitdata = { + SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ {} }; -- cgit v1.2.2 From a9605391cfab2bf9a73e51faac5147622f73c6d5 Mon Sep 17 00:00:00 2001 From: Florian Fainelli Date: Mon, 21 Dec 2009 16:36:10 -0800 Subject: ALSA: sound/core/pcm_timer.c: use lib/gcd.c Make sound/core/pcm_timer.c use lib/gcd.c Signed-off-by: Florian Fainelli Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 1 + sound/core/pcm_timer.c | 17 +---------------- 2 files changed, 2 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index c15682a2f9db..475455c76610 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -5,6 +5,7 @@ config SND_TIMER config SND_PCM tristate select SND_TIMER + select GCD config SND_HWDEP tristate diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index ca8068b63d6c..b01d9481d632 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -20,6 +20,7 @@ */ #include +#include #include #include #include @@ -28,22 +29,6 @@ * Timer functions */ -/* Greatest common divisor */ -static unsigned long gcd(unsigned long a, unsigned long b) -{ - unsigned long r; - if (a < b) { - r = a; - a = b; - b = r; - } - while ((r = a % b) != 0) { - a = b; - b = r; - } - return b; -} - void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) { unsigned long rate, mult, fsize, l, post; -- cgit v1.2.2 From 41116e926cb92292fa4fcbe888ae8133fa0038e6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 09:00:14 +0100 Subject: ALSA: cs46xx - Fix suspend/resume with new DSP Fix the basic suspend/resume of snd-cs46xx drivers with new DSP. References: https://bugzilla.redhat.com/show_bug.cgi?id=498287 https://bugzilla.redhat.com/show_bug.cgi?id=160751 Tested-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos.c | 42 +++++++++++++++++++++++++++++++++---- sound/pci/cs46xx/dsp_spos.h | 4 ++++ sound/pci/cs46xx/dsp_spos_scb_lib.c | 33 +++++++++++++---------------- 4 files changed, 58 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1be96ead4244..e6b4a879ae2e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3597,7 +3597,7 @@ static struct cs_card_type __devinitdata cards[] = { #ifdef CONFIG_PM static unsigned int saved_regs[] = { BA0_ACOSV, - BA0_ASER_FADDR, + /*BA0_ASER_FADDR,*/ BA0_ASER_MASTER, BA1_PVOL, BA1_CVOL, diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index f4f0c8f5dad7..3e5ca8fb519f 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -298,6 +298,9 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) if (ins->scbs[i].deleted) continue; cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); +#ifdef CONFIG_PM + kfree(ins->scbs[i].data); +#endif } kfree(ins->code.data); @@ -974,13 +977,11 @@ static struct dsp_scb_descriptor * _map_scb (struct snd_cs46xx *chip, char * nam index = find_free_scb_index (ins); + memset(&ins->scbs[index], 0, sizeof(ins->scbs[index])); strcpy(ins->scbs[index].scb_name, name); ins->scbs[index].address = dest; ins->scbs[index].index = index; - ins->scbs[index].proc_info = NULL; ins->scbs[index].ref_count = 1; - ins->scbs[index].deleted = 0; - spin_lock_init(&ins->scbs[index].lock); desc = (ins->scbs + index); ins->scbs[index].scb_symbol = add_symbol (chip, name, dest, SYMBOL_PARAMETER); @@ -1022,17 +1023,29 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) return desc; } +#define SCB_BYTES (0x10 * 4) + struct dsp_scb_descriptor * cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest) { struct dsp_scb_descriptor * desc; +#ifdef CONFIG_PM + /* copy the data for resume */ + scb_data = kmemdup(scb_data, SCB_BYTES, GFP_KERNEL); + if (!scb_data) + return NULL; +#endif + desc = _map_scb (chip,name,dest); if (desc) { desc->data = scb_data; _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); +#ifdef CONFIG_PM + kfree(scb_data); +#endif } return desc; @@ -1988,7 +2001,28 @@ int cs46xx_dsp_resume(struct snd_cs46xx * chip) continue; _dsp_create_scb(chip, s->data, s->address); } - + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + if (s->updated) + cs46xx_dsp_spos_update_scb(chip, s); + if (s->volume_set) + cs46xx_dsp_scb_set_volume(chip, s, + s->volume[0], s->volume[1]); + } + if (ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) { + cs46xx_dsp_enable_spdif_hw(chip); + snd_cs46xx_poke(chip, (ins->ref_snoop_scb->address + 2) << 2, + (OUTPUT_SNOOP_BUFFER + 0x10) << 0x10); + if (ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) + cs46xx_poke_via_dsp(chip, SP_SPDOUT_CSUV, + ins->spdif_csuv_stream); + } + if (chip->dsp_spos_instance->spdif_status_in) { + cs46xx_poke_via_dsp(chip, SP_ASER_COUNTDOWN, 0x80000005); + cs46xx_poke_via_dsp(chip, SP_SPDIN_CONTROL, 0x800003ff); + } return 0; } #endif diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h index f9e169d33c03..ca47a8114c7f 100644 --- a/sound/pci/cs46xx/dsp_spos.h +++ b/sound/pci/cs46xx/dsp_spos.h @@ -212,6 +212,7 @@ static inline void cs46xx_dsp_spos_update_scb (struct snd_cs46xx * chip, (scb->address + SCBsubListPtr) << 2, (scb->sub_list_ptr->address << 0x10) | (scb->next_scb_ptr->address)); + scb->updated = 1; } static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, @@ -222,6 +223,9 @@ static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val); snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val); + scb->volume_set = 1; + scb->volume[0] = left; + scb->volume[1] = right; } #endif /* __DSP_SPOS_H__ */ #endif /* CONFIG_SND_CS46XX_NEW_DSP */ diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index dd7c41b037b4..00b148a10239 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -115,7 +115,6 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; - unsigned long flags; if ( scb->parent_scb_ptr ) { /* unlink parent SCB */ @@ -153,8 +152,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor scb->next_scb_ptr = ins->the_null_scb; } - spin_lock_irqsave(&chip->reg_lock, flags); - /* update parent first entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,scb->parent_scb_ptr); @@ -162,7 +159,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor cs46xx_dsp_spos_update_scb(chip,scb); scb->parent_scb_ptr = NULL; - spin_unlock_irqrestore(&chip->reg_lock, flags); } } @@ -197,9 +193,9 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * goto _end; #endif - spin_lock_irqsave(&scb->lock, flags); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,scb); - spin_unlock_irqrestore(&scb->lock, flags); + spin_unlock_irqrestore(&chip->reg_lock, flags); cs46xx_dsp_proc_free_scb_desc(scb); if (snd_BUG_ON(!scb->scb_symbol)) @@ -207,6 +203,10 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * remove_symbol (chip,scb->scb_symbol); ins->scbs[scb->index].deleted = 1; +#ifdef CONFIG_PM + kfree(ins->scbs[scb->index].data); + ins->scbs[scb->index].data = NULL; +#endif if (scb->index < ins->scb_highest_frag_index) ins->scb_highest_frag_index = scb->index; @@ -1508,20 +1508,17 @@ int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip, chip->dsp_spos_instance->npcm_channels <= 0)) return -EIO; - spin_lock(&pcm_channel->src_scb->lock); - + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } - spin_lock_irqsave(&chip->reg_lock, flags); pcm_channel->unlinked = 1; - spin_unlock_irqrestore(&chip->reg_lock, flags); _dsp_unlink_scb (chip,pcm_channel->pcm_reader_scb); + spin_unlock_irqrestore(&chip->reg_lock, flags); - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1533,10 +1530,10 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, struct dsp_scb_descriptor * src_scb = pcm_channel->src_scb; unsigned long flags; - spin_lock(&pcm_channel->src_scb->lock); + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked == 0) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } @@ -1552,8 +1549,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, snd_BUG_ON(pcm_channel->pcm_reader_scb->parent_scb_ptr); pcm_channel->pcm_reader_scb->parent_scb_ptr = parent_scb; - spin_lock_irqsave(&chip->reg_lock, flags); - /* update SCB entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,pcm_channel->pcm_reader_scb); @@ -1562,8 +1557,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, pcm_channel->unlinked = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); - - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1596,13 +1589,17 @@ cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * s int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) { + unsigned long flags; + if (snd_BUG_ON(!src->parent_scb_ptr)) return -EINVAL; /* mute SCB */ cs46xx_dsp_scb_set_volume (chip,src,0,0); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,src); + spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } -- cgit v1.2.2 From 75d1aeb9d6899b10420d10284e8ea894b2794224 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 11:56:32 +0100 Subject: ALSA: hda - Add Bass Speaker switch for HP dv7 The bass speaker is controlled via GPIO5. Tested-by: Wael Nasreddine Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 57 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0bafea9d5106..a4526d008042 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5402,6 +5402,54 @@ static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, return 0; } +/* HP dv7 bass switch - GPIO5 */ +#define stac_hp_bass_gpio_info snd_ctl_boolean_mono_info +static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20); + return 0; +} + +static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_data; + + gpio_data = (spec->gpio_data & ~0x20) | + (ucontrol->value.integer.value[0] ? 0x20 : 0); + if (gpio_data == spec->gpio_data) + return 0; + spec->gpio_data = gpio_data; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); + return 1; +} + +static struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = stac_hp_bass_gpio_info, + .get = stac_hp_bass_gpio_get, + .put = stac_hp_bass_gpio_put, +}; + +static int stac_add_hp_bass_switch(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!stac_control_new(spec, &stac_hp_bass_sw_ctrl, + "Bass Speaker Playback Switch", 0)) + return -ENOMEM; + + spec->gpio_mask |= 0x20; + spec->gpio_dir |= 0x20; + spec->gpio_data |= 0x20; + return 0; +} + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5642,6 +5690,15 @@ again: return err; } + /* enable bass on HP dv7 */ + if (spec->board_config == STAC_HP_DV5) { + unsigned int cap; + cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP); + cap &= AC_GPIO_IO_COUNT; + if (cap >= 6) + stac_add_hp_bass_switch(codec); + } + codec->proc_widget_hook = stac92hd7x_proc_hook; return 0; -- cgit v1.2.2 From b6aa179334743c6152bd63f1fa368d6db3720db9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 16 Dec 2009 17:10:09 +0100 Subject: ASoC: sh: FSI:: don't check platform_get_irq's return value against zero MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit platform_get_irq returns -ENXIO on failure, so !irq was probably always true. Better use (int)irq <= 0. Note that a return value of zero is still handled as error even though this could mean irq0. This is a followup to 305b3228f9ff4d59f49e6d34a7034d44ee8ce2f0 that changed the return value of platform_get_irq from 0 to -ENXIO on error. Signed-off-by: Uwe Kleine-König Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9c49c11c43ce..42813b808389 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -876,7 +876,7 @@ static int fsi_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); irq = platform_get_irq(pdev, 0); - if (!res || !irq) { + if (!res || (int)irq <= 0) { dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); ret = -ENODEV; goto exit; -- cgit v1.2.2 From 1628af5adf64cc2960bce81009f119de822f876e Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Tue, 22 Dec 2009 09:26:10 +0100 Subject: ASoC: add missing parameter to mx27vis_hifi_hw_free() Commit 2ccafed4 added an extra parameter to the DAI .set_pll() method, but it missed this call in sound/soc/imx/mx27vis_wm8974.c. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/mx27vis_wm8974.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c index 0267d2d91685..07d2a248438c 100644 --- a/sound/soc/imx/mx27vis_wm8974.c +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -180,7 +180,8 @@ static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); + return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, + 0, 0); } /* -- cgit v1.2.2 From 21949f00a022e090a7e8bc9a01dfca88273c6146 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Dec 2009 08:31:59 +0100 Subject: ALSA: hda - Fix NID association for capture mixers Fix the wrong implementation of NID <-> kctl mapping for capture mixers introduced by the ocmmit 5b0cb1d850c26893b1468b3a519433a1b7a176be. So far, the driver returns an error at probe. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 26 -------------------------- sound/pci/hda/hda_local.h | 2 -- sound/pci/hda/patch_analog.c | 3 +-- sound/pci/hda/patch_cirrus.c | 12 ++++++++---- sound/pci/hda/patch_cmedia.c | 3 +-- sound/pci/hda/patch_realtek.c | 3 +-- sound/pci/hda/patch_via.c | 3 +-- 7 files changed, 12 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c848ec0f085e..29c90d748c91 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3537,32 +3537,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) } EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); -/** - * snd_hda_add_nids - assign nids to controls from the array - * @codec: the HDA codec - * @kctl: struct snd_kcontrol - * @index: index to kctl - * @nids: the array of hda_nid_t - * @size: count of hda_nid_t items - * - * This helper function assigns NIDs in the given array to a control element. - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, - unsigned int index, hda_nid_t *nids, unsigned int size) -{ - int err; - - for ( ; size > 0; size--, nids++) { - err = snd_hda_add_nid(codec, kctl, index, *nids); - if (err < 0) - return err; - } - return 0; -} -EXPORT_SYMBOL_HDA(snd_hda_add_nids); - #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d505d052972e..7cee364976ff 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -343,8 +343,6 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); -int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, - unsigned int index, hda_nid_t *nids, unsigned int size); /* * unsolicited event handler diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 92b72d4f3984..45ee352df329 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -244,8 +244,7 @@ static int ad198x_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 093cfbb55e9e..7de782a5b8f4 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -753,6 +753,7 @@ static int build_input(struct hda_codec *codec) spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; + int n; if (!spec->capture_bind[i]) return -ENOMEM; kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); @@ -762,10 +763,13 @@ static int build_input(struct hda_codec *codec) err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - err = snd_hda_add_nids(codec, kctl, 0, spec->adc_nid, - spec->num_inputs); - if (err < 0) - return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]); + if (err < 0) + return err; + } } if (spec->num_inputs > 1 && !spec->mic_detect) { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index cc1c22370a60..ff60908f4554 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -345,8 +345,7 @@ static int cmi9880_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->adc_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->adc_nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e7cdc6a7d61d..a45199014986 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2551,8 +2551,7 @@ static int alc_build_controls(struct hda_codec *codec) hda_nid_t *nids = spec->capsrc_nids; if (!nids) nids = spec->adc_nids; - err = snd_hda_add_nids(codec, kctl, i, nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index de4839e46762..9ddc37300f6b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1907,8 +1907,7 @@ static int via_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->mux_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]); if (err < 0) return err; } -- cgit v1.2.2 From f62faedbed546f4e0c1ba204999e7c206059f305 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Dec 2009 09:27:51 +0100 Subject: ALSA: hda - Set mixer name after codec patch Postpone the mixer name setup after the codec patch since the codec patch may change the codec name string in itself. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9cfdb771928c..950ee5cfcacf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1086,11 +1086,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) if (err < 0) return err; } - /* audio codec should override the mixer name */ - if (codec->afg || !*codec->bus->card->mixername) - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), - "%s %s", codec->vendor_name, codec->chip_name); if (is_generic_config(codec)) { err = snd_hda_parse_generic_codec(codec); @@ -1109,6 +1104,11 @@ int snd_hda_codec_configure(struct hda_codec *codec) patched: if (!err && codec->patch_ops.unsol_event) err = init_unsol_queue(codec->bus); + /* audio codec should override the mixer name */ + if (!err && (codec->afg || !*codec->bus->card->mixername)) + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); return err; } EXPORT_SYMBOL_HDA(snd_hda_codec_configure); -- cgit v1.2.2 From 48e3cbb3f67a27d9c2db075f3d0f700246c40caa Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Tue, 22 Dec 2009 10:13:24 -0500 Subject: ASoC: Do not write to invalid registers on the wm9712. This patch fixes a bug where "virtual" registers were being written to the ac97 bus. This was causing unrelated registers to become corrupted (headphone 0x04, touchscreen 0x78, etc). This patch duplicates protection that was included in the wm9713 driver. Signed-off-by: Eric Millbrandt Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm9712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 0ac1215dcd9b..e237bf615129 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -463,7 +463,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + if (reg < 0x7c) + soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; -- cgit v1.2.2 From 95e70e87533f9d117d369495ee633cb7d18dc802 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Wed, 23 Dec 2009 17:28:45 +0100 Subject: ALSA: hda - Add STAC9205 PCI_QUIRK for Dell Vostro 1700 Signed-off-by: Anisse Astier Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 417fb22ae83c..eeda7beeb57a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2104,6 +2104,7 @@ static unsigned int ref9205_pin_configs[12] = { 10280204 1028021F 10280228 (Dell Vostro 1500) + 10280229 (Dell Vostro 1700) */ static unsigned int dell_9205_m42_pin_configs[12] = { 0x0321101F, 0x03A11020, 0x400003FA, 0x90170310, @@ -2189,6 +2190,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, "Dell Vostro 1500", STAC_9205_DELL_M42), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229, + "Dell Vostro 1700", STAC_9205_DELL_M42), /* Gateway */ SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD), SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), -- cgit v1.2.2 From 18f98ab54735f66ea84bf679b70fcec5e8b3df66 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Dec 2009 13:22:04 +0900 Subject: ASoC: fsi-ak4642: Remove ak4642_add_i2c_device I2C devices should be registered when platform board setting in latest ASoC. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi-ak4642.c | 30 ------------------------------ 1 file changed, 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index c7af09729c6e..5263ab18f827 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -42,42 +42,12 @@ static struct snd_soc_device fsi_snd_devdata = { .codec_dev = &soc_codec_dev_ak4642, }; -#define AK4642_BUS 0 -#define AK4642_ADR 0x12 -static int ak4642_add_i2c_device(void) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = AK4642_ADR; - strlcpy(info.type, "ak4642", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(AK4642_BUS); - if (!adapter) { - printk(KERN_DEBUG "can't get i2c adapter\n"); - return -ENODEV; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - printk(KERN_DEBUG "can't add i2c device\n"); - return -ENODEV; - } - - return 0; -} - static struct platform_device *fsi_snd_device; static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - ak4642_add_i2c_device(); - fsi_snd_device = platform_device_alloc("soc-audio", -1); if (!fsi_snd_device) goto out; -- cgit v1.2.2 From b3172f222ab5afdc91ea058bd11c42cf169728f3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 24 Dec 2009 01:13:51 +0100 Subject: ASoC: fix params_rate() macro use in several codecs Sevelar ASoC codec drivers wrongly assume, that the params_rate() macro returns one of SNDRV_PCM_RATE_* defines instead of the actual numerical sampling rate. Fix them. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8510.c | 14 +++++++------- sound/soc/codecs/wm8940.c | 14 +++++++------- sound/soc/codecs/wm8974.c | 14 +++++++------- 3 files changed, 21 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 265e68c75df8..af8cb6995a1f 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -424,23 +424,23 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, /* filter coefficient */ switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: adn |= 0x5 << 1; break; - case SNDRV_PCM_RATE_11025: + case 11025: adn |= 0x4 << 1; break; - case SNDRV_PCM_RATE_16000: + case 16000: adn |= 0x3 << 1; break; - case SNDRV_PCM_RATE_22050: + case 22050: adn |= 0x2 << 1; break; - case SNDRV_PCM_RATE_32000: + case 32000: adn |= 0x1 << 1; break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 3d850b97037a..31e39ffd1d8e 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -378,23 +378,23 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, iface |= (1 << 9); switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: addcntrl |= (0x5 << 1); break; - case SNDRV_PCM_RATE_11025: + case 11025: addcntrl |= (0x4 << 1); break; - case SNDRV_PCM_RATE_16000: + case 16000: addcntrl |= (0x3 << 1); break; - case SNDRV_PCM_RATE_22050: + case 22050: addcntrl |= (0x2 << 1); break; - case SNDRV_PCM_RATE_32000: + case 32000: addcntrl |= (0x1 << 1); break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a808675388fc..8812751da8c9 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -482,23 +482,23 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream, /* filter coefficient */ switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: adn |= 0x5 << 1; break; - case SNDRV_PCM_RATE_11025: + case 11025: adn |= 0x4 << 1; break; - case SNDRV_PCM_RATE_16000: + case 16000: adn |= 0x3 << 1; break; - case SNDRV_PCM_RATE_22050: + case 22050: adn |= 0x2 << 1; break; - case SNDRV_PCM_RATE_32000: + case 32000: adn |= 0x1 << 1; break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } -- cgit v1.2.2 From 8b90ca08821fee79e181bfcbc3bbd41ef5637136 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 24 Dec 2009 01:17:46 +0100 Subject: ALSA: Fix indentation in pcm_native.c Signed-off-by: Guennadi Liakhovetski Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 29ab46a12e11..25b0641e6b8c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1918,13 +1918,13 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, hw->rate_min, hw->rate_max); - if (err < 0) - return err; + if (err < 0) + return err; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, hw->period_bytes_min, hw->period_bytes_max); - if (err < 0) - return err; + if (err < 0) + return err; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, hw->periods_min, hw->periods_max); -- cgit v1.2.2 From 44eba3e82b35ae796826a65d8040001582adc10a Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 23 Dec 2009 18:02:41 +0100 Subject: ALSA: jazz16: refine dma and irq selection Narrow the dma and irq selection after the DOS driver. Add ALSA configuration description as well. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sb/jazz16.c | 21 ++++++++++++++++++++- sound/isa/sb/sb8_main.c | 3 ++- 2 files changed, 22 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index d52966b75846..8d21a3feda3a 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -189,10 +189,29 @@ static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) if (port[dev] == SNDRV_AUTO_PORT) { snd_printk(KERN_ERR "please specify port\n"); return 0; + } else if (port[dev] == 0x200 || (port[dev] & ~0x270)) { + snd_printk(KERN_ERR "incorrect port specified\n"); + return 0; + } + if (dma8[dev] != SNDRV_AUTO_DMA && + dma8[dev] != 1 && dma8[dev] != 3) { + snd_printk(KERN_ERR "dma8 must be 1 or 3\n"); + return 0; } if (dma16[dev] != SNDRV_AUTO_DMA && dma16[dev] != 5 && dma16[dev] != 7) { - snd_printk(KERN_ERR "dma16 must be 5 or 7"); + snd_printk(KERN_ERR "dma16 must be 5 or 7\n"); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x030) != 0x300) { + snd_printk(KERN_ERR "incorrect mpu_port specified\n"); + return 0; + } + if (mpu_irq[dev] != SNDRV_AUTO_DMA && + mpu_irq[dev] != 2 && mpu_irq[dev] != 3 && + mpu_irq[dev] != 5 && mpu_irq[dev] != 7) { + snd_printk(KERN_ERR "mpu_irq must be 2, 3, 5 or 7\n"); return 0; } return 1; diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 3222aed5fac6..7d84c9f34dc9 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -510,7 +510,8 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } switch (chip->hardware) { case SB_HW_JAZZ16: - runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (chip->dma16 == 5 || chip->dma16 == 7) + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; runtime->hw.rate_min = 4000; runtime->hw.rate_max = 50000; -- cgit v1.2.2 From ef18beded8ddbaafdf4914bab209f77e60ae3a18 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 25 Dec 2009 13:14:27 +0800 Subject: ALSA: hda - HDMI sticky stream tag support When we run the following commands in turn (with CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0), speaker-test -Dhw:0,3 -c2 -twav # HDMI speaker-test -Dhw:0,0 -c2 -twav # Analog The second command will produce sound in the analog lineout _as well as_ HDMI sink. The root cause is, device 0 "reuses" the same stream tag that was used by device 3, and the "intelhdmi - sticky stream id" patch leaves the HDMI codec in a functional state. So the HDMI codec happily accepts the audio samples which reuse its stream tag. The proposed solution is to remember the last device each azx_dev was assigned to, and prefer to 1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used 2) or assign a never-used azx_dev for HDMI With this patch and the above two speaker-test commands, HDMI codec will use stream tag 8 and Analog codec will use 5. The stream tag used by HDMI codec won't be reused by others, as long as we don't run out of the 4 playback azx_dev's. The legacy Analog codec will continue to use stream tag 5 because its device id is 0 (this is a bit tricky). Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ff8ad46cc50e..ec9c348336cc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -356,6 +356,7 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ + int device; /* last device number assigned to */ unsigned int opened :1; unsigned int running :1; @@ -1441,10 +1442,13 @@ static int __devinit azx_codec_configure(struct azx *chip) */ /* assign a stream for the PCM */ -static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream) +static inline struct azx_dev * +azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) { int dev, i, nums; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct azx_dev *res = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev = chip->playback_index_offset; nums = chip->playback_streams; } else { @@ -1453,10 +1457,15 @@ static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream) } for (i = 0; i < nums; i++, dev++) if (!chip->azx_dev[dev].opened) { - chip->azx_dev[dev].opened = 1; - return &chip->azx_dev[dev]; + res = &chip->azx_dev[dev]; + if (res->device == substream->pcm->device) + break; } - return NULL; + if (res) { + res->opened = 1; + res->device = substream->pcm->device; + } + return res; } /* release the assigned stream */ @@ -1505,7 +1514,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) int err; mutex_lock(&chip->open_mutex); - azx_dev = azx_assign_device(chip, substream->stream); + azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { mutex_unlock(&chip->open_mutex); return -EBUSY; -- cgit v1.2.2 From 729d55ba972348234759f8e40abf8de020f0d505 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Dec 2009 22:49:01 +0100 Subject: ALSA: hda - Disable tigger at pin-sensing on AD codecs Analog Device codecs seem to have problems with the triggering of pin-sensing although their pincaps give the trigger requirements. Some reported that constant CPU load on HP laptops with AD codecs. For avoiding this regression, add a flag to codec struct to notify explicitly that the codec doesn't suppot the trigger at pin-sensing. Tested-by: Maciej Rutecki Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 ++++++---- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_analog.c | 16 ++++++++++++++++ 3 files changed, 23 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 950ee5cfcacf..f98b47cd6cfb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1327,11 +1327,13 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); */ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) { - u32 pincap = snd_hda_query_pin_caps(codec, nid); - - if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + u32 pincap; + if (!codec->no_trigger_sense) { + pincap = snd_hda_query_pin_caps(codec, nid); + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + } return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1d541b7f5547..0a770a28e71f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -817,6 +817,7 @@ struct hda_codec { unsigned int pin_amp_workaround:1; /* pin out-amp takes index * (e.g. Conexant codecs) */ + unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a36137e13ec..69a941c7b158 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1186,6 +1186,8 @@ static int patch_ad1986a(struct hda_codec *codec) */ spec->multiout.no_share_stream = 1; + codec->no_trigger_sense = 1; + return 0; } @@ -1371,6 +1373,8 @@ static int patch_ad1983(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + codec->no_trigger_sense = 1; + return 0; } @@ -1813,6 +1817,9 @@ static int patch_ad1981(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1981_hp_unsol_event; break; } + + codec->no_trigger_sense = 1; + return 0; } @@ -3118,6 +3125,8 @@ static int patch_ad1988(struct hda_codec *codec) #endif spec->vmaster_nid = 0x04; + codec->no_trigger_sense = 1; + return 0; } @@ -3330,6 +3339,8 @@ static int patch_ad1884(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + codec->no_trigger_sense = 1; + return 0; } @@ -4287,6 +4298,8 @@ static int patch_ad1884a(struct hda_codec *codec) break; } + codec->no_trigger_sense = 1; + return 0; } @@ -4623,6 +4636,9 @@ static int patch_ad1882(struct hda_codec *codec) spec->mixers[2] = ad1882_6stack_mixers; break; } + + codec->no_trigger_sense = 1; + return 0; } -- cgit v1.2.2 From a252c81a69c4f9a5a8782f33b91bd837e9dcd406 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Dec 2009 22:56:20 +0100 Subject: ALSA: hda - use snd_hda_jack_detect() again in patch_sigmatel.c Use snd_hda_jack_detect() again for jack-sensing. The triggering problem can be worked around with codec->no_trigger_sense flag now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index eeda7beeb57a..2291a8396817 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4453,14 +4453,7 @@ static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) { if (!nid) return 0; - /* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT - * codecs behave wrongly when SET_PIN_SENSE is triggered, although - * the pincap gives TRIG_REQ bit. - */ - if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE) - return 1; - return 0; + return snd_hda_jack_detect(codec, nid); } static void stac92xx_line_out_detect(struct hda_codec *codec, @@ -4962,6 +4955,7 @@ static int patch_stac9200(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac9200_pin_nids); spec->pin_nids = stac9200_pin_nids; @@ -5024,6 +5018,7 @@ static int patch_stac925x(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; @@ -5108,6 +5103,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); @@ -5255,6 +5251,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; @@ -5418,6 +5415,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->patch_ops = stac92xx_patch_ops; spec->num_pins = STAC92HD71BXX_NUM_PINS; @@ -5661,6 +5659,7 @@ static int patch_stac922x(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac922x_pin_nids); spec->pin_nids = stac922x_pin_nids; @@ -5764,6 +5763,7 @@ static int patch_stac927x(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); @@ -5898,6 +5898,7 @@ static int patch_stac9205(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac9205_pin_nids); spec->pin_nids = stac9205_pin_nids; @@ -6053,6 +6054,7 @@ static int patch_stac9872(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); spec->pin_nids = stac9872_pin_nids; -- cgit v1.2.2 From 903b0eb39e3284cbf4d5a3773858e9586d16cbda Mon Sep 17 00:00:00 2001 From: Peter Huewe Date: Sat, 26 Dec 2009 03:27:45 +0100 Subject: ALSA: sound/arm: Fix build failure caused by missing struct aaci definition This patch fixes a build failure introduced by the patch ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params [1] by adding/moving the aaci struct to the right position. The patch mentioned above merged common source parts into one function, but unfortunately left out the aaci struct and consequently caused a build failure e.g. for arm versatile_config [2] References: [1] http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=d3aee7996c30f928bbbbfd0994148e35d2e83084 [2] http://kisskb.ellerman.id.au/kisskb/buildresult/1893605/ Patch against Linus' tree. Signed-off-by: Peter Huewe Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index c5699863643b..656e474dca47 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -441,6 +441,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int err; + struct aaci *aaci = substream->private_data; aaci_pcm_hw_free(substream); if (aacirun->pcm_open) { @@ -560,7 +561,6 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; unsigned int channels = params_channels(params); int ret; @@ -659,7 +659,6 @@ static struct snd_pcm_ops aaci_playback_ops = { static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; int ret; -- cgit v1.2.2 From 043958e602ac2cbf918c0dab1e4e2a7f9751ebf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 26 Dec 2009 10:36:12 +0100 Subject: ALSA: hda - Add more hints for GPIO setup of IDT/STAC codecs gpio_led, gpio_led_polarity and gpio_mute are added now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 247be19e17b8..69dd5a4e52f2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4184,9 +4184,23 @@ static void stac_store_hints(struct hda_codec *codec) p = snd_hda_get_hint(codec, "eapd_mask"); if (p) spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "gpio_mute"); + if (p) + spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask; val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; + p = snd_hda_get_hint(codec, "gpio_led_polarity"); + if (p) + spec->gpio_led_polarity = simple_strtoul(p, NULL, 0); + p = snd_hda_get_hint(codec, "gpio_led"); + if (p) { + spec->gpio_led = simple_strtoul(p, NULL, 0); + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + if (spec->gpio_led_polarity) + spec->gpio_data |= spec->gpio_led; + } } static int stac92xx_init(struct hda_codec *codec) -- cgit v1.2.2 From 411fe85c7653f51403c2a6fd9026b0db2ab19478 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 10:25:58 +0100 Subject: ALSA: hda - Don't cache beep controls The beep control verbs don't need to be cached for resume. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 5fe34a8d8c81..ca3c57a5f888 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -42,7 +42,7 @@ static void snd_hda_generate_beep(struct work_struct *work) return; /* generate tone */ - snd_hda_codec_write_cache(codec, beep->nid, 0, + snd_hda_codec_write(codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, beep->tone); } @@ -119,7 +119,7 @@ static void snd_hda_do_detach(struct hda_beep *beep) beep->dev = NULL; cancel_work_sync(&beep->beep_work); /* turn off beep for sure */ - snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + snd_hda_codec_write(beep->codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, 0); } @@ -192,7 +192,7 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) beep->enabled = enable; if (!enable) { /* turn off beep */ - snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + snd_hda_codec_write(beep->codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, 0); } if (beep->mode == HDA_BEEP_MODE_SWREG) { -- cgit v1.2.2 From 54f7190b23080c7ac32078ed6a346bdc591ebef1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 13:27:39 +0100 Subject: ALSA: hda - Fix Oops at reloading beep devices The recent change for supporting dynamic beep device allocation caused a problem resulting in Oops at reloading the driver. Also, it ignores the error from input device registration. This patch fixes the wrong check in snd_hda_detach_beep_device(), and returns an error when the input device registration fails properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index ca3c57a5f888..e4581a42ace5 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -239,8 +239,12 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) mutex_init(&beep->mutex); if (beep->mode == HDA_BEEP_MODE_ON) { - beep->enabled = 1; - snd_hda_do_register(&beep->register_work); + int err = snd_hda_do_attach(beep); + if (err < 0) { + kfree(beep); + codec->beep = NULL; + return err; + } } return 0; @@ -253,7 +257,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) if (beep) { cancel_work_sync(&beep->register_work); cancel_delayed_work(&beep->unregister_work); - if (beep->enabled) + if (beep->dev) snd_hda_do_detach(beep); codec->beep = NULL; kfree(beep); -- cgit v1.2.2 From 92ee6162c48fab24f0676969f0f147fc12f8f21c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:18:59 +0100 Subject: ALSA: hda - Add snd_hda_shutup_pins() helper function Add a common helper function for clearing pin controls before suspend. Use the pincfg array instead of looking through all widget tree. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 19 +++++++++++++++++++ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_sigmatel.c | 12 +----------- 3 files changed, 21 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b3554df740ff..94ae69f20925 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -899,6 +899,25 @@ static void restore_pincfgs(struct hda_codec *codec) } } +/** + * snd_hda_shutup_pins - Shut up all pins + * @codec: the HDA codec + * + * Clear all pin controls to shup up before suspend for avoiding click noise. + * The controls aren't cached so that they can be resumed properly. + */ +void snd_hda_shutup_pins(struct hda_codec *codec) +{ + int i; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + /* use read here for syncing after issuing each verb */ + snd_hda_codec_read(codec, pin->nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } +} +EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); + static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0d08ad5bd898..11c4aa8ee996 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -898,6 +898,7 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, unsigned int cfg); int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, hda_nid_t nid, unsigned int cfg); /* for hwdep */ +void snd_hda_shutup_pins(struct hda_codec *codec); /* * Mixer diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 69dd5a4e52f2..dc1d9f124578 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4385,18 +4385,8 @@ static void stac92xx_free_kctls(struct hda_codec *codec) static void stac92xx_shutup(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int i; - hda_nid_t nid; - /* reset each pin before powering down DAC/ADC to avoid click noise */ - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - unsigned int wid_type = get_wcaps_type(wcaps); - if (wid_type == AC_WID_PIN) - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } + snd_hda_shutup_pins(codec); if (spec->eapd_mask) stac_gpio_set(codec, spec->gpio_mask, -- cgit v1.2.2 From a4e09aa3cf592d9f084ff4ceb216be40c4c265dc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:22:24 +0100 Subject: ALSA: hda - Fix click noises at suspend/free with Realtek codecs Call snd_hda_shutup_pins() at suspend and free for avoiding click noises. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6361e6b3c9c5..cd6d139b4fd5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3693,6 +3693,11 @@ static int alc_build_pcms(struct hda_codec *codec) return 0; } +static inline void alc_shutup(struct hda_codec *codec) +{ + snd_hda_shutup_pins(codec); +} + static void alc_free_kctls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -3713,6 +3718,7 @@ static void alc_free(struct hda_codec *codec) if (!spec) return; + alc_shutup(codec); alc_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); @@ -3722,6 +3728,7 @@ static void alc_free(struct hda_codec *codec) static int alc_suspend(struct hda_codec *codec, pm_message_t state) { struct alc_spec *spec = codec->spec; + alc_shutup(codec); if (spec && spec->power_hook) spec->power_hook(codec, 0); return 0; -- cgit v1.2.2 From b82855a0d76ebda1cc14c00040560d77bfa042ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:24:56 +0100 Subject: ALSA: hda - Add sanity check for storing the user-defined pin configs Check whether the given NID is a pin widget before storing the user-defined pin configs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 94ae69f20925..d02ea8926e7e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -824,6 +824,9 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, struct hda_pincfg *pin; unsigned int oldcfg; + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return -EINVAL; + oldcfg = snd_hda_codec_get_pincfg(codec, nid); pin = look_up_pincfg(codec, list, nid); if (!pin) { -- cgit v1.2.2 From 014c41fce1bd5cec381e70fc6f58fdfc96cdaf69 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 13:53:24 +0100 Subject: ALSA: hda - Use strict_strtoul() Rewrite the codes to use strict_strtoul() instead of simple_strtoul(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 7 ++++-- sound/pci/hda/patch_sigmatel.c | 48 +++++++++++++++++++++++------------------- 2 files changed, 31 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 40ccb419b6e9..b36919c0d363 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -293,8 +293,11 @@ static ssize_t type##_store(struct device *dev, \ { \ struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ struct hda_codec *codec = hwdep->private_data; \ - char *after; \ - codec->type = simple_strtoul(buf, &after, 0); \ + unsigned long val; \ + int err = strict_strtoul(buf, 0, &val); \ + if (err < 0) \ + return err; \ + codec->type = val; \ return count; \ } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dc1d9f124578..e28c810bc00c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4159,43 +4159,47 @@ static void stac92xx_power_down(struct hda_codec *codec) static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, int enable); +static inline int get_int_hint(struct hda_codec *codec, const char *key, + int *valp) +{ + const char *p; + p = snd_hda_get_hint(codec, key); + if (p) { + unsigned long val; + if (!strict_strtoul(p, 0, &val)) { + *valp = val; + return 1; + } + } + return 0; +} + /* override some hints from the hwdep entry */ static void stac_store_hints(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - const char *p; int val; val = snd_hda_get_bool_hint(codec, "hp_detect"); if (val >= 0) spec->hp_detect = val; - p = snd_hda_get_hint(codec, "gpio_mask"); - if (p) { - spec->gpio_mask = simple_strtoul(p, NULL, 0); + if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) { spec->eapd_mask = spec->gpio_dir = spec->gpio_data = spec->gpio_mask; } - p = snd_hda_get_hint(codec, "gpio_dir"); - if (p) - spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "gpio_data"); - if (p) - spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "eapd_mask"); - if (p) - spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "gpio_mute"); - if (p) - spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) + spec->gpio_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) + spec->eapd_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) + spec->gpio_mute &= spec->gpio_mask; val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; - p = snd_hda_get_hint(codec, "gpio_led_polarity"); - if (p) - spec->gpio_led_polarity = simple_strtoul(p, NULL, 0); - p = snd_hda_get_hint(codec, "gpio_led"); - if (p) { - spec->gpio_led = simple_strtoul(p, NULL, 0); + get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity); + if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { spec->gpio_mask |= spec->gpio_led; spec->gpio_dir |= spec->gpio_led; if (spec->gpio_led_polarity) -- cgit v1.2.2 From dfb12eeb0f04b37e5eb3858864d074af4ecd2ac7 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 15:48:40 -0500 Subject: ALSA: atiixp: Specify codec for Foxconn RC4107MA-RS2 BugLink: https://bugs.launchpad.net/ubuntu/+bug/498863 This mainboard needs ac97_codec=0. Cc: stable@kernel.org Tested-by: Apoorv Parle Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752dff2a44..42b4fbbd8e2b 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -297,6 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = { MODULE_DEVICE_TABLE(pci, snd_atiixp_ids); static struct snd_pci_quirk atiixp_quirks[] __devinitdata = { + SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0), SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0), { } /* terminator */ }; -- cgit v1.2.2 From 9980c6209ebc2a02eb3ca51a4fae1e17f645c868 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Sun, 27 Dec 2009 22:26:47 +0100 Subject: ALSA: test off by one in setsamplerate() With `while (i++ < MAX_WRITE_RETRY)' i reaches MAX_WRITE_RETRY + 1 after the loop Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b5ca02e2038c..e66ef2b69b5d 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate) rptr.retwords[2] != M && rptr.retwords[3] != N && i++ < MAX_WRITE_RETRY); - if (i == MAX_WRITE_RETRY) { + if (i > MAX_WRITE_RETRY) { snd_printdd("sent samplerate %d: %d failed\n", *intdec, rate); return -EIO; -- cgit v1.2.2 From ea52bf260ecbb175339af3178c15788df21b7516 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 18:48:29 -0500 Subject: ALSA: hda: Add powerdown for Analog Devices HDA codecs This patch ports powerdown fixes to AD198x. Currently we only turn off Front and HP for suspend, but this is easily extended for additional nids. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 68 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 68 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 45ee352df329..cecd3c108990 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -441,6 +441,11 @@ static int ad198x_build_pcms(struct hda_codec *codec) return 0; } +static inline void ad198x_shutup(struct hda_codec *codec) +{ + snd_hda_shutup_pins(codec); +} + static void ad198x_free_kctls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -454,6 +459,46 @@ static void ad198x_free_kctls(struct hda_codec *codec) snd_array_free(&spec->kctls); } +static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front, + hda_nid_t hp) +{ + struct ad198x_spec *spec = codec->spec; + snd_hda_codec_write(codec, front, 0, AC_VERB_SET_EAPD_BTLENABLE, + !spec->inv_eapd ? 0x00 : 0x02); + snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_EAPD_BTLENABLE, + !spec->inv_eapd ? 0x00 : 0x02); +} + +static void ad198x_power_eapd(struct hda_codec *codec) +{ + /* We currently only handle front, HP */ + switch (codec->vendor_id) { + case 0x11d41882: + case 0x11d4882a: + case 0x11d41884: + case 0x11d41984: + case 0x11d41883: + case 0x11d4184a: + case 0x11d4194a: + case 0x11d4194b: + ad198x_power_eapd_write(codec, 0x12, 0x11); + break; + case 0x11d41981: + case 0x11d41983: + ad198x_power_eapd_write(codec, 0x05, 0x06); + break; + case 0x11d41986: + ad198x_power_eapd_write(codec, 0x1b, 0x1a); + break; + case 0x11d41988: + case 0x11d4198b: + case 0x11d4989a: + case 0x11d4989b: + ad198x_power_eapd_write(codec, 0x29, 0x22); + break; + } +} + static void ad198x_free(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -461,11 +506,29 @@ static void ad198x_free(struct hda_codec *codec) if (!spec) return; + ad198x_shutup(codec); ad198x_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); } +#ifdef SND_HDA_NEEDS_RESUME +static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) +{ + ad198x_shutup(codec); + ad198x_power_eapd(codec); + return 0; +} + +static int ad198x_resume(struct hda_codec *codec) +{ + ad198x_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + return 0; +} +#endif + static struct hda_codec_ops ad198x_patch_ops = { .build_controls = ad198x_build_controls, .build_pcms = ad198x_build_pcms, @@ -474,6 +537,11 @@ static struct hda_codec_ops ad198x_patch_ops = { #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = ad198x_check_power_status, #endif +#ifdef SND_HDA_NEEDS_RESUME + .suspend = ad198x_suspend, + .resume = ad198x_resume, +#endif + .reboot_notify = ad198x_shutup, }; -- cgit v1.2.2 From c97259df3f2e163c72f4d0685c61fb2e026dc989 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 18:52:08 -0500 Subject: ALSA: hda: Refactor powerdown for Realtek HDA codecs This patch converts the alc889 Aspire-specific powerdown to a generic one. Like the previous effort, it currently only handles Front and PCM but can be easily extended to cover other nids. The existing hook for alc889 Aspire-specific remains enabled. Upon further testing, I've added its use for ALC861_AUTO as well. Following patches will enable them for other quirks. Tested-by: Dr. David Alan Gilbert Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 60 +++++++++++++++++++++++++++---------------- 1 file changed, 38 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd6d139b4fd5..141ff446104a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -338,7 +338,7 @@ struct alc_spec { void (*init_hook)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); #ifdef CONFIG_SND_HDA_POWER_SAVE - void (*power_hook)(struct hda_codec *codec, int power); + void (*power_hook)(struct hda_codec *codec); #endif /* for pin sensing */ @@ -391,7 +391,7 @@ struct alc_config_preset { void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_amp_list *loopbacks; - void (*power_hook)(struct hda_codec *codec, int power); + void (*power_hook)(struct hda_codec *codec); #endif }; @@ -1835,16 +1835,6 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x1b; } -#ifdef CONFIG_SND_HDA_POWER_SAVE -static void alc889_power_eapd(struct hda_codec *codec, int power) -{ - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); -} -#endif - /* * ALC880 3-stack model * @@ -3725,12 +3715,40 @@ static void alc_free(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE +static void alc_power_eapd(struct hda_codec *codec) +{ + /* We currently only handle front, HP */ + switch (codec->vendor_id) { + case 0x10ec0260: + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + break; + case 0x10ec0262: + case 0x10ec0267: + case 0x10ec0268: + case 0x10ec0269: + case 0x10ec0272: + case 0x10ec0660: + case 0x10ec0662: + case 0x10ec0663: + case 0x10ec0862: + case 0x10ec0889: + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + break; + } +} + static int alc_suspend(struct hda_codec *codec, pm_message_t state) { struct alc_spec *spec = codec->spec; alc_shutup(codec); if (spec && spec->power_hook) - spec->power_hook(codec, 0); + spec->power_hook(codec); return 0; } #endif @@ -3738,16 +3756,9 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct alc_spec *spec = codec->spec; -#endif codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (spec && spec->power_hook) - spec->power_hook(codec, 1); -#endif return 0; } #endif @@ -3767,6 +3778,7 @@ static struct hda_codec_ops alc_patch_ops = { .suspend = alc_suspend, .check_power_status = alc_check_power_status, #endif + .reboot_notify = alc_shutup, }; @@ -9547,7 +9559,7 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc889_acer_aspire_8930g_setup, .init_hook = alc_automute_amp, #ifdef CONFIG_SND_HDA_POWER_SAVE - .power_hook = alc889_power_eapd, + .power_hook = alc_power_eapd, #endif }, [ALC888_ACER_ASPIRE_7730G] = { @@ -14984,8 +14996,12 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; codec->patch_ops = alc_patch_ops; - if (board_config == ALC861_AUTO) + if (board_config == ALC861_AUTO) { spec->init_hook = alc861_auto_init; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = alc_power_eapd; +#endif + } #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; -- cgit v1.2.2 From 78b8d5d2ee280c463908fd75f3bdf246bcb6ac8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Dec 2009 12:24:22 +0100 Subject: ALSA: usb-audio - Avoid Oops after disconnect As the release of substreams may be done asynchronously from the disconnection, close callback needs to check the shutdown flag before actually accessing the usb interface. Reference: Novell bnc#505027 http://bugzilla.novell.com/show_bug.cgi?id=565027 Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4963defee18a..9edef4684978 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1936,7 +1936,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; - if (subs->interface >= 0) { + if (!as->chip->shutdown && subs->interface >= 0) { usb_set_interface(subs->dev, subs->interface, 0); subs->interface = -1; } -- cgit v1.2.2 From 7d2b451e65d255427c108e990507964ac39c13ee Mon Sep 17 00:00:00 2001 From: Sergiy Kovalchuk Date: Sun, 27 Dec 2009 09:13:41 -0800 Subject: ALSA: usb-audio - Added functionality for E-mu 0404USB/0202USB/TrackerPre Added functionality: 1) Extension Units support (all XU settings now available at alsamixer, kmix, etc): - "AnalogueIn soft limiter" switch; - "Sample rate" selector (values 0,1,2,3,4,5 corresponds to 44.1 48 ... 192 kHz); - "DigitalIn CLK source" selector (internal/external) (**); - "DigitalOut format SPDIF/AC3" switch (**); (**)E-mu-0404usb only. 2) Automatic device sample rate adjustment depending on substream samplerate for both capture and playback substream. [minor coding-style fixes by tiwai] Signed-off-by: Sergiy Kovalchuk Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 49 ++++++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 13 +++++++++ sound/usb/usbmixer.c | 75 +++++++++++++++++++++++++++++++++++++++++++++++++--- 3 files changed, 133 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 31b63ea098b7..286fa14e48bd 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1270,6 +1270,47 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, return 0; } +/* + * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * not for interface. + */ +static void set_format_emu_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + unsigned char emu_samplerate_id = 0; + + /* When capture is active + * sample rate shouldn't be changed + * by playback substream + */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (subs->stream->substream[SNDRV_PCM_STREAM_CAPTURE].interface != -1) + return; + } + + switch (fmt->rate_min) { + case 48000: + emu_samplerate_id = EMU_QUIRK_SR_48000HZ; + break; + case 88200: + emu_samplerate_id = EMU_QUIRK_SR_88200HZ; + break; + case 96000: + emu_samplerate_id = EMU_QUIRK_SR_96000HZ; + break; + case 176400: + emu_samplerate_id = EMU_QUIRK_SR_176400HZ; + break; + case 192000: + emu_samplerate_id = EMU_QUIRK_SR_192000HZ; + break; + default: + emu_samplerate_id = EMU_QUIRK_SR_44100HZ; + break; + } + snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); +} + /* * find a matching format and set up the interface */ @@ -1383,6 +1424,14 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->cur_audiofmt = fmt; + switch (subs->stream->chip->usb_id) { + case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ + case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + set_format_emu_quirk(subs, fmt); + break; + } + #if 0 printk(KERN_DEBUG "setting done: format = %d, rate = %d..%d, channels = %d\n", diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9826337c76b8..152216738cce 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -208,6 +208,16 @@ struct snd_usb_midi_endpoint_info { /* */ +/*E-mu USB samplerate control quirk*/ +enum { + EMU_QUIRK_SR_44100HZ = 0, + EMU_QUIRK_SR_48000HZ, + EMU_QUIRK_SR_88200HZ, + EMU_QUIRK_SR_96000HZ, + EMU_QUIRK_SR_176400HZ, + EMU_QUIRK_SR_192000HZ +}; + #define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) @@ -233,6 +243,9 @@ void snd_usbmidi_input_stop(struct list_head* p); void snd_usbmidi_input_start(struct list_head* p); void snd_usbmidi_disconnect(struct list_head *p); +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id); + /* * retrieve usb_interface descriptor from the host interface * (conditional for compatibility with the older API) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index c998220b99c6..f5596cfdbde1 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -186,6 +186,21 @@ enum { USB_PROC_DCR_RELEASE = 6, }; +/*E-mu 0202(0404) eXtension Unit(XU) control*/ +enum { + USB_XU_CLOCK_RATE = 0xe301, + USB_XU_CLOCK_SOURCE = 0xe302, + USB_XU_DIGITAL_IO_STATUS = 0xe303, + USB_XU_DEVICE_OPTIONS = 0xe304, + USB_XU_DIRECT_MONITORING = 0xe305, + USB_XU_METERING = 0xe306 +}; +enum { + USB_XU_CLOCK_SOURCE_SELECTOR = 0x02, /* clock source*/ + USB_XU_CLOCK_RATE_SELECTOR = 0x03, /* clock rate */ + USB_XU_DIGITAL_FORMAT_SELECTOR = 0x01, /* the spdif format */ + USB_XU_SOFT_LIMIT_SELECTOR = 0x03 /* soft limiter */ +}; /* * manual mapping of mixer names @@ -1330,7 +1345,32 @@ static struct procunit_info procunits[] = { { USB_PROC_DCR, "DCR", dcr_proc_info }, { 0 }, }; - +/* + * predefined data for extension units + */ +static struct procunit_value_info clock_rate_xu_info[] = { + { USB_XU_CLOCK_RATE_SELECTOR, "Selector", USB_MIXER_U8, 0 }, + { 0 } +}; +static struct procunit_value_info clock_source_xu_info[] = { + { USB_XU_CLOCK_SOURCE_SELECTOR, "External", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_value_info spdif_format_xu_info[] = { + { USB_XU_DIGITAL_FORMAT_SELECTOR, "SPDIF/AC3", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_value_info soft_limit_xu_info[] = { + { USB_XU_SOFT_LIMIT_SELECTOR, " ", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_info extunits[] = { + { USB_XU_CLOCK_RATE, "Clock rate", clock_rate_xu_info }, + { USB_XU_CLOCK_SOURCE, "DigitalIn CLK source", clock_source_xu_info }, + { USB_XU_DIGITAL_IO_STATUS, "DigitalOut format:", spdif_format_xu_info }, + { USB_XU_DEVICE_OPTIONS, "AnalogueIn Soft Limit", soft_limit_xu_info }, + { 0 } +}; /* * build a processing/extension unit */ @@ -1391,8 +1431,18 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned cval->max = dsc[15]; cval->res = 1; cval->initialized = 1; - } else - get_min_max(cval, valinfo->min_value); + } else { + if (type == USB_XU_CLOCK_RATE) { + /* E-Mu USB 0404/0202/TrackerPre + * samplerate control quirk + */ + cval->min = 0; + cval->max = 5; + cval->res = 1; + cval->initialized = 1; + } else + get_min_max(cval, valinfo->min_value); + } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (! kctl) { @@ -1433,7 +1483,7 @@ static int parse_audio_processing_unit(struct mixer_build *state, int unitid, un static int parse_audio_extension_unit(struct mixer_build *state, int unitid, unsigned char *desc) { - return build_audio_procunit(state, unitid, desc, NULL, "Extension Unit"); + return build_audio_procunit(state, unitid, desc, extunits, "Extension Unit"); } @@ -2109,6 +2159,23 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) return 0; } +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id) +{ + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid = 12; /* SamleRate ExtensionUnit ID */ + + list_for_each_entry(mixer, &chip->mixer_list, list) { + cval = mixer->id_elems[unitid]; + if (cval) { + set_cur_ctl_value(cval, cval->control << 8, samplerate_id); + snd_usb_mixer_notify_id(mixer, unitid); + } + break; + } +} + int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error) { -- cgit v1.2.2 From adc8d31326c32a2a1e145ab80accbc3c6570b117 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 27 Dec 2009 12:19:57 -0500 Subject: ALSA: usb-audio: make buffer pointer based on bytes instead on frames Since there are devices that do not align the size of their data packets to frame boundaries, the driver needs to be able to keep track of partial frames. This patch prepares for support for such devices by changing the hwptr_done variable from a frame counter to a byte counter. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 76 +++++++++++++++++++++++++--------------------------- 1 file changed, 37 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 286fa14e48bd..8fcb5d5a94b6 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -173,7 +173,7 @@ struct snd_usb_substream { unsigned int running: 1; /* running status */ - unsigned int hwptr_done; /* processed frame position in the buffer */ + unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ @@ -342,7 +342,7 @@ static int retire_capture_urb(struct snd_usb_substream *subs, unsigned long flags; unsigned char *cp; int i; - unsigned int stride, len, oldptr; + unsigned int stride, frames, bytes, oldptr; int period_elapsed = 0; stride = runtime->frame_bits >> 3; @@ -353,29 +353,28 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - len = urb->iso_frame_desc[i].actual_length / stride; - if (! len) - continue; + frames = urb->iso_frame_desc[i].actual_length / stride; + bytes = frames * stride; /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; - subs->hwptr_done += len; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - subs->transfer_done += len; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; } spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ - if (oldptr + len > runtime->buffer_size) { - unsigned int cnt = runtime->buffer_size - oldptr; - unsigned int blen = cnt * stride; - memcpy(runtime->dma_area + oldptr * stride, cp, blen); - memcpy(runtime->dma_area, cp + blen, len * stride - blen); + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); } else { - memcpy(runtime->dma_area + oldptr * stride, cp, len * stride); + memcpy(runtime->dma_area + oldptr, cp, bytes); } } if (period_elapsed) @@ -562,24 +561,24 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { - int i, stride, offs; - unsigned int counts; + int i, stride; + unsigned int counts, frames, bytes; unsigned long flags; int period_elapsed = 0; struct snd_urb_ctx *ctx = urb->context; stride = runtime->frame_bits >> 3; - offs = 0; + frames = 0; urb->dev = ctx->subs->dev; /* we need to set this at each time */ urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { counts = snd_usb_audio_next_packet_size(subs); /* set up descriptor */ - urb->iso_frame_desc[i].offset = offs * stride; + urb->iso_frame_desc[i].offset = frames * stride; urb->iso_frame_desc[i].length = counts * stride; - offs += counts; + frames += counts; urb->number_of_packets++; subs->transfer_done += counts; if (subs->transfer_done >= runtime->period_size) { @@ -589,7 +588,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ - offs -= subs->transfer_done; + frames -= subs->transfer_done; counts -= subs->transfer_done; urb->iso_frame_desc[i].length = counts * stride; @@ -599,7 +598,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (i < ctx->packets) { /* add a transfer delimiter */ urb->iso_frame_desc[i].offset = - offs * stride; + frames * stride; urb->iso_frame_desc[i].length = 0; urb->number_of_packets++; } @@ -609,26 +608,25 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (period_elapsed) /* finish at the period boundary */ break; } - if (subs->hwptr_done + offs > runtime->buffer_size) { + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { /* err, the transferred area goes over buffer boundary. */ - unsigned int len = runtime->buffer_size - subs->hwptr_done; + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - len * stride); - memcpy(urb->transfer_buffer + len * stride, - runtime->dma_area, - (offs - len) * stride); + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); } else { memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - offs * stride); + runtime->dma_area + subs->hwptr_done, bytes); } - subs->hwptr_done += offs; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - runtime->delay += offs; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = offs * stride; + urb->transfer_buffer_length = bytes; if (period_elapsed) snd_pcm_period_elapsed(subs->pcm_substream); return 0; @@ -901,18 +899,18 @@ static int wait_clear_urbs(struct snd_usb_substream *subs) /* - * return the current pcm pointer. just return the hwptr_done value. + * return the current pcm pointer. just based on the hwptr_done value. */ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs; - snd_pcm_uframes_t hwptr_done; + unsigned int hwptr_done; subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; spin_unlock(&subs->lock); - return hwptr_done; + return hwptr_done / (substream->runtime->frame_bits >> 3); } -- cgit v1.2.2 From 98e89f606c38a310a20342f90e0c453e6afadf18 Mon Sep 17 00:00:00 2001 From: "John S. Gruber" Date: Sun, 27 Dec 2009 12:19:58 -0500 Subject: ALSA: usb-audio: relax urb data align. restriction HVR-950Q and HVR-850 only Addressing audio quality problem. In sound/usb/usbaudio.c, for the Hauppage HVR-950Q and HVR-850 only, change retire_capture_urb to allow transfers on audio sub-slot boundaries rather than audio slots boundaries. With these devices the left and right channel samples can be split between two different urbs. Throwing away extra channel samples causes a sound quality problem for stereo streams as the left and right channels are swapped repeatedly, perhaps many times per second. Urbs unaligned on sub-slot boundaries are still truncated to the next lowest stride (audio slot) to retain synchronization on samples even though left/right channel synchronization may be lost in this case. Detect the quirk using a case statement in snd_usb_audio_probe. BugLink: https://bugs.launchpad.net/ubuntu/+bug/495745 Signed-off-by: John S. Gruber Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 31 +++++++++++++++++++++++++++++-- sound/usb/usbaudio.h | 1 + 2 files changed, 30 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8fcb5d5a94b6..617515f6ec7b 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -169,6 +169,7 @@ struct snd_usb_substream { unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int fill_max: 1; /* fill max packet size always */ + unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int running: 1; /* running status */ @@ -353,14 +354,25 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - frames = urb->iso_frame_desc[i].actual_length / stride; - bytes = frames * stride; + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; @@ -2238,6 +2250,7 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo subs->stream = as; subs->direction = stream; subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { subs->ops = audio_urb_ops[stream]; } else { @@ -3618,6 +3631,20 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } + switch (chip->usb_id) { + case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ + case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ + chip->txfr_quirk = 1; + break; + default: + chip->txfr_quirk = 0; + } err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 152216738cce..d180554b81f0 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -125,6 +125,7 @@ struct snd_usb_audio { struct snd_card *card; u32 usb_id; int shutdown; + unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ int num_interfaces; int num_suspended_intf; -- cgit v1.2.2 From 52a7a5835173af61b9f6c3038212370d9717526f Mon Sep 17 00:00:00 2001 From: "John S. Gruber" Date: Sun, 27 Dec 2009 12:19:59 -0500 Subject: ALSA: usb-audio: use usbquirk.h for detection of HVR-950Q/850 Detect the HVR-950Q HVR-850 urb data alignment quirk using usbquirk.h rather than using a case statement in snd_usb_audio_probe. Signed-off-by: John S. Gruber Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 30 ++++++------- sound/usb/usbaudio.h | 1 + sound/usb/usbquirks.h | 114 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 130 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 617515f6ec7b..4ada98e16309 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3203,6 +3203,18 @@ static int ignore_interface_quirk(struct snd_usb_audio *chip, return 0; } +/* + * Allow alignment on audio sub-slot (channel samples) rather than + * on audio slots (audio frames) + */ +static int create_align_transfer_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) +{ + chip->txfr_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + /* * boot quirks @@ -3377,7 +3389,8 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, - [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, + [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { @@ -3631,20 +3644,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } - switch (chip->usb_id) { - case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ - case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ - chip->txfr_quirk = 1; - break; - default: - chip->txfr_quirk = 0; - } + chip->txfr_quirk = 0; err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d180554b81f0..9d8cea48fc5f 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -161,6 +161,7 @@ enum quirk_type { QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UAXX, + QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index bd6706c2d534..65bbd22f2e0c 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2074,6 +2074,120 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Hauppauge HVR-950Q and HVR-850 */ +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7201), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7202), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7203), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7204), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7205), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7250), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7230), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-850", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v1.2.2 From afe1c2cd71eb4e0fade720b5709722e7124f29c0 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 25 Dec 2009 14:10:06 +0800 Subject: ASoC: ad1836: reset and restore clock control mode in suspend/resume entry Tests show frequent suspend/resume(frequent poweroff/on ad1836 internal components) maybe make ad1836 clock mode wrong sometimes after wakeup. This patch reset/restore ad1836 clock mode while executing PM, then ad1836 can always resume to right clock status. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 2c18e3d1b71e..83add2f3afba 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,6 +223,36 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } +#ifdef CONFIG_PM +static int ad1836_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* reset clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} + +static int ad1836_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* restore clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 |= AD1836_ADC_AUX; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} +#else +#define ad1836_soc_suspend NULL +#define ad1836_soc_resume NULL +#endif + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -404,6 +434,8 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, + .suspend = ad1836_soc_suspend, + .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); -- cgit v1.2.2 From 08ba864e2789a94c259b8d0aee13a5a183edd46e Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 25 Dec 2009 14:10:07 +0800 Subject: ASoC: ad1938: fix typo, rename mask to rx_mask for ad1938_set_tdm_slot Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1938.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 5d489186c05b..735c3562d20d 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -145,7 +145,7 @@ static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) } static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, - unsigned int mask, int slots, int width) + unsigned int rx_mask, int slots, int width) { struct snd_soc_codec *codec = dai->codec; int dac_reg = codec->read(codec, AD1938_DAC_CTRL1); -- cgit v1.2.2 From 5b61735534193ab357636d5b56c098f0bbe8bac8 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 25 Dec 2009 14:10:08 +0800 Subject: ASoC: ad1938: let soc-core dapm handle PLL power PM architecture of ad1938 is simple, we don't need a bundle of functions like ad1938_pll_powerctrl, ad1938_set_bias_level for only PLL. A dapm supply will handle on/off of PLL. Since soc-core can poweron/off PLL on-demand, we don't need to poweron/off PLL in suspend/resume entries too. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1938.c | 62 +++-------------------------------------------- 1 file changed, 3 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 735c3562d20d..47d9ac0ec9d9 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -97,6 +97,7 @@ static const struct snd_kcontrol_new ad1938_snd_controls[] = { static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("PLL_PWR", AD1938_PLL_CLK_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), @@ -107,6 +108,8 @@ static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { }; static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "PLL_PWR" }, + { "ADC", NULL, "PLL_PWR" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", "DAC1 Switch", "DAC" }, @@ -134,16 +137,6 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) return 0; } -static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) -{ - int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0); - reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | - AD1938_PLL_POWERDOWN; - codec->write(codec, AD1938_PLL_CLK_CTRL0, reg); - - return 0; -} - static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int width) { @@ -306,24 +299,6 @@ static int ad1938_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ad1938_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - ad1938_pll_powerctrl(codec, 1); - break; - case SND_SOC_BIAS_PREPARE: - break; - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - ad1938_pll_powerctrl(codec, 0); - break; - } - codec->bias_level = level; - return 0; -} - /* * interface to read/write ad1938 register */ @@ -514,7 +489,6 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; - codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -559,7 +533,6 @@ static int ad1938_register(struct ad1938_priv *ad1938) static void ad1938_unregister(struct ad1938_priv *ad1938) { - ad1938_set_bias_level(&ad1938->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&ad1938_dai); snd_soc_unregister_codec(&ad1938->codec); kfree(ad1938); @@ -593,7 +566,6 @@ static int ad1938_probe(struct platform_device *pdev) ARRAY_SIZE(ad1938_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); pcm_err: return ret; @@ -610,37 +582,9 @@ static int ad1938_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM -static int ad1938_suspend(struct platform_device *pdev, - pm_message_t state) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - ad1938_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int ad1938_resume(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->suspend_bias_level == SND_SOC_BIAS_ON) - ad1938_set_bias_level(codec, SND_SOC_BIAS_ON); - - return 0; -} -#else -#define ad1938_suspend NULL -#define ad1938_resume NULL -#endif - struct snd_soc_codec_device soc_codec_dev_ad1938 = { .probe = ad1938_probe, .remove = ad1938_remove, - .suspend = ad1938_suspend, - .resume = ad1938_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1938); -- cgit v1.2.2 From 1c418d1f623438147a485db987de296ab372e0f3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 28 Dec 2009 14:09:05 +0900 Subject: ASoC: fsi: Add over_period flag to prevent the misunderstanding Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7506ef6d287a..b311a9eaf021 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -373,14 +373,16 @@ static int fsi_data_push(struct fsi_priv *fsi) int fifo_free; int width; u8 *start; - int i; + int i, over_period; if (!fsi || !fsi->substream || !fsi->substream->runtime) return -EINVAL; - runtime = fsi->substream->runtime; + over_period = 0; + substream = fsi->substream; + runtime = substream->runtime; /* FSI FIFO has limit. * So, this driver can not send periods data at a time @@ -388,7 +390,7 @@ static int fsi_data_push(struct fsi_priv *fsi) if (fsi->byte_offset >= fsi->period_len * (fsi->periods + 1)) { - substream = fsi->substream; + over_period = 1; fsi->periods = (fsi->periods + 1) % runtime->periods; if (0 == fsi->periods) @@ -429,7 +431,7 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi_irq_enable(fsi, 1); - if (substream) + if (over_period) snd_pcm_period_elapsed(substream); return 0; @@ -443,14 +445,16 @@ static int fsi_data_pop(struct fsi_priv *fsi) int fifo_fill; int width; u8 *start; - int i; + int i, over_period; if (!fsi || !fsi->substream || !fsi->substream->runtime) return -EINVAL; - runtime = fsi->substream->runtime; + over_period = 0; + substream = fsi->substream; + runtime = substream->runtime; /* FSI FIFO has limit. * So, this driver can not send periods data at a time @@ -458,7 +462,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) if (fsi->byte_offset >= fsi->period_len * (fsi->periods + 1)) { - substream = fsi->substream; + over_period = 1; fsi->periods = (fsi->periods + 1) % runtime->periods; if (0 == fsi->periods) @@ -498,7 +502,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi_irq_enable(fsi, 0); - if (substream) + if (over_period) snd_pcm_period_elapsed(substream); return 0; -- cgit v1.2.2 From 142e8174b3c493f40469d3ecee0e404645e9c483 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 28 Dec 2009 14:09:11 +0900 Subject: ASoC: fsi: Add fsi_get_dai to get snd_soc_dai Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b311a9eaf021..d078151e1de6 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -210,11 +210,17 @@ static int fsi_is_port_a(struct fsi_priv *fsi) return fsi->master->base == fsi->base; } -static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_dai *dai = machine->cpu_dai; + + return machine->cpu_dai; +} + +static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +{ + struct snd_soc_dai *dai = fsi_get_dai(substream); return dai->private_data; } -- cgit v1.2.2 From 59c3b003ddd3c815de1aa015920710a9e4bf195b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 28 Dec 2009 14:09:16 +0900 Subject: ASoC: fsi: Add over/under run error settlement Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index d078151e1de6..123cd6f45e0c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -67,6 +67,7 @@ /* DOFF_ST */ #define ERR_OVER 0x00000010 #define ERR_UNDER 0x00000001 +#define ST_ERR (ERR_OVER | ERR_UNDER) /* CLK_RST */ #define B_CLK 0x00000010 @@ -375,11 +376,12 @@ static int fsi_data_push(struct fsi_priv *fsi) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; + u32 status; int send; int fifo_free; int width; u8 *start; - int i, over_period; + int i, ret, over_period; if (!fsi || !fsi->substream || @@ -435,23 +437,33 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi->byte_offset += send * width; + ret = 0; + status = fsi_reg_read(fsi, DOFF_ST); + if (status & ERR_OVER) { + struct snd_soc_dai *dai = fsi_get_dai(substream); + dev_err(dai->dev, "over run error\n"); + fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR); + ret = -EIO; + } + fsi_irq_enable(fsi, 1); if (over_period) snd_pcm_period_elapsed(substream); - return 0; + return ret; } static int fsi_data_pop(struct fsi_priv *fsi) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; + u32 status; int free; int fifo_fill; int width; u8 *start; - int i, over_period; + int i, ret, over_period; if (!fsi || !fsi->substream || @@ -506,12 +518,21 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi->byte_offset += fifo_fill * width; + ret = 0; + status = fsi_reg_read(fsi, DIFF_ST); + if (status & ERR_UNDER) { + struct snd_soc_dai *dai = fsi_get_dai(substream); + dev_err(dai->dev, "under run error\n"); + fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR); + ret = -EIO; + } + fsi_irq_enable(fsi, 0); if (over_period) snd_pcm_period_elapsed(substream); - return 0; + return ret; } static irqreturn_t fsi_interrupt(int irq, void *data) -- cgit v1.2.2 From 8998c89907f84f7e25536c1c670a134c831e682f Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Thu, 31 Dec 2009 10:30:34 +0800 Subject: ASoC: soc-cache: cleanup training whitespace and coding style Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d2505e8b06c9..02c235711bb8 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -182,7 +182,7 @@ static struct { { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, - .spi_write = snd_soc_7_9_spi_write + .spi_write = snd_soc_7_9_spi_write, }, { .addr_bits = 8, .data_bits = 8, -- cgit v1.2.2 From 7427b4b9a63fd7e051d642ff0f12ef8337c08bb3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:19 +0200 Subject: ASoC: tlv320dac33: Change nsample switch to FIFO mode enum In order to have support for more FIFO modes supported by tlv320dac33, the switch for enabling/disabling the FIFO use has to be replaced with an enum. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 49 +++++++++++++++++++++++++++--------------- 1 file changed, 32 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 5037454974b6..b67961dd2a12 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -59,6 +59,12 @@ enum dac33_state { DAC33_FLUSH, }; +enum dac33_fifo_modes { + DAC33_FIFO_BYPASS = 0, + DAC33_FIFO_MODE1, + DAC33_FIFO_LAST_MODE, +}; + #define DAC33_NUM_SUPPLIES 3 static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { "AVDD", @@ -82,7 +88,7 @@ struct tlv320dac33_priv { * this */ unsigned int nsample_max; /* nsample should not be higher than * this */ - unsigned int nsample_switch; /* Use FIFO or bypass FIFO switch */ + enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */ unsigned int nsample; /* burst read amount from host */ enum dac33_state state; @@ -381,39 +387,48 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol, return ret; } -static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol, +static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct tlv320dac33_priv *dac33 = codec->private_data; - ucontrol->value.integer.value[0] = dac33->nsample_switch; + ucontrol->value.integer.value[0] = dac33->fifo_mode; return 0; } -static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol, +static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct tlv320dac33_priv *dac33 = codec->private_data; int ret = 0; - if (dac33->nsample_switch == ucontrol->value.integer.value[0]) + if (dac33->fifo_mode == ucontrol->value.integer.value[0]) return 0; /* Do not allow changes while stream is running*/ if (codec->active) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || - ucontrol->value.integer.value[0] > 1) + ucontrol->value.integer.value[0] >= DAC33_FIFO_LAST_MODE) ret = -EINVAL; else - dac33->nsample_switch = ucontrol->value.integer.value[0]; + dac33->fifo_mode = ucontrol->value.integer.value[0]; return ret; } +/* Codec operation modes */ +static const char *dac33_fifo_mode_texts[] = { + "Bypass", "Mode 1" +}; + +static const struct soc_enum dac33_fifo_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts), + dac33_fifo_mode_texts); + /* * DACL/R digital volume control: * from 0 dB to -63.5 in 0.5 dB steps @@ -436,8 +451,8 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = { static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, dac33_get_nsample, dac33_set_nsample), - SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0, - dac33_get_nsample_switch, dac33_set_nsample_switch), + SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum, + dac33_get_fifo_mode, dac33_set_fifo_mode), }; /* Analog bypass */ @@ -586,7 +601,7 @@ static void dac33_shutdown(struct snd_pcm_substream *substream, unsigned int pwr_ctrl; /* Stop pending workqueue */ - if (dac33->nsample_switch) + if (dac33->fifo_mode) cancel_work_sync(&dac33->work); mutex_lock(&dac33->mutex); @@ -714,7 +729,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_oscwait(codec); - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { /* 50-51 : ASRC Control registers */ dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ @@ -734,7 +749,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ } - if (dac33->nsample_switch) + if (dac33->fifo_mode) fifoctrl_a &= ~DAC33_FBYPAS; else fifoctrl_a |= DAC33_FBYPAS; @@ -742,13 +757,13 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); - if (dac33->nsample_switch) + if (dac33->fifo_mode) reg_tmp &= ~DAC33_BCLKON; else reg_tmp |= DAC33_BCLKON; dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { /* 20: BCLK divide ratio */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); @@ -828,7 +843,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; queue_work(dac33->dac33_wq, &dac33->work); } @@ -836,7 +851,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; queue_work(dac33->dac33_wq, &dac33->work); } @@ -1125,7 +1140,7 @@ static int dac33_i2c_probe(struct i2c_client *client, dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; /* Disable FIFO use by default */ - dac33->nsample_switch = 0; + dac33->fifo_mode = DAC33_FIFO_BYPASS; tlv320dac33_codec = codec; -- cgit v1.2.2 From d4f102d437c069a64f3a4c7a6cd50360e034541f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:20 +0200 Subject: ASoC: tlv320dac33: Introduce prefill and playback state handlers Ensure that the code is going to be readable, when new FIFO modes are introduced later. Move the prefill and playback state handling to inlined functions. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 46 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 40 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index b67961dd2a12..f7c7bbceb3db 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -543,6 +543,44 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, return 0; } +static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) +{ + struct snd_soc_codec *codec; + + codec = &dac33->codec; + + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + break; + default: + dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", + dac33->fifo_mode); + break; + } +} + +static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) +{ + struct snd_soc_codec *codec; + + codec = &dac33->codec; + + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + break; + default: + dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", + dac33->fifo_mode); + break; + } +} + static void dac33_work(struct work_struct *work) { struct snd_soc_codec *codec; @@ -556,14 +594,10 @@ static void dac33_work(struct work_struct *work) switch (dac33->state) { case DAC33_PREFILL: dac33->state = DAC33_PLAYBACK; - dac33_write16(codec, DAC33_NSAMPLE_MSB, - DAC33_THRREG(dac33->nsample)); - dac33_write16(codec, DAC33_PREFILL_MSB, - DAC33_THRREG(dac33->alarm_threshold)); + dac33_prefill_handler(dac33); break; case DAC33_PLAYBACK: - dac33_write16(codec, DAC33_NSAMPLE_MSB, - DAC33_THRREG(dac33->nsample)); + dac33_playback_handler(dac33); break; case DAC33_IDLE: break; -- cgit v1.2.2 From aec242dc3719e19bd7c1561f8a56a4eb37bb3987 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:21 +0200 Subject: ASoC: tlv320dac33: Clean up the hardware configuration code Use switch instead of if statements to configure FIFO bypass and mode1. With this change adding new FIFO mode is going to be easier, and cleaner. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 61 +++++++++++++++++++++++++++++++----------- 1 file changed, 45 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7c7bbceb3db..c684aa23bd51 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -707,7 +707,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) struct snd_soc_codec *codec = socdev->card->codec; struct tlv320dac33_priv *dac33 = codec->private_data; unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; - u8 aictrl_a, fifoctrl_a; + u8 aictrl_a, aictrl_b, fifoctrl_a; switch (substream->runtime->rate) { case 44100: @@ -764,6 +764,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_oscwait(codec); if (dac33->fifo_mode) { + /* Generic for all FIFO modes */ /* 50-51 : ASRC Control registers */ dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ @@ -773,38 +774,66 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) /* Set interrupts to high active */ dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH); - - dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, - DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); - dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); } else { + /* FIFO bypass mode */ /* 50-51 : ASRC Control registers */ dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP); dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ } - if (dac33->fifo_mode) + /* Interrupt behaviour configuration */ + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, + DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); + dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); + break; + default: + /* in FIFO bypass mode, the interrupts are not used */ + break; + } + + aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + /* + * For mode1: + * Disable the FIFO bypass (Enable the use of FIFO) + * Select nSample mode + * BCLK is only running when data is needed by DAC33 + */ fifoctrl_a &= ~DAC33_FBYPAS; - else + fifoctrl_a &= ~DAC33_FAUTO; + aictrl_b &= ~DAC33_BCLKON; + break; + default: + /* + * For FIFO bypass mode: + * Enable the FIFO bypass (Disable the FIFO use) + * Set the BCLK as continous + */ fifoctrl_a |= DAC33_FBYPAS; - dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); + aictrl_b |= DAC33_BCLKON; + break; + } + dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); - reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); - if (dac33->fifo_mode) - reg_tmp &= ~DAC33_BCLKON; - else - reg_tmp |= DAC33_BCLKON; - dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); - if (dac33->fifo_mode) { + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: /* 20: BCLK divide ratio */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); dac33_write16(codec, DAC33_ATHR_MSB, DAC33_THRREG(dac33->alarm_threshold)); - } else { + break; + default: + /* BYPASS mode */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + break; } mutex_unlock(&dac33->mutex); -- cgit v1.2.2 From 28e05d987028023b09652bfe3ac597de6dba5e60 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:22 +0200 Subject: ASoC: tlv320dac33: Add new FIFO mode: mode 7 Mode 7 of tlv320dac33 operates in the following way: The codec is in master mode. Host configures upper and lower thresholds in tlv320dac33 During playback the codec will clock in the data until the upper threshold is reached in FIFO. At this point the codec stops the colocks on the serial bus. When the FIFO fill is reaching the lower threshold limit the codec will enable the clocks on the serial bus, and clocks in data till the upper threshold is reached. In this mode, we can also request interrupts for threshold events (upper, lower and alarm), which could be used for power management. At this point the interrupts are not enabled for this mode, but it can be taken into use in the future, when the surrounding code makes it possible to use it. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c684aa23bd51..bc35f3ff8717 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,6 +62,7 @@ enum dac33_state { enum dac33_fifo_modes { DAC33_FIFO_BYPASS = 0, DAC33_FIFO_MODE1, + DAC33_FIFO_MODE7, DAC33_FIFO_LAST_MODE, }; @@ -422,7 +423,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, /* Codec operation modes */ static const char *dac33_fifo_mode_texts[] = { - "Bypass", "Mode 1" + "Bypass", "Mode 1", "Mode 7" }; static const struct soc_enum dac33_fifo_mode_enum = @@ -556,6 +557,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); break; + case DAC33_FIFO_MODE7: + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(20)); + break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", dac33->fifo_mode); @@ -574,6 +579,9 @@ static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) dac33_write16(codec, DAC33_NSAMPLE_MSB, DAC33_THRREG(dac33->nsample)); break; + case DAC33_FIFO_MODE7: + /* At the moment we are not using interrupts in mode7 */ + break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", dac33->fifo_mode); @@ -788,6 +796,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); break; + case DAC33_FIFO_MODE7: + /* Disable all interrupts */ + dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0); + break; default: /* in FIFO bypass mode, the interrupts are not used */ break; @@ -807,6 +819,17 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) fifoctrl_a &= ~DAC33_FAUTO; aictrl_b &= ~DAC33_BCLKON; break; + case DAC33_FIFO_MODE7: + /* + * For mode1: + * Disable the FIFO bypass (Enable the use of FIFO) + * Select Threshold mode + * BCLK is only running when data is needed by DAC33 + */ + fifoctrl_a &= ~DAC33_FBYPAS; + fifoctrl_a |= DAC33_FAUTO; + aictrl_b &= ~DAC33_BCLKON; + break; default: /* * For FIFO bypass mode: @@ -830,6 +853,16 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write16(codec, DAC33_ATHR_MSB, DAC33_THRREG(dac33->alarm_threshold)); break; + case DAC33_FIFO_MODE7: + /* + * Configure the threshold levels, and leave 10 sample space + * at the bottom, and also at the top of the FIFO + */ + dac33_write16(codec, DAC33_UTHR_MSB, + DAC33_THRREG(DAC33_BUFFER_SIZE_SAMPLES - 10)); + dac33_write16(codec, DAC33_LTHR_MSB, + DAC33_THRREG(10)); + break; default: /* BYPASS mode */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); -- cgit v1.2.2 From adcb8bc02d86259c117a03b54e9918e5ad3121af Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:23 +0200 Subject: ASoC: tlv320dac33: Safety check for codec slave mode The currently available FIFO modes (mode1 and mode7) require master mode from the codec. Do not allow the slave configuration when the FIFO is in use. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index bc35f3ff8717..3ef3255cd1e7 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -993,6 +993,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; u8 aictrl_a, aictrl_b; aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); @@ -1005,7 +1006,11 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, break; case SND_SOC_DAIFMT_CBS_CFS: /* Codec Slave */ - aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); + if (dac33->fifo_mode) { + dev_err(codec->dev, "FIFO mode requires master mode\n"); + return -EINVAL; + } else + aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); break; default: return -EINVAL; -- cgit v1.2.2 From 633154d3a7bbd542465b905392bf76b780f00b4f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Dec 2009 13:42:43 +0000 Subject: ASoC: Remove unneeded suspend checks from CODEC drivers Better integration of the core with the device model means that we now no longer get the ASoC suspend and resume callbacks without the card having been set up. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8753.c | 8 -------- sound/soc/codecs/wm8990.c | 8 -------- 2 files changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d6850dacda29..c2444e7c8480 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1507,10 +1507,6 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - /* we only need to suspend if we are a valid card */ - if (!codec->card) - return 0; - wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1523,10 +1519,6 @@ static int wm8753_resume(struct platform_device *pdev) u8 data[2]; u16 *cache = codec->reg_cache; - /* we only need to resume if we are a valid card */ - if (!codec->card) - return 0; - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) { if (i + 1 == WM8753_RESET) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 341481e0e830..a54dc77b7f34 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1319,10 +1319,6 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - /* we only need to suspend if we are a valid card */ - if (!codec->card) - return 0; - wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1335,10 +1331,6 @@ static int wm8990_resume(struct platform_device *pdev) u8 data[2]; u16 *cache = codec->reg_cache; - /* we only need to resume if we are a valid card */ - if (!codec->card) - return 0; - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { if (i + 1 == WM8990_RESET) -- cgit v1.2.2 From 40ca114265a281d51b261771df551a373fc8ff3c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Dec 2009 13:44:28 +0000 Subject: ASoC: Use snprintf() when generating stream names Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8b900a842677..9b36c5eec75c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1276,8 +1276,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = card->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, - num); + snprintf(new_name, sizeof(new_name), "%s %s-%d", + dai_link->stream_name, codec_dai->name, num); if (codec_dai->playback.channels_min) playback = 1; -- cgit v1.2.2 From a126fd5691e6cd680758b72e6ea288bb83b9deb6 Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Mon, 4 Jan 2010 14:30:03 +0200 Subject: ASoc: tpa6130a2: Remove unnecessary variable Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0eb33d49942e..8e98ccfab75c 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -267,12 +267,8 @@ static const struct snd_kcontrol_new tpa6130a2_controls[] = { */ static void tpa6130a2_channel_enable(u8 channel, int enable) { - struct tpa6130a2_data *data; u8 val; - BUG_ON(tpa6130a2_client == NULL); - data = i2c_get_clientdata(tpa6130a2_client); - if (enable) { /* Enable channel */ /* Enable amplifier */ -- cgit v1.2.2 From ecbec242961ec66e900b5649ded1e40f5d5edc41 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 4 Jan 2010 16:29:49 +0100 Subject: ASoC: fixup oops in generic AC97 codec glue Initialize the glue by calling snd_soc_new_ac97_codec() as is done in other ASoC AC97 codecs. Fixes an oops caused by dereferencing uninitialized members in snd_soc_new_pcms(). Run-tested on Au1250. Signed-off-by: Manuel Lauss Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 69bd0acc81c8..a1bbe16b7f96 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n"); + goto err; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) -- cgit v1.2.2 From 5baf831541c61546c00e8d6f294cb10ed5d25e7d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:13:42 +0000 Subject: ASoC: Fix variable shadowing warning in TLV320AIC3x Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/tlv320aic3x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5a8f53ce2250..e4b946a19ea3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -849,20 +849,20 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, * The term had to be converted to get * rid of the division by 10000; d = 0 here */ - int clk = (1000 * j * r) / p; + int tmp_clk = (1000 * j * r) / p; /* Check whether this values get closer than * the best ones we had before */ - if (abs(codec_clk - clk) < + if (abs(codec_clk - tmp_clk) < abs(codec_clk - last_clk)) { pll_j = j; pll_d = 0; pll_r = r; pll_p = p; - last_clk = clk; + last_clk = tmp_clk; } /* Early exit for exact matches */ - if (clk == codec_clk) + if (tmp_clk == codec_clk) goto found; } } -- cgit v1.2.2 From d11c5ab186310389b8e573be00279bab0a565d30 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:14:07 +0000 Subject: ASoC: Only restore non-default registers for WM8731 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8731.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 3a497810f939..5a2619dbf283 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -456,6 +456,9 @@ static int wm8731_resume(struct platform_device *pdev) /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { + if (cache[i] == wm8731_reg[i]) + continue; + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); -- cgit v1.2.2 From e0fb28e079b50f891b6c9db1c2bb25fef3268cf4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:14:23 +0000 Subject: ASoC: Only restore non-default registers for WM8776 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8776.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ab2c0da18091..44e7d9d82f87 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -406,6 +406,8 @@ static int wm8776_resume(struct platform_device *pdev) /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { + if (cache[i] == wm8776_reg[i]) + continue; data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); -- cgit v1.2.2 From 10505634bfa74871118a21eef8617acad00e4019 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:14:45 +0000 Subject: ASoC: Only restore non-default registers for WM8961 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8961.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index a8007d58813f..d2342c5e0425 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1022,6 +1022,9 @@ static int wm8961_resume(struct platform_device *pdev) int i; for (i = 0; i < codec->reg_cache_size; i++) { + if (reg_cache[i] == wm8961_reg_defaults[i]) + continue; + if (i == WM8961_SOFTWARE_RESET) continue; -- cgit v1.2.2 From 53242c68333570631a15a69842851b458eca3d99 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:15:56 +0000 Subject: ASoC: Implement suspend and resume for WM8993 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 67 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 5e32f2ed5fc2..cd2bc05f78cc 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -227,6 +227,7 @@ struct wm8993_priv { int class_w_users; unsigned int fll_fref; unsigned int fll_fout; + int fll_src; }; static unsigned int wm8993_read_hw(struct snd_soc_codec *codec, u8 reg) @@ -506,6 +507,7 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, wm8993->fll_fref = Fref; wm8993->fll_fout = Fout; + wm8993->fll_src = source; return 0; } @@ -1480,9 +1482,74 @@ static int wm8993_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int wm8993_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8993_priv *wm8993 = codec->private_data; + int fll_fout = wm8993->fll_fout; + int fll_fref = wm8993->fll_fref; + int ret; + + /* Stop the FLL in an orderly fashion */ + ret = wm8993_set_fll(codec->dai, 0, 0, 0, 0); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to stop FLL\n"); + return ret; + } + + wm8993->fll_fout = fll_fout; + wm8993->fll_fref = fll_fref; + + wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8993_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8993_priv *wm8993 = codec->private_data; + u16 *cache = wm8993->reg_cache; + int i, ret; + + /* Restore the register settings */ + for (i = 1; i < WM8993_MAX_REGISTER; i++) { + if (cache[i] == wm8993_reg_defaults[i]) + continue; + snd_soc_write(codec, i, cache[i]); + } + + wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restart the FLL? */ + if (wm8993->fll_fout) { + int fll_fout = wm8993->fll_fout; + int fll_fref = wm8993->fll_fref; + + wm8993->fll_fref = 0; + wm8993->fll_fout = 0; + + ret = wm8993_set_fll(codec->dai, 0, wm8993->fll_src, + fll_fref, fll_fout); + if (ret != 0) + dev_err(codec->dev, "Failed to restart FLL\n"); + } + + return 0; +} +#else +#define wm8993_suspend NULL +#define wm8993_resume NULL +#endif + struct snd_soc_codec_device soc_codec_dev_wm8993 = { .probe = wm8993_probe, .remove = wm8993_remove, + .suspend = wm8993_suspend, + .resume = wm8993_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8993); -- cgit v1.2.2 From 741b20cfb9109760937f403d18d731bfde31f56f Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 17 Dec 2009 17:34:39 +0100 Subject: ALSA: pcm_lib.c - convert second xrun_debug() parameter to use defines To increase code readability, convert send xrun_debug() argument to use defines. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_lib.c | 27 +++++++++++++++++---------- 1 file changed, 17 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f410832a25..9621236b2fef 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,15 +126,22 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#define XRUN_DEBUG_BASIC (1<<0) +#define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ +#define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ +#define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ +#define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ + #ifdef CONFIG_SND_PCM_XRUN_DEBUG -#define xrun_debug(substream, mask) ((substream)->pstr->xrun_debug & (mask)) +#define xrun_debug(substream, mask) \ + ((substream)->pstr->xrun_debug & (mask)) #else #define xrun_debug(substream, mask) 0 #endif -#define dump_stack_on_xrun(substream) do { \ - if (xrun_debug(substream, 2)) \ - dump_stack(); \ +#define dump_stack_on_xrun(substream) do { \ + if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ + dump_stack(); \ } while (0) static void pcm_debug_name(struct snd_pcm_substream *substream, @@ -154,7 +161,7 @@ static void xrun(struct snd_pcm_substream *substream) if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - if (xrun_debug(substream, 1)) { + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd(KERN_DEBUG "XRUN: %s\n", name); @@ -215,7 +222,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, #define hw_ptr_error(substream, fmt, args...) \ do { \ - if (xrun_debug(substream, 1)) { \ + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -237,7 +244,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - if (xrun_debug(substream, 8)) { + if (xrun_debug(substream, XRUN_DEBUG_PERIODUPDATE)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " @@ -290,7 +297,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, 4)) + if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) goto no_jiffies_check; /* Skip the jiffies check for hardwares with BATCH flag. @@ -369,7 +376,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - if (xrun_debug(substream, 16)) { + if (xrun_debug(substream, XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " @@ -403,7 +410,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, 4)) + if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) goto no_jiffies_check; if (delta < runtime->delay) goto no_jiffies_check; -- cgit v1.2.2 From 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 11:47:57 +0100 Subject: ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions In some debug cases, it might be usefull to see previous ring buffer positions to determine position problems from the lowlevel drivers. Signed-off-by: Jaroslav Kysela --- sound/core/pcm.c | 4 ++ sound/core/pcm_lib.c | 140 ++++++++++++++++++++++++++++++++++++++++++--------- 2 files changed, 121 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6884ae031f6f..df57a0e30bf2 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -921,6 +921,10 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) snd_free_pages((void*)runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + if (runtime->hwptr_log) + kfree(runtime->hwptr_log); +#endif kfree(runtime); substream->runtime = NULL; put_pid(substream->pid); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9621236b2fef..1990afb8a735 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,34 +126,34 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +static void pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len) +{ + snprintf(name, len, "pcmC%dD%d%c:%d", + substream->pcm->card->number, + substream->pcm->device, + substream->stream ? 'c' : 'p', + substream->number); +} + #define XRUN_DEBUG_BASIC (1<<0) #define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ #define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ #define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ #define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ +#define XRUN_DEBUG_LOG (1<<5) /* show last 10 positions on err */ +#define XRUN_DEBUG_LOGONCE (1<<6) /* do above only once */ #ifdef CONFIG_SND_PCM_XRUN_DEBUG + #define xrun_debug(substream, mask) \ ((substream)->pstr->xrun_debug & (mask)) -#else -#define xrun_debug(substream, mask) 0 -#endif #define dump_stack_on_xrun(substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ dump_stack(); \ } while (0) -static void pcm_debug_name(struct snd_pcm_substream *substream, - char *name, size_t len) -{ - snprintf(name, len, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} - static void xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -169,6 +169,102 @@ static void xrun(struct snd_pcm_substream *substream) } } +#define hw_ptr_error(substream, fmt, args...) \ + do { \ + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + if (printk_ratelimit()) { \ + snd_printd("PCM: " fmt, ##args); \ + } \ + dump_stack_on_xrun(substream); \ + } \ + } while (0) + +#define XRUN_LOG_CNT 10 + +struct hwptr_log_entry { + unsigned long jiffies; + snd_pcm_uframes_t pos; + snd_pcm_uframes_t period_size; + snd_pcm_uframes_t buffer_size; + snd_pcm_uframes_t old_hw_ptr; + snd_pcm_uframes_t hw_ptr_base; + snd_pcm_uframes_t hw_ptr_interrupt; +}; + +struct snd_pcm_hwptr_log { + unsigned int idx; + unsigned int hit: 1; + struct hwptr_log_entry entries[XRUN_LOG_CNT]; +}; + +static void xrun_log(struct snd_pcm_substream *substream, + snd_pcm_uframes_t pos) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hwptr_log *log = runtime->hwptr_log; + struct hwptr_log_entry *entry; + + if (log == NULL) { + log = kzalloc(sizeof(*log), GFP_ATOMIC); + if (log == NULL) + return; + runtime->hwptr_log = log; + } else { + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + } + entry = &log->entries[log->idx]; + entry->jiffies = jiffies; + entry->pos = pos; + entry->period_size = runtime->period_size; + entry->buffer_size = runtime->buffer_size;; + entry->old_hw_ptr = runtime->status->hw_ptr; + entry->hw_ptr_base = runtime->hw_ptr_base; + entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; + log->idx = (log->idx + 1) % XRUN_LOG_CNT; +} + +static void xrun_log_show(struct snd_pcm_substream *substream) +{ + struct snd_pcm_hwptr_log *log = substream->runtime->hwptr_log; + struct hwptr_log_entry *entry; + char name[16]; + unsigned int idx; + int cnt; + + if (log == NULL) + return; + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + pcm_debug_name(substream, name, sizeof(name)); + for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { + entry = &log->entries[idx]; + if (entry->period_size == 0) + break; + snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, entry->jiffies, (unsigned long)entry->pos, + (unsigned long)entry->period_size, + (unsigned long)entry->buffer_size, + (unsigned long)entry->old_hw_ptr, + (unsigned long)entry->hw_ptr_base, + (unsigned long)entry->hw_ptr_interrupt); + idx++; + idx %= XRUN_LOG_CNT; + } + log->hit = 1; +} + +#else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ + +#define xrun_debug(substream, mask) 0 +#define xrun(substream) do { } while (0) +#define hw_ptr_error(substream, fmt, args...) do { } while (0) +#define xrun_log(substream, pos) do { } while (0) +#define xrun_log_show(substream) do { } while (0) + +#endif + static snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) @@ -182,6 +278,7 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, if (printk_ratelimit()) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " "buffer size = 0x%lx, period size = 0x%lx\n", name, pos, runtime->buffer_size, @@ -190,6 +287,8 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, pos = 0; } pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); return pos; } @@ -220,16 +319,6 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -#define hw_ptr_error(substream, fmt, args...) \ - do { \ - if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ - if (printk_ratelimit()) { \ - snd_printd("PCM: " fmt, ##args); \ - } \ - dump_stack_on_xrun(substream); \ - } \ - } while (0) - static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -270,6 +359,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", @@ -315,6 +405,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! [Q] " "(pos=%ld, delta=%ld, period=%ld, " @@ -334,6 +425,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { + xrun_log_show(substream); hw_ptr_error(substream, "Lost interrupts? " "(stream=%i, delta=%ld, intr_ptr=%ld)\n", @@ -397,6 +489,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) if (delta < 0) { delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value [2] " "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", @@ -416,6 +509,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) goto no_jiffies_check; delta -= runtime->delay; if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! " "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", -- cgit v1.2.2 From f240406babfe1526998e10583ea5eccc2676a433 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 5 Jan 2010 17:19:34 +0100 Subject: ALSA: pcm_lib - cleanup & merge hw_ptr update functions Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them. The main change is hw_ptr_interrupt variable removal to simplify code logic. This variable can be computed directly from hw_ptr. Ensure that updated hw_ptr is not lower than previous one (it was possible with old code in some obscure situations when interrupt was delayed or the lowlevel driver returns wrong ring buffer position value). Signed-off-by: Jaroslav Kysela --- sound/core/oss/pcm_oss.c | 32 ++++-- sound/core/pcm_lib.c | 279 ++++++++++++++++------------------------------- sound/core/pcm_native.c | 2 - 3 files changed, 120 insertions(+), 193 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d9c96353121a..255ad910077a 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -632,6 +632,13 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes); } +static inline +snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) +{ + snd_pcm_uframes_t ptr = runtime->status->hw_ptr; + return ptr - (ptr % runtime->period_size); +} + /* define extended formats in the recent OSS versions (if any) */ /* linear formats */ #define AFMT_S32_LE 0x00001000 @@ -1102,7 +1109,7 @@ static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream) return err; } runtime->oss.prepare = 0; - runtime->oss.prev_hw_ptr_interrupt = 0; + runtime->oss.prev_hw_ptr_period = 0; runtime->oss.period_ptr = 0; runtime->oss.buffer_used = 0; @@ -1950,7 +1957,8 @@ static int snd_pcm_oss_get_caps(struct snd_pcm_oss_file *pcm_oss_file) return result; } -static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, snd_pcm_uframes_t hw_ptr) +static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, + snd_pcm_uframes_t hw_ptr) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t appl_ptr; @@ -1986,7 +1994,8 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr if (runtime->oss.trigger) goto _skip1; if (atomic_read(&psubstream->mmap_count)) - snd_pcm_oss_simulate_fill(psubstream, runtime->hw_ptr_interrupt); + snd_pcm_oss_simulate_fill(psubstream, + get_hw_ptr_period(runtime)); runtime->oss.trigger = 1; runtime->start_threshold = 1; cmd = SNDRV_PCM_IOCTL_START; @@ -2105,11 +2114,12 @@ static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream info.ptr = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr % runtime->buffer_size); if (atomic_read(&substream->mmap_count)) { snd_pcm_sframes_t n; - n = (delay = runtime->hw_ptr_interrupt) - runtime->oss.prev_hw_ptr_interrupt; + delay = get_hw_ptr_period(runtime); + n = delay - runtime->oss.prev_hw_ptr_period; if (n < 0) n += runtime->boundary; info.blocks = n / runtime->period_size; - runtime->oss.prev_hw_ptr_interrupt = delay; + runtime->oss.prev_hw_ptr_period = delay; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_pcm_oss_simulate_fill(substream, delay); info.bytes = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr) & INT_MAX; @@ -2673,18 +2683,22 @@ static int snd_pcm_oss_playback_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_playback_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_playback_avail(runtime) >= + runtime->oss.period_frames; } static int snd_pcm_oss_capture_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_capture_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_capture_avail(runtime) >= + runtime->oss.period_frames; } static unsigned int snd_pcm_oss_poll(struct file *file, poll_table * wait) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1990afb8a735..70a4f7428d78 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -172,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream) #define hw_ptr_error(substream, fmt, args...) \ do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + xrun_log_show(substream); \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -188,7 +189,6 @@ struct hwptr_log_entry { snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t old_hw_ptr; snd_pcm_uframes_t hw_ptr_base; - snd_pcm_uframes_t hw_ptr_interrupt; }; struct snd_pcm_hwptr_log { @@ -220,7 +220,6 @@ static void xrun_log(struct snd_pcm_substream *substream, entry->buffer_size = runtime->buffer_size;; entry->old_hw_ptr = runtime->status->hw_ptr; entry->hw_ptr_base = runtime->hw_ptr_base; - entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; log->idx = (log->idx + 1) % XRUN_LOG_CNT; } @@ -241,14 +240,13 @@ static void xrun_log_show(struct snd_pcm_substream *substream) entry = &log->entries[idx]; if (entry->period_size == 0) break; - snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + snd_printd("hwptr log: %s: j=%lu, pos=%ld/%ld/%ld, " + "hwptr=%ld/%ld\n", name, entry->jiffies, (unsigned long)entry->pos, (unsigned long)entry->period_size, (unsigned long)entry->buffer_size, (unsigned long)entry->old_hw_ptr, - (unsigned long)entry->hw_ptr_base, - (unsigned long)entry->hw_ptr_interrupt); + (unsigned long)entry->hw_ptr_base); idx++; idx %= XRUN_LOG_CNT; } @@ -265,33 +263,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static snd_pcm_uframes_t -snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) -{ - snd_pcm_uframes_t pos; - - pos = substream->ops->pointer(substream); - if (pos == SNDRV_PCM_POS_XRUN) - return pos; /* XRUN */ - if (pos >= runtime->buffer_size) { - if (printk_ratelimit()) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - xrun_log_show(substream); - snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " - "buffer size = 0x%lx, period size = 0x%lx\n", - name, pos, runtime->buffer_size, - runtime->period_size); - } - pos = 0; - } - pos -= pos % runtime->min_align; - if (xrun_debug(substream, XRUN_DEBUG_LOG)) - xrun_log(substream, pos); - return pos; -} - static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { @@ -319,72 +290,88 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) +static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, + unsigned int in_interrupt) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_ptr_interrupt, hw_base; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t hdelta, delta; unsigned long jdelta; old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); + pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; } - if (xrun_debug(substream, XRUN_DEBUG_PERIODUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); + if (pos >= runtime->buffer_size) { + if (printk_ratelimit()) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); + snd_printd(KERN_ERR "BUG: %s, pos = %ld, " + "buffer size = %ld, period size = %ld\n", + name, pos, runtime->buffer_size, + runtime->period_size); + } + pos = 0; } + pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; - hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - delta = new_hw_ptr - hw_ptr_interrupt; - if (hw_ptr_interrupt >= runtime->boundary) { - hw_ptr_interrupt -= runtime->boundary; - if (hw_base < runtime->boundary / 2) - /* hw_base was already lapped; recalc delta */ - delta = new_hw_ptr - hw_ptr_interrupt; - } - if (delta < 0) { - if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value " - "(stream=%i, pos=%ld, intr_ptr=%ld)\n", - substream->stream, (long)pos, - (long)hw_ptr_interrupt); -#if 1 - /* simply skipping the hwptr update seems more - * robust in some cases, e.g. on VMware with - * inaccurate timer source - */ - return 0; /* skip this update */ -#else - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; -#endif - } else { + if (in_interrupt) { + /* we know that one period was processed */ + /* delta = "expected next hw_ptr" for in_interrupt != 0 */ + delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) + + runtime->period_size; + if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; + goto __delta; } } + /* new_hw_ptr might be lower than old_hw_ptr in case when */ + /* pointer crosses the end of the ring buffer */ + if (new_hw_ptr < old_hw_ptr) { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + __delta: + delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary; + if (xrun_debug(substream, in_interrupt ? + XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("%s_update: %s: pos=%u/%u/%u, " + "hwptr=%ld/%ld/%ld/%ld\n", + in_interrupt ? "period" : "hwptr", + name, + (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr, + (unsigned long)runtime->hw_ptr_base); + } + /* something must be really wrong */ + if (delta >= runtime->buffer_size) { + hw_ptr_error(substream, + "Unexpected hw_pointer value %s" + "(stream=%i, pos=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "[P]", + substream->stream, (long)pos, + (long)new_hw_ptr, (long)old_hw_ptr); + return 0; + } /* Do jiffies check only in xrun_debug mode */ if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) @@ -396,7 +383,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) */ if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; - hdelta = new_hw_ptr - old_hw_ptr; + hdelta = delta; if (hdelta < runtime->delay) goto no_jiffies_check; hdelta -= runtime->delay; @@ -405,45 +392,49 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); - xrun_log_show(substream); + /* move new_hw_ptr according jiffies not pos variable */ + new_hw_ptr = old_hw_ptr; + /* use loop to avoid checks for delta overflows */ + /* the delta value is small or zero in most cases */ + while (delta > 0) { + new_hw_ptr += runtime->period_size; + if (new_hw_ptr >= runtime->boundary) + new_hw_ptr -= runtime->boundary; + delta--; + } + /* align hw_base to buffer_size */ + hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); + delta = 0; hw_ptr_error(substream, - "hw_ptr skipping! [Q] " + "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " - "jdelta=%lu/%lu/%lu)\n", + "jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", + in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, - ((hdelta * HZ) / runtime->rate), delta); - hw_ptr_interrupt = runtime->hw_ptr_interrupt + - runtime->period_size * delta; - if (hw_ptr_interrupt >= runtime->boundary) - hw_ptr_interrupt -= runtime->boundary; - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; + ((hdelta * HZ) / runtime->rate), delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr); } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { - xrun_log_show(substream); hw_ptr_error(substream, - "Lost interrupts? " - "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + "Lost interrupts? %s" + "(stream=%i, delta=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "", substream->stream, (long)delta, - (long)hw_ptr_interrupt); - /* rebase hw_ptr_interrupt */ - hw_ptr_interrupt = - new_hw_ptr - new_hw_ptr % runtime->period_size; + (long)new_hw_ptr, + (long)old_hw_ptr); } - runtime->hw_ptr_interrupt = hw_ptr_interrupt; + + if (runtime->status->hw_ptr == new_hw_ptr) + return 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; @@ -456,83 +447,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) /* CAUTION: call it with irq disabled */ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; - snd_pcm_sframes_t delta; - unsigned long jdelta; - - old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); - if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); - return -EPIPE; - } - if (xrun_debug(substream, XRUN_DEBUG_HWPTRUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); - } - - hw_base = runtime->hw_ptr_base; - new_hw_ptr = hw_base + pos; - - delta = new_hw_ptr - old_hw_ptr; - jdelta = jiffies - runtime->hw_ptr_jiffies; - if (delta < 0) { - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value [2] " - "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", - substream->stream, (long)pos, - (long)old_hw_ptr, jdelta); - return 0; - } - hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) - hw_base = 0; - new_hw_ptr = hw_base + pos; - } - /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) - goto no_jiffies_check; - if (delta < runtime->delay) - goto no_jiffies_check; - delta -= runtime->delay; - if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { - xrun_log_show(substream); - hw_ptr_error(substream, - "hw_ptr skipping! " - "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", - (long)pos, (long)delta, - (long)runtime->period_size, jdelta, - ((delta * HZ) / runtime->rate)); - return 0; - } - no_jiffies_check: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, new_hw_ptr); - - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - - runtime->hw_ptr_base = hw_base; - runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_jiffies = jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) - snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_hw_ptr0(substream, 0); } /** @@ -1744,7 +1659,7 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || - snd_pcm_update_hw_ptr_interrupt(substream) < 0) + snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; if (substream->timer_running) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 29ab46a12e11..8e777f71717c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1247,8 +1247,6 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; - runtime->hw_ptr_interrupt = runtime->status->hw_ptr - - runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v1.2.2 From 1250932e48d3b698415b1f04775433cf1da688d6 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 7 Jan 2010 15:36:31 +0100 Subject: ALSA: pcm_lib - optimize wake_up() calls for PCM I/O As noted by pl bossart , the PCM I/O routines (snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls until all samples are not processed. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_lib.c | 30 ++++++++++++++++++++---------- sound/core/pcm_native.c | 6 ++++-- 2 files changed, 24 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 70a4f7428d78..a63226232ef4 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -263,8 +263,8 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t avail; @@ -285,7 +285,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) + if (!runtime->nowake && avail >= runtime->control->avail_min) wake_up(&runtime->sleep); return 0; } @@ -441,7 +441,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_state(substream, runtime); } /* CAUTION: call it with irq disabled */ @@ -1792,6 +1792,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1813,15 +1814,17 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -1850,8 +1853,10 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } @@ -2009,6 +2014,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2037,15 +2043,17 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -2068,8 +2076,10 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8e777f71717c..27284f628361 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -516,6 +516,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, struct snd_pcm_sw_params *params) { struct snd_pcm_runtime *runtime; + int err; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; @@ -540,6 +541,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (params->silence_threshold > runtime->buffer_size) return -EINVAL; } + err = 0; snd_pcm_stream_lock_irq(substream); runtime->tstamp_mode = params->tstamp_mode; runtime->period_step = params->period_step; @@ -553,10 +555,10 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - wake_up(&runtime->sleep); + err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); - return 0; + return err; } static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream, -- cgit v1.2.2 From 7b3a177b0d4f92b3431b8dca777313a07533a710 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 8 Jan 2010 08:43:01 +0100 Subject: ALSA: pcm_lib: fix "something must be really wrong" condition When runtime->periods == 1 or when pointer crosses end of ring buffer, the delta might be greater than buffer_size. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a63226232ef4..c7b35b20e659 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -362,7 +362,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (unsigned long)runtime->hw_ptr_base); } /* something must be really wrong */ - if (delta >= runtime->buffer_size) { + if (delta >= runtime->buffer_size + runtime->period_size) { hw_ptr_error(substream, "Unexpected hw_pointer value %s" "(stream=%i, pos=%ld, new_hw_ptr=%ld, " -- cgit v1.2.2 From dd3533eca859a6debb1565503ec03e68354e08e0 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 1 Jan 2010 19:05:43 +0100 Subject: ALSA: ac97_codec: merge WM9703 and WM9705 ops The WM9705 and WM9703 ops are the same actually so use the same code for both. Signed-off-by: Krzysztof Helt Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 139cf3b2b9d7..e288a5595f34 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -544,25 +544,10 @@ static int patch_wolfson04(struct snd_ac97 * ac97) return 0; } -static int patch_wolfson_wm9705_specific(struct snd_ac97 * ac97) -{ - int err, i; - for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0) - return err; - } - snd_ac97_write_cache(ac97, 0x72, 0x0808); - return 0; -} - -static struct snd_ac97_build_ops patch_wolfson_wm9705_ops = { - .build_specific = patch_wolfson_wm9705_specific, -}; - static int patch_wolfson05(struct snd_ac97 * ac97) { /* WM9705, WM9710 */ - ac97->build_ops = &patch_wolfson_wm9705_ops; + ac97->build_ops = &patch_wolfson_wm9703_ops; #ifdef CONFIG_TOUCHSCREEN_WM9705 /* WM9705 touchscreen uses AUX and VIDEO for touch */ ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; -- cgit v1.2.2 From cd9d95a55550555da8e587ead9cbba5f98a371a3 Mon Sep 17 00:00:00 2001 From: Ken Prox Date: Fri, 8 Jan 2010 09:01:47 +0100 Subject: ALSA: hda - conexant - Fixed microphone mixer for HP Compaq Presario F700 Added patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea. Signed-off-by: Ken Prox Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 50 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1ab2958a290b..b20c640f7502 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1720,6 +1720,22 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_f700_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1810,6 +1826,32 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_f700_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1829,6 +1871,7 @@ enum { CXT5051_HP, /* no docking */ CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ + CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_MODELS }; @@ -1837,6 +1880,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", + [CXT5051_F700] = "hp 700" }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1846,6 +1890,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), {} }; @@ -1896,6 +1941,11 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; break; + case CXT5051_F700: + spec->init_verbs[0] = cxt5051_f700_init_verbs; + spec->mixers[0] = cxt5051_f700_mixers; + spec->no_auto_mic = 1; + break; } return 0; -- cgit v1.2.2 From 75f8991d0e6969407d51501d5a0537f104075c99 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 7 Jan 2010 13:46:25 +0100 Subject: ALSA: hda - Configure XO-1.5 microphones at capture time The XO-1.5 has a microphone LED designed to indicate to the user when something is being recorded. This light is controlled by the microphone bias voltage and it is currently coming on all the time. This patch defers the microphone port configuration until when recording is actually taking place, fixing the behaviour of the LED. Signed-off-by: Daniel Drake Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 125 ++++++++++++++++++++++++++++------------- 1 file changed, 85 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 01e46ba72690..3521f33d43c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -111,8 +111,12 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; - unsigned char ext_mic_bias; unsigned int dell_vostro; + + unsigned int ext_mic_present; + unsigned int recording; + void (*capture_prepare)(struct hda_codec *codec); + void (*capture_cleanup)(struct hda_codec *codec); }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -185,6 +189,8 @@ static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; + if (spec->capture_prepare) + spec->capture_prepare(codec); snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], stream_tag, 0, format); return 0; @@ -196,6 +202,8 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct conexant_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + if (spec->capture_cleanup) + spec->capture_cleanup(codec); return 0; } @@ -2016,53 +2024,53 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* toggle input of built-in and mic jack appropriately */ -static void cxt5066_automic(struct hda_codec *codec) +/* OLPC defers mic widget control until when capture is started because the + * microphone LED comes on as soon as these settings are put in place. if we + * did this before recording, it would give the false indication that recording + * is happening when it is not. */ +static void cxt5066_olpc_select_mic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct hda_verb ext_mic_present[] = { - /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, - - /* switch to external mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + if (!spec->recording) + return; - /* disable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static struct hda_verb ext_mic_absent[] = { - /* enable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* external mic, port B */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); - /* switch to internal mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + /* internal mic, port C */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? 0 : PIN_VREF80); +} - /* disable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_olpc_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; unsigned int present; - present = snd_hda_jack_detect(codec, 0x1a); - if (present) { + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) snd_printdd("CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { + else snd_printdd("CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } + + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 1); + spec->ext_mic_present = !!present; + + cxt5066_olpc_select_mic(codec); } /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_vostro_automic(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; unsigned int present; struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2113,7 +2121,7 @@ static void cxt5066_hp_automute(struct hda_codec *codec) } /* unsolicited event for jack sensing */ -static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { @@ -2121,7 +2129,7 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_automic(codec); + cxt5066_olpc_automic(codec); break; } } @@ -2197,6 +2205,31 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, return 1; } +static void cxt5066_olpc_capture_prepare(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* mark as recording and configure the microphone widget so that the + * recording LED comes on. */ + spec->recording = 1; + cxt5066_olpc_select_mic(codec); +} + +static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + const struct hda_verb disable_mics[] = { + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {}, + }; + + snd_hda_sequence_write(codec, disable_mics); + spec->recording = 0; +} + static struct hda_input_mux cxt5066_capture_source = { .num_items = 4, .items = { @@ -2347,10 +2380,10 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port C: internal microphone */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port D: unused */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, @@ -2479,12 +2512,19 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_hp_automute(codec); if (spec->dell_vostro) cxt5066_vostro_automic(codec); - else - cxt5066_automic(codec); } return 0; } +static int cxt5066_olpc_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + cxt5066_hp_automute(codec); + cxt5066_olpc_automic(codec); + return 0; +} + enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ @@ -2521,7 +2561,7 @@ static int patch_cxt5066(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = cxt5066_init; + codec->patch_ops.init = conexant_init; spec->dell_automute = 0; spec->multiout.max_channels = 2; @@ -2534,7 +2574,6 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; - spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2561,20 +2600,26 @@ static int patch_cxt5066(struct hda_codec *codec) spec->dell_automute = 1; break; case CXT5066_OLPC_XO_1_5: - codec->patch_ops.unsol_event = cxt5066_unsol_event; + codec->patch_ops.init = cxt5066_olpc_init; + codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; - spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; /* input source automatically selected */ spec->input_mux = NULL; + + /* our capture hooks which allow us to turn on the microphone LED + * at the right time */ + spec->capture_prepare = cxt5066_olpc_capture_prepare; + spec->capture_cleanup = cxt5066_olpc_capture_cleanup; break; case CXT5066_DELL_VOSTO: + codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_vostro_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; -- cgit v1.2.2 From c4cfe66c4c2d5a91b3734ffb4e2bad0badd5c874 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 7 Jan 2010 13:47:04 +0100 Subject: ALSA: hda - support OLPC XO-1.5 DC input The XO's audio hardware is wired up to allow DC sensors (e.g. light sensors, thermistors, etc) to be plugged in through the microphone jack. Add sound mixer controls to allow this mode to be enabled and tweaked. Signed-off-by: Daniel Drake Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 213 ++++++++++++++++++++++++++++++++++++----- 1 file changed, 190 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3521f33d43c3..685015a53292 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -117,6 +117,16 @@ struct conexant_spec { unsigned int recording; void (*capture_prepare)(struct hda_codec *codec); void (*capture_cleanup)(struct hda_codec *codec); + + /* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors) + * through the microphone jack. + * When the user enables this through a mixer switch, both internal and + * external microphones are disabled. Gain is fixed at 0dB. In this mode, + * we also allow the bias to be configured through a separate mixer + * control. */ + unsigned int dc_enable; + unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */ + unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -2024,6 +2034,26 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } +static const struct hda_input_mux cxt5066_olpc_dc_bias = { + .num_items = 3, + .items = { + { "Off", PIN_IN }, + { "50%", PIN_VREF50 }, + { "80%", PIN_VREF80 }, + }, +}; + +static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* Even though port F is the DC input, the bias is controlled on port B. + * we also leave that port as an active input (but unselected) in DC mode + * just in case that is necessary to make the bias setting take effect. */ + return snd_hda_codec_write_cache(codec, 0x1a, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index); +} + /* OLPC defers mic widget control until when capture is started because the * microphone LED comes on as soon as these settings are put in place. if we * did this before recording, it would give the false indication that recording @@ -2034,6 +2064,27 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec) if (!spec->recording) return; + if (spec->dc_enable) { + /* in DC mode we ignore presence detection and just use the jack + * through our special DC port */ + const struct hda_verb enable_dc_mode[] = { + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {}, + }; + + snd_hda_sequence_write(codec, enable_dc_mode); + /* port B input disabled (and bias set) through the following call */ + cxt5066_set_olpc_dc_bias(codec); + return; + } + + /* disable DC (port F) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + /* external mic, port B */ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); @@ -2049,6 +2100,9 @@ static void cxt5066_olpc_automic(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->dc_enable) /* don't do presence detection in DC mode */ + return; + present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; if (present) @@ -2123,13 +2177,16 @@ static void cxt5066_hp_automute(struct hda_codec *codec) /* unsolicited event for jack sensing */ static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_olpc_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } @@ -2159,6 +2216,15 @@ static const struct hda_input_mux cxt5066_analog_mic_boost = { }, }; +static int cxt5066_set_mic_boost(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + return snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + cxt5066_analog_mic_boost.items[spec->mic_boost].index); +} + static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2169,15 +2235,8 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int val; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT; - - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, inout); - - ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->mic_boost; return 0; } @@ -2185,23 +2244,101 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; unsigned int idx; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + spec->mic_boost = idx; + if (!spec->dc_enable) + cxt5066_set_mic_boost(codec); + return 1; +} + +static void cxt5066_enable_dc(struct hda_codec *codec) +{ + const struct hda_verb enable_dc_mode[] = { + /* disable gain */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* switch to DC input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 3}, + {} + }; + + /* configure as input source */ + snd_hda_sequence_write(codec, enable_dc_mode); + cxt5066_olpc_select_mic(codec); /* also sets configured bias */ +} + +static void cxt5066_disable_dc(struct hda_codec *codec) +{ + /* reconfigure input source */ + cxt5066_set_mic_boost(codec); + /* automic also selects the right mic if we're recording */ + cxt5066_olpc_automic(codec); +} + +static int cxt5066_olpc_dc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = spec->dc_enable; + return 0; +} - if (!imux->num_items) +static int cxt5066_olpc_dc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + int dc_enable = !!ucontrol->value.integer.value[0]; + + if (dc_enable == spec->dc_enable) return 0; + + spec->dc_enable = dc_enable; + if (dc_enable) + cxt5066_enable_dc(codec); + else + cxt5066_disable_dc(codec); + + return 1; +} + +static int cxt5066_olpc_dc_bias_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_olpc_dc_bias, uinfo); +} + +static int cxt5066_olpc_dc_bias_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->dc_input_bias; + return 0; +} + +static int cxt5066_olpc_dc_bias_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout | - imux->items[idx].index); - + spec->dc_input_bias = idx; + if (spec->dc_enable) + cxt5066_set_olpc_dc_bias(codec); return 1; } @@ -2223,6 +2360,9 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) /* disble internal mic, port C */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {}, }; @@ -2282,6 +2422,24 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { {} }; +static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Mode Enable Switch", + .info = snd_ctl_boolean_mono_info, + .get = cxt5066_olpc_dc_get, + .put = cxt5066_olpc_dc_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Input Bias Enum", + .info = cxt5066_olpc_dc_bias_enum_info, + .get = cxt5066_olpc_dc_bias_enum_get, + .put = cxt5066_olpc_dc_bias_enum_put, + }, + {} +}; + static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2294,11 +2452,10 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Ext Mic Boost Capture Enum", + .name = "Analog Mic Boost Capture Enum", .info = cxt5066_mic_boost_mux_enum_info, .get = cxt5066_mic_boost_mux_enum_get, .put = cxt5066_mic_boost_mux_enum_put, - .private_value = 0x17, }, HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), @@ -2392,7 +2549,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - /* Port F: unused */ + /* Port F: external DC input through microphone port */ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port G: internal speakers */ @@ -2513,15 +2670,22 @@ static int cxt5066_init(struct hda_codec *codec) if (spec->dell_vostro) cxt5066_vostro_automic(codec); } + cxt5066_set_mic_boost(codec); return 0; } static int cxt5066_olpc_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: init\n"); conexant_init(codec); cxt5066_hp_automute(codec); - cxt5066_olpc_automic(codec); + if (!spec->dc_enable) { + cxt5066_set_mic_boost(codec); + cxt5066_olpc_automic(codec); + } else { + cxt5066_enable_dc(codec); + } return 0; } @@ -2604,8 +2768,10 @@ static int patch_cxt5066(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_olpc_dc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; @@ -2627,6 +2793,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; spec->dell_vostro = 1; + spec->mic_boost = 3; /* default 30dB gain */ snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ -- cgit v1.2.2 From 444c1953d496d272208902ff7010dc70d1f887f0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 3 Jan 2010 12:39:27 +0100 Subject: sound: oss: off by one bug The problem is that in the original code sound_nblocks could go up to 1024 which would be an array overflow. This was found with a static checker and has been compile tested only. Signed-off-by: Dan Carpenter Signed-off-by: Jaroslav Kysela --- sound/oss/dev_table.c | 16 +++++++++------- sound/oss/sound_config.h | 2 ++ sound/oss/soundcard.c | 4 ++-- 3 files changed, 13 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index 08274c995d06..727bdb9ba2dc 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -67,14 +67,15 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, return -(EBUSY); } d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver))); - - if (sound_nblocks < 1024) - sound_nblocks++; + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations))); + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; - if (sound_nblocks < 1024) - sound_nblocks++; if (d == NULL || op == NULL) { printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name); sound_unload_audiodev(num); @@ -128,9 +129,10 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver, until you unload sound! */ op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations))); + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; - if (sound_nblocks < 1024) - sound_nblocks++; if (op == NULL) { printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name); return -ENOMEM; diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 55271fbe7f49..9d35c4c65b9b 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -142,4 +142,6 @@ static inline int translate_mode(struct file *file) #define TIMER_ARMED 121234 #define TIMER_NOT_ARMED 1 +#define MAX_MEM_BLOCKS 1024 + #endif diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 61aaedae6b7e..c62530943888 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -56,7 +56,7 @@ /* * Table for permanently allocated memory (used when unloading the module) */ -void * sound_mem_blocks[1024]; +void * sound_mem_blocks[MAX_MEM_BLOCKS]; int sound_nblocks = 0; /* Persistent DMA buffers */ @@ -574,7 +574,7 @@ static int __init oss_init(void) NULL, "%s%d", dev_list[i].name, j); } - if (sound_nblocks >= 1024) + if (sound_nblocks >= MAX_MEM_BLOCKS - 1) printk(KERN_ERR "Sound warning: Deallocation table was too small.\n"); return 0; -- cgit v1.2.2 From edf12b4af6e1d2b7c42c75ff00e55a9c52c06d70 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 4 Jan 2010 22:23:34 +0100 Subject: sbawe: fix memory detection part 2 The patch "sbawe: fix memory detection" fixed detection for memoryless SB32 cards but broke detection of memory above 512KB. This patch fixes the regression. The patch has been tested on the SB32 card (CT3670) with 0MB, 2MB and 8MB memory installed. Signed-off-by: Krzysztof Helt Signed-off-by: Jaroslav Kysela --- sound/isa/sb/emu8000.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 751762f1c59a..0c40951b6523 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -377,12 +377,13 @@ init_arrays(struct snd_emu8000 *emu) static void __devinit size_dram(struct snd_emu8000 *emu) { - int i, size; + int i, size, detected_size; if (emu->dram_checked) return; size = 0; + detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -393,6 +394,8 @@ size_dram(struct snd_emu8000 *emu) while (size < EMU8000_MAX_DRAM) { + size += 512 * 1024; /* increment 512kbytes */ + /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -414,7 +417,7 @@ size_dram(struct snd_emu8000 *emu) if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) break; /* no memory at this address */ - size += 512 * 1024; /* increment 512kbytes */ + detected_size = size; snd_emu8000_read_wait(emu); @@ -442,9 +445,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, size/1024); + emu->port1, detected_size/1024); - emu->mem_size = size; + emu->mem_size = detected_size; emu->dram_checked = 1; } -- cgit v1.2.2 From 5ee518ecbcb5934e284ea51a19a939c891f5f7ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 7 Jan 2010 16:29:20 +0000 Subject: ASoC: Fix WM8350 DSP mode B configuration We need to set the LRCLK inversion bit to select DSP mode. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8350.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index ebbf11b653a4..718ef912e758 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) iface |= 0x3 << 8; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x3 << 8; /* lg not sure which mode */ + iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV; break; default: return -EINVAL; -- cgit v1.2.2 From af9a75dd1a1f8a9aa406466cc8bb16208120488a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sat, 9 Jan 2010 01:22:29 -0500 Subject: ALSA: ac97: Add Dell Dimension 2400 to Headphone/Line Jack Sense blacklist This model needs both 'Headphone Jack Sense' and 'Line Jack Sense' muted for audible playback, so just add it to the ad1981 jack sense blacklist. Cc: stable@kernel.org Tested-by: Pete Signed-off-by: Daniel T Chen Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 139cf3b2b9d7..d9266bae2849 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1870,6 +1870,7 @@ static unsigned int ad1981_jacks_blacklist[] = { 0x10140554, /* Thinkpad T42p/R50p */ 0x10140567, /* Thinkpad T43p 2668-G7U */ 0x10140581, /* Thinkpad X41-2527 */ + 0x10280160, /* Dell Dimension 2400 */ 0x104380b0, /* Asus A7V8X-MX */ 0x11790241, /* Toshiba Satellite A-15 S127 */ 0x144dc01a, /* Samsung NP-X20C004/SEG */ -- cgit v1.2.2 From c68db7175f4dcb3d5789bb50bea6376fb81f87fe Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 10 Jan 2010 17:21:14 +0100 Subject: ALSA: ac97: add AC97 STMicroelectronics' codecs Add the STMicroelectronics ST7597 codec and an unknown codec from the same manufacturer found on the Creative SB 128 card (CT4810). Signed-off-by: Krzysztof Helt Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 10 ++++++++++ sound/pci/ac97/ac97_id.h | 2 ++ 2 files changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index c11920623009..a7630e9edf8a 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -83,6 +83,7 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { { 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL }, { 0x50534300, 0xffffff00, "Philips", NULL, NULL }, { 0x53494c00, 0xffffff00, "Silicon Laboratory", NULL, NULL }, +{ 0x53544d00, 0xffffff00, "STMicroelectronics", NULL, NULL }, { 0x54524100, 0xffffff00, "TriTech", NULL, NULL }, { 0x54584e00, 0xffffff00, "Texas Instruments", NULL, NULL }, { 0x56494100, 0xffffff00, "VIA Technologies", NULL, NULL }, @@ -161,6 +162,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, +{ 0x53544d02, 0xffffffff, "ST7597", NULL, NULL }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, { 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, @@ -213,6 +215,14 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) { /* filter some registers for buggy codecs */ switch (ac97->id) { + case AC97_ID_ST_AC97_ID4: + if (reg == 0x08) + return 0; + /* fall through */ + case AC97_ID_ST7597: + if (reg == 0x22 || reg == 0x7a) + return 1; + /* fall through */ case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h index c129492c82b3..d603147c4a96 100644 --- a/sound/pci/ac97/ac97_id.h +++ b/sound/pci/ac97/ac97_id.h @@ -62,3 +62,5 @@ #define AC97_ID_CM9761_78 0x434d4978 #define AC97_ID_CM9761_82 0x434d4982 #define AC97_ID_CM9761_83 0x434d4983 +#define AC97_ID_ST7597 0x53544d02 +#define AC97_ID_ST_AC97_ID4 0x53544d04 -- cgit v1.2.2 From 2138301e1687bd4f22aa2b4df4829b6ffdae19bc Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Fri, 8 Jan 2010 17:48:31 +0200 Subject: ASoC: tpa6130a2: Support for tpa6140's regulators tpa6140a2 uses different names for the regulators. Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8e98ccfab75c..8b27281e62a1 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -41,6 +41,11 @@ static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { "Vdd", }; +static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { + "HPVdd", + "AVdd", +}; + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; @@ -420,8 +425,21 @@ static int tpa6130a2_probe(struct i2c_client *client, gpio_direction_output(data->power_gpio, 0); } - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + switch (pdata->id) { + case TPA6130A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + break; + case TPA6140A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6140a2_supply_names[i];; + break; + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + } ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), data->supplies); -- cgit v1.2.2 From 6b98515a620592636d2f8e0d3e2942d1cb4847ec Mon Sep 17 00:00:00 2001 From: Alan Cox Date: Mon, 4 Jan 2010 16:22:59 +0000 Subject: sound_oss: remove use of old BKL ioctl path Signed-off-by: Alan Cox Signed-off-by: Takashi Iwai --- sound/oss/soundcard.c | 35 ++++++++++++++++++++++------------- 1 file changed, 22 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 61aaedae6b7e..6c3267bf05d0 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -328,11 +328,11 @@ static int sound_mixer_ioctl(int mixdev, unsigned int cmd, void __user *arg) return mixer_devs[mixdev]->ioctl(mixdev, cmd, arg); } -static int sound_ioctl(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) +static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { int len = 0, dtype; - int dev = iminor(inode); + int dev = iminor(file->f_dentry->d_inode); + long ret = -EINVAL; void __user *p = (void __user *)arg; if (_SIOC_DIR(cmd) != _SIOC_NONE && _SIOC_DIR(cmd) != 0) { @@ -353,6 +353,7 @@ static int sound_ioctl(struct inode *inode, struct file *file, if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); + lock_kernel(); if (_IOC_TYPE(cmd) == 'M' && num_mixers > 0 && /* Mixer ioctl */ (dev & 0x0f) != SND_DEV_CTL) { dtype = dev & 0x0f; @@ -360,24 +361,31 @@ static int sound_ioctl(struct inode *inode, struct file *file, case SND_DEV_DSP: case SND_DEV_DSP16: case SND_DEV_AUDIO: - return sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev, + ret = sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev, cmd, p); - + break; default: - return sound_mixer_ioctl(dev >> 4, cmd, p); + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; } + unlock_kernel(); + return ret; } + switch (dev & 0x0f) { case SND_DEV_CTL: if (cmd == SOUND_MIXER_GETLEVELS) - return get_mixer_levels(p); - if (cmd == SOUND_MIXER_SETLEVELS) - return set_mixer_levels(p); - return sound_mixer_ioctl(dev >> 4, cmd, p); + ret = get_mixer_levels(p); + else if (cmd == SOUND_MIXER_SETLEVELS) + ret = set_mixer_levels(p); + else + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; case SND_DEV_SEQ: case SND_DEV_SEQ2: - return sequencer_ioctl(dev, file, cmd, p); + ret = sequencer_ioctl(dev, file, cmd, p); + break; case SND_DEV_DSP: case SND_DEV_DSP16: @@ -390,7 +398,8 @@ static int sound_ioctl(struct inode *inode, struct file *file, break; } - return -EINVAL; + unlock_kernel(); + return ret; } static unsigned int sound_poll(struct file *file, poll_table * wait) @@ -490,7 +499,7 @@ const struct file_operations oss_sound_fops = { .read = sound_read, .write = sound_write, .poll = sound_poll, - .ioctl = sound_ioctl, + .unlocked_ioctl = sound_ioctl, .mmap = sound_mmap, .open = sound_open, .release = sound_release, -- cgit v1.2.2 From 9c0afc861a7228f718cb6a79fa7f9d46bf9ff300 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Jan 2010 14:00:11 +0100 Subject: ALSA: hda - Fix ALC861-VD capture source mixer The capture source or input source mixer element wasn't created properly for ALC861-VD codec due to the wrong NID passed to alc_auto_create_input_ctls(). References: Novell bnc#568305 http://bugzilla.novell.com/show_bug.cgi?id=568305 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7465053d6bb..e3caa78ccd54 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -15493,7 +15493,7 @@ static struct alc_config_preset alc861vd_presets[] = { static int alc861vd_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0); + return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0); } -- cgit v1.2.2 From 03e7a35c0ef7a462385fb6a301dfc1b287cac6de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Jan 2010 14:01:19 +0000 Subject: Revert "ASoC: ad1836: reset and restore clock control mode in suspend/resume entry" This reverts commit afe1c2cd71eb4e0fade720b5709722e7124f29c0 since it doesn't build. --- sound/soc/codecs/ad1836.c | 32 -------------------------------- 1 file changed, 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 83add2f3afba..2c18e3d1b71e 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,36 +223,6 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } -#ifdef CONFIG_PM -static int ad1836_soc_suspend(struct platform_device *pdev, - pm_message_t state) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - /* reset clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} - -static int ad1836_soc_resume(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - /* restore clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 |= AD1836_ADC_AUX; - - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} -#else -#define ad1836_soc_suspend NULL -#define ad1836_soc_resume NULL -#endif - static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -434,8 +404,6 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, - .suspend = ad1836_soc_suspend, - .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); -- cgit v1.2.2 From 735fe4cfbc3cedea41bd0ed31955054dae6beb46 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Jan 2010 14:13:00 +0000 Subject: ASoC: Add missing __devexit and __devinit annotations Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 6 +++--- sound/soc/codecs/tlv320dac33.c | 6 +++--- sound/soc/codecs/tpa6130a2.c | 6 +++--- 3 files changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index fbf3ab482015..cf2975a7294a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -471,8 +471,8 @@ init_err: } #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static int da7210_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int __devinit da7210_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct da7210_priv *da7210; struct snd_soc_codec *codec; @@ -495,7 +495,7 @@ static int da7210_i2c_probe(struct i2c_client *i2c, return ret; } -static int da7210_i2c_remove(struct i2c_client *client) +static int __devexit da7210_i2c_remove(struct i2c_client *client) { struct da7210_priv *da7210 = i2c_get_clientdata(client); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 3ef3255cd1e7..2df9c20b7d52 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1191,8 +1191,8 @@ struct snd_soc_dai dac33_dai = { }; EXPORT_SYMBOL_GPL(dac33_dai); -static int dac33_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) +static int __devinit dac33_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) { struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; @@ -1345,7 +1345,7 @@ error_reg: return ret; } -static int dac33_i2c_remove(struct i2c_client *client) +static int __devexit dac33_i2c_remove(struct i2c_client *client) { struct tlv320dac33_priv *dac33; diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8b27281e62a1..958d49c969ac 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -379,8 +379,8 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); -static int tpa6130a2_probe(struct i2c_client *client, - const struct i2c_device_id *id) +static int __devinit tpa6130a2_probe(struct i2c_client *client, + const struct i2c_device_id *id) { struct device *dev; struct tpa6130a2_data *data; @@ -479,7 +479,7 @@ err_gpio: return ret; } -static int tpa6130a2_remove(struct i2c_client *client) +static int __devexit tpa6130a2_remove(struct i2c_client *client) { struct tpa6130a2_data *data = i2c_get_clientdata(client); -- cgit v1.2.2 From ed69c6a8eef679f2783848ed624897a937a434ac Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 13 Jan 2010 08:12:31 +0100 Subject: ALSA: pcm_lib - fix wrong delta print for jiffies check The previous jiffies delta was 0 in all cases. Use hw_ptr variable to store and print original value. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_lib.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0ee7e807c964..5417f7dce834 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -394,6 +394,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, + HZ/100); /* move new_hw_ptr according jiffies not pos variable */ new_hw_ptr = old_hw_ptr; + hw_base = delta; /* use loop to avoid checks for delta overflows */ /* the delta value is small or zero in most cases */ while (delta > 0) { @@ -403,8 +404,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, delta--; } /* align hw_base to buffer_size */ - hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); - delta = 0; hw_ptr_error(substream, "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " @@ -412,9 +411,12 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, - ((hdelta * HZ) / runtime->rate), delta, + ((hdelta * HZ) / runtime->rate), hw_base, (unsigned long)old_hw_ptr, (unsigned long)new_hw_ptr); + /* reset values to proper state */ + delta = 0; + hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { -- cgit v1.2.2 From d2f2fcd2541bae004db7f4798ffd9d2cb75ae817 Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Tue, 12 Jan 2010 17:03:35 -0800 Subject: ALSA: hda_intel: ALSA HD Audio patch for Intel Cougar Point DeviceIDs This patch adds the Intel Cougar Point (PCH) HD Audio Controller DeviceIDs. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f516e668d88..6d331c4cf185 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -125,6 +125,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH9}," "{Intel, ICH10}," "{Intel, PCH}," + "{Intel, CPT}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -2677,6 +2678,8 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, /* PCH */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, + /* CPT */ + { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_ICH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ -- cgit v1.2.2 From fd63df2264f2518fa67dca596d493a330537494d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 13 Jan 2010 12:37:49 +0200 Subject: ASoC: TWL4030: Replace comma with semicolon in probe function The codec structure initialization statements should be separated by semicolons. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a27f7b56726..74f0d65f0784 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2192,7 +2192,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); -- cgit v1.2.2 From 617b14c50eb95b36360b2b3232c6cf20b910e2f8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 13 Jan 2010 11:25:05 +0100 Subject: ASoC: ak4104: allow more sample rates The transmitter supports all sample rates up to 192KHz, so the driver should not give a limit. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 3a14c6fc4f5e..b9ef7e45891d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -185,9 +185,7 @@ struct snd_soc_dai ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_32000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- cgit v1.2.2 From 4dee8baa18d611b6dc854e1cc193550ff6f687be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jan 2010 17:20:08 +0100 Subject: ALSA: hda - Fix Toshiba NB20x quirk entry The alc664-mode4 model doesn't seem to fit with Toshiba NB205 correctly. NB205 uses the pin 0x17 connected with the mixer 0x0f for the speaker output, which isn't controlled by mode4 model at all. Rather model=auto works fine as is on the latest driver, so let it back again. Tested-by: Nickolas Lloyd Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e3caa78ccd54..bff60cea7777 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17251,7 +17251,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), -- cgit v1.2.2 From a76221d47ef2b73ff16c0fef00a784026308ea02 Mon Sep 17 00:00:00 2001 From: Alex Murray Date: Wed, 13 Jan 2010 23:15:03 +1030 Subject: ALSA: hda - Improved MacBook (Pro) 5,1 / 5,2 support This patch adds support for automatically muting the speakers when headphones are inserted, as well as relabelling the headphone widgets from the non-standard "HP" to the standard "Headphone" for the mb5 model. Signed-off-by: Alex Murray Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++-- 1 file changed, 26 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bff60cea7777..11b989bacd3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7094,8 +7094,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -7496,6 +7496,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -7680,6 +7681,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_mb5_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_mb5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -9126,6 +9148,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_mb5_unsol_event, + .init_hook = alc885_mb5_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, -- cgit v1.2.2 From d1458279bf9c575a52fd22818ca19c463f380aba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:16:52 +0100 Subject: ALSA: Add snd_pci_quirk_lookup_id() Added a new function to look up a quirk entry with the given PCI SSID instead of a pci device pointer. This can be used when the searched ID is overridden for debugging or such a purpose. Signed-off-by: Takashi Iwai --- sound/core/misc.c | 32 +++++++++++++++++++++++++++----- 1 file changed, 27 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/misc.c b/sound/core/misc.c index 23a032c6d487..3da4f92427d8 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -101,8 +101,9 @@ EXPORT_SYMBOL_GPL(__snd_printk); #ifdef CONFIG_PCI #include /** - * snd_pci_quirk_lookup - look up a PCI SSID quirk list - * @pci: pci_dev handle + * snd_pci_quirk_lookup_id - look up a PCI SSID quirk list + * @vendor: PCI SSV id + * @device: PCI SSD id * @list: quirk list, terminated by a null entry * * Look through the given quirk list and finds a matching entry @@ -112,18 +113,39 @@ EXPORT_SYMBOL_GPL(__snd_printk); * Returns the matched entry pointer, or NULL if nothing matched. */ const struct snd_pci_quirk * -snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list) { const struct snd_pci_quirk *q; for (q = list; q->subvendor; q++) { - if (q->subvendor != pci->subsystem_vendor) + if (q->subvendor != vendor) continue; if (!q->subdevice || - (pci->subsystem_device & q->subdevice_mask) == q->subdevice) + (device & q->subdevice_mask) == q->subdevice) return q; } return NULL; } +EXPORT_SYMBOL(snd_pci_quirk_lookup_id); + +/** + * snd_pci_quirk_lookup - look up a PCI SSID quirk list + * @pci: pci_dev handle + * @list: quirk list, terminated by a null entry + * + * Look through the given quirk list and finds a matching entry + * with the same PCI SSID. When subdevice is 0, all subdevice + * values may match. + * + * Returns the matched entry pointer, or NULL if nothing matched. + */ +const struct snd_pci_quirk * +snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +{ + return snd_pci_quirk_lookup_id(pci->subsystem_vendor, + pci->subsystem_device, + list); +} EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif -- cgit v1.2.2 From 408bffd01cfcda2907b07fb86b3666e3db86fd82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:19:46 +0100 Subject: ALSA: ctxfi - Add subsystem option Added a new option "subsystem" to override the PCI SSID for identifying the card type. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 23 +++++++++++++++-------- sound/pci/ctxfi/ctatc.h | 2 +- sound/pci/ctxfi/xfi.c | 5 ++++- 3 files changed, 20 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index cb65bd0dd35b..903594e6ed79 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1225,10 +1225,11 @@ static int atc_dev_free(struct snd_device *dev) return ct_atc_destroy(atc); } -static int __devinit atc_identify_card(struct ct_atc *atc) +static int __devinit atc_identify_card(struct ct_atc *atc, unsigned int ssid) { const struct snd_pci_quirk *p; const struct snd_pci_quirk *list; + u16 vendor_id, device_id; switch (atc->chip_type) { case ATC20K1: @@ -1242,13 +1243,19 @@ static int __devinit atc_identify_card(struct ct_atc *atc) default: return -ENOENT; } - p = snd_pci_quirk_lookup(atc->pci, list); + if (ssid) { + vendor_id = ssid >> 16; + device_id = ssid & 0xffff; + } else { + vendor_id = atc->pci->subsystem_vendor; + device_id = atc->pci->subsystem_device; + } + p = snd_pci_quirk_lookup_id(vendor_id, device_id, list); if (p) { if (p->value < 0) { printk(KERN_ERR "ctxfi: " "Device %04x:%04x is black-listed\n", - atc->pci->subsystem_vendor, - atc->pci->subsystem_device); + vendor_id, device_id); return -ENOENT; } atc->model = p->value; @@ -1261,8 +1268,7 @@ static int __devinit atc_identify_card(struct ct_atc *atc) atc->model_name = ct_subsys_name[atc->model]; snd_printd("ctxfi: chip %s model %s (%04x:%04x) is found\n", atc->chip_name, atc->model_name, - atc->pci->subsystem_vendor, - atc->pci->subsystem_device); + vendor_id, device_id); return 0; } @@ -1636,7 +1642,8 @@ static struct ct_atc atc_preset __devinitdata = { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, - int chip_type, struct ct_atc **ratc) + int chip_type, unsigned int ssid, + struct ct_atc **ratc) { struct ct_atc *atc; static struct snd_device_ops ops = { @@ -1662,7 +1669,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, mutex_init(&atc->atc_mutex); /* Find card model */ - err = atc_identify_card(atc); + err = atc_identify_card(atc, ssid); if (err < 0) { printk(KERN_ERR "ctatc: Card not recognised\n"); goto error1; diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 9fd8a5708943..7167c0185d52 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -148,7 +148,7 @@ struct ct_atc { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, int chip_type, - struct ct_atc **ratc); + unsigned int subsysid, struct ct_atc **ratc); int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc); #endif /* CTATC_H */ diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 76541748e7bc..ed44ed788b60 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -32,6 +32,7 @@ module_param(multiple, uint, S_IRUGO); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static unsigned int subsystem[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Creative X-Fi driver"); @@ -39,6 +40,8 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Creative X-Fi driver"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); +module_param_array(subsystem, int, NULL, 0444); +MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); static struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ @@ -85,7 +88,7 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) multiple = 2; } err = ct_atc_create(card, pci, reference_rate, multiple, - pci_id->driver_data, &atc); + pci_id->driver_data, subsystem[dev], &atc); if (err < 0) goto error; -- cgit v1.2.2 From 738ada47cf60830d37bb70ffb0b0281d19fc4c7f Mon Sep 17 00:00:00 2001 From: Thomas Weber Date: Tue, 12 Jan 2010 17:07:18 +0100 Subject: ASoC: TWL4030: Fix typo in comment in header file Signed-off-by: Thomas Weber Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index dd6396ec9c79..f206d242ca31 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -25,7 +25,7 @@ /* Register descriptions are here */ #include -/* Sgadow register used by the audio driver */ +/* Shadow register used by the audio driver */ #define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -- cgit v1.2.2 From c7a8eb103248a110cdbe0530d8c5ce987f099eee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 12:39:02 +0100 Subject: ALSA: hda - Fix missing capture mixer for ALC861/660 codecs The capture-related mixer elements are missing with ALC861/ALC660 codecs when quirks are present, due to missing call of set_capture_mixer(). Reference: Novell bnc#567340 http://bugzilla.novell.com/show_bug.cgi?id=567340 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 11b989bacd3c..abae1007cea2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14879,6 +14879,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + if (!spec->cap_mixer) + set_capture_mixer(codec); set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); spec->vmaster_nid = 0x03; -- cgit v1.2.2 From c181a13a41ef32c9037393f4b42b780e1a36eb91 Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Tue, 12 Jan 2010 20:20:39 -0200 Subject: ALSA: use subsys_initcall for sound core instead of module_init This is needed for built-in drivers which are built before the sound directory, like thinkpad_acpi. Otherwise, registering a card fails. Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai --- sound/core/sound.c | 4 ++-- sound/sound_core.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/sound.c b/sound/core/sound.c index 7872a02f6ca9..563d1967a0ad 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void) unregister_chrdev(major, "alsa"); } -module_init(alsa_sound_init) -module_exit(alsa_sound_exit) +subsys_initcall(alsa_sound_init); +module_exit(alsa_sound_exit); diff --git a/sound/sound_core.c b/sound/sound_core.c index dbca7c909a31..7c2d677a2df5 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void) class_destroy(sound_class); } -module_init(init_soundcore); +subsys_initcall(init_soundcore); module_exit(cleanup_soundcore); -- cgit v1.2.2 From d38cce7046cfd0011f69d5dcf6a22525438154f6 Mon Sep 17 00:00:00 2001 From: Kunal Gangakhedkar Date: Fri, 15 Jan 2010 21:01:47 +0530 Subject: ALSA: hda - Fix mute led GPIO on HP dv-series notebooks On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type "HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO) either. As per the documentation of find_mute_led_gpio(), these strings occur in HP B-series systems - so, before scanning the SMBIOS strings, we need to check if we're dealing with a B-series system. Need to get confirmation from HP if this logic takes care of all the systems. I'm trying to poke a friend there. Signed-off-by: Kunal Gangakhedkar Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 61 +++++++++++++++++++++++++++++++----------- 1 file changed, 45 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2291a8396817..799ba2570902 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4730,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_blike_system(u32 subsystem_id); + +static void set_hp_led_gpio(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + break; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + break; + } +} + /* * This method searches for the mute LED GPIO configuration * provided as OEM string in SMBIOS. The format of that string @@ -4741,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) * * So, HP B-series like systems may have HP_Mute_LED_0 (current models) * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + * + * + * The dv-series laptops don't seem to have the HP_Mute_LED* strings in + * SMBIOS - at least the ones I have seen do not have them - which include + * my own system (HP Pavilion dv6-1110ax) and my cousin's + * HP Pavilion dv9500t CTO. + * Need more information on whether it is true across the entire series. + * -- kunal */ static int find_mute_led_gpio(struct hda_codec *codec) { @@ -4751,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec) while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (sscanf(dev->name, "HP_Mute_LED_%d_%d", - &spec->gpio_led_polarity, - &spec->gpio_led) == 2) { + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { spec->gpio_led = 1 << spec->gpio_led; return 1; } if (sscanf(dev->name, "HP_Mute_LED_%d", - &spec->gpio_led_polarity) == 1) { - switch (codec->vendor_id) { - case 0x111d7608: - /* GPIO 0 */ - spec->gpio_led = 0x01; - return 1; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - /* GPIO 3 */ - spec->gpio_led = 0x08; - return 1; - } + &spec->gpio_led_polarity) == 1) { + set_hp_led_gpio(codec); + return 1; } } + + /* + * Fallback case - if we don't find the DMI strings, + * we statically set the GPIO - if not a B-series system. + */ + if (!hp_blike_system(codec->subsystem_id)) { + set_hp_led_gpio(codec); + spec->gpio_led_polarity = 1; + return 1; + } } return 0; } @@ -5548,6 +5575,8 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); + snd_printdd("Found board config: %d\n", spec->board_config); + switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ -- cgit v1.2.2 From 6aababdf20bb8892023bb8df136514d7679e4959 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 15 Jan 2010 17:36:48 +0100 Subject: ASoC: cs4270: allow passing freq=0 in set_dai_sysclk() For setups with variable MCLKs, the current logic of limiting the available sampling rates at startup time is not sufficient. We need to be able to change the setting at a later point, and so the codec must offer all possible rates until the hw_params are given. This patches allows that by passing 0 as 'freq' argument to cs4270_set_dai_sysclk(). Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 38 +++++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b5457542a0e..593bfc7a6986 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -200,6 +200,11 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = { * This function must be called by the machine driver's 'startup' function, * otherwise the list of supported sample rates will not be available in * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. */ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) @@ -213,20 +218,27 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, cs4270->mclk = freq; - for (i = 0; i < NUM_MCLK_RATIOS; i++) { - unsigned int rate = freq / cs4270_mode_ratios[i].ratio; - rates |= snd_pcm_rate_to_rate_bit(rate); - if (rate < rate_min) - rate_min = rate; - if (rate > rate_max) - rate_max = rate; - } - /* FIXME: soc should support a rate list */ - rates &= ~SNDRV_PCM_RATE_KNOT; + if (cs4270->mclk) { + for (i = 0; i < NUM_MCLK_RATIOS; i++) { + unsigned int rate = freq / cs4270_mode_ratios[i].ratio; + rates |= snd_pcm_rate_to_rate_bit(rate); + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; - if (!rates) { - dev_err(codec->dev, "could not find a valid sample rate\n"); - return -EINVAL; + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = SNDRV_PCM_RATE_8000_192000; + rate_min = 8000; + rate_max = 192000; } codec_dai->playback.rates = rates; -- cgit v1.2.2 From a421296840379aee7d00ec4a28ecfe7e697a0a44 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 15 Jan 2010 17:36:49 +0100 Subject: ASoC: support more sample rates on raumfeld devices Add support for sample rates other than 44100Khz on raumfeld audio devices. At startup time, call snd_soc_dai_set_sysclk() with 0 as 'freq' argument so it offers all the sample rates. Later, the function is called again to give proper constraints. Use the external audio clock generator to provide double data rate clocks as the PXA's internal baud generator does anything but what's described in the datasheets. Signed-off-by: Daniel Mack Cc: Mark Brown Cc: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/pxa/raumfeld.c | 61 +++++++++++++++++++++++++++++++----------------- 1 file changed, 40 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index acfce1c0f1c9..7e3f41696c41 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -41,7 +41,9 @@ static struct i2c_board_info max9486_hwmon_info = { }; #define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_225792 0x32 +#define MAX9485_MCLK_FREQ_245760 0x33 static void set_max9485_clk(char clk) { @@ -71,9 +73,17 @@ static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - set_max9485_clk(MAX9485_MCLK_FREQ_112896); + /* set freq to 0 to enable all possible codec sample rates */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); +} - return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* set freq to 0 to enable all possible codec sample rates */ + snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); } static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, @@ -86,20 +96,24 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, int ret = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; + break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | @@ -128,7 +142,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -137,6 +151,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, static struct snd_soc_ops raumfeld_cs4270_ops = { .startup = raumfeld_cs4270_startup, + .shutdown = raumfeld_cs4270_shutdown, .hw_params = raumfeld_cs4270_hw_params, }; @@ -181,20 +196,24 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, int fmt, ret = 0, clk = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; + break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; @@ -217,7 +236,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; -- cgit v1.2.2 From 8380222ec9458d38a4e0cc3cb688ad7fff311df4 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 25 Nov 2009 16:41:04 +0100 Subject: ASoC: Add a new imx-ssi sound driver The old driver has the number of SSI units in the system hardcoded, does not make use of the device model and works only on i.MX21/27. This driver replaces it. It works in DMA mode on i.MX21/27 and using an FIQ handler on other systems. It also supports AC97 mode of the SSI units. Signed-off-by: Sascha Hauer Acked-by: Javier Martin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 12 +- sound/soc/imx/imx-pcm-dma-mx2.c | 313 +++++++++++++++++ sound/soc/imx/imx-pcm-fiq.c | 277 +++++++++++++++ sound/soc/imx/imx-ssi.c | 762 ++++++++++++++++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 238 +++++++++++++ 6 files changed, 1602 insertions(+), 20 deletions(-) create mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c create mode 100644 sound/soc/imx/imx-pcm-fiq.c create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index a700562e8692..84a25e61bed8 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,21 +1,13 @@ -config SND_MX1_MX2_SOC - tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs" - depends on ARCH_MX2 || ARCH_MX1 +config SND_IMX_SOC + tristate "SoC Audio for Freecale i.MX CPUs" + depends on ARCH_MXC select SND_PCM + select FIQ + select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to - the MX1 or MX2 SSI interface. + the i.MX SSI interface. config SND_MXC_SOC_SSI tristate -config SND_SOC_MX27VIS_WM8974 - tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" - depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 - select SND_MXC_SOC_SSI - select SND_SOC_WM8974 - help - Say Y if you want to add support for SoC audio on Visstrim SM10 - board with WM8974. - - diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index c2ffd2c8df5a..4bde34a3a878 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,10 +1,10 @@ # i.MX Platform Support -snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o -snd-soc-mxc-ssi-objs := mxc-ssi.o +snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o imx-pcm-dma-mx2.o -obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o -obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o +ifdef CONFIG_MACH_MX27 +snd-soc-imx-objs += imx-pcm-dma-mx2.o +endif + +obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support -snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o -obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c new file mode 100644 index 000000000000..19452e44afdc --- /dev/null +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -0,0 +1,313 @@ +/* + * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int sg_count; + struct scatterlist *sg_list; + int period; + int periods; + unsigned long dma_addr; + int dma; + struct snd_pcm_substream *substream; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + int period_time; +}; + +/* Called by the DMA framework when a period has elapsed */ +static void imx_ssi_dma_progression(int channel, void *data, + struct scatterlist *sg) +{ + struct snd_pcm_substream *substream = data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (!sg) + return; + + runtime = iprtd->substream->runtime; + + iprtd->offset = sg->dma_address - runtime->dma_addr; + + snd_pcm_period_elapsed(iprtd->substream); +} + +static void imx_ssi_dma_callback(int channel, void *data) +{ + pr_err("%s shouldn't be called\n", __func__); +} + +static void snd_imx_dma_err_callback(int channel, void *data, int err) +{ + pr_err("DMA error callback called\n"); + + pr_err("DMA timeout on channel %d -%s%s%s%s\n", + channel, + err & IMX_DMA_ERR_BURST ? " burst" : "", + err & IMX_DMA_ERR_REQUEST ? " request" : "", + err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", + err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); +} + +static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int ret; + + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); + if (iprtd->dma < 0) { + pr_err("Failed to claim the audio DMA\n"); + return -ENODEV; + } + + ret = imx_dma_setup_handlers(iprtd->dma, + imx_ssi_dma_callback, + snd_imx_dma_err_callback, substream); + if (ret) + goto out; + + ret = imx_dma_setup_progression_handler(iprtd->dma, + imx_ssi_dma_progression); + if (ret) { + pr_err("Failed to setup the DMA handler\n"); + goto out; + } + + ret = imx_dma_config_channel(iprtd->dma, + IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, + dma_params->dma, 1); + if (ret < 0) { + pr_err("Cannot configure DMA channel: %d\n", ret); + goto out; + } + + imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2); + + return 0; +out: + imx_dma_free(iprtd->dma); + return ret; +} + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int i; + unsigned long dma_addr; + + imx_ssi_dma_alloc(substream); + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / + params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + if (iprtd->sg_count != iprtd->periods) { + kfree(iprtd->sg_list); + + iprtd->sg_list = kcalloc(iprtd->periods + 1, + sizeof(struct scatterlist), GFP_KERNEL); + if (!iprtd->sg_list) + return -ENOMEM; + iprtd->sg_count = iprtd->periods + 1; + } + + sg_init_table(iprtd->sg_list, iprtd->sg_count); + dma_addr = runtime->dma_addr; + + for (i = 0; i < iprtd->periods; i++) { + iprtd->sg_list[i].page_link = 0; + iprtd->sg_list[i].offset = 0; + iprtd->sg_list[i].dma_address = dma_addr; + iprtd->sg_list[i].length = iprtd->period; + dma_addr += iprtd->period; + } + + /* close the loop */ + iprtd->sg_list[iprtd->sg_count - 1].offset = 0; + iprtd->sg_list[iprtd->sg_count - 1].length = 0; + iprtd->sg_list[iprtd->sg_count - 1].page_link = + ((unsigned long) iprtd->sg_list | 0x01) & ~0x02; + return 0; +} + +static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (iprtd->dma >= 0) { + imx_dma_free(iprtd->dma); + iprtd->dma = -EINVAL; + } + + kfree(iprtd->sg_list); + iprtd->sg_list = NULL; + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int err; + + iprtd->substream = substream; + iprtd->buf = (unsigned int *)substream->dma_buffer.area; + iprtd->period_cnt = 0; + + pr_debug("%s: buf: %p period: %d periods: %d\n", + __func__, iprtd->buf, iprtd->period, iprtd->periods); + + err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, + IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_MODE_WRITE : DMA_MODE_READ); + if (err) + return err; + + return 0; +} + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + imx_dma_enable(iprtd->dma); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + imx_dma_disable(iprtd->dma); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .hw_free = snd_imx_pcm_hw_free, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static struct snd_soc_platform imx_soc_platform_dma = { + .name = "imx-audio", + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + ssi->dma_params_tx.burstsize = DMA_TXFIFO_BURST; + ssi->dma_params_rx.burstsize = DMA_RXFIFO_BURST; + + return &imx_soc_platform_dma; +} + diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c new file mode 100644 index 000000000000..5532579ece4d --- /dev/null +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -0,0 +1,277 @@ +/* + * imx-pcm-fiq.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include + +#include + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int period; + int periods; + unsigned long dma_addr; + int dma; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + struct timer_list timer; + int period_time; +}; + +static void imx_ssi_timer_callback(unsigned long data) +{ + struct snd_pcm_substream *substream = (void *)data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(®s); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + iprtd->offset = regs.ARM_r8 & 0xffff; + else + iprtd->offset = regs.ARM_r9 & 0xffff; + + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + snd_pcm_period_elapsed(substream); +} + +static struct fiq_handler fh = { + .name = DRV_NAME, +}; + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(®s); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16; + else + regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16; + + set_fiq_regs(®s); + + return 0; +} + +static int fiq_enable; +static int imx_pcm_fiq; + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + if (++fiq_enable == 1) + enable_fiq(imx_pcm_fiq); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + del_timer(&iprtd->timer); + if (--fiq_enable == 0) + disable_fiq(imx_pcm_fiq); + + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + init_timer(&iprtd->timer); + iprtd->timer.data = (unsigned long)substream; + iprtd->timer.function = imx_ssi_timer_callback; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + del_timer_sync(&iprtd->timer); + kfree(iprtd); + + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .close = snd_imx_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret; + + ret = imx_pcm_new(card, dai, pcm); + if (ret) + return ret; + + if (dai->playback.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; + } + + if (dai->capture.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; + } + + set_fiq_handler(&imx_ssi_fiq_start, + &imx_ssi_fiq_end - &imx_ssi_fiq_start); + + return 0; +} + +static struct snd_soc_platform imx_soc_platform_fiq = { + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_fiq_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + int ret = 0; + + ret = claim_fiq(&fh); + if (ret) { + dev_err(&pdev->dev, "failed to claim fiq: %d", ret); + return ERR_PTR(ret); + } + + mxc_set_irq_fiq(ssi->irq, 1); + + imx_pcm_fiq = ssi->irq; + + imx_ssi_fiq_base = (unsigned long)ssi->base; + + ssi->dma_params_tx.burstsize = 4; + ssi->dma_params_rx.burstsize = 6; + + return &imx_soc_platform_fiq; +} + +void imx_ssi_fiq_exit(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + mxc_set_irq_fiq(ssi->irq, 0); + release_fiq(&fh); +} + diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c new file mode 100644 index 000000000000..c57a11f66954 --- /dev/null +++ b/sound/soc/imx/imx-ssi.c @@ -0,0 +1,762 @@ +/* + * imx-ssi.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * + * The i.MX SSI core has some nasty limitations in AC97 mode. While most + * sane processor vendors have a FIFO per AC97 slot, the i.MX has only + * one FIFO which combines all valid receive slots. We cannot even select + * which slots we want to receive. The WM9712 with which this driver + * was developped with always sends GPIO status data in slot 12 which + * we receive in our (PCM-) data stream. The only chance we have is to + * manually skip this data in the FIQ handler. With sampling rates different + * from 48000Hz not every frame has valid receive data, so the ratio + * between pcm data and GPIO status data changes. Our FIQ handler is not + * able to handle this, hence this driver only works with 48000Hz sampling + * rate. + * Reading and writing AC97 registers is another challange. The core + * provides us status bits when the read register is updated with *another* + * value. When we read the same register two times (and the register still + * contains the same value) these status bits are not set. We work + * around this by not polling these bits but only wait a fixed delay. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "imx-ssi.h" + +#define SSI_SACNT_DEFAULT (SSI_SACNT_AC97EN | SSI_SACNT_FV) + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 sccr; + + sccr = readl(ssi->base + SSI_STCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_STCCR); + + sccr = readl(ssi->base + SSI_SRCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_SRCCR); + + writel(tx_mask, ssi->base + SSI_STMSK); + writel(rx_mask, ssi->base + SSI_SRMSK); + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + * Note: We don't use the I2S modes but instead manually configure the + * SSI for I2S because the I2S mode is only a register preset. + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 strcr = 0, scr; + + scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + scr |= SSI_SCR_NET; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TFSL; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + strcr |= SSI_STCR_TFSI; + strcr &= ~SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + strcr &= ~SSI_STCR_TFSI; + strcr |= SSI_STCR_TSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr |= SSI_STCR_TFDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + strcr |= SSI_STCR_TXDIR; + break; + } + + strcr |= SSI_STCR_TFEN0; + + writel(strcr, ssi->base + SSI_STCR); + writel(strcr, ssi->base + SSI_SRCR); + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 scr; + + scr = readl(ssi->base + SSI_SCR); + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) + scr |= SSI_SCR_SYS_CLK_EN; + else + scr &= ~SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 stccr, srccr; + + stccr = readl(ssi->base + SSI_STCCR); + srccr = readl(ssi->base + SSI_SRCCR); + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + writel(stccr, ssi->base + SSI_STCCR); + writel(srccr, ssi->base + SSI_SRCCR); + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +static int imx_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 reg, sccr; + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg = SSI_STCCR; + cpu_dai->dma_data = &ssi->dma_params_tx; + } else { + reg = SSI_SRCCR; + cpu_dai->dma_data = &ssi->dma_params_rx; + } + + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + sccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + sccr |= SSI_SRCCR_WL(24); + break; + } + + writel(sccr, ssi->base + reg); + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + unsigned int sier_bits, sier; + unsigned int scr; + + scr = readl(ssi->base + SSI_SCR); + sier = readl(ssi->base + SSI_SIER); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_TDMAE; + else + sier_bits = SSI_SIER_TIE | SSI_SIER_TFE0_EN; + } else { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_RDMAE; + else + sier_bits = SSI_SIER_RIE | SSI_SIER_RFF0_EN; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE; + else + scr |= SSI_SCR_RE; + sier |= sier_bits; + + if (++ssi->enabled == 1) + scr |= SSI_SCR_SSIEN; + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + sier &= ~sier_bits; + + if (--ssi->enabled == 0) + scr &= ~SSI_SCR_SSIEN; + + break; + default: + return -EINVAL; + } + + if (!(ssi->flags & IMX_SSI_USE_AC97)) + /* rx/tx are always enabled to access ac97 registers */ + writel(scr, ssi->base + SSI_SCR); + + writel(sier, ssi->base + SSI_SIER); + + return 0; +} + +static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .hw_params = imx_ssi_hw_params, + .set_fmt = imx_ssi_set_dai_fmt, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, + .trigger = imx_ssi_trigger, +}; + +static struct snd_soc_dai imx_ssi_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} + +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (dai->playback.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + +void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +struct snd_soc_platform imx_soc_platform = { + .name = "imx-audio", +}; +EXPORT_SYMBOL_GPL(imx_soc_platform); + +static struct snd_soc_dai imx_ac97_dai = { + .name = "AC97", + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +static void setup_channel_to_ac97(struct imx_ssi *imx_ssi) +{ + void __iomem *base = imx_ssi->base; + + writel(0x0, base + SSI_SCR); + writel(0x0, base + SSI_STCR); + writel(0x0, base + SSI_SRCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR); + + writel(SSI_SFCSR_RFWM0(8) | + SSI_SFCSR_TFWM0(8) | + SSI_SFCSR_RFWM1(8) | + SSI_SFCSR_TFWM1(8), base + SSI_SFCSR); + + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR); + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN, base + SSI_SCR); + writel(SSI_SOR_WAIT(3), base + SSI_SOR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN | + SSI_SCR_TE | SSI_SCR_RE, + base + SSI_SCR); + + writel(SSI_SACNT_DEFAULT, base + SSI_SACNT); + writel(0xff, base + SSI_SACCDIS); + writel(0x300, base + SSI_SACCEN); +} + +static struct imx_ssi *ac97_ssi; + +static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + unsigned int lreg; + unsigned int lval; + + if (reg > 0x7f) + return; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + lreg = reg << 12; + writel(lreg, base + SSI_SACADD); + + lval = val << 4; + writel(lval , base + SSI_SACDAT); + + writel(SSI_SACNT_DEFAULT | SSI_SACNT_WR, base + SSI_SACNT); + udelay(100); +} + +static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + + unsigned short val = -1; + unsigned int lreg; + + lreg = (reg & 0x7f) << 12 ; + writel(lreg, base + SSI_SACADD); + writel(SSI_SACNT_DEFAULT | SSI_SACNT_RD, base + SSI_SACNT); + + udelay(100); + + val = (readl(base + SSI_SACDAT) >> 4) & 0xffff; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + return val; +} + +static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_reset) + imx_ssi->ac97_reset(ac97); +} + +static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_warm_reset) + imx_ssi->ac97_warm_reset(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = imx_ssi_ac97_read, + .write = imx_ssi_ac97_write, + .reset = imx_ssi_ac97_reset, + .warm_reset = imx_ssi_ac97_warm_reset +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +struct snd_soc_dai *imx_ssi_pcm_dai[2]; +EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); + +static int imx_ssi_probe(struct platform_device *pdev) +{ + struct resource *res; + struct imx_ssi *ssi; + struct imx_ssi_platform_data *pdata = pdev->dev.platform_data; + struct snd_soc_platform *platform; + int ret = 0; + unsigned int val; + + ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); + if (!ssi) + return -ENOMEM; + + if (pdata) { + ssi->ac97_reset = pdata->ac97_reset; + ssi->ac97_warm_reset = pdata->ac97_warm_reset; + ssi->flags = pdata->flags; + } + + imx_ssi_pcm_dai[pdev->id] = &ssi->dai; + + ssi->irq = platform_get_irq(pdev, 0); + + ssi->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi->clk)) { + ret = PTR_ERR(ssi->clk); + dev_err(&pdev->dev, "Cannot get the clock: %d\n", + ret); + goto failed_clk; + } + clk_enable(ssi->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + goto failed_get_resource; + } + + if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + ret = -EBUSY; + goto failed_get_resource; + } + + ssi->base = ioremap(res->start, resource_size(res)); + if (!ssi->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENODEV; + goto failed_ioremap; + } + + if (ssi->flags & IMX_SSI_USE_AC97) { + if (ac97_ssi) { + ret = -EBUSY; + goto failed_ac97; + } + ac97_ssi = ssi; + setup_channel_to_ac97(ssi); + memcpy(&ssi->dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); + } else + memcpy(&ssi->dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + writel(0x0, ssi->base + SSI_SIER); + + ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; + ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); + if (res) + ssi->dma_params_tx.dma = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); + if (res) + ssi->dma_params_rx.dma = res->start; + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + if ((cpu_is_mx27() || cpu_is_mx21()) && + !(ssi->flags & IMX_SSI_USE_AC97)) { + ssi->flags |= IMX_SSI_DMA; + platform = imx_ssi_dma_mx2_init(pdev, ssi); + } else + platform = imx_ssi_fiq_init(pdev, ssi); + + imx_soc_platform.pcm_ops = platform->pcm_ops; + imx_soc_platform.pcm_new = platform->pcm_new; + imx_soc_platform.pcm_free = platform->pcm_free; + + val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | + SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); + writel(val, ssi->base + SSI_SFCSR); + + ret = snd_soc_register_dai(&ssi->dai); + if (ret) { + dev_err(&pdev->dev, "register DAI failed\n"); + goto failed_register; + } + + platform_set_drvdata(pdev, ssi); + + return 0; + +failed_register: +failed_ac97: + iounmap(ssi->base); +failed_ioremap: + release_mem_region(res->start, resource_size(res)); +failed_get_resource: + clk_disable(ssi->clk); + clk_put(ssi->clk); +failed_clk: + kfree(ssi); + + return ret; +} + +static int __devexit imx_ssi_remove(struct platform_device *pdev) +{ + struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + struct imx_ssi *ssi = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&ssi->dai); + + if (ssi->flags & IMX_SSI_USE_AC97) + ac97_ssi = NULL; + + if (!(ssi->flags & IMX_SSI_DMA)) + imx_ssi_fiq_exit(pdev, ssi); + + iounmap(ssi->base); + release_mem_region(res->start, resource_size(res)); + clk_disable(ssi->clk); + clk_put(ssi->clk); + kfree(ssi); + + return 0; +} + +static struct platform_driver imx_ssi_driver = { + .probe = imx_ssi_probe, + .remove = __devexit_p(imx_ssi_remove), + + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, +}; + +static int __init imx_ssi_init(void) +{ + int ret; + + ret = snd_soc_register_platform(&imx_soc_platform); + if (ret) { + pr_err("failed to register soc platform: %d\n", ret); + return ret; + } + + ret = platform_driver_register(&imx_ssi_driver); + if (ret) { + snd_soc_unregister_platform(&imx_soc_platform); + return ret; + } + + return 0; +} + +static void __exit imx_ssi_exit(void) +{ + platform_driver_unregister(&imx_ssi_driver); + snd_soc_unregister_platform(&imx_soc_platform); +} + +module_init(imx_ssi_init); +module_exit(imx_ssi_exit); + +/* Module information */ +MODULE_AUTHOR("Sascha Hauer, "); +MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h new file mode 100644 index 000000000000..cb2c81f1a6fc --- /dev/null +++ b/sound/soc/imx/imx-ssi.h @@ -0,0 +1,238 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _IMX_SSI_H +#define _IMX_SSI_H + +#define SSI_STX0 0x00 +#define SSI_STX1 0x04 +#define SSI_SRX0 0x08 +#define SSI_SRX1 0x0c + +#define SSI_SCR 0x10 +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_CLK_IST_SHIFT 9 +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_I2S_MODE_MASK (3 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR 0x14 +#define SSI_SISR_MASK ((1 << 19) - 1) +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER 0x18 +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR 0x1c +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR 0x20 +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_SRCCR 0x28 +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 17) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + +#define SSI_STCCR 0x24 +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 17) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SFCSR 0x2c +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_RX_FIFO_1_COUNT_SHIFT 28 +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_TX_FIFO_1_COUNT_SHIFT 24 +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_RX_FIFO_0_COUNT_SHIFT 12 +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_TX_FIFO_0_COUNT_SHIFT 8 +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) +#define SSI_SFCSR_RFWM0_MASK (0xf << 4) +#define SSI_SFCSR_TFWM0_MASK (0xf << 0) + +#define SSI_STR 0x30 +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR 0x34 +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_WAIT_MASK (0x3 << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT 0x38 +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (1 << 4) +#define SSI_SACNT_RD (1 << 3) +#define SSI_SACNT_TIF (1 << 2) +#define SSI_SACNT_FV (1 << 1) +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SACADD 0x3c +#define SSI_SACDAT 0x40 +#define SSI_SATAG 0x44 +#define SSI_STMSK 0x48 +#define SSI_SRMSK 0x4c +#define SSI_SACCST 0x50 +#define SSI_SACCEN 0x54 +#define SSI_SACCDIS 0x58 + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + +extern struct snd_soc_dai *imx_ssi_pcm_dai[2]; +extern struct snd_soc_platform imx_soc_platform; + +#define DRV_NAME "imx-ssi" + +struct imx_pcm_dma_params { + int dma; + unsigned long dma_addr; + int burstsize; +}; + +struct imx_ssi { + struct snd_soc_dai dai; + struct platform_device *ac97_dev; + + struct snd_soc_device imx_ac97; + struct clk *clk; + void __iomem *base; + int irq; + int fiq_enable; + unsigned int offset; + + unsigned int flags; + + void (*ac97_reset) (struct snd_ac97 *ac97); + void (*ac97_warm_reset)(struct snd_ac97 *ac97); + + struct imx_pcm_dma_params dma_params_rx; + struct imx_pcm_dma_params dma_params_tx; + + int enabled; +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi); +void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi); + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm); +void imx_pcm_free(struct snd_pcm *pcm); + +/* + * Do not change this as the FIQ handler depends on this size + */ +#define IMX_SSI_DMABUF_SIZE (64 * 1024) + +#define DMA_RXFIFO_BURST 0x4 +#define DMA_TXFIFO_BURST 0x6 + +#endif /* _IMX_SSI_H */ -- cgit v1.2.2 From 157a777c8e809bd0c703e3f7617b3539df30feff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Jan 2010 17:50:29 +0000 Subject: ASoC: Fix i.MX audio build for i.MX3x Don't unconditionally include the i.MX2x DMA driver, the arch/arm functions it uses aren't available for i.MX3x. Signed-off-by: Mark Brown Acked-by: Javier Martin --- sound/soc/imx/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 4bde34a3a878..d05cc95c5cc4 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,5 +1,5 @@ # i.MX Platform Support -snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o imx-pcm-dma-mx2.o +snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o ifdef CONFIG_MACH_MX27 snd-soc-imx-objs += imx-pcm-dma-mx2.o -- cgit v1.2.2 From 48dbc41988d07c7a9ba83afd31543d8ecb2beecc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Jan 2010 17:56:52 +0000 Subject: ASoC: Convert new i.MX SSI driver to use static DAI array While dynamically allocated DAIs are the way forward the core doesn't yet support anything except matching with a pointer to the actual DAI so convert to doing that so that machine drivers don't have to jump through hoops to register themselves. Signed-off-by: Mark Brown Acked-by: Javier Martin --- sound/soc/imx/imx-ssi.c | 40 ++++++++++++++++++++-------------------- sound/soc/imx/imx-ssi.h | 3 +-- 2 files changed, 21 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index c57a11f66954..ccb7ec9ce997 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -60,7 +60,7 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 sccr; sccr = readl(ssi->base + SSI_STCCR); @@ -87,7 +87,7 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 strcr = 0, scr; scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); @@ -160,7 +160,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 scr; scr = readl(ssi->base + SSI_SCR); @@ -188,7 +188,7 @@ static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 stccr, srccr; stccr = readl(ssi->base + SSI_STCCR); @@ -237,7 +237,7 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 reg, sccr; /* Tx/Rx config */ @@ -274,7 +274,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; unsigned int sier_bits, sier; unsigned int scr; @@ -570,7 +570,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = { }; EXPORT_SYMBOL_GPL(soc_ac97_ops); -struct snd_soc_dai *imx_ssi_pcm_dai[2]; +struct snd_soc_dai imx_ssi_pcm_dai[2]; EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); static int imx_ssi_probe(struct platform_device *pdev) @@ -581,6 +581,10 @@ static int imx_ssi_probe(struct platform_device *pdev) struct snd_soc_platform *platform; int ret = 0; unsigned int val; + struct snd_soc_dai *dai = &imx_ssi_pcm_dai[pdev->id]; + + if (dai->id >= ARRAY_SIZE(imx_ssi_pcm_dai)) + return -EINVAL; ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); if (!ssi) @@ -592,8 +596,6 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->flags = pdata->flags; } - imx_ssi_pcm_dai[pdev->id] = &ssi->dai; - ssi->irq = platform_get_irq(pdev, 0); ssi->clk = clk_get(&pdev->dev, NULL); @@ -631,13 +633,9 @@ static int imx_ssi_probe(struct platform_device *pdev) } ac97_ssi = ssi; setup_channel_to_ac97(ssi); - memcpy(&ssi->dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); + memcpy(dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); } else - memcpy(&ssi->dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); - - ssi->dai.id = pdev->id; - ssi->dai.dev = &pdev->dev; - ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + memcpy(dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); writel(0x0, ssi->base + SSI_SIER); @@ -652,9 +650,10 @@ static int imx_ssi_probe(struct platform_device *pdev) if (res) ssi->dma_params_rx.dma = res->start; - ssi->dai.id = pdev->id; - ssi->dai.dev = &pdev->dev; - ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + dai->id = pdev->id; + dai->dev = &pdev->dev; + dai->name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + dai->private_data = ssi; if ((cpu_is_mx27() || cpu_is_mx21()) && !(ssi->flags & IMX_SSI_USE_AC97)) { @@ -671,7 +670,7 @@ static int imx_ssi_probe(struct platform_device *pdev) SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); writel(val, ssi->base + SSI_SFCSR); - ret = snd_soc_register_dai(&ssi->dai); + ret = snd_soc_register_dai(dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); goto failed_register; @@ -699,8 +698,9 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev) { struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); + struct snd_soc_dai *dai = &imx_ssi_pcm_dai[pdev->id]; - snd_soc_unregister_dai(&ssi->dai); + snd_soc_unregister_dai(dai); if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index cb2c81f1a6fc..55f26ebcd8c2 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -183,7 +183,7 @@ #define IMX_SSI_RX_DIV_PSR 4 #define IMX_SSI_RX_DIV_PM 5 -extern struct snd_soc_dai *imx_ssi_pcm_dai[2]; +extern struct snd_soc_dai imx_ssi_pcm_dai[2]; extern struct snd_soc_platform imx_soc_platform; #define DRV_NAME "imx-ssi" @@ -195,7 +195,6 @@ struct imx_pcm_dma_params { }; struct imx_ssi { - struct snd_soc_dai dai; struct platform_device *ac97_dev; struct snd_soc_device imx_ac97; -- cgit v1.2.2 From d08a68bfca5a6464eb9167be0659bf0676f02555 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jan 2010 16:56:19 +0000 Subject: ASoC: i.MX SSI driver does not yet support master mode The clocks for the SSI block need handling before this can work. Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index ccb7ec9ce997..56f46a75d297 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -133,15 +133,11 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* DAI clock master masks */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; - break; - case SND_SOC_DAIFMT_CBM_CFS: - strcr |= SSI_STCR_TFDIR; - break; - case SND_SOC_DAIFMT_CBS_CFM: - strcr |= SSI_STCR_TXDIR; + case SND_SOC_DAIFMT_CBM_CFM: break; + default: + /* Master mode not implemented, needs handling of clocks. */ + return -EINVAL; } strcr |= SSI_STCR_TFEN0; -- cgit v1.2.2 From e919c24b6422a095bed3929074bd74ae1dbf251f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 17 Jan 2010 11:08:38 +0000 Subject: ASoC: Remove old i.MX driver code This has been superceeded by Sascha's new driver but was not removed in the patch series due to cutdowns for review. Signed-off-by: Mark Brown --- sound/soc/imx/mx1_mx2-pcm.c | 488 ----------------------- sound/soc/imx/mx1_mx2-pcm.h | 26 -- sound/soc/imx/mx27vis_wm8974.c | 317 --------------- sound/soc/imx/mxc-ssi.c | 860 ----------------------------------------- sound/soc/imx/mxc-ssi.h | 238 ------------ 5 files changed, 1929 deletions(-) delete mode 100644 sound/soc/imx/mx1_mx2-pcm.c delete mode 100644 sound/soc/imx/mx1_mx2-pcm.h delete mode 100644 sound/soc/imx/mx27vis_wm8974.c delete mode 100644 sound/soc/imx/mxc-ssi.c delete mode 100644 sound/soc/imx/mxc-ssi.h (limited to 'sound') diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c deleted file mode 100644 index bffffcd5ff34..000000000000 --- a/sound/soc/imx/mx1_mx2-pcm.c +++ /dev/null @@ -1,488 +0,0 @@ -/* - * mx1_mx2-pcm.c -- ALSA SoC interface for Freescale i.MX1x, i.MX2x CPUs - * - * Copyright 2009 Vista Silicon S.L. - * Author: Javier Martin - * javier.martin@vista-silicon.com - * - * Based on mxc-pcm.c by Liam Girdwood. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "mx1_mx2-pcm.h" - - -static const struct snd_pcm_hardware mx1_mx2_pcm_hardware = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .buffer_bytes_max = 32 * 1024, - .period_bytes_min = 64, - .period_bytes_max = 8 * 1024, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -struct mx1_mx2_runtime_data { - int dma_ch; - int active; - unsigned int period; - unsigned int periods; - int tx_spin; - spinlock_t dma_lock; - struct mx1_mx2_pcm_dma_params *dma_params; -}; - - -/** - * This function stops the current dma transfer for playback - * and clears the dma pointers. - * - * @param substream pointer to the structure of the current stream. - * - */ -static int audio_stop_dma(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - unsigned long flags; - - spin_lock_irqsave(&prtd->dma_lock, flags); - - pr_debug("%s\n", __func__); - - prtd->active = 0; - prtd->period = 0; - prtd->periods = 0; - - /* this stops the dma channel and clears the buffer ptrs */ - - imx_dma_disable(prtd->dma_ch); - - spin_unlock_irqrestore(&prtd->dma_lock, flags); - - return 0; -} - -/** - * This function is called whenever a new audio block needs to be - * transferred to the codec. The function receives the address and the size - * of the new block and start a new DMA transfer. - * - * @param substream pointer to the structure of the current stream. - * - */ -static int dma_new_period(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - unsigned int dma_size; - unsigned int offset; - int ret = 0; - dma_addr_t mem_addr; - unsigned int dev_addr; - - if (prtd->active) { - dma_size = frames_to_bytes(runtime, runtime->period_size); - offset = dma_size * prtd->period; - - pr_debug("%s: period (%d) out of (%d)\n", __func__, - prtd->period, - runtime->periods); - pr_debug("period_size %d frames\n offset %d bytes\n", - (unsigned int)runtime->period_size, - offset); - pr_debug("dma_size %d bytes\n", dma_size); - - snd_BUG_ON(dma_size > mx1_mx2_pcm_hardware.period_bytes_max); - - mem_addr = (dma_addr_t)(runtime->dma_addr + offset); - dev_addr = prtd->dma_params->per_address; - pr_debug("%s: mem_addr is %x\n dev_addr is %x\n", - __func__, mem_addr, dev_addr); - - ret = imx_dma_setup_single(prtd->dma_ch, mem_addr, - dma_size, dev_addr, - prtd->dma_params->transfer_type); - if (ret < 0) { - printk(KERN_ERR "Error %d configuring DMA\n", ret); - return ret; - } - imx_dma_enable(prtd->dma_ch); - - pr_debug("%s: transfer enabled\nmem_addr = %x\n", - __func__, (unsigned int) mem_addr); - pr_debug("dev_addr = %x\ndma_size = %d\n", - (unsigned int) dev_addr, dma_size); - - prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ - prtd->period++; - prtd->period %= runtime->periods; - } - return ret; -} - - -/** - * This is a callback which will be called - * when a TX transfer finishes. The call occurs - * in interrupt context. - * - * @param dat pointer to the structure of the current stream. - * - */ -static void audio_dma_irq(int channel, void *data) -{ - struct snd_pcm_substream *substream; - struct snd_pcm_runtime *runtime; - struct mx1_mx2_runtime_data *prtd; - unsigned int dma_size; - unsigned int previous_period; - unsigned int offset; - - substream = data; - runtime = substream->runtime; - prtd = runtime->private_data; - previous_period = prtd->periods; - dma_size = frames_to_bytes(runtime, runtime->period_size); - offset = dma_size * previous_period; - - prtd->tx_spin = 0; - prtd->periods++; - prtd->periods %= runtime->periods; - - pr_debug("%s: irq per %d offset %x\n", __func__, prtd->periods, offset); - - /* - * If we are getting a callback for an active stream then we inform - * the PCM middle layer we've finished a period - */ - if (prtd->active) - snd_pcm_period_elapsed(substream); - - /* - * Trig next DMA transfer - */ - dma_new_period(substream); -} - -/** - * This function configures the hardware to allow audio - * playback operations. It is called by ALSA framework. - * - * @param substream pointer to the structure of the current stream. - * - * @return 0 on success, -1 otherwise. - */ -static int -snd_mx1_mx2_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - - prtd->period = 0; - prtd->periods = 0; - - return 0; -} - -static int mx1_mx2_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - if (ret < 0) { - printk(KERN_ERR "%s: Error %d failed to malloc pcm pages \n", - __func__, ret); - return ret; - } - - pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_addr 0x(%x)\n", - __func__, (unsigned int)runtime->dma_addr); - pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_area 0x(%x)\n", - __func__, (unsigned int)runtime->dma_area); - pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_bytes 0x(%x)\n", - __func__, (unsigned int)runtime->dma_bytes); - - return ret; -} - -static int mx1_mx2_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - - imx_dma_free(prtd->dma_ch); - - snd_pcm_lib_free_pages(substream); - - return 0; -} - -static int mx1_mx2_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct mx1_mx2_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->tx_spin = 0; - /* requested stream startup */ - prtd->active = 1; - pr_debug("%s: starting dma_new_period\n", __func__); - ret = dma_new_period(substream); - break; - case SNDRV_PCM_TRIGGER_STOP: - /* requested stream shutdown */ - pr_debug("%s: stopping dma transfer\n", __func__); - ret = audio_stop_dma(substream); - break; - default: - ret = -EINVAL; - break; - } - - return ret; -} - -static snd_pcm_uframes_t -mx1_mx2_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - unsigned int offset = 0; - - /* tx_spin value is used here to check if a transfer is active */ - if (prtd->tx_spin) { - offset = (runtime->period_size * (prtd->periods)) + - (runtime->period_size >> 1); - if (offset >= runtime->buffer_size) - offset = runtime->period_size >> 1; - } else { - offset = (runtime->period_size * (prtd->periods)); - if (offset >= runtime->buffer_size) - offset = 0; - } - pr_debug("%s: pointer offset %x\n", __func__, offset); - - return offset; -} - -static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mx1_mx2_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; - int ret; - - snd_soc_set_runtime_hwparams(substream, &mx1_mx2_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(struct mx1_mx2_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - - runtime->private_data = prtd; - - if (!dma_data) - return -ENODEV; - - prtd->dma_params = dma_data; - - pr_debug("%s: Requesting dma channel (%s)\n", __func__, - prtd->dma_params->name); - ret = imx_dma_request_by_prio(prtd->dma_params->name, DMA_PRIO_HIGH); - if (ret < 0) { - printk(KERN_ERR "Error %d requesting dma channel\n", ret); - return ret; - } - prtd->dma_ch = ret; - imx_dma_config_burstlen(prtd->dma_ch, - prtd->dma_params->watermark_level); - - ret = imx_dma_config_channel(prtd->dma_ch, - prtd->dma_params->per_config, - prtd->dma_params->mem_config, - prtd->dma_params->event_id, 0); - - if (ret) { - pr_debug(KERN_ERR "Error %d configuring dma channel %d\n", - ret, prtd->dma_ch); - return ret; - } - - pr_debug("%s: Setting tx dma callback function\n", __func__); - ret = imx_dma_setup_handlers(prtd->dma_ch, - audio_dma_irq, NULL, - (void *)substream); - if (ret < 0) { - printk(KERN_ERR "Error %d setting dma callback function\n", ret); - return ret; - } - return 0; - - out: - return ret; -} - -static int mx1_mx2_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - - kfree(prtd); - - return 0; -} - -static int mx1_mx2_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops mx1_mx2_pcm_ops = { - .open = mx1_mx2_pcm_open, - .close = mx1_mx2_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = mx1_mx2_pcm_hw_params, - .hw_free = mx1_mx2_pcm_hw_free, - .prepare = snd_mx1_mx2_prepare, - .trigger = mx1_mx2_pcm_trigger, - .pointer = mx1_mx2_pcm_pointer, - .mmap = mx1_mx2_pcm_mmap, -}; - -static u64 mx1_mx2_pcm_dmamask = 0xffffffff; - -static int mx1_mx2_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = mx1_mx2_pcm_hardware.buffer_bytes_max; - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - /* Reserve uncached-buffered memory area for DMA */ - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - pr_debug("%s: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - __func__, (void *) buf->area, (void *) buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static void mx1_mx2_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static int mx1_mx2_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &mx1_mx2_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - pr_debug("%s: preallocate playback buffer\n", __func__); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - pr_debug("%s: preallocate capture buffer\n", __func__); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform mx1_mx2_soc_platform = { - .name = "mx1_mx2-audio", - .pcm_ops = &mx1_mx2_pcm_ops, - .pcm_new = mx1_mx2_pcm_new, - .pcm_free = mx1_mx2_pcm_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(mx1_mx2_soc_platform); - -static int __init mx1_mx2_soc_platform_init(void) -{ - return snd_soc_register_platform(&mx1_mx2_soc_platform); -} -module_init(mx1_mx2_soc_platform_init); - -static void __exit mx1_mx2_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&mx1_mx2_soc_platform); -} -module_exit(mx1_mx2_soc_platform_exit); - -MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com"); -MODULE_DESCRIPTION("Freescale i.MX2x, i.MX1x PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mx1_mx2-pcm.h b/sound/soc/imx/mx1_mx2-pcm.h deleted file mode 100644 index 2e528106570b..000000000000 --- a/sound/soc/imx/mx1_mx2-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * mx1_mx2-pcm.h :- ASoC platform header for Freescale i.MX1x, i.MX2x - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _MX1_MX2_PCM_H -#define _MX1_MX2_PCM_H - -/* DMA information for mx1_mx2 platforms */ -struct mx1_mx2_pcm_dma_params { - char *name; /* stream identifier */ - unsigned int transfer_type; /* READ or WRITE DMA transfer */ - dma_addr_t per_address; /* physical address of SSI fifo */ - int event_id; /* fixed DMA number for SSI fifo */ - int watermark_level; /* SSI fifo watermark level */ - int per_config; /* DMA Config flags for peripheral */ - int mem_config; /* DMA Config flags for RAM */ - }; - -/* platform data */ -extern struct snd_soc_platform mx1_mx2_soc_platform; - -#endif diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c deleted file mode 100644 index 0267d2d91685..000000000000 --- a/sound/soc/imx/mx27vis_wm8974.c +++ /dev/null @@ -1,317 +0,0 @@ -/* - * mx27vis_wm8974.c -- SoC audio for mx27vis - * - * Copyright 2009 Vista Silicon S.L. - * Author: Javier Martin - * javier.martin@vista-silicon.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include - - -#include "../codecs/wm8974.h" -#include "mx1_mx2-pcm.h" -#include "mxc-ssi.h" -#include -#include - -#define IGNORED_ARG 0 - - -static struct snd_soc_card mx27vis; - -/** - * This function connects SSI1 (HPCR1) as slave to - * SSI1 external signals (PPCR1) - * As slave, HPCR1 must set TFSDIR and TCLKDIR as inputs from - * port 4 - */ -void audmux_connect_1_4(void) -{ - pr_debug("AUDMUX: normal operation mode\n"); - /* Reset HPCR1 and PPCR1 */ - - DAM_HPCR1 = 0x00000000; - DAM_PPCR1 = 0x00000000; - - /* set to synchronous */ - DAM_HPCR1 |= AUDMUX_HPCR_SYN; - DAM_PPCR1 |= AUDMUX_PPCR_SYN; - - - /* set Rx sources 1 <--> 4 */ - DAM_HPCR1 |= AUDMUX_HPCR_RXDSEL(3); /* port 4 */ - DAM_PPCR1 |= AUDMUX_PPCR_RXDSEL(0); /* port 1 */ - - /* set Tx frame and Clock direction and source 4 --> 1 output */ - DAM_HPCR1 |= AUDMUX_HPCR_TFSDIR | AUDMUX_HPCR_TCLKDIR; - DAM_HPCR1 |= AUDMUX_HPCR_TFCSEL(3); /* TxDS and TxCclk from port 4 */ - - return; -} - -static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0, bclk = 0, fmt = 0, mclk = 0; - int ret = 0; - - /* - * The WM8974 is better at generating accurate audio clocks than the - * MX27 SSI controller, so we will use it as master when we can. - */ - switch (params_rate(params)) { - case 8000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - mclk = WM8974_MCLKDIV_12; - pll_out = 24576000; - break; - case 16000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - pll_out = 12288000; - break; - case 48000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_4; - pll_out = 12288000; - break; - case 96000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_2; - pll_out = 12288000; - break; - case 11025: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_16; - pll_out = 11289600; - break; - case 22050: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_8; - pll_out = 11289600; - break; - case 44100: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_4; - mclk = WM8974_MCLKDIV_2; - pll_out = 11289600; - break; - case 88200: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = codec_dai->ops->set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_SYNC | fmt); - if (ret < 0) { - printk(KERN_ERR "Error from codec DAI configuration\n"); - return ret; - } - - /* set cpu DAI configuration */ - ret = cpu_dai->ops->set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_SYNC | fmt); - if (ret < 0) { - printk(KERN_ERR "Error from cpu DAI configuration\n"); - return ret; - } - - /* Put DC field of STCCR to 1 (not zero) */ - ret = cpu_dai->ops->set_tdm_slot(cpu_dai, 0, 2); - - /* set the SSI system clock as input */ - ret = cpu_dai->ops->set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "Error when setting system SSI clk\n"); - return ret; - } - - /* set codec BCLK division for sample rate */ - ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_BCLKDIV, bclk); - if (ret < 0) { - printk(KERN_ERR "Error when setting BCLK division\n"); - return ret; - } - - - /* codec PLL input is 25 MHz */ - ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, - 25000000, pll_out); - if (ret < 0) { - printk(KERN_ERR "Error when setting PLL input\n"); - return ret; - } - - /*set codec MCLK division for sample rate */ - ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_MCLKDIV, mclk); - if (ret < 0) { - printk(KERN_ERR "Error when setting MCLK division\n"); - return ret; - } - - return 0; -} - -static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); -} - -/* - * mx27vis WM8974 HiFi DAI opserations. - */ -static struct snd_soc_ops mx27vis_hifi_ops = { - .hw_params = mx27vis_hifi_hw_params, - .hw_free = mx27vis_hifi_hw_free, -}; - - -static int mx27vis_suspend(struct platform_device *pdev, pm_message_t state) -{ - return 0; -} - -static int mx27vis_resume(struct platform_device *pdev) -{ - return 0; -} - -static int mx27vis_probe(struct platform_device *pdev) -{ - int ret = 0; - - ret = get_ssi_clk(0, &pdev->dev); - - if (ret < 0) { - printk(KERN_ERR "%s: cant get ssi clock\n", __func__); - return ret; - } - - - return 0; -} - -static int mx27vis_remove(struct platform_device *pdev) -{ - put_ssi_clk(0); - return 0; -} - -static struct snd_soc_dai_link mx27vis_dai[] = { -{ /* Hifi Playback*/ - .name = "WM8974", - .stream_name = "WM8974 HiFi", - .cpu_dai = &imx_ssi_pcm_dai[0], - .codec_dai = &wm8974_dai, - .ops = &mx27vis_hifi_ops, -}, -}; - -static struct snd_soc_card mx27vis = { - .name = "mx27vis", - .platform = &mx1_mx2_soc_platform, - .probe = mx27vis_probe, - .remove = mx27vis_remove, - .suspend_pre = mx27vis_suspend, - .resume_post = mx27vis_resume, - .dai_link = mx27vis_dai, - .num_links = ARRAY_SIZE(mx27vis_dai), -}; - -static struct snd_soc_device mx27vis_snd_devdata = { - .card = &mx27vis, - .codec_dev = &soc_codec_dev_wm8974, -}; - -static struct platform_device *mx27vis_snd_device; - -/* Temporal definition of board specific behaviour */ -void gpio_ssi_active(int ssi_num) -{ - int ret = 0; - - unsigned int ssi1_pins[] = { - PC20_PF_SSI1_FS, - PC21_PF_SSI1_RXD, - PC22_PF_SSI1_TXD, - PC23_PF_SSI1_CLK, - }; - unsigned int ssi2_pins[] = { - PC24_PF_SSI2_FS, - PC25_PF_SSI2_RXD, - PC26_PF_SSI2_TXD, - PC27_PF_SSI2_CLK, - }; - if (ssi_num == 0) - ret = mxc_gpio_setup_multiple_pins(ssi1_pins, - ARRAY_SIZE(ssi1_pins), "USB OTG"); - else - ret = mxc_gpio_setup_multiple_pins(ssi2_pins, - ARRAY_SIZE(ssi2_pins), "USB OTG"); - if (ret) - printk(KERN_ERR "Error requesting ssi %x pins\n", ssi_num); -} - - -static int __init mx27vis_init(void) -{ - int ret; - - mx27vis_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_snd_device, &mx27vis_snd_devdata); - mx27vis_snd_devdata.dev = &mx27vis_snd_device->dev; - ret = platform_device_add(mx27vis_snd_device); - - if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_snd_device); - } - - /* WM8974 uses SSI1 (HPCR1) via AUDMUX port 4 for audio (PPCR1) */ - gpio_ssi_active(0); - audmux_connect_1_4(); - - return ret; -} - -static void __exit mx27vis_exit(void) -{ - /* We should call some "ssi_gpio_inactive()" properly */ -} - -module_init(mx27vis_init); -module_exit(mx27vis_exit); - - -MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com"); -MODULE_DESCRIPTION("ALSA SoC WM8974 mx27vis"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c deleted file mode 100644 index ccdefe60e752..000000000000 --- a/sound/soc/imx/mxc-ssi.c +++ /dev/null @@ -1,860 +0,0 @@ -/* - * mxc-ssi.c -- SSI driver for Freescale IMX - * - * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Based on mxc-alsa-mc13783 (C) 2006 Freescale. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * TODO: - * Need to rework SSI register defs when new defs go into mainline. - * Add support for TDM and FIFO 1. - * Add support for i.mx3x DMA interface. - * - */ - - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "mxc-ssi.h" -#include "mx1_mx2-pcm.h" - -#define SSI1_PORT 0 -#define SSI2_PORT 1 - -static int ssi_active[2] = {0, 0}; - -/* DMA information for mx1_mx2 platforms */ -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out0 = { - .name = "SSI1 PCM Stereo out 0", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI1_BASE_ADDR + STX0, - .event_id = DMA_REQ_SSI1_TX0, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out1 = { - .name = "SSI1 PCM Stereo out 1", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI1_BASE_ADDR + STX1, - .event_id = DMA_REQ_SSI1_TX1, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in0 = { - .name = "SSI1 PCM Stereo in 0", - .transfer_type = DMA_MODE_READ, - .per_address = SSI1_BASE_ADDR + SRX0, - .event_id = DMA_REQ_SSI1_RX0, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in1 = { - .name = "SSI1 PCM Stereo in 1", - .transfer_type = DMA_MODE_READ, - .per_address = SSI1_BASE_ADDR + SRX1, - .event_id = DMA_REQ_SSI1_RX1, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out0 = { - .name = "SSI2 PCM Stereo out 0", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI2_BASE_ADDR + STX0, - .event_id = DMA_REQ_SSI2_TX0, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out1 = { - .name = "SSI2 PCM Stereo out 1", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI2_BASE_ADDR + STX1, - .event_id = DMA_REQ_SSI2_TX1, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in0 = { - .name = "SSI2 PCM Stereo in 0", - .transfer_type = DMA_MODE_READ, - .per_address = SSI2_BASE_ADDR + SRX0, - .event_id = DMA_REQ_SSI2_RX0, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in1 = { - .name = "SSI2 PCM Stereo in 1", - .transfer_type = DMA_MODE_READ, - .per_address = SSI2_BASE_ADDR + SRX1, - .event_id = DMA_REQ_SSI2_RX1, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct clk *ssi_clk0, *ssi_clk1; - -int get_ssi_clk(int ssi, struct device *dev) -{ - switch (ssi) { - case 0: - ssi_clk0 = clk_get(dev, "ssi1"); - if (IS_ERR(ssi_clk0)) - return PTR_ERR(ssi_clk0); - return 0; - case 1: - ssi_clk1 = clk_get(dev, "ssi2"); - if (IS_ERR(ssi_clk1)) - return PTR_ERR(ssi_clk1); - return 0; - default: - return -EINVAL; - } -} -EXPORT_SYMBOL(get_ssi_clk); - -void put_ssi_clk(int ssi) -{ - switch (ssi) { - case 0: - clk_put(ssi_clk0); - ssi_clk0 = NULL; - break; - case 1: - clk_put(ssi_clk1); - ssi_clk1 = NULL; - break; - } -} -EXPORT_SYMBOL(put_ssi_clk); - -/* - * SSI system clock configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - u32 scr; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - scr = SSI1_SCR; - pr_debug("%s: SCR for SSI1 is %x\n", __func__, scr); - } else { - scr = SSI2_SCR; - pr_debug("%s: SCR for SSI2 is %x\n", __func__, scr); - } - - if (scr & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - - switch (clk_id) { - case IMX_SSP_SYS_CLK: - if (dir == SND_SOC_CLOCK_OUT) { - scr |= SSI_SCR_SYS_CLK_EN; - pr_debug("%s: clk of is output\n", __func__); - } else { - scr &= ~SSI_SCR_SYS_CLK_EN; - pr_debug("%s: clk of is input\n", __func__); - } - break; - default: - return -EINVAL; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - pr_debug("%s: writeback of SSI1_SCR\n", __func__); - SSI1_SCR = scr; - } else { - pr_debug("%s: writeback of SSI2_SCR\n", __func__); - SSI2_SCR = scr; - } - - return 0; -} - -/* - * SSI Clock dividers - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - u32 stccr, srccr; - - pr_debug("%s\n", __func__); - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - if (SSI1_SCR & SSI_SCR_SSIEN) - return 0; - srccr = SSI1_STCCR; - stccr = SSI1_STCCR; - } else { - if (SSI2_SCR & SSI_SCR_SSIEN) - return 0; - srccr = SSI2_STCCR; - stccr = SSI2_STCCR; - } - - switch (div_id) { - case IMX_SSI_TX_DIV_2: - stccr &= ~SSI_STCCR_DIV2; - stccr |= div; - break; - case IMX_SSI_TX_DIV_PSR: - stccr &= ~SSI_STCCR_PSR; - stccr |= div; - break; - case IMX_SSI_TX_DIV_PM: - stccr &= ~0xff; - stccr |= SSI_STCCR_PM(div); - break; - case IMX_SSI_RX_DIV_2: - stccr &= ~SSI_STCCR_DIV2; - stccr |= div; - break; - case IMX_SSI_RX_DIV_PSR: - stccr &= ~SSI_STCCR_PSR; - stccr |= div; - break; - case IMX_SSI_RX_DIV_PM: - stccr &= ~0xff; - stccr |= SSI_STCCR_PM(div); - break; - default: - return -EINVAL; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STCCR = stccr; - SSI1_SRCCR = srccr; - } else { - SSI2_STCCR = stccr; - SSI2_SRCCR = srccr; - } - return 0; -} - -/* - * SSI Network Mode or TDM slots configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) -{ - u32 stmsk, srmsk, stccr; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - if (SSI1_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - stccr = SSI1_STCCR; - } else { - if (SSI2_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - stccr = SSI2_STCCR; - } - - stmsk = srmsk = mask; - stccr &= ~SSI_STCCR_DC_MASK; - stccr |= SSI_STCCR_DC(slots - 1); - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STMSK = stmsk; - SSI1_SRMSK = srmsk; - SSI1_SRCCR = SSI1_STCCR = stccr; - } else { - SSI2_STMSK = stmsk; - SSI2_SRMSK = srmsk; - SSI2_SRCCR = SSI2_STCCR = stccr; - } - - return 0; -} - -/* - * SSI DAI format configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S. - */ -static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 stcr = 0, srcr = 0, scr; - - /* - * This is done to avoid this function to modify - * previous set values in stcr - */ - stcr = SSI1_STCR; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - scr = SSI1_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); - else - scr = SSI2_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); - - if (scr & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - - /* DAI mode */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - /* data on rising edge of bclk, frame low 1clk before data */ - stcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; - srcr |= SSI_SRCR_RFSI | SSI_SRCR_REFS | SSI_SRCR_RXBIT0; - break; - case SND_SOC_DAIFMT_LEFT_J: - /* data on rising edge of bclk, frame high with data */ - stcr |= SSI_STCR_TXBIT0; - srcr |= SSI_SRCR_RXBIT0; - break; - case SND_SOC_DAIFMT_DSP_B: - /* data on rising edge of bclk, frame high with data */ - stcr |= SSI_STCR_TFSL; - srcr |= SSI_SRCR_RFSL; - break; - case SND_SOC_DAIFMT_DSP_A: - /* data on rising edge of bclk, frame high 1clk before data */ - stcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; - srcr |= SSI_SRCR_RFSL | SSI_SRCR_REFS; - break; - } - - /* DAI clock inversion */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: - stcr |= SSI_STCR_TFSI; - stcr &= ~SSI_STCR_TSCKP; - srcr |= SSI_SRCR_RFSI; - srcr &= ~SSI_SRCR_RSCKP; - break; - case SND_SOC_DAIFMT_IB_NF: - stcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); - srcr &= ~(SSI_SRCR_RSCKP | SSI_SRCR_RFSI); - break; - case SND_SOC_DAIFMT_NB_IF: - stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; - srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP; - break; - case SND_SOC_DAIFMT_NB_NF: - stcr &= ~SSI_STCR_TFSI; - stcr |= SSI_STCR_TSCKP; - srcr &= ~SSI_SRCR_RFSI; - srcr |= SSI_SRCR_RSCKP; - break; - } - - /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; - srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; - break; - case SND_SOC_DAIFMT_CBM_CFS: - stcr |= SSI_STCR_TFDIR; - srcr |= SSI_SRCR_RFDIR; - break; - case SND_SOC_DAIFMT_CBS_CFM: - stcr |= SSI_STCR_TXDIR; - srcr |= SSI_SRCR_RXDIR; - break; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STCR = stcr; - SSI1_SRCR = srcr; - SSI1_SCR = scr; - } else { - SSI2_STCR = stcr; - SSI2_SRCR = srcr; - SSI2_SCR = scr; - } - - return 0; -} - -static int imx_ssi_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set up TX DMA params */ - switch (cpu_dai->id) { - case IMX_DAI_SSI0: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out0; - break; - case IMX_DAI_SSI1: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out1; - break; - case IMX_DAI_SSI2: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out0; - break; - case IMX_DAI_SSI3: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out1; - } - pr_debug("%s: (playback)\n", __func__); - } else { - /* set up RX DMA params */ - switch (cpu_dai->id) { - case IMX_DAI_SSI0: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in0; - break; - case IMX_DAI_SSI1: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in1; - break; - case IMX_DAI_SSI2: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in0; - break; - case IMX_DAI_SSI3: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in1; - } - pr_debug("%s: (capture)\n", __func__); - } - - /* - * we cant really change any SSI values after SSI is enabled - * need to fix in software for max flexibility - lrg - */ - if (cpu_dai->active) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - - /* reset the SSI port - Sect 45.4.4 */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - - if (!ssi_clk0) - return -EINVAL; - - if (ssi_active[SSI1_PORT]++) { - pr_debug("%s: exit before reset\n", __func__); - return 0; - } - - /* SSI1 Reset */ - SSI1_SCR = 0; - - SSI1_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | - SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | - SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | - SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); - } else { - - if (!ssi_clk1) - return -EINVAL; - - if (ssi_active[SSI2_PORT]++) { - pr_debug("%s: exit before reset\n", __func__); - return 0; - } - - /* SSI2 Reset */ - SSI2_SCR = 0; - - SSI2_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | - SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | - SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | - SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); - } - - return 0; -} - -int imx_ssi_hw_tx_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - u32 stccr, stcr, sier; - - pr_debug("%s\n", __func__); - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - stccr = SSI1_STCCR & ~SSI_STCCR_WL_MASK; - stcr = SSI1_STCR; - sier = SSI1_SIER; - } else { - stccr = SSI2_STCCR & ~SSI_STCCR_WL_MASK; - stcr = SSI2_STCR; - sier = SSI2_SIER; - } - - /* DAI data (word) size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - stccr |= SSI_STCCR_WL(16); - break; - case SNDRV_PCM_FORMAT_S20_3LE: - stccr |= SSI_STCCR_WL(20); - break; - case SNDRV_PCM_FORMAT_S24_LE: - stccr |= SSI_STCCR_WL(24); - break; - } - - /* enable interrupts */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - stcr |= SSI_STCR_TFEN0; - else - stcr |= SSI_STCR_TFEN1; - sier |= SSI_SIER_TDMAE; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STCR = stcr; - SSI1_STCCR = stccr; - SSI1_SIER = sier; - } else { - SSI2_STCR = stcr; - SSI2_STCCR = stccr; - SSI2_SIER = sier; - } - - return 0; -} - -int imx_ssi_hw_rx_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - u32 srccr, srcr, sier; - - pr_debug("%s\n", __func__); - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - srccr = SSI1_SRCCR & ~SSI_SRCCR_WL_MASK; - srcr = SSI1_SRCR; - sier = SSI1_SIER; - } else { - srccr = SSI2_SRCCR & ~SSI_SRCCR_WL_MASK; - srcr = SSI2_SRCR; - sier = SSI2_SIER; - } - - /* DAI data (word) size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - srccr |= SSI_SRCCR_WL(16); - break; - case SNDRV_PCM_FORMAT_S20_3LE: - srccr |= SSI_SRCCR_WL(20); - break; - case SNDRV_PCM_FORMAT_S24_LE: - srccr |= SSI_SRCCR_WL(24); - break; - } - - /* enable interrupts */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - srcr |= SSI_SRCR_RFEN0; - else - srcr |= SSI_SRCR_RFEN1; - sier |= SSI_SIER_RDMAE; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_SRCR = srcr; - SSI1_SRCCR = srccr; - SSI1_SIER = sier; - } else { - SSI2_SRCR = srcr; - SSI2_SRCCR = srccr; - SSI2_SIER = sier; - } - - return 0; -} - -/* - * Should only be called when port is inactive (i.e. SSIEN = 0), - * although can be called multiple times by upper layers. - */ -int imx_ssi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - int ret; - - /* cant change any parameters when SSI is running */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - if (SSI1_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - } else { - if (SSI2_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - } - - /* - * Configure both tx and rx params with the same settings. This is - * really a harware restriction because SSI must be disabled until - * we can change those values. If there is an active audio stream in - * one direction, enabling the other direction with different - * settings would mean disturbing the running one. - */ - ret = imx_ssi_hw_tx_params(substream, params); - if (ret < 0) - return ret; - return imx_ssi_hw_rx_params(substream, params); -} - -int imx_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - pr_debug("%s\n", __func__); - - /* Enable clks here to follow SSI recommended init sequence */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - ret = clk_enable(ssi_clk0); - if (ret < 0) - printk(KERN_ERR "Unable to enable ssi_clk0\n"); - } else { - ret = clk_enable(ssi_clk1); - if (ret < 0) - printk(KERN_ERR "Unable to enable ssi_clk1\n"); - } - - return 0; -} - -static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - u32 scr; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - scr = SSI1_SCR; - else - scr = SSI2_SCR; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - scr |= SSI_SCR_TE | SSI_SCR_SSIEN; - else - scr |= SSI_SCR_RE | SSI_SCR_SSIEN; - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - scr &= ~SSI_SCR_TE; - else - scr &= ~SSI_SCR_RE; - break; - default: - return -EINVAL; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - SSI1_SCR = scr; - else - SSI2_SCR = scr; - - return 0; -} - -static void imx_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - /* shutdown SSI if neither Tx or Rx is active */ - if (!cpu_dai->active) { - - if (cpu_dai->id == IMX_DAI_SSI0 || - cpu_dai->id == IMX_DAI_SSI2) { - - if (--ssi_active[SSI1_PORT] > 1) - return; - - SSI1_SCR = 0; - clk_disable(ssi_clk0); - } else { - if (--ssi_active[SSI2_PORT]) - return; - SSI2_SCR = 0; - clk_disable(ssi_clk1); - } - } -} - -#ifdef CONFIG_PM -static int imx_ssi_suspend(struct platform_device *dev, - struct snd_soc_dai *dai) -{ - return 0; -} - -static int imx_ssi_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - return 0; -} - -#else -#define imx_ssi_suspend NULL -#define imx_ssi_resume NULL -#endif - -#define IMX_SSI_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ - SNDRV_PCM_RATE_96000) - -#define IMX_SSI_BITS \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE) - -static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { - .startup = imx_ssi_startup, - .shutdown = imx_ssi_shutdown, - .trigger = imx_ssi_trigger, - .prepare = imx_ssi_prepare, - .hw_params = imx_ssi_hw_params, - .set_sysclk = imx_ssi_set_dai_sysclk, - .set_clkdiv = imx_ssi_set_dai_clkdiv, - .set_fmt = imx_ssi_set_dai_fmt, - .set_tdm_slot = imx_ssi_set_dai_tdm_slot, -}; - -struct snd_soc_dai imx_ssi_pcm_dai[] = { -{ - .name = "imx-i2s-1-0", - .id = IMX_DAI_SSI0, - .suspend = imx_ssi_suspend, - .resume = imx_ssi_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -{ - .name = "imx-i2s-2-0", - .id = IMX_DAI_SSI1, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -{ - .name = "imx-i2s-1-1", - .id = IMX_DAI_SSI2, - .suspend = imx_ssi_suspend, - .resume = imx_ssi_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -{ - .name = "imx-i2s-2-1", - .id = IMX_DAI_SSI3, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -}; -EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); - -static int __init imx_ssi_init(void) -{ - return snd_soc_register_dais(imx_ssi_pcm_dai, - ARRAY_SIZE(imx_ssi_pcm_dai)); -} - -static void __exit imx_ssi_exit(void) -{ - snd_soc_unregister_dais(imx_ssi_pcm_dai, - ARRAY_SIZE(imx_ssi_pcm_dai)); -} - -module_init(imx_ssi_init); -module_exit(imx_ssi_exit); -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com"); -MODULE_DESCRIPTION("i.MX ASoC I2S driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.h b/sound/soc/imx/mxc-ssi.h deleted file mode 100644 index 12bbdc9c7ecd..000000000000 --- a/sound/soc/imx/mxc-ssi.h +++ /dev/null @@ -1,238 +0,0 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _IMX_SSI_H -#define _IMX_SSI_H - -#include - -/* SSI regs definition - MOVE to /arch/arm/plat-mxc/include/mach/ when stable */ -#define SSI1_IO_BASE_ADDR IO_ADDRESS(SSI1_BASE_ADDR) -#define SSI2_IO_BASE_ADDR IO_ADDRESS(SSI2_BASE_ADDR) - -#define STX0 0x00 -#define STX1 0x04 -#define SRX0 0x08 -#define SRX1 0x0c -#define SCR 0x10 -#define SISR 0x14 -#define SIER 0x18 -#define STCR 0x1c -#define SRCR 0x20 -#define STCCR 0x24 -#define SRCCR 0x28 -#define SFCSR 0x2c -#define STR 0x30 -#define SOR 0x34 -#define SACNT 0x38 -#define SACADD 0x3c -#define SACDAT 0x40 -#define SATAG 0x44 -#define STMSK 0x48 -#define SRMSK 0x4c - -#define SSI1_STX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX0))) -#define SSI1_STX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX1))) -#define SSI1_SRX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX0))) -#define SSI1_SRX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX1))) -#define SSI1_SCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SCR))) -#define SSI1_SISR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SISR))) -#define SSI1_SIER (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SIER))) -#define SSI1_STCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCR))) -#define SSI1_SRCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCR))) -#define SSI1_STCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCCR))) -#define SSI1_SRCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCCR))) -#define SSI1_SFCSR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SFCSR))) -#define SSI1_STR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STR))) -#define SSI1_SOR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SOR))) -#define SSI1_SACNT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACNT))) -#define SSI1_SACADD (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACADD))) -#define SSI1_SACDAT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACDAT))) -#define SSI1_SATAG (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SATAG))) -#define SSI1_STMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STMSK))) -#define SSI1_SRMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRMSK))) - - -#define SSI2_STX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX0))) -#define SSI2_STX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX1))) -#define SSI2_SRX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX0))) -#define SSI2_SRX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX1))) -#define SSI2_SCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SCR))) -#define SSI2_SISR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SISR))) -#define SSI2_SIER (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SIER))) -#define SSI2_STCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCR))) -#define SSI2_SRCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCR))) -#define SSI2_STCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCCR))) -#define SSI2_SRCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCCR))) -#define SSI2_SFCSR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SFCSR))) -#define SSI2_STR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STR))) -#define SSI2_SOR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SOR))) -#define SSI2_SACNT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACNT))) -#define SSI2_SACADD (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACADD))) -#define SSI2_SACDAT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACDAT))) -#define SSI2_SATAG (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SATAG))) -#define SSI2_STMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STMSK))) -#define SSI2_SRMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRMSK))) - -#define SSI_SCR_CLK_IST (1 << 9) -#define SSI_SCR_TCH_EN (1 << 8) -#define SSI_SCR_SYS_CLK_EN (1 << 7) -#define SSI_SCR_I2S_MODE_NORM (0 << 5) -#define SSI_SCR_I2S_MODE_MSTR (1 << 5) -#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) -#define SSI_SCR_SYN (1 << 4) -#define SSI_SCR_NET (1 << 3) -#define SSI_SCR_RE (1 << 2) -#define SSI_SCR_TE (1 << 1) -#define SSI_SCR_SSIEN (1 << 0) - -#define SSI_SISR_CMDAU (1 << 18) -#define SSI_SISR_CMDDU (1 << 17) -#define SSI_SISR_RXT (1 << 16) -#define SSI_SISR_RDR1 (1 << 15) -#define SSI_SISR_RDR0 (1 << 14) -#define SSI_SISR_TDE1 (1 << 13) -#define SSI_SISR_TDE0 (1 << 12) -#define SSI_SISR_ROE1 (1 << 11) -#define SSI_SISR_ROE0 (1 << 10) -#define SSI_SISR_TUE1 (1 << 9) -#define SSI_SISR_TUE0 (1 << 8) -#define SSI_SISR_TFS (1 << 7) -#define SSI_SISR_RFS (1 << 6) -#define SSI_SISR_TLS (1 << 5) -#define SSI_SISR_RLS (1 << 4) -#define SSI_SISR_RFF1 (1 << 3) -#define SSI_SISR_RFF0 (1 << 2) -#define SSI_SISR_TFE1 (1 << 1) -#define SSI_SISR_TFE0 (1 << 0) - -#define SSI_SIER_RDMAE (1 << 22) -#define SSI_SIER_RIE (1 << 21) -#define SSI_SIER_TDMAE (1 << 20) -#define SSI_SIER_TIE (1 << 19) -#define SSI_SIER_CMDAU_EN (1 << 18) -#define SSI_SIER_CMDDU_EN (1 << 17) -#define SSI_SIER_RXT_EN (1 << 16) -#define SSI_SIER_RDR1_EN (1 << 15) -#define SSI_SIER_RDR0_EN (1 << 14) -#define SSI_SIER_TDE1_EN (1 << 13) -#define SSI_SIER_TDE0_EN (1 << 12) -#define SSI_SIER_ROE1_EN (1 << 11) -#define SSI_SIER_ROE0_EN (1 << 10) -#define SSI_SIER_TUE1_EN (1 << 9) -#define SSI_SIER_TUE0_EN (1 << 8) -#define SSI_SIER_TFS_EN (1 << 7) -#define SSI_SIER_RFS_EN (1 << 6) -#define SSI_SIER_TLS_EN (1 << 5) -#define SSI_SIER_RLS_EN (1 << 4) -#define SSI_SIER_RFF1_EN (1 << 3) -#define SSI_SIER_RFF0_EN (1 << 2) -#define SSI_SIER_TFE1_EN (1 << 1) -#define SSI_SIER_TFE0_EN (1 << 0) - -#define SSI_STCR_TXBIT0 (1 << 9) -#define SSI_STCR_TFEN1 (1 << 8) -#define SSI_STCR_TFEN0 (1 << 7) -#define SSI_STCR_TFDIR (1 << 6) -#define SSI_STCR_TXDIR (1 << 5) -#define SSI_STCR_TSHFD (1 << 4) -#define SSI_STCR_TSCKP (1 << 3) -#define SSI_STCR_TFSI (1 << 2) -#define SSI_STCR_TFSL (1 << 1) -#define SSI_STCR_TEFS (1 << 0) - -#define SSI_SRCR_RXBIT0 (1 << 9) -#define SSI_SRCR_RFEN1 (1 << 8) -#define SSI_SRCR_RFEN0 (1 << 7) -#define SSI_SRCR_RFDIR (1 << 6) -#define SSI_SRCR_RXDIR (1 << 5) -#define SSI_SRCR_RSHFD (1 << 4) -#define SSI_SRCR_RSCKP (1 << 3) -#define SSI_SRCR_RFSI (1 << 2) -#define SSI_SRCR_RFSL (1 << 1) -#define SSI_SRCR_REFS (1 << 0) - -#define SSI_STCCR_DIV2 (1 << 18) -#define SSI_STCCR_PSR (1 << 15) -#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) -#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) -#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) -#define SSI_STCCR_WL_MASK (0xf << 13) -#define SSI_STCCR_DC_MASK (0x1f << 8) -#define SSI_STCCR_PM_MASK (0xff << 0) - -#define SSI_SRCCR_DIV2 (1 << 18) -#define SSI_SRCCR_PSR (1 << 15) -#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) -#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) -#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) -#define SSI_SRCCR_WL_MASK (0xf << 13) -#define SSI_SRCCR_DC_MASK (0x1f << 8) -#define SSI_SRCCR_PM_MASK (0xff << 0) - - -#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) -#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) -#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) -#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) -#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) -#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) -#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) -#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) - -#define SSI_STR_TEST (1 << 15) -#define SSI_STR_RCK2TCK (1 << 14) -#define SSI_STR_RFS2TFS (1 << 13) -#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) -#define SSI_STR_TXD2RXD (1 << 7) -#define SSI_STR_TCK2RCK (1 << 6) -#define SSI_STR_TFS2RFS (1 << 5) -#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) - -#define SSI_SOR_CLKOFF (1 << 6) -#define SSI_SOR_RX_CLR (1 << 5) -#define SSI_SOR_TX_CLR (1 << 4) -#define SSI_SOR_INIT (1 << 3) -#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) -#define SSI_SOR_SYNRST (1 << 0) - -#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) -#define SSI_SACNT_WR (x << 4) -#define SSI_SACNT_RD (x << 3) -#define SSI_SACNT_TIF (x << 2) -#define SSI_SACNT_FV (x << 1) -#define SSI_SACNT_AC97EN (x << 0) - -/* Watermarks for FIFO's */ -#define TXFIFO_WATERMARK 0x4 -#define RXFIFO_WATERMARK 0x4 - -/* i.MX DAI SSP ID's */ -#define IMX_DAI_SSI0 0 /* SSI1 FIFO 0 */ -#define IMX_DAI_SSI1 1 /* SSI1 FIFO 1 */ -#define IMX_DAI_SSI2 2 /* SSI2 FIFO 0 */ -#define IMX_DAI_SSI3 3 /* SSI2 FIFO 1 */ - -/* SSI clock sources */ -#define IMX_SSP_SYS_CLK 0 - -/* SSI audio dividers */ -#define IMX_SSI_TX_DIV_2 0 -#define IMX_SSI_TX_DIV_PSR 1 -#define IMX_SSI_TX_DIV_PM 2 -#define IMX_SSI_RX_DIV_2 3 -#define IMX_SSI_RX_DIV_PSR 4 -#define IMX_SSI_RX_DIV_PM 5 - - -/* SSI Div 2 */ -#define IMX_SSI_DIV_2_OFF (~SSI_STCCR_DIV2) -#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2 - -extern struct snd_soc_dai imx_ssi_pcm_dai[4]; -extern int get_ssi_clk(int ssi, struct device *dev); -extern void put_ssi_clk(int ssi); -#endif -- cgit v1.2.2 From eaa9b3a748539651f50e3a234c8854e1b42a839a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Jan 2010 13:09:33 +0100 Subject: ALSA: hda - Fix capture on Sony VAIO with single input Sony VAIO VGN-P11G with ALC262 codec has only one input pin, and the recording doesn't work with model=auto because ALC262 parser sets the wrong cap NIDs to choose the route and the default route for the sole input pin wasn't initialized properly. This patch solves these issues. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++------ 1 file changed, 54 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index abae1007cea2..3f92def752fd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1230,6 +1230,8 @@ static void alc_init_auto_mic(struct hda_codec *codec) return; /* invalid entry */ } } + if (!ext || !fixed) + return; if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) return; /* no unsol support */ snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n", @@ -4812,6 +4814,49 @@ static void fixup_automic_adc(struct hda_codec *codec) spec->auto_mic = 0; /* disable auto-mic to be sure */ } +/* choose the ADC/MUX containing the input pin and initialize the setup */ +static void fixup_single_adc(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + int i; + + /* search for the input pin; there must be only one */ + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (spec->autocfg.input_pins[i]) { + pin = spec->autocfg.input_pins[i]; + break; + } + } + if (!pin) + return; + + /* set the default connection to that pin */ + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t cap = spec->capsrc_nids ? + spec->capsrc_nids[i] : spec->adc_nids[i]; + int idx; + + idx = get_connection_index(codec, cap, pin); + if (idx < 0) + continue; + /* use only this ADC */ + if (spec->capsrc_nids) + spec->capsrc_nids += i; + spec->adc_nids += i; + spec->num_adc_nids = 1; + /* select or unmute this route */ + if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { + snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, + HDA_AMP_MUTE, 0); + } else { + snd_hda_codec_write_cache(codec, cap, 0, + AC_VERB_SET_CONNECT_SEL, idx); + } + return; + } +} + static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4824,14 +4869,15 @@ static void set_capture_mixer(struct hda_codec *codec) alc_capture_mixer3 }, }; if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { - int mux; - if (spec->auto_mic) { - mux = 0; + int mux = 0; + if (spec->auto_mic) fixup_automic_adc(codec); - } else if (spec->input_mux && spec->input_mux->num_items > 1) - mux = 1; - else - mux = 0; + else if (spec->input_mux) { + if (spec->input_mux->num_items > 1) + mux = 1; + else if (spec->input_mux->num_items == 1) + fixup_single_adc(codec); + } spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; } } @@ -11203,7 +11249,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, } #define alc262_auto_create_input_ctls \ - alc880_auto_create_input_ctls + alc882_auto_create_input_ctls /* * generic initialization of ADC, input mixers and output mixers -- cgit v1.2.2 From b05f5c13d5bc2fa9945c9534f8881396555290a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 17 Jan 2010 16:45:06 +0000 Subject: ASoC: Mark new i.MX drivers as BROKEN until arch/arm merged Currently they don't build due to cross tree dependencies, they will be reenabled once the arch/arm side has merged. Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 84a25e61bed8..5f006f0d03dc 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,6 +1,6 @@ config SND_IMX_SOC tristate "SoC Audio for Freecale i.MX CPUs" - depends on ARCH_MXC + depends on ARCH_MXC && BROKEN select SND_PCM select FIQ select SND_SOC_AC97_BUS -- cgit v1.2.2 From 808c569f3609b37642d1e08373e3de829b99d0f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:16:24 +0100 Subject: ALSA: Remove warning message for invalid OSS minor ranges When a card instance with a higher card number is registered, warning messages are spewed eventually with stack traces due to the invalid minor number for OSS device registration. For example, thinkpad-acpi registers the card number 29 as default, and you'll see always these messages. This is rather confusing (and worries users), thus better to return simply the error code. Signed-off-by: Takashi Iwai --- sound/core/sound_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 7fe12264ff80..0c164e5e4322 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev) default: return -EINVAL; } - if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS)) + if (minor < 0 || minor >= SNDRV_OSS_MINORS) return -EINVAL; return minor; } -- cgit v1.2.2 From 3e879d7bac705be4813a0ec9560cbe31db4b269f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:49:50 +0100 Subject: ALSA: pcm - Remove unneeded ifdef pgprot_noncached Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a870fe696578..5df0d21f18b3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3162,9 +3162,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, long size; unsigned long offset; -#ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); -#endif area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; -- cgit v1.2.2 From c32d977b8157bf67cdf47729ce7dd054a26eb534 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:58:57 +0100 Subject: ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffers pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd need non-cached behavior more or less, even for the intermediate ring- buffers. Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 9 +++++++++ sound/drivers/vx/vx_pcm.c | 2 ++ sound/mips/sgio2audio.c | 3 +++ sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 + sound/usb/ua101.c | 2 ++ sound/usb/usbaudio.c | 2 ++ 6 files changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 5df0d21f18b3..88fff44702a4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3176,6 +3176,15 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ +/* mmap callback with pgprot_noncached */ +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); + return snd_pcm_default_mmap(substream, area); +} +EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached); + /* * mmap DMA buffer */ diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index c8385d26a16f..35a2f71a6af5 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -905,6 +905,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; @@ -1125,6 +1126,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9b486beeb932..6aff217379d9 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -691,6 +691,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -703,6 +704,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -715,6 +717,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; /* diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 0afa683c900e..0d668f471620 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -277,6 +277,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = { .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 16dc7bd5e120..4f4ccdf70dd0 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -911,6 +911,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops playback_pcm_ops = { @@ -923,6 +924,7 @@ static struct snd_pcm_ops playback_pcm_ops = { .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct uac_format_type_i_discrete_descriptor * diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4ada98e16309..b8e0b8fda607 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1997,6 +1997,7 @@ static struct snd_pcm_ops snd_usb_playback_ops = { .trigger = snd_usb_pcm_playback_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_usb_capture_ops = { @@ -2009,6 +2010,7 @@ static struct snd_pcm_ops snd_usb_capture_ops = { .trigger = snd_usb_pcm_capture_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; -- cgit v1.2.2 From a32f66746c635ebf2341d99b3d4c0cc1c11b2cbf Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 18 Jan 2010 15:40:56 +0100 Subject: sound: seq_timer: simplify snd_seq_timer_set_tick_resolution() parameters As snd_seq_timer_set_tick_resolution() is always called with the same three fields of struct snd_seq_timer, it suffices to give that as the only parameter. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/core/seq/seq_timer.c | 27 +++++++++++++-------------- 1 file changed, 13 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index f745c317d6af..160b1bd0cd62 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -33,22 +33,21 @@ #define SKEW_BASE 0x10000 /* 16bit shift */ -static void snd_seq_timer_set_tick_resolution(struct snd_seq_timer_tick *tick, - int tempo, int ppq) +static void snd_seq_timer_set_tick_resolution(struct snd_seq_timer *tmr) { - if (tempo < 1000000) - tick->resolution = (tempo * 1000) / ppq; + if (tmr->tempo < 1000000) + tmr->tick.resolution = (tmr->tempo * 1000) / tmr->ppq; else { /* might overflow.. */ unsigned int s; - s = tempo % ppq; - s = (s * 1000) / ppq; - tick->resolution = (tempo / ppq) * 1000; - tick->resolution += s; + s = tmr->tempo % tmr->ppq; + s = (s * 1000) / tmr->ppq; + tmr->tick.resolution = (tmr->tempo / tmr->ppq) * 1000; + tmr->tick.resolution += s; } - if (tick->resolution <= 0) - tick->resolution = 1; - snd_seq_timer_update_tick(tick, 0); + if (tmr->tick.resolution <= 0) + tmr->tick.resolution = 1; + snd_seq_timer_update_tick(&tmr->tick, 0); } /* create new timer (constructor) */ @@ -96,7 +95,7 @@ void snd_seq_timer_defaults(struct snd_seq_timer * tmr) /* setup defaults */ tmr->ppq = 96; /* 96 PPQ */ tmr->tempo = 500000; /* 120 BPM */ - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); tmr->running = 0; tmr->type = SNDRV_SEQ_TIMER_ALSA; @@ -180,7 +179,7 @@ int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo) spin_lock_irqsave(&tmr->lock, flags); if ((unsigned int)tempo != tmr->tempo) { tmr->tempo = tempo; - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); } spin_unlock_irqrestore(&tmr->lock, flags); return 0; @@ -205,7 +204,7 @@ int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq) } tmr->ppq = ppq; - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); spin_unlock_irqrestore(&tmr->lock, flags); return 0; } -- cgit v1.2.2 From d1db38c015a392b0ea8c15ab95abb3ee768b8d47 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 18 Jan 2010 15:44:04 +0100 Subject: sound: virtuoso: add Xonar DS support Add experimental support for the Asus Xonar DS. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/Kconfig | 1 + sound/pci/oxygen/Makefile | 2 +- sound/pci/oxygen/virtuoso.c | 3 + sound/pci/oxygen/wm8766.h | 73 +++ sound/pci/oxygen/wm8776.h | 177 +++++++ sound/pci/oxygen/xonar.h | 2 + sound/pci/oxygen/xonar_wm87x6.c | 1021 +++++++++++++++++++++++++++++++++++++++ 7 files changed, 1278 insertions(+), 1 deletion(-) create mode 100644 sound/pci/oxygen/wm8766.h create mode 100644 sound/pci/oxygen/wm8776.h create mode 100644 sound/pci/oxygen/xonar_wm87x6.c (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 351654cf7b09..1298c68d6bf0 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -789,6 +789,7 @@ config SND_VIRTUOSO Say Y here to include support for sound cards based on the Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, Essence ST (Deluxe), and Essence STX. + Support for the DS is experimental. Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 389941cf6100..acd8f15f7bff 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -2,7 +2,7 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o snd-hifier-objs := hifier.o snd-oxygen-objs := oxygen.o snd-virtuoso-objs := virtuoso.o xonar_lib.o \ - xonar_pcm179x.o xonar_cs43xx.o xonar_hdmi.o + xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o obj-$(CONFIG_SND_HIFIER) += snd-hifier.o diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6accaf9580b2..563b6f50821f 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -49,6 +49,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x834f) }, { OXYGEN_PCI_SUBID(0x1043, 0x835c) }, { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, + { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; @@ -61,6 +62,8 @@ static int __devinit get_xonar_model(struct oxygen *chip, return 0; if (get_xonar_cs43xx_model(chip, id) >= 0) return 0; + if (get_xonar_wm87x6_model(chip, id) >= 0) + return 0; return -EINVAL; } diff --git a/sound/pci/oxygen/wm8766.h b/sound/pci/oxygen/wm8766.h new file mode 100644 index 000000000000..e0e849a7eaeb --- /dev/null +++ b/sound/pci/oxygen/wm8766.h @@ -0,0 +1,73 @@ +#ifndef WM8766_H_INCLUDED +#define WM8766_H_INCLUDED + +#define WM8766_LDA1 0x00 +#define WM8766_RDA1 0x01 +#define WM8766_DAC_CTRL 0x02 +#define WM8766_INT_CTRL 0x03 +#define WM8766_LDA2 0x04 +#define WM8766_RDA2 0x05 +#define WM8766_LDA3 0x06 +#define WM8766_RDA3 0x07 +#define WM8766_MASTDA 0x08 +#define WM8766_DAC_CTRL2 0x09 +#define WM8766_DAC_CTRL3 0x0a +#define WM8766_MUTE1 0x0c +#define WM8766_MUTE2 0x0f +#define WM8766_RESET 0x1f + +/* LDAx/RDAx/MASTDA */ +#define WM8766_ATT_MASK 0x0ff +#define WM8766_UPDATE 0x100 +/* DAC_CTRL */ +#define WM8766_MUTEALL 0x001 +#define WM8766_DEEMPALL 0x002 +#define WM8766_PWDN 0x004 +#define WM8766_ATC 0x008 +#define WM8766_IZD 0x010 +#define WM8766_PL_LEFT_MASK 0x060 +#define WM8766_PL_LEFT_MUTE 0x000 +#define WM8766_PL_LEFT_LEFT 0x020 +#define WM8766_PL_LEFT_RIGHT 0x040 +#define WM8766_PL_LEFT_LRMIX 0x060 +#define WM8766_PL_RIGHT_MASK 0x180 +#define WM8766_PL_RIGHT_MUTE 0x000 +#define WM8766_PL_RIGHT_LEFT 0x080 +#define WM8766_PL_RIGHT_RIGHT 0x100 +#define WM8766_PL_RIGHT_LRMIX 0x180 +/* INT_CTRL */ +#define WM8766_FMT_MASK 0x003 +#define WM8766_FMT_RJUST 0x000 +#define WM8766_FMT_LJUST 0x001 +#define WM8766_FMT_I2S 0x002 +#define WM8766_FMT_DSP 0x003 +#define WM8766_LRP 0x004 +#define WM8766_BCP 0x008 +#define WM8766_IWL_MASK 0x030 +#define WM8766_IWL_16 0x000 +#define WM8766_IWL_20 0x010 +#define WM8766_IWL_24 0x020 +#define WM8766_IWL_32 0x030 +#define WM8766_PHASE_MASK 0x1c0 +/* DAC_CTRL2 */ +#define WM8766_ZCD 0x001 +#define WM8766_DZFM_MASK 0x006 +#define WM8766_DMUTE_MASK 0x038 +#define WM8766_DEEMP_MASK 0x1c0 +/* DAC_CTRL3 */ +#define WM8766_DACPD_MASK 0x00e +#define WM8766_PWRDNALL 0x010 +#define WM8766_MS 0x020 +#define WM8766_RATE_MASK 0x1c0 +#define WM8766_RATE_128 0x000 +#define WM8766_RATE_192 0x040 +#define WM8766_RATE_256 0x080 +#define WM8766_RATE_384 0x0c0 +#define WM8766_RATE_512 0x100 +#define WM8766_RATE_768 0x140 +/* MUTE1 */ +#define WM8766_MPD1 0x040 +/* MUTE2 */ +#define WM8766_MPD2 0x020 + +#endif diff --git a/sound/pci/oxygen/wm8776.h b/sound/pci/oxygen/wm8776.h new file mode 100644 index 000000000000..1a96f5615727 --- /dev/null +++ b/sound/pci/oxygen/wm8776.h @@ -0,0 +1,177 @@ +#ifndef WM8776_H_INCLUDED +#define WM8776_H_INCLUDED + +/* + * the following register names are from: + * wm8776.h -- WM8776 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#define WM8776_HPLVOL 0x00 +#define WM8776_HPRVOL 0x01 +#define WM8776_HPMASTER 0x02 +#define WM8776_DACLVOL 0x03 +#define WM8776_DACRVOL 0x04 +#define WM8776_DACMASTER 0x05 +#define WM8776_PHASESWAP 0x06 +#define WM8776_DACCTRL1 0x07 +#define WM8776_DACMUTE 0x08 +#define WM8776_DACCTRL2 0x09 +#define WM8776_DACIFCTRL 0x0a +#define WM8776_ADCIFCTRL 0x0b +#define WM8776_MSTRCTRL 0x0c +#define WM8776_PWRDOWN 0x0d +#define WM8776_ADCLVOL 0x0e +#define WM8776_ADCRVOL 0x0f +#define WM8776_ALCCTRL1 0x10 +#define WM8776_ALCCTRL2 0x11 +#define WM8776_ALCCTRL3 0x12 +#define WM8776_NOISEGATE 0x13 +#define WM8776_LIMITER 0x14 +#define WM8776_ADCMUX 0x15 +#define WM8776_OUTMUX 0x16 +#define WM8776_RESET 0x17 + + +/* HPLVOL/HPRVOL/HPMASTER */ +#define WM8776_HPATT_MASK 0x07f +#define WM8776_HPZCEN 0x080 +#define WM8776_UPDATE 0x100 + +/* DACLVOL/DACRVOL/DACMASTER */ +#define WM8776_DATT_MASK 0x0ff +/*#define WM8776_UPDATE 0x100*/ + +/* PHASESWAP */ +#define WM8776_PH_MASK 0x003 + +/* DACCTRL1 */ +#define WM8776_DZCEN 0x001 +#define WM8776_ATC 0x002 +#define WM8776_IZD 0x004 +#define WM8776_TOD 0x008 +#define WM8776_PL_LEFT_MASK 0x030 +#define WM8776_PL_LEFT_MUTE 0x000 +#define WM8776_PL_LEFT_LEFT 0x010 +#define WM8776_PL_LEFT_RIGHT 0x020 +#define WM8776_PL_LEFT_LRMIX 0x030 +#define WM8776_PL_RIGHT_MASK 0x0c0 +#define WM8776_PL_RIGHT_MUTE 0x000 +#define WM8776_PL_RIGHT_LEFT 0x040 +#define WM8776_PL_RIGHT_RIGHT 0x080 +#define WM8776_PL_RIGHT_LRMIX 0x0c0 + +/* DACMUTE */ +#define WM8776_DMUTE 0x001 + +/* DACCTRL2 */ +#define WM8776_DEEMPH 0x001 +#define WM8776_DZFM_MASK 0x006 +#define WM8776_DZFM_NONE 0x000 +#define WM8776_DZFM_LR 0x002 +#define WM8776_DZFM_BOTH 0x004 +#define WM8776_DZFM_EITHER 0x006 + +/* DACIFCTRL */ +#define WM8776_DACFMT_MASK 0x003 +#define WM8776_DACFMT_RJUST 0x000 +#define WM8776_DACFMT_LJUST 0x001 +#define WM8776_DACFMT_I2S 0x002 +#define WM8776_DACFMT_DSP 0x003 +#define WM8776_DACLRP 0x004 +#define WM8776_DACBCP 0x008 +#define WM8776_DACWL_MASK 0x030 +#define WM8776_DACWL_16 0x000 +#define WM8776_DACWL_20 0x010 +#define WM8776_DACWL_24 0x020 +#define WM8776_DACWL_32 0x030 + +/* ADCIFCTRL */ +#define WM8776_ADCFMT_MASK 0x003 +#define WM8776_ADCFMT_RJUST 0x000 +#define WM8776_ADCFMT_LJUST 0x001 +#define WM8776_ADCFMT_I2S 0x002 +#define WM8776_ADCFMT_DSP 0x003 +#define WM8776_ADCLRP 0x004 +#define WM8776_ADCBCP 0x008 +#define WM8776_ADCWL_MASK 0x030 +#define WM8776_ADCWL_16 0x000 +#define WM8776_ADCWL_20 0x010 +#define WM8776_ADCWL_24 0x020 +#define WM8776_ADCWL_32 0x030 +#define WM8776_ADCMCLK 0x040 +#define WM8776_ADCHPD 0x100 + +/* MSTRCTRL */ +#define WM8776_ADCRATE_MASK 0x007 +#define WM8776_ADCRATE_256 0x002 +#define WM8776_ADCRATE_384 0x003 +#define WM8776_ADCRATE_512 0x004 +#define WM8776_ADCRATE_768 0x005 +#define WM8776_ADCOSR 0x008 +#define WM8776_DACRATE_MASK 0x070 +#define WM8776_DACRATE_128 0x000 +#define WM8776_DACRATE_192 0x010 +#define WM8776_DACRATE_256 0x020 +#define WM8776_DACRATE_384 0x030 +#define WM8776_DACRATE_512 0x040 +#define WM8776_DACRATE_768 0x050 +#define WM8776_DACMS 0x080 +#define WM8776_ADCMS 0x100 + +/* PWRDOWN */ +#define WM8776_PDWN 0x001 +#define WM8776_ADCPD 0x002 +#define WM8776_DACPD 0x004 +#define WM8776_HPPD 0x008 +#define WM8776_AINPD 0x040 + +/* ADCLVOL/ADCRVOL */ +#define WM8776_AGMASK 0x0ff +#define WM8776_ZCA 0x100 + +/* ALCCTRL1 */ +#define WM8776_LCT_MASK 0x00f +#define WM8776_MAXGAIN_MASK 0x070 +#define WM8776_LCSEL_MASK 0x180 +#define WM8776_LCSEL_LIMITER 0x000 +#define WM8776_LCSEL_ALC_RIGHT 0x080 +#define WM8776_LCSEL_ALC_LEFT 0x100 +#define WM8776_LCSEL_ALC_STEREO 0x180 + +/* ALCCTRL2 */ +#define WM8776_HLD_MASK 0x00f +#define WM8776_ALCZC 0x080 +#define WM8776_LCEN 0x100 + +/* ALCCTRL3 */ +#define WM8776_ATK_MASK 0x00f +#define WM8776_DCY_MASK 0x0f0 + +/* NOISEGATE */ +#define WM8776_NGAT 0x001 +#define WM8776_NGTH_MASK 0x01c + +/* LIMITER */ +#define WM8776_MAXATTEN_MASK 0x00f +#define WM8776_TRANWIN_MASK 0x070 + +/* ADCMUX */ +#define WM8776_AMX_MASK 0x01f +#define WM8776_MUTERA 0x040 +#define WM8776_MUTELA 0x080 +#define WM8776_LRBOTH 0x100 + +/* OUTMUX */ +#define WM8776_MX_DAC 0x001 +#define WM8776_MX_AUX 0x002 +#define WM8776_MX_BYPASS 0x004 + +#endif diff --git a/sound/pci/oxygen/xonar.h b/sound/pci/oxygen/xonar.h index 89b3ed814d64..b35343b0a9a5 100644 --- a/sound/pci/oxygen/xonar.h +++ b/sound/pci/oxygen/xonar.h @@ -35,6 +35,8 @@ int get_xonar_pcm179x_model(struct oxygen *chip, const struct pci_device_id *id); int get_xonar_cs43xx_model(struct oxygen *chip, const struct pci_device_id *id); +int get_xonar_wm87x6_model(struct oxygen *chip, + const struct pci_device_id *id); /* HDMI helper functions */ diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c new file mode 100644 index 000000000000..7754db166d9e --- /dev/null +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -0,0 +1,1021 @@ +/* + * card driver for models with WM8776/WM8766 DACs (Xonar DS) + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see . + */ + +/* + * Xonar DS + * -------- + * + * CMI8788: + * + * SPI 0 -> WM8766 (surround, center/LFE, back) + * SPI 1 -> WM8776 (front, input) + * + * GPIO 4 <- headphone detect + * GPIO 6 -> route input jack to input 1/2 (1/0) + * GPIO 7 -> enable output to speakers + * GPIO 8 -> enable output to speakers + */ + +#include +#include +#include +#include +#include +#include +#include +#include "xonar.h" +#include "wm8776.h" +#include "wm8766.h" + +#define GPIO_DS_HP_DETECT 0x0010 +#define GPIO_DS_INPUT_ROUTE 0x0040 +#define GPIO_DS_OUTPUT_ENABLE 0x0180 + +#define LC_CONTROL_LIMITER 0x40000000 +#define LC_CONTROL_ALC 0x20000000 + +struct xonar_wm87x6 { + struct xonar_generic generic; + u16 wm8776_regs[0x17]; + u16 wm8766_regs[0x10]; + struct snd_kcontrol *lc_controls[13]; +}; + +static void wm8776_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (1 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_LO, + (reg << 9) | value); + if (reg < ARRAY_SIZE(data->wm8776_regs)) { + if (reg >= WM8776_HPLVOL || reg <= WM8776_DACMASTER) + value &= ~WM8776_UPDATE; + data->wm8776_regs[reg] = value; + } +} + +static void wm8776_write_cached(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if (reg >= ARRAY_SIZE(data->wm8776_regs) || + value != data->wm8776_regs[reg]) + wm8776_write(chip, reg, value); +} + +static void wm8766_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_LO, + (reg << 9) | value); + if (reg < ARRAY_SIZE(data->wm8766_regs)) + data->wm8766_regs[reg] = value; +} + +static void wm8766_write_cached(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if (reg >= ARRAY_SIZE(data->wm8766_regs) || + value != data->wm8766_regs[reg]) { + if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) || + (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA)) + value &= ~WM8766_UPDATE; + wm8766_write(chip, reg, value); + } +} + +static void wm8776_registers_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + wm8776_write(chip, WM8776_RESET, 0); + wm8776_write(chip, WM8776_DACCTRL1, WM8776_DZCEN | + WM8776_PL_LEFT_LEFT | WM8776_PL_RIGHT_RIGHT); + wm8776_write(chip, WM8776_DACMUTE, chip->dac_mute ? WM8776_DMUTE : 0); + wm8776_write(chip, WM8776_DACIFCTRL, + WM8776_DACFMT_LJUST | WM8776_DACWL_24); + wm8776_write(chip, WM8776_ADCIFCTRL, + data->wm8776_regs[WM8776_ADCIFCTRL]); + wm8776_write(chip, WM8776_MSTRCTRL, data->wm8776_regs[WM8776_MSTRCTRL]); + wm8776_write(chip, WM8776_PWRDOWN, data->wm8776_regs[WM8776_PWRDOWN]); + wm8776_write(chip, WM8776_HPLVOL, data->wm8776_regs[WM8776_HPLVOL]); + wm8776_write(chip, WM8776_HPRVOL, data->wm8776_regs[WM8776_HPRVOL] | + WM8776_UPDATE); + wm8776_write(chip, WM8776_ADCLVOL, data->wm8776_regs[WM8776_ADCLVOL]); + wm8776_write(chip, WM8776_ADCRVOL, data->wm8776_regs[WM8776_ADCRVOL]); + wm8776_write(chip, WM8776_ADCMUX, data->wm8776_regs[WM8776_ADCMUX]); + wm8776_write(chip, WM8776_DACLVOL, chip->dac_volume[0]); + wm8776_write(chip, WM8776_DACRVOL, chip->dac_volume[1] | WM8776_UPDATE); +} + +static void wm8766_registers_init(struct oxygen *chip) +{ + wm8766_write(chip, WM8766_RESET, 0); + wm8766_write(chip, WM8766_INT_CTRL, WM8766_FMT_LJUST | WM8766_IWL_24); + wm8766_write(chip, WM8766_DAC_CTRL2, + WM8766_ZCD | (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); + wm8766_write(chip, WM8766_LDA1, chip->dac_volume[2]); + wm8766_write(chip, WM8766_RDA1, chip->dac_volume[3]); + wm8766_write(chip, WM8766_LDA2, chip->dac_volume[4]); + wm8766_write(chip, WM8766_RDA2, chip->dac_volume[5]); + wm8766_write(chip, WM8766_LDA3, chip->dac_volume[6]); + wm8766_write(chip, WM8766_RDA3, chip->dac_volume[7] | WM8766_UPDATE); +} + +static void wm8776_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->wm8776_regs[WM8776_HPLVOL] = (0x79 - 60) | WM8776_HPZCEN; + data->wm8776_regs[WM8776_HPRVOL] = (0x79 - 60) | WM8776_HPZCEN; + data->wm8776_regs[WM8776_ADCIFCTRL] = + WM8776_ADCFMT_LJUST | WM8776_ADCWL_24 | WM8776_ADCMCLK; + data->wm8776_regs[WM8776_MSTRCTRL] = + WM8776_ADCRATE_256 | WM8776_DACRATE_256; + data->wm8776_regs[WM8776_PWRDOWN] = WM8776_HPPD; + data->wm8776_regs[WM8776_ADCLVOL] = 0xa5 | WM8776_ZCA; + data->wm8776_regs[WM8776_ADCRVOL] = 0xa5 | WM8776_ZCA; + data->wm8776_regs[WM8776_ADCMUX] = 0x001; + wm8776_registers_init(chip); +} + +static void xonar_ds_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->generic.anti_pop_delay = 300; + data->generic.output_enable_bit = GPIO_DS_OUTPUT_ENABLE; + + wm8776_init(chip); + wm8766_registers_init(chip); + + oxygen_write16_masked(chip, OXYGEN_GPIO_CONTROL, GPIO_DS_INPUT_ROUTE, + GPIO_DS_HP_DETECT | GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, GPIO_DS_HP_DETECT); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + + xonar_enable_output(chip); + + snd_component_add(chip->card, "WM8776"); + snd_component_add(chip->card, "WM8766"); +} + +static void xonar_ds_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + +static void xonar_ds_suspend(struct oxygen *chip) +{ + xonar_ds_cleanup(chip); +} + +static void xonar_ds_resume(struct oxygen *chip) +{ + wm8776_registers_init(chip); + wm8766_registers_init(chip); + xonar_enable_output(chip); +} + +static void wm8776_adc_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + if (channel == PCM_A) { + hardware->rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + hardware->rate_max = 96000; + } +} + +static void set_wm87x6_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ +} + +static void set_wm8776_adc_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + u16 reg; + + reg = WM8776_ADCRATE_256 | WM8776_DACRATE_256; + if (params_rate(params) > 48000) + reg |= WM8776_ADCOSR; + wm8776_write_cached(chip, WM8776_MSTRCTRL, reg); +} + +static void update_wm8776_volume(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + u8 to_change; + + if (chip->dac_volume[0] == chip->dac_volume[1]) { + if (chip->dac_volume[0] != data->wm8776_regs[WM8776_DACLVOL] || + chip->dac_volume[1] != data->wm8776_regs[WM8776_DACRVOL]) { + wm8776_write(chip, WM8776_DACMASTER, + chip->dac_volume[0] | WM8776_UPDATE); + data->wm8776_regs[WM8776_DACLVOL] = chip->dac_volume[0]; + data->wm8776_regs[WM8776_DACRVOL] = chip->dac_volume[0]; + } + } else { + to_change = (chip->dac_volume[0] != + data->wm8776_regs[WM8776_DACLVOL]) << 0; + to_change |= (chip->dac_volume[1] != + data->wm8776_regs[WM8776_DACLVOL]) << 1; + if (to_change & 1) + wm8776_write(chip, WM8776_DACLVOL, chip->dac_volume[0] | + ((to_change & 2) ? 0 : WM8776_UPDATE)); + if (to_change & 2) + wm8776_write(chip, WM8776_DACRVOL, + chip->dac_volume[1] | WM8776_UPDATE); + } +} + +static void update_wm87x6_volume(struct oxygen *chip) +{ + static const u8 wm8766_regs[6] = { + WM8766_LDA1, WM8766_RDA1, + WM8766_LDA2, WM8766_RDA2, + WM8766_LDA3, WM8766_RDA3, + }; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + u8 to_change; + + update_wm8776_volume(chip); + if (chip->dac_volume[2] == chip->dac_volume[3] && + chip->dac_volume[2] == chip->dac_volume[4] && + chip->dac_volume[2] == chip->dac_volume[5] && + chip->dac_volume[2] == chip->dac_volume[6] && + chip->dac_volume[2] == chip->dac_volume[7]) { + to_change = 0; + for (i = 0; i < 6; ++i) + if (chip->dac_volume[2] != + data->wm8766_regs[wm8766_regs[i]]) + to_change = 1; + if (to_change) { + wm8766_write(chip, WM8766_MASTDA, + chip->dac_volume[2] | WM8766_UPDATE); + for (i = 0; i < 6; ++i) + data->wm8766_regs[wm8766_regs[i]] = + chip->dac_volume[2]; + } + } else { + to_change = 0; + for (i = 0; i < 6; ++i) + to_change |= (chip->dac_volume[2 + i] != + data->wm8766_regs[wm8766_regs[i]]) << i; + for (i = 0; i < 6; ++i) + if (to_change & (1 << i)) + wm8766_write(chip, wm8766_regs[i], + chip->dac_volume[2 + i] | + ((to_change & (0x3e << i)) + ? 0 : WM8766_UPDATE)); + } +} + +static void update_wm8776_mute(struct oxygen *chip) +{ + wm8776_write_cached(chip, WM8776_DACMUTE, + chip->dac_mute ? WM8776_DMUTE : 0); +} + +static void update_wm87x6_mute(struct oxygen *chip) +{ + update_wm8776_mute(chip); + wm8766_write_cached(chip, WM8766_DAC_CTRL2, WM8766_ZCD | + (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); +} + +static void xonar_ds_gpio_changed(struct oxygen *chip) +{ + u16 bits; + + bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); + snd_printk(KERN_INFO "HP detect: %d\n", !!(bits & GPIO_DS_HP_DETECT)); +} + +static int wm8776_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u16 bit = ctl->private_value & 0xffff; + unsigned int reg_index = (ctl->private_value >> 16) & 0xff; + bool invert = (ctl->private_value >> 24) & 1; + + value->value.integer.value[0] = + ((data->wm8776_regs[reg_index] & bit) != 0) ^ invert; + return 0; +} + +static int wm8776_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u16 bit = ctl->private_value & 0xffff; + u16 reg_value; + unsigned int reg_index = (ctl->private_value >> 16) & 0xff; + bool invert = (ctl->private_value >> 24) & 1; + int changed; + + mutex_lock(&chip->mutex); + reg_value = data->wm8776_regs[reg_index] & ~bit; + if (value->value.integer.value[0] ^ invert) + reg_value |= bit; + changed = reg_value != data->wm8776_regs[reg_index]; + if (changed) + wm8776_write(chip, reg_index, reg_value); + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_field_enum_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const hld[16] = { + "0 ms", "2.67 ms", "5.33 ms", "10.6 ms", + "21.3 ms", "42.7 ms", "85.3 ms", "171 ms", + "341 ms", "683 ms", "1.37 s", "2.73 s", + "5.46 s", "10.9 s", "21.8 s", "43.7 s", + }; + static const char *const atk_lim[11] = { + "0.25 ms", "0.5 ms", "1 ms", "2 ms", + "4 ms", "8 ms", "16 ms", "32 ms", + "64 ms", "128 ms", "256 ms", + }; + static const char *const atk_alc[11] = { + "8.40 ms", "16.8 ms", "33.6 ms", "67.2 ms", + "134 ms", "269 ms", "538 ms", "1.08 s", + "2.15 s", "4.3 s", "8.6 s", + }; + static const char *const dcy_lim[11] = { + "1.2 ms", "2.4 ms", "4.8 ms", "9.6 ms", + "19.2 ms", "38.4 ms", "76.8 ms", "154 ms", + "307 ms", "614 ms", "1.23 s", + }; + static const char *const dcy_alc[11] = { + "33.5 ms", "67.0 ms", "134 ms", "268 ms", + "536 ms", "1.07 s", "2.14 s", "4.29 s", + "8.58 s", "17.2 s", "34.3 s", + }; + static const char *const tranwin[8] = { + "0 us", "62.5 us", "125 us", "250 us", + "500 us", "1 ms", "2 ms", "4 ms", + }; + u8 max; + const char *const *names; + + max = (ctl->private_value >> 12) & 0xf; + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = max + 1; + if (info->value.enumerated.item > max) + info->value.enumerated.item = max; + switch ((ctl->private_value >> 24) & 0x1f) { + case WM8776_ALCCTRL2: + names = hld; + break; + case WM8776_ALCCTRL3: + if (((ctl->private_value >> 20) & 0xf) == 0) { + if (ctl->private_value & LC_CONTROL_LIMITER) + names = atk_lim; + else + names = atk_alc; + } else { + if (ctl->private_value & LC_CONTROL_LIMITER) + names = dcy_lim; + else + names = dcy_alc; + } + break; + case WM8776_LIMITER: + names = tranwin; + break; + default: + return -ENXIO; + } + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int wm8776_field_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = (ctl->private_value >> 8) & 0xf; + info->value.integer.max = (ctl->private_value >> 12) & 0xf; + return 0; +} + +static void wm8776_field_set_from_ctl(struct snd_kcontrol *ctl) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int value, reg_index, mode; + u8 min, max, shift; + u16 mask, reg_value; + bool invert; + + if ((data->wm8776_regs[WM8776_ALCCTRL1] & WM8776_LCSEL_MASK) == + WM8776_LCSEL_LIMITER) + mode = LC_CONTROL_LIMITER; + else + mode = LC_CONTROL_ALC; + if (!(ctl->private_value & mode)) + return; + + value = ctl->private_value & 0xf; + min = (ctl->private_value >> 8) & 0xf; + max = (ctl->private_value >> 12) & 0xf; + mask = (ctl->private_value >> 16) & 0xf; + shift = (ctl->private_value >> 20) & 0xf; + reg_index = (ctl->private_value >> 24) & 0x1f; + invert = (ctl->private_value >> 29) & 0x1; + + if (invert) + value = max - (value - min); + reg_value = data->wm8776_regs[reg_index]; + reg_value &= ~(mask << shift); + reg_value |= value << shift; + wm8776_write_cached(chip, reg_index, reg_value); +} + +static int wm8776_field_set(struct snd_kcontrol *ctl, unsigned int value) +{ + struct oxygen *chip = ctl->private_data; + u8 min, max; + int changed; + + min = (ctl->private_value >> 8) & 0xf; + max = (ctl->private_value >> 12) & 0xf; + if (value < min || value > max) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = value != (ctl->private_value & 0xf); + if (changed) { + ctl->private_value = (ctl->private_value & ~0xf) | value; + wm8776_field_set_from_ctl(ctl); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_field_enum_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.enumerated.item[0] = ctl->private_value & 0xf; + return 0; +} + +static int wm8776_field_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.integer.value[0] = ctl->private_value & 0xf; + return 0; +} + +static int wm8776_field_enum_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + return wm8776_field_set(ctl, value->value.enumerated.item[0]); +} + +static int wm8776_field_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + return wm8776_field_set(ctl, value->value.integer.value[0]); +} + +static int wm8776_hp_vol_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0x79 - 60; + info->value.integer.max = 0x7f; + return 0; +} + +static int wm8776_hp_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = + data->wm8776_regs[WM8776_HPLVOL] & WM8776_HPATT_MASK; + value->value.integer.value[1] = + data->wm8776_regs[WM8776_HPRVOL] & WM8776_HPATT_MASK; + mutex_unlock(&chip->mutex); + return 0; +} + +static int wm8776_hp_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u8 to_update; + + mutex_lock(&chip->mutex); + to_update = (value->value.integer.value[0] != + (data->wm8776_regs[WM8776_HPLVOL] & WM8776_HPATT_MASK)) + << 0; + to_update |= (value->value.integer.value[1] != + (data->wm8776_regs[WM8776_HPRVOL] & WM8776_HPATT_MASK)) + << 1; + if (value->value.integer.value[0] == value->value.integer.value[1]) { + if (to_update) { + wm8776_write(chip, WM8776_HPMASTER, + value->value.integer.value[0] | + WM8776_HPZCEN | WM8776_UPDATE); + data->wm8776_regs[WM8776_HPLVOL] = + value->value.integer.value[0] | WM8776_HPZCEN; + data->wm8776_regs[WM8776_HPRVOL] = + value->value.integer.value[0] | WM8776_HPZCEN; + } + } else { + if (to_update & 1) + wm8776_write(chip, WM8776_HPLVOL, + value->value.integer.value[0] | + WM8776_HPZCEN | + ((to_update & 2) ? 0 : WM8776_UPDATE)); + if (to_update & 2) + wm8776_write(chip, WM8776_HPRVOL, + value->value.integer.value[1] | + WM8776_HPZCEN | WM8776_UPDATE); + } + mutex_unlock(&chip->mutex); + return to_update != 0; +} + +static int wm8776_input_mux_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mux_bit = ctl->private_value; + + value->value.integer.value[0] = + !!(data->wm8776_regs[WM8776_ADCMUX] & mux_bit); + return 0; +} + +static int wm8776_input_mux_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mux_bit = ctl->private_value; + u16 reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8776_regs[WM8776_ADCMUX]; + if (value->value.integer.value[0]) { + reg &= ~0x003; + reg |= mux_bit; + } else + reg &= ~mux_bit; + changed = reg != data->wm8776_regs[WM8776_ADCMUX]; + if (changed) { + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + reg & 1 ? GPIO_DS_INPUT_ROUTE : 0, + GPIO_DS_INPUT_ROUTE); + wm8776_write(chip, WM8776_ADCMUX, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_input_vol_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0xa5; + info->value.integer.max = 0xff; + return 0; +} + +static int wm8776_input_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = + data->wm8776_regs[WM8776_ADCLVOL] & WM8776_AGMASK; + value->value.integer.value[1] = + data->wm8776_regs[WM8776_ADCRVOL] & WM8776_AGMASK; + mutex_unlock(&chip->mutex); + return 0; +} + +static int wm8776_input_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + int changed = 0; + + mutex_lock(&chip->mutex); + changed = (value->value.integer.value[0] != + (data->wm8776_regs[WM8776_ADCLVOL] & WM8776_AGMASK)) || + (value->value.integer.value[1] != + (data->wm8776_regs[WM8776_ADCRVOL] & WM8776_AGMASK)); + wm8776_write_cached(chip, WM8776_ADCLVOL, + value->value.integer.value[0] | WM8776_ZCA); + wm8776_write_cached(chip, WM8776_ADCRVOL, + value->value.integer.value[1] | WM8776_ZCA); + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_level_control_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "None", "Peak Limiter", "Automatic Level Control" + }; + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int wm8776_level_control_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + if (!(data->wm8776_regs[WM8776_ALCCTRL2] & WM8776_LCEN)) + value->value.enumerated.item[0] = 0; + else if ((data->wm8776_regs[WM8776_ALCCTRL1] & WM8776_LCSEL_MASK) == + WM8776_LCSEL_LIMITER) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + +static void activate_control(struct oxygen *chip, + struct snd_kcontrol *ctl, unsigned int mode) +{ + unsigned int access; + + if (ctl->private_value & mode) + access = 0; + else + access = SNDRV_CTL_ELEM_ACCESS_INACTIVE; + if ((ctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) != access) { + ctl->vd[0].access ^= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + } +} + +static int wm8776_level_control_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mode = 0, i; + u16 ctrl1, ctrl2; + int changed; + + if (value->value.enumerated.item[0] >= 3) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != ctl->private_value; + if (changed) { + ctl->private_value = value->value.enumerated.item[0]; + ctrl1 = data->wm8776_regs[WM8776_ALCCTRL1]; + ctrl2 = data->wm8776_regs[WM8776_ALCCTRL2]; + switch (value->value.enumerated.item[0]) { + default: + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 & ~WM8776_LCEN); + break; + case 1: + wm8776_write_cached(chip, WM8776_ALCCTRL1, + (ctrl1 & ~WM8776_LCSEL_MASK) | + WM8776_LCSEL_LIMITER); + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 | WM8776_LCEN); + mode = LC_CONTROL_LIMITER; + break; + case 2: + wm8776_write_cached(chip, WM8776_ALCCTRL1, + (ctrl1 & ~WM8776_LCSEL_MASK) | + WM8776_LCSEL_ALC_STEREO); + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 | WM8776_LCEN); + mode = LC_CONTROL_ALC; + break; + } + for (i = 0; i < ARRAY_SIZE(data->lc_controls); ++i) + activate_control(chip, data->lc_controls[i], mode); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "None", "High-pass Filter" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + value->value.enumerated.item[0] = + !(data->wm8776_regs[WM8776_ADCIFCTRL] & WM8776_ADCHPD); + return 0; +} + +static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8776_regs[WM8776_ADCIFCTRL] & ~WM8776_ADCHPD; + if (!value->value.enumerated.item[0]) + reg |= WM8776_ADCHPD; + changed = reg != data->wm8776_regs[WM8776_ADCIFCTRL]; + if (changed) + wm8776_write(chip, WM8776_ADCIFCTRL, reg); + mutex_unlock(&chip->mutex); + return changed; +} + +#define WM8776_BIT_SWITCH(xname, reg, bit, invert, flags) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = wm8776_bit_switch_get, \ + .put = wm8776_bit_switch_put, \ + .private_value = ((reg) << 16) | (bit) | ((invert) << 24) | (flags), \ +} +#define _WM8776_FIELD_CTL(xname, reg, shift, initval, min, max, mask, flags) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .private_value = (initval) | ((min) << 8) | ((max) << 12) | \ + ((mask) << 16) | ((shift) << 20) | ((reg) << 24) | (flags) +#define WM8776_FIELD_CTL_ENUM(xname, reg, shift, init, min, max, mask, flags) {\ + _WM8776_FIELD_CTL(xname " Capture Enum", \ + reg, shift, init, min, max, mask, flags), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_INACTIVE, \ + .info = wm8776_field_enum_info, \ + .get = wm8776_field_enum_get, \ + .put = wm8776_field_enum_put, \ +} +#define WM8776_FIELD_CTL_VOLUME(a, b, c, d, e, f, g, h, tlv_p) { \ + _WM8776_FIELD_CTL(a " Capture Volume", b, c, d, e, f, g, h), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_INACTIVE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = wm8776_field_volume_info, \ + .get = wm8776_field_volume_get, \ + .put = wm8776_field_volume_put, \ + .tlv = { .p = tlv_p }, \ +} + +static const DECLARE_TLV_DB_SCALE(wm87x6_dac_db_scale, -6000, 50, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_adc_db_scale, -2100, 50, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_hp_db_scale, -6000, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_lct_db_scale, -1600, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxgain_db_scale, 0, 400, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_ngth_db_scale, -7800, 600, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxatten_lim_db_scale, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxatten_alc_db_scale, -2100, 400, 0); + +static const struct snd_kcontrol_new ds_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .info = wm8776_hp_vol_info, + .get = wm8776_hp_vol_get, + .put = wm8776_hp_vol_put, + .tlv = { .p = wm8776_hp_db_scale }, + }, + WM8776_BIT_SWITCH("Headphone Playback Switch", + WM8776_PWRDOWN, WM8776_HPPD, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Capture Volume", + .info = wm8776_input_vol_info, + .get = wm8776_input_vol_get, + .put = wm8776_input_vol_put, + .tlv = { .p = wm8776_adc_db_scale }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Switch", + .info = snd_ctl_boolean_mono_info, + .get = wm8776_input_mux_get, + .put = wm8776_input_mux_put, + .private_value = 1 << 0, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Switch", + .info = snd_ctl_boolean_mono_info, + .get = wm8776_input_mux_get, + .put = wm8776_input_mux_put, + .private_value = 1 << 1, + }, + WM8776_BIT_SWITCH("Aux", WM8776_ADCMUX, 1 << 2, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Level Control Capture Enum", + .info = wm8776_level_control_info, + .get = wm8776_level_control_get, + .put = wm8776_level_control_put, + .private_value = 0, + }, +}; +static const struct snd_kcontrol_new lc_controls[] = { + WM8776_FIELD_CTL_VOLUME("Limiter Threshold", + WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, + LC_CONTROL_LIMITER, wm8776_lct_db_scale), + WM8776_FIELD_CTL_ENUM("Limiter Attack Time", + WM8776_ALCCTRL3, 0, 2, 0, 10, 0xf, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_ENUM("Limiter Decay Time", + WM8776_ALCCTRL3, 4, 3, 0, 10, 0xf, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_ENUM("Limiter Transient Window", + WM8776_LIMITER, 4, 2, 0, 7, 0x7, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_VOLUME("Limiter Maximum Attenuation", + WM8776_LIMITER, 0, 6, 3, 12, 0xf, + LC_CONTROL_LIMITER, + wm8776_maxatten_lim_db_scale), + WM8776_FIELD_CTL_VOLUME("ALC Target Level", + WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, + LC_CONTROL_ALC, wm8776_lct_db_scale), + WM8776_FIELD_CTL_ENUM("ALC Attack Time", + WM8776_ALCCTRL3, 0, 2, 0, 10, 0xf, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_ENUM("ALC Decay Time", + WM8776_ALCCTRL3, 4, 3, 0, 10, 0xf, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_VOLUME("ALC Maximum Gain", + WM8776_ALCCTRL1, 4, 7, 1, 7, 0x7, + LC_CONTROL_ALC, wm8776_maxgain_db_scale), + WM8776_FIELD_CTL_VOLUME("ALC Maximum Attenuation", + WM8776_LIMITER, 0, 10, 10, 15, 0xf, + LC_CONTROL_ALC, wm8776_maxatten_alc_db_scale), + WM8776_FIELD_CTL_ENUM("ALC Hold Time", + WM8776_ALCCTRL2, 0, 0, 0, 15, 0xf, + LC_CONTROL_ALC), + WM8776_BIT_SWITCH("Noise Gate Capture Switch", + WM8776_NOISEGATE, WM8776_NGAT, 0, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_VOLUME("Noise Gate Threshold", + WM8776_NOISEGATE, 2, 0, 0, 7, 0x7, + LC_CONTROL_ALC, wm8776_ngth_db_scale), +}; + +static int xonar_ds_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + return 0; +} + +static int xonar_ds_mixer_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + for (i = 0; i < ARRAY_SIZE(ds_controls); ++i) { + ctl = snd_ctl_new1(&ds_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + } + BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); + for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { + ctl = snd_ctl_new1(&lc_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + data->lc_controls[i] = ctl; + } + return 0; +} + +static const struct oxygen_model model_xonar_ds = { + .shortname = "Xonar DS", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_ds_init, + .control_filter = xonar_ds_control_filter, + .mixer_init = xonar_ds_mixer_init, + .cleanup = xonar_ds_cleanup, + .suspend = xonar_ds_suspend, + .resume = xonar_ds_resume, + .pcm_hardware_filter = wm8776_adc_hardware_filter, + .get_i2s_mclk = oxygen_default_i2s_mclk, + .set_dac_params = set_wm87x6_dac_params, + .set_adc_params = set_wm8776_adc_params, + .update_dac_volume = update_wm87x6_volume, + .update_dac_mute = update_wm87x6_mute, + .gpio_changed = xonar_ds_gpio_changed, + .dac_tlv = wm87x6_dac_db_scale, + .model_data_size = sizeof(struct xonar_wm87x6), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1, + .dac_channels = 8, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_SPI, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +int __devinit get_xonar_wm87x6_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + switch (id->subdevice) { + case 0x838e: + chip->model = model_xonar_ds; + break; + default: + return -EINVAL; + } + return 0; +} -- cgit v1.2.2 From a5b5a0649a84db1a0cc1e19997572be8ef3b8c81 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 19 Jan 2010 11:15:45 +0200 Subject: ASoC: tlv320dac33: Correct the prefill number of samples Set the prefill number of samples as the same as the lower threshold in mode7. In this way the codec will read the same amount of data on startup and during the running playback. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2df9c20b7d52..65683aa3920c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -559,7 +559,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: dac33_write16(codec, DAC33_PREFILL_MSB, - DAC33_THRREG(20)); + DAC33_THRREG(10)); break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", -- cgit v1.2.2 From 4feabefe53eb3742f0b2773a43200d1686f3a288 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jan 2010 15:38:44 +0100 Subject: ALSA: hda - Fix parsing pin node 0x21 on ALC259 ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled properly in alc268_new_analog_output(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3f92def752fd..79cdae324c5e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12541,6 +12541,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x21: dac = 0x03; break; default: -- cgit v1.2.2 From 3fb4a508b8e7957aa899f32cd6d9d462e102c7ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jan 2010 15:46:37 +0100 Subject: ALSA: hda - Turn on EAPD only if available for Realtek codecs Some codecs disable widgets used for output pins and reserve as vendor- spec widgets. Thus we need to check the widget type and pin cap before actually sending SET_EAPD verbs in the auto-configuration mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++------------ 1 file changed, 17 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79cdae324c5e..6ae610c0111e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1093,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010); } +/* turn on/off EAPD control (only if available) */ +static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on) +{ + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return; + if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, + on ? 2 : 0); +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -1110,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case ALC_INIT_DEFAULT: switch (codec->vendor_id) { case 0x10ec0260: - snd_hda_codec_write(codec, 0x0f, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); + set_eapd(codec, 0x0f, 1); + set_eapd(codec, 0x10, 1); break; case 0x10ec0262: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0270: case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: case 0x10ec0862: case 0x10ec0889: - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); + set_eapd(codec, 0x14, 1); + set_eapd(codec, 0x15, 1); break; } switch (codec->vendor_id) { @@ -1836,10 +1843,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE static void alc889_power_eapd(struct hda_codec *codec, int power) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); + set_eapd(codec, 0x14, power); + set_eapd(codec, 0x15, power); } #endif -- cgit v1.2.2 From dc99be47667c56046555e89e62f1ac17fa06329a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jan 2010 08:35:06 +0100 Subject: ALSA: hda - Fix HP T5735 automute This patch fixes the aut-mute setup on HP T5735 with ALC262 codec. Instead of wrong amp, use pin control toggling for muting the speaker now. Tested-by: Lee Trager Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ae610c0111e..d00e6d1da085 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10382,7 +10382,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + spec->autocfg.speaker_pins[0] = 0x14; } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { @@ -11793,9 +11793,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hp_t5735_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_inithook, }, [ALC262_HP_RP5700] = { .mixers = { alc262_hp_rp5700_mixer }, -- cgit v1.2.2 From 6cd6cede8c33364d8e1abb5ea35adf627e3781b0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 20 Jan 2010 09:39:35 +0200 Subject: ASoC: tlv320dac33: BCLK divider fix The BCLK divider was not configured in case of mode7. This leads to unpredictable behavior when switching between FIFO modes. Configure the BCLK divider depending on the fifo_mode (FIFO is in use, or FIFO bypass). Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 65683aa3920c..e1aa66ff7f1c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -845,11 +845,14 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); - switch (dac33->fifo_mode) { - case DAC33_FIFO_MODE1: - /* 20: BCLK divide ratio */ + /* BCLK divide ratio */ + if (dac33->fifo_mode) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + else + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: dac33_write16(codec, DAC33_ATHR_MSB, DAC33_THRREG(dac33->alarm_threshold)); break; @@ -864,8 +867,6 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) DAC33_THRREG(10)); break; default: - /* BYPASS mode */ - dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); break; } -- cgit v1.2.2 From 6aceabb459c07a3fb4873c8306de8143c56241b2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 20 Jan 2010 09:39:36 +0200 Subject: ASoC: tlv320dac33: Burst mode BCLK divider configuration Add possibility to configure the burst mode BCLK divider through platform data structure. The BCLK divider changes the actual speed of the serial bus in burst mode, which is faster than the sampling frequency of the running stream. In this way platforms can experiment with the optimal burst speed without the need to modify the codec driver itself. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index e1aa66ff7f1c..1b35d0cf3364 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -91,6 +91,7 @@ struct tlv320dac33_priv { * this */ enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */ unsigned int nsample; /* burst read amount from host */ + u8 burst_bclkdiv; /* BCLK divider value in burst mode */ enum dac33_state state; }; @@ -845,9 +846,18 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); - /* BCLK divide ratio */ + /* + * BCLK divide ratio + * 0: 1.5 + * 1: 1 + * 2: 2 + * ... + * 254: 254 + * 255: 255 + */ if (dac33->fifo_mode) - dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, + dac33->burst_bclkdiv); else dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); @@ -1239,6 +1249,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, dac33); dac33->power_gpio = pdata->power_gpio; + dac33->burst_bclkdiv = pdata->burst_bclkdiv; dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; /* Disable FIFO use by default */ -- cgit v1.2.2 From c91a988dc6551c66418690e36b2a23cdb0255da8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 10:32:15 +0100 Subject: ALSA: pcm_core: Fix wake_up() optimization This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela --- sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 20 ++++++++++---------- sound/core/pcm_native.c | 3 +++ 3 files changed, 14 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index df57a0e30bf2..0d428d0896db 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -894,6 +894,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, memset((void*)runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); + init_waitqueue_head(&runtime->tsleep); runtime->status->state = SNDRV_PCM_STATE_OPEN; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5417f7dce834..e2a817eac2a9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -285,8 +285,8 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (!runtime->nowake && avail >= runtime->control->avail_min) - wake_up(&runtime->sleep); + if (avail >= runtime->control->avail_min) + wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); return 0; } @@ -1692,7 +1692,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, long tout; init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->sleep, &wait); + add_wait_queue(&runtime->tsleep, &wait); for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; @@ -1735,7 +1735,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, break; } _endloop: - remove_wait_queue(&runtime->sleep, &wait); + remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; } @@ -1794,7 +1794,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1816,7 +1816,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -1855,7 +1855,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); @@ -2016,7 +2016,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2045,7 +2045,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -2078,7 +2078,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 27284f628361..56ec35e8510b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -919,6 +919,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) runtime->status->state = state; } wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_stop = { @@ -1004,6 +1005,7 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) SNDRV_TIMER_EVENT_MPAUSE, &runtime->trigger_tstamp); wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; if (substream->timer) @@ -1061,6 +1063,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_suspend = { -- cgit v1.2.2 From b91b8fa02482a5a18f598ee5d2cd42970051731b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Jan 2010 18:18:35 +0000 Subject: ASoC: Remove console DAPM debug code The same information is now visible via debugfs and with large modern devices dumping everything to the console can be very resource intensive, causing more harm than good. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 80 ++-------------------------------------------------- 1 file changed, 3 insertions(+), 77 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index de22c2f1842e..d8e93749321e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -44,13 +44,6 @@ #include #include -/* debug */ -#ifdef DEBUG -#define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -#else -#define dump_dapm(codec, action) -#endif - /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -1063,66 +1056,6 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return 0; } -#ifdef DEBUG -static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) -{ - struct snd_soc_dapm_widget *w; - struct snd_soc_dapm_path *p = NULL; - int in, out; - - printk("DAPM %s %s\n", codec->name, action); - - list_for_each_entry(w, &codec->dapm_widgets, list) { - - /* only display widgets that effect routing */ - switch (w->id) { - case snd_soc_dapm_pre: - case snd_soc_dapm_post: - case snd_soc_dapm_vmid: - continue; - case snd_soc_dapm_mux: - case snd_soc_dapm_value_mux: - case snd_soc_dapm_output: - case snd_soc_dapm_input: - case snd_soc_dapm_switch: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_spk: - case snd_soc_dapm_line: - case snd_soc_dapm_micbias: - case snd_soc_dapm_dac: - case snd_soc_dapm_adc: - case snd_soc_dapm_pga: - case snd_soc_dapm_mixer: - case snd_soc_dapm_mixer_named_ctl: - case snd_soc_dapm_supply: - case snd_soc_dapm_aif_in: - case snd_soc_dapm_aif_out: - if (w->name) { - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - printk("%s: %s in %d out %d\n", w->name, - w->power ? "On":"Off",in, out); - - list_for_each_entry(p, &w->sources, list_sink) { - if (p->connect) - printk(" in %s %s\n", p->name ? p->name : "static", - p->source->name); - } - list_for_each_entry(p, &w->sinks, list_source) { - if (p->connect) - printk(" out %s %s\n", p->name ? p->name : "static", - p->sink->name); - } - } - break; - } - } -} -#endif - #ifdef CONFIG_DEBUG_FS static int dapm_widget_power_open_file(struct inode *inode, struct file *file) { @@ -1254,10 +1187,8 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, path->connect = 0; /* old connection must be powered down */ } - if (found) { + if (found) dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(widget->codec, "mux power update"); - } return 0; } @@ -1285,10 +1216,8 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, break; } - if (found) { + if (found) dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(widget->codec, "mixer power update"); - } return 0; } @@ -1404,9 +1333,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, */ int snd_soc_dapm_sync(struct snd_soc_codec *codec) { - int ret = dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(codec, "sync"); - return ret; + return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); @@ -2163,7 +2090,6 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, dapm_power_widgets(codec, event); mutex_unlock(&codec->mutex); - dump_dapm(codec, __func__); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); -- cgit v1.2.2 From a96ca3387382498ec8b501db5acef3ed9eb1bd36 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 19 Jan 2010 22:49:43 +0000 Subject: ASoC: Support turning off bias when the CODEC is idle Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d8e93749321e..6c3351095786 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1012,13 +1012,28 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) sys_power = 0; break; case SND_SOC_DAPM_STREAM_NOP: - sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + sys_power = 0; + break; + default: + sys_power = 1; + break; + } break; default: break; } } + if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); + if (ret != 0) + pr_err("Failed to turn on bias: %d\n", ret); + } + /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { @@ -1042,6 +1057,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pr_err("Failed to apply standby bias: %d\n", ret); } + /* If we're in standby and can support bias off then do that */ + if (codec->bias_level == SND_SOC_BIAS_STANDBY && + codec->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + if (ret != 0) + pr_err("Failed to turn off bias: %d\n", ret); + } + /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { ret = snd_soc_dapm_set_bias_level(socdev, -- cgit v1.2.2 From 821dd91ec7838e1313d783384ea9ce43510d4013 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 21 Jan 2010 11:33:20 +0000 Subject: ASoC: Use BIAS_OFF when idle for wm_hubs devices This provides a small power saving when audio is inactive. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index d73c30536a2c..a67319d9ca7e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -753,6 +753,12 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, WM8993_LINEOUT2_MODE, WM8993_LINEOUT2_MODE); + /* If the line outputs are differential then we aren't presenting + * VMID as an output and can disable it. + */ + if (lineout1_diff && lineout2_diff) + codec->idle_bias_off = 1; + if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); -- cgit v1.2.2 From fd0b092a7b14559e2ff17ef3aaefb5d8adc7e15f Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 14:54:38 +0100 Subject: ALSA: hda - AD1988 codec - fix SPDIF-input mixer initialization (unmute) The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate pin to get captured samples instead zeros. Tested on Lenovo Thinkstation. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cecd3c108990..865715e3f938 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2458,6 +2458,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { { } }; +static struct hda_verb ad1988_spdif_in_init_verbs[] = { + /* unmute SPDIF input pin */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + /* AD1989 has no ADC -> SPDIF route */ static struct hda_verb ad1989_spdif_init_verbs[] = { /* SPDIF-1 out pin */ @@ -3193,8 +3199,11 @@ static int patch_ad1988(struct hda_codec *codec) ad1988_spdif_init_verbs; } } - if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) + if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) { spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1988_spdif_in_init_verbs; + } codec->patch_ops = ad198x_patch_ops; switch (board_config) { -- cgit v1.2.2 From 5f6c3de6a79820de124fa2bb1b77d43a09410e42 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 23 Jan 2010 22:19:29 +0100 Subject: ALSA: hda - Minor fixes for Compaq Presario F700 quirk Minor fixes for HP Compaq Presario F700 quirks with Cxt5051 codec: - changed the capture mixer elements to the standard name. - fixed the quirk name string without a space - sorted the quirk list - updated the documentation Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 685015a53292..084600e40829 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1742,8 +1742,8 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { }; static struct snd_kcontrol_new cxt5051_f700_mixers[] = { - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1901,17 +1901,17 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", - [CXT5051_F700] = "hp 700" + [CXT5051_F700] = "hp-700", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), + SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), - SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), {} }; -- cgit v1.2.2 From 4e4ac60030293cb3d1e4bacf7c8be9aebdb8df61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 23 Jan 2010 22:29:54 +0100 Subject: ALSA: hda - Fix HP dv6736 capture mixer name Use the standard "Capture" mixer name for HP dv6736 with Cxt5051 codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 084600e40829..08c5b32dcd63 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1726,8 +1726,8 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { }; static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, -- cgit v1.2.2 From faddaa5d1c0cd29629c9c7e7a9d41ecb3149a064 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 23 Jan 2010 22:31:36 +0100 Subject: ALSA: hda - Add support for Toshiba Satellite M300 Added the support for Toshiba Satellite M300 with Conexant 5051 codec. Since the laptop has no port C connection and the pin reports always the jack sense true, we need to ignore port-C unsol event. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 38 +++++++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 08c5b32dcd63..56dda9c7f899 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -46,6 +46,8 @@ #define CXT5051_PORTB_EVENT 0x38 #define CXT5051_PORTC_EVENT 0x39 +#define AUTO_MIC_PORTB (1 << 1) +#define AUTO_MIC_PORTC (1 << 2) struct conexant_jack { @@ -74,7 +76,7 @@ struct conexant_spec { */ unsigned int cur_eapd; unsigned int hp_present; - unsigned int no_auto_mic; + unsigned int auto_mic; unsigned int need_dac_fix; /* capture */ @@ -1626,7 +1628,7 @@ static void cxt5051_portb_automic(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int present; - if (spec->no_auto_mic) + if (!(spec->auto_mic & AUTO_MIC_PORTB)) return; present = snd_hda_jack_detect(codec, 0x17); snd_hda_codec_write(codec, 0x14, 0, @@ -1641,7 +1643,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec) unsigned int present; hda_nid_t new_adc; - if (spec->no_auto_mic) + if (!(spec->auto_mic & AUTO_MIC_PORTC)) return; present = snd_hda_jack_detect(codec, 0x18); if (present) @@ -1757,6 +1759,24 @@ static struct snd_kcontrol_new cxt5051_f700_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { + HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1893,6 +1913,7 @@ enum { CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_F700, /* HP Compaq Presario F700 */ + CXT5051_TOSHIBA, /* Toshiba M300 & co */ CXT5051_MODELS }; @@ -1902,12 +1923,14 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", + [CXT5051_TOSHIBA] = "toshiba", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), + SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba M30x", CXT5051_TOSHIBA), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), @@ -1950,6 +1973,7 @@ static int patch_cxt5051(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, cxt5051_models, cxt5051_cfg_tbl); + spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC; switch (board_config) { case CXT5051_HP: spec->mixers[0] = cxt5051_hp_mixers; @@ -1957,7 +1981,7 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_HP_DV6736: spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; spec->mixers[0] = cxt5051_hp_dv6736_mixers; - spec->no_auto_mic = 1; + spec->auto_mic = 0; break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; @@ -1965,7 +1989,11 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_F700: spec->init_verbs[0] = cxt5051_f700_init_verbs; spec->mixers[0] = cxt5051_f700_mixers; - spec->no_auto_mic = 1; + spec->auto_mic = 0; + break; + case CXT5051_TOSHIBA: + spec->mixers[0] = cxt5051_toshiba_mixers; + spec->auto_mic = AUTO_MIC_PORTB; break; } -- cgit v1.2.2 From 2c7a3fb3f81df7318c70d2b8ecbd87f008e28d52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 10:47:02 +0100 Subject: ALSA: hda - Merge playback controls for Cx5051 codec models All cx5051 codec models have the same Master playback mixer definitions. Merge them together. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 62 +++++++++--------------------------------- 1 file changed, 13 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56dda9c7f899..e24bec6ca23a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1689,13 +1689,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec, conexant_report_jack(codec, nid); } -static struct snd_kcontrol_new cxt5051_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), +static struct snd_kcontrol_new cxt5051_playback_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1705,7 +1699,16 @@ static struct snd_kcontrol_new cxt5051_mixers[] = { .put = cxt5051_hp_master_sw_put, .private_value = 0x1a, }, + {} +}; +static struct snd_kcontrol_new cxt5051_capture_mixers[] = { + HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), {} }; @@ -1714,48 +1717,18 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("External Mic Volume", 0x15, 0x00, HDA_INPUT), HDA_CODEC_MUTE("External Mic Switch", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; static struct snd_kcontrol_new cxt5051_f700_mixers[] = { HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; @@ -1764,16 +1737,6 @@ static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; @@ -1958,8 +1921,9 @@ static int patch_cxt5051(struct hda_codec *codec) spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT; spec->num_adc_nids = 1; /* not 2; via auto-mic switch */ spec->adc_nids = cxt5051_adc_nids; - spec->num_mixers = 1; - spec->mixers[0] = cxt5051_mixers; + spec->num_mixers = 2; + spec->mixers[0] = cxt5051_capture_mixers; + spec->mixers[1] = cxt5051_playback_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5051_init_verbs; spec->spdif_route = 0; -- cgit v1.2.2 From 6953e5524a2ee0dcf57a83d8a6728d1262c54c37 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 11:00:27 +0100 Subject: ALSA: hda - initialize mic port on cxt5051 codec dynamically Initialize the mic ports B & C on Conexant 5051 codec dynamically according to the mic jack detection, instead of static init arrays. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e24bec6ca23a..4fbb398ccd67 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1765,8 +1765,6 @@ static struct hda_verb cxt5051_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, { } /* end */ }; @@ -1792,7 +1790,6 @@ static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, { } /* end */ }; @@ -1824,8 +1821,6 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, { } /* end */ }; @@ -1852,15 +1847,34 @@ static struct hda_verb cxt5051_f700_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, { } /* end */ }; +static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, + unsigned int event) +{ + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | event); +#ifdef CONFIG_SND_HDA_INPUT_JACK + conexant_add_jack(codec, nid, SND_JACK_MICROPHONE); + conexant_report_jack(codec, nid); +#endif +} + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + conexant_init(codec); conexant_init_jacks(codec); + + if (spec->auto_mic & AUTO_MIC_PORTB) + cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT); + if (spec->auto_mic & AUTO_MIC_PORTC) + cxt5051_init_mic_port(codec, 0x18, CXT5051_PORTC_EVENT); + if (codec->patch_ops.unsol_event) { cxt5051_hp_automute(codec); cxt5051_portb_automic(codec); -- cgit v1.2.2 From ecda0cff9df77d3f7d388bd4966e61f1947d2c95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 11:14:36 +0100 Subject: ALSA: hda - Fix SPDIF output widget for Cxt5051 codec Fixed the wrongly set up for SPDIF output on Conexant 5051 codec. It must point to the audio out widget instead of a pin. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4fbb398ccd67..250b74f8136e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -42,7 +42,7 @@ /* Conexant 5051 specific */ -#define CXT5051_SPDIF_OUT 0x1C +#define CXT5051_SPDIF_OUT 0x12 #define CXT5051_PORTB_EVENT 0x38 #define CXT5051_PORTC_EVENT 0x39 -- cgit v1.2.2 From 23d2df5b0db67fa90d3caf4b2d2f21ca33ec9c11 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 11:19:27 +0100 Subject: ALSA: hda - Change headphone pin control with master volume on cx5051 The HP pin (0x16) control has to be changed dynamically depending on the master volume switch as well as the speaker pin (0x1a). Otherwise the headphone still sounds with master off. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 250b74f8136e..9077e4174ee6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1605,6 +1605,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; unsigned int pinctl; + /* headphone pin */ + pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0; + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + /* speaker pin */ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); -- cgit v1.2.2 From 973b8cb0ead3e0b1dd3ee7b2df52e4dff1ffc707 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=C5=81ukasz=20Wojni=C5=82owicz?= Date: Sun, 24 Jan 2010 14:12:37 +0100 Subject: ALSA: hda - add possibility to choose speakers configuration for 4930g MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Now one can choose speaker configuration in e.g. PulseAudio mixer Signed-off-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d00e6d1da085..da34095c707f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9478,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, + .const_channel_count = 6, .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, -- cgit v1.2.2 From 40aa7030e5213a43e9e0554fd7f95534ea310bf3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 22 Jan 2010 18:00:03 +0100 Subject: ASoC: fix a memory-leak in wm8903 Remember to free the temporary register-cache. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8903.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce5515e3f2b0..3595bd57c4eb 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev) struct i2c_client *i2c = codec->control_data; int i; u16 *reg_cache = codec->reg_cache; - u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults), + u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults), GFP_KERNEL); /* Bring the codec back up to standby first to minimise pop/clicks */ @@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev) for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++) if (tmp_cache[i] != reg_cache[i]) snd_soc_write(codec, i, tmp_cache[i]); + kfree(tmp_cache); } else { dev_err(&i2c->dev, "Failed to allocate temporary cache\n"); } -- cgit v1.2.2 From 95f475f7a2e5d60fe9eeb7a2700753036a6ee6a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jan 2010 15:41:11 +0100 Subject: ALSA: hda - Remove coef output in Realtek proc files The output of COEF index/value in the proc file for Realtek codecs is rather useless since the value varies together with the index. Let's get rid of it again. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 31 ------------------------------- 1 file changed, 31 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c53faa959390..a3d223894642 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -841,27 +841,6 @@ static void add_verb(struct alc_spec *spec, const struct hda_verb *verb) spec->init_verbs[spec->num_init_verbs++] = verb; } -#ifdef CONFIG_PROC_FS -/* - * hook for proc - */ -static void print_realtek_coef(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) -{ - int coeff; - - if (nid != 0x20) - return; - coeff = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0); - snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff); - coeff = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_COEF_INDEX, 0); - snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff); -} -#else -#define print_realtek_coef NULL -#endif - /* * set up from the preset table */ @@ -5078,7 +5057,6 @@ static int patch_alc880(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -6688,7 +6666,6 @@ static int patch_alc260(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc260_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -10306,7 +10283,6 @@ static int patch_alc882(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc882_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -12170,7 +12146,6 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc262_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -13237,8 +13212,6 @@ static int patch_alc268(struct hda_codec *codec) if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; - codec->proc_widget_hook = print_realtek_coef; - return 0; } @@ -13955,7 +13928,6 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc269_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -15083,7 +15055,6 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -16063,7 +16034,6 @@ static int patch_alc861vd(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861vd_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -18198,7 +18168,6 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } -- cgit v1.2.2 From 0aea778efa0d632b62eb35122cbb3b9fae548c61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jan 2010 15:44:11 +0100 Subject: ALSA: hda - Remove the COEF setup for ALC267/ALC268 The COEF setup for model=auto seems problematic on some laptops, resulting in the silent speaker output. Better to disable it for now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a3d223894642..b2f543d3b833 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1145,6 +1145,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0888: alc888_coef_init(codec); break; +#if 0 /* XXX: This may cause the silent output on speaker on some machines */ case 0x10ec0267: case 0x10ec0268: snd_hda_codec_write(codec, 0x20, 0, @@ -1157,6 +1158,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) AC_VERB_SET_PROC_COEF, tmp | 0x3000); break; +#endif /* XXX */ } break; } -- cgit v1.2.2 From 895d4509d069f0706427ca75fcf0929ed136d0d7 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 22 Jan 2010 19:09:03 +0100 Subject: ASoC: add DAI and platform / DMA drivers for SH SIU Several SuperH platforms, including sh7722, sh7343, sh7354, sh7367 include a Sound Interface Unit (SIU). This patch adds DAI and platform / DMA drivers for this interface. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 2 + sound/soc/sh/siu.h | 193 +++++++++++ sound/soc/sh/siu_dai.c | 847 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/siu_pcm.c | 616 +++++++++++++++++++++++++++++++++++ 5 files changed, 1664 insertions(+) create mode 100644 sound/soc/sh/siu.h create mode 100644 sound/soc/sh/siu_dai.c create mode 100644 sound/soc/sh/siu_pcm.c (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 8072a6d1c4db..3f1cd5503342 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -26,6 +26,12 @@ config SND_SOC_SH4_FSI help This option enables FSI sound support +config SND_SOC_SH4_SIU + tristate + depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK + select DMADEVICES + select SH_DMAE + ## ## Boards ## diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 1d0ec0af74b7..5a97d2539d84 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -6,9 +6,11 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o snd-soc-hac-objs := hac.o snd-soc-ssi-objs := ssi.o snd-soc-fsi-objs := fsi.o +snd-soc-siu-objs := siu_pcm.o siu_dai.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o +obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h new file mode 100644 index 000000000000..9cc04ab2bce7 --- /dev/null +++ b/sound/soc/sh/siu.h @@ -0,0 +1,193 @@ +/* + * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2006 Carlos Munoz + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef SIU_H +#define SIU_H + +/* Common kernel and user-space firmware-building defines and types */ + +#define YRAM0_SIZE (0x0040 / 4) /* 16 */ +#define YRAM1_SIZE (0x0080 / 4) /* 32 */ +#define YRAM2_SIZE (0x0040 / 4) /* 16 */ +#define YRAM3_SIZE (0x0080 / 4) /* 32 */ +#define YRAM4_SIZE (0x0080 / 4) /* 32 */ +#define YRAM_DEF_SIZE (YRAM0_SIZE + YRAM1_SIZE + YRAM2_SIZE + \ + YRAM3_SIZE + YRAM4_SIZE) +#define YRAM_FIR_SIZE (0x0400 / 4) /* 256 */ +#define YRAM_IIR_SIZE (0x0200 / 4) /* 128 */ + +#define XRAM0_SIZE (0x0400 / 4) /* 256 */ +#define XRAM1_SIZE (0x0200 / 4) /* 128 */ +#define XRAM2_SIZE (0x0200 / 4) /* 128 */ + +/* PRAM program array size */ +#define PRAM0_SIZE (0x0100 / 4) /* 64 */ +#define PRAM1_SIZE ((0x2000 - 0x0100) / 4) /* 1984 */ + +#include + +struct siu_spb_param { + __u32 ab1a; /* input FIFO address */ + __u32 ab0a; /* output FIFO address */ + __u32 dir; /* 0=the ather except CPUOUTPUT, 1=CPUINPUT */ + __u32 event; /* SPB program starting conditions */ + __u32 stfifo; /* STFIFO register setting value */ + __u32 trdat; /* TRDAT register setting value */ +}; + +struct siu_firmware { + __u32 yram_fir_coeff[YRAM_FIR_SIZE]; + __u32 pram0[PRAM0_SIZE]; + __u32 pram1[PRAM1_SIZE]; + __u32 yram0[YRAM0_SIZE]; + __u32 yram1[YRAM1_SIZE]; + __u32 yram2[YRAM2_SIZE]; + __u32 yram3[YRAM3_SIZE]; + __u32 yram4[YRAM4_SIZE]; + __u32 spbpar_num; + struct siu_spb_param spbpar[32]; +}; + +#ifdef __KERNEL__ + +#include +#include +#include + +#include + +#include +#include +#include + +#define SIU_PERIOD_BYTES_MAX 8192 /* DMA transfer/period size */ +#define SIU_PERIOD_BYTES_MIN 256 /* DMA transfer/period size */ +#define SIU_PERIODS_MAX 64 /* Max periods in buffer */ +#define SIU_PERIODS_MIN 4 /* Min periods in buffer */ +#define SIU_BUFFER_BYTES_MAX (SIU_PERIOD_BYTES_MAX * SIU_PERIODS_MAX) + +/* SIU ports: only one can be used at a time */ +enum { + SIU_PORT_A, + SIU_PORT_B, + SIU_PORT_NUM, +}; + +/* SIU clock configuration */ +enum { + SIU_CLKA_PLL, + SIU_CLKA_EXT, + SIU_CLKB_PLL, + SIU_CLKB_EXT +}; + +struct siu_info { + int port_id; + u32 __iomem *pram; + u32 __iomem *xram; + u32 __iomem *yram; + u32 __iomem *reg; + struct siu_firmware fw; +}; + +struct siu_stream { + struct tasklet_struct tasklet; + struct snd_pcm_substream *substream; + snd_pcm_format_t format; + size_t buf_bytes; + size_t period_bytes; + int cur_period; /* Period currently in dma */ + u32 volume; + snd_pcm_sframes_t xfer_cnt; /* Number of frames */ + u8 rw_flg; /* transfer status */ + /* DMA status */ + struct dma_chan *chan; /* DMA channel */ + struct dma_async_tx_descriptor *tx_desc; + dma_cookie_t cookie; + struct sh_dmae_slave param; +}; + +struct siu_port { + unsigned long play_cap; /* Used to track full duplex */ + struct snd_pcm *pcm; + struct siu_stream playback; + struct siu_stream capture; + u32 stfifo; /* STFIFO value from firmware */ + u32 trdat; /* TRDAT value from firmware */ +}; + +extern struct siu_port *siu_ports[SIU_PORT_NUM]; + +static inline struct siu_port *siu_port_info(struct snd_pcm_substream *substream) +{ + struct platform_device *pdev = + to_platform_device(substream->pcm->card->dev); + return siu_ports[pdev->id]; +} + +/* Register access */ +static inline void siu_write32(u32 __iomem *addr, u32 val) +{ + __raw_writel(val, addr); +} + +static inline u32 siu_read32(u32 __iomem *addr) +{ + return __raw_readl(addr); +} + +/* SIU registers */ +#define SIU_IFCTL (0x000 / sizeof(u32)) +#define SIU_SRCTL (0x004 / sizeof(u32)) +#define SIU_SFORM (0x008 / sizeof(u32)) +#define SIU_CKCTL (0x00c / sizeof(u32)) +#define SIU_TRDAT (0x010 / sizeof(u32)) +#define SIU_STFIFO (0x014 / sizeof(u32)) +#define SIU_DPAK (0x01c / sizeof(u32)) +#define SIU_CKREV (0x020 / sizeof(u32)) +#define SIU_EVNTC (0x028 / sizeof(u32)) +#define SIU_SBCTL (0x040 / sizeof(u32)) +#define SIU_SBPSET (0x044 / sizeof(u32)) +#define SIU_SBFSTS (0x068 / sizeof(u32)) +#define SIU_SBDVCA (0x06c / sizeof(u32)) +#define SIU_SBDVCB (0x070 / sizeof(u32)) +#define SIU_SBACTIV (0x074 / sizeof(u32)) +#define SIU_DMAIA (0x090 / sizeof(u32)) +#define SIU_DMAIB (0x094 / sizeof(u32)) +#define SIU_DMAOA (0x098 / sizeof(u32)) +#define SIU_DMAOB (0x09c / sizeof(u32)) +#define SIU_DMAML (0x0a0 / sizeof(u32)) +#define SIU_SPSTS (0x0cc / sizeof(u32)) +#define SIU_SPCTL (0x0d0 / sizeof(u32)) +#define SIU_BRGASEL (0x100 / sizeof(u32)) +#define SIU_BRRA (0x104 / sizeof(u32)) +#define SIU_BRGBSEL (0x108 / sizeof(u32)) +#define SIU_BRRB (0x10c / sizeof(u32)) + +extern struct snd_soc_platform siu_platform; +extern struct snd_soc_dai siu_i2s_dai; + +int siu_init_port(int port, struct siu_port **port_info, struct snd_card *card); +void siu_free_port(struct siu_port *port_info); + +#endif + +#endif /* SIU_H */ diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c new file mode 100644 index 000000000000..5452d19607e1 --- /dev/null +++ b/sound/soc/sh/siu_dai.c @@ -0,0 +1,847 @@ +/* + * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2006 Carlos Munoz + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include +#include +#include + +#include +#include + +#include +#include + +#include "siu.h" + +/* Board specifics */ +#if defined(CONFIG_CPU_SUBTYPE_SH7722) +# define SIU_MAX_VOLUME 0x1000 +#else +# define SIU_MAX_VOLUME 0x7fff +#endif + +#define PRAM_SIZE 0x2000 +#define XRAM_SIZE 0x800 +#define YRAM_SIZE 0x800 + +#define XRAM_OFFSET 0x4000 +#define YRAM_OFFSET 0x6000 +#define REG_OFFSET 0xc000 + +#define PLAYBACK_ENABLED 1 +#define CAPTURE_ENABLED 2 + +#define VOLUME_CAPTURE 0 +#define VOLUME_PLAYBACK 1 +#define DFLT_VOLUME_LEVEL 0x08000800 + +/* + * SPDIF is only available on port A and on some SIU implementations it is only + * available for input. Due to the lack of hardware to test it, SPDIF is left + * disabled in this driver version + */ +struct format_flag { + u32 i2s; + u32 pcm; + u32 spdif; + u32 mask; +}; + +struct port_flag { + struct format_flag playback; + struct format_flag capture; +}; + +static struct port_flag siu_flags[SIU_PORT_NUM] = { + [SIU_PORT_A] = { + .playback = { + .i2s = 0x50000000, + .pcm = 0x40000000, + .spdif = 0x80000000, /* not on all SIU versions */ + .mask = 0xd0000000, + }, + .capture = { + .i2s = 0x05000000, + .pcm = 0x04000000, + .spdif = 0x08000000, + .mask = 0x0d000000, + }, + }, + [SIU_PORT_B] = { + .playback = { + .i2s = 0x00500000, + .pcm = 0x00400000, + .spdif = 0, /* impossible - turn off */ + .mask = 0x00500000, + }, + .capture = { + .i2s = 0x00050000, + .pcm = 0x00040000, + .spdif = 0, /* impossible - turn off */ + .mask = 0x00050000, + }, + }, +}; + +static void siu_dai_start(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + + dev_dbg(port_info->pcm->card->dev, "%s\n", __func__); + + /* Turn on SIU clock */ + pm_runtime_get_sync(siu_i2s_dai.dev); + + /* Issue software reset to siu */ + siu_write32(base + SIU_SRCTL, 0); + + /* Wait for the reset to take effect */ + udelay(1); + + port_info->stfifo = 0; + port_info->trdat = 0; + + /* portA, portB, SIU operate */ + siu_write32(base + SIU_SRCTL, 0x301); + + /* portA=256fs, portB=256fs */ + siu_write32(base + SIU_CKCTL, 0x40400000); + + /* portA's BRG does not divide SIUCKA */ + siu_write32(base + SIU_BRGASEL, 0); + siu_write32(base + SIU_BRRA, 0); + + /* portB's BRG divides SIUCKB by half */ + siu_write32(base + SIU_BRGBSEL, 1); + siu_write32(base + SIU_BRRB, 0); + + siu_write32(base + SIU_IFCTL, 0x44440000); + + /* portA: 32 bit/fs, master; portB: 32 bit/fs, master */ + siu_write32(base + SIU_SFORM, 0x0c0c0000); + + /* + * Volume levels: looks like the DSP firmware implements volume controls + * differently from what's described in the datasheet + */ + siu_write32(base + SIU_SBDVCA, port_info->playback.volume); + siu_write32(base + SIU_SBDVCB, port_info->capture.volume); +} + +static void siu_dai_stop(void) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + + /* SIU software reset */ + siu_write32(base + SIU_SRCTL, 0); + + /* Turn off SIU clock */ + pm_runtime_put_sync(siu_i2s_dai.dev); +} + +static void siu_dai_spbAselect(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_firmware *fw = &info->fw; + u32 *ydef = fw->yram0; + u32 idx; + + /* path A use */ + if (!info->port_id) + idx = 1; /* portA */ + else + idx = 2; /* portB */ + + ydef[0] = (fw->spbpar[idx].ab1a << 16) | + (fw->spbpar[idx].ab0a << 8) | + (fw->spbpar[idx].dir << 7) | 3; + ydef[1] = fw->yram0[1]; /* 0x03000300 */ + ydef[2] = (16 / 2) << 24; + ydef[3] = fw->yram0[3]; /* 0 */ + ydef[4] = fw->yram0[4]; /* 0 */ + ydef[7] = fw->spbpar[idx].event; + port_info->stfifo |= fw->spbpar[idx].stfifo; + port_info->trdat |= fw->spbpar[idx].trdat; +} + +static void siu_dai_spbBselect(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_firmware *fw = &info->fw; + u32 *ydef = fw->yram0; + u32 idx; + + /* path B use */ + if (!info->port_id) + idx = 7; /* portA */ + else + idx = 8; /* portB */ + + ydef[5] = (fw->spbpar[idx].ab1a << 16) | + (fw->spbpar[idx].ab0a << 8) | 1; + ydef[6] = fw->spbpar[idx].event; + port_info->stfifo |= fw->spbpar[idx].stfifo; + port_info->trdat |= fw->spbpar[idx].trdat; +} + +static void siu_dai_open(struct siu_stream *siu_stream) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 srctl, ifctl; + + srctl = siu_read32(base + SIU_SRCTL); + ifctl = siu_read32(base + SIU_IFCTL); + + switch (info->port_id) { + case SIU_PORT_A: + /* portA operates */ + srctl |= 0x200; + ifctl &= ~0xc2; + break; + case SIU_PORT_B: + /* portB operates */ + srctl |= 0x100; + ifctl &= ~0x31; + break; + } + + siu_write32(base + SIU_SRCTL, srctl); + /* Unmute and configure portA */ + siu_write32(base + SIU_IFCTL, ifctl); +} + +/* + * At the moment only fixed Left-upper, Left-lower, Right-upper, Right-lower + * packing is supported + */ +static void siu_dai_pcmdatapack(struct siu_stream *siu_stream) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 dpak; + + dpak = siu_read32(base + SIU_DPAK); + + switch (info->port_id) { + case SIU_PORT_A: + dpak &= ~0xc0000000; + break; + case SIU_PORT_B: + dpak &= ~0x00c00000; + break; + } + + siu_write32(base + SIU_DPAK, dpak); +} + +static int siu_dai_spbstart(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_firmware *fw = &info->fw; + u32 *ydef = fw->yram0; + int cnt; + u32 __iomem *add; + u32 *ptr; + + /* Load SPB Program in PRAM */ + ptr = fw->pram0; + add = info->pram; + for (cnt = 0; cnt < PRAM0_SIZE; cnt++, add++, ptr++) + siu_write32(add, *ptr); + + ptr = fw->pram1; + add = info->pram + (0x0100 / sizeof(u32)); + for (cnt = 0; cnt < PRAM1_SIZE; cnt++, add++, ptr++) + siu_write32(add, *ptr); + + /* XRAM initialization */ + add = info->xram; + for (cnt = 0; cnt < XRAM0_SIZE + XRAM1_SIZE + XRAM2_SIZE; cnt++, add++) + siu_write32(add, 0); + + /* YRAM variable area initialization */ + add = info->yram; + for (cnt = 0; cnt < YRAM_DEF_SIZE; cnt++, add++) + siu_write32(add, ydef[cnt]); + + /* YRAM FIR coefficient area initialization */ + add = info->yram + (0x0200 / sizeof(u32)); + for (cnt = 0; cnt < YRAM_FIR_SIZE; cnt++, add++) + siu_write32(add, fw->yram_fir_coeff[cnt]); + + /* YRAM IIR coefficient area initialization */ + add = info->yram + (0x0600 / sizeof(u32)); + for (cnt = 0; cnt < YRAM_IIR_SIZE; cnt++, add++) + siu_write32(add, 0); + + siu_write32(base + SIU_TRDAT, port_info->trdat); + port_info->trdat = 0x0; + + + /* SPB start condition: software */ + siu_write32(base + SIU_SBACTIV, 0); + /* Start SPB */ + siu_write32(base + SIU_SBCTL, 0xc0000000); + /* Wait for program to halt */ + cnt = 0x10000; + while (--cnt && siu_read32(base + SIU_SBCTL) != 0x80000000) + cpu_relax(); + + if (!cnt) + return -EBUSY; + + /* SPB program start address setting */ + siu_write32(base + SIU_SBPSET, 0x00400000); + /* SPB hardware start(FIFOCTL source) */ + siu_write32(base + SIU_SBACTIV, 0xc0000000); + + return 0; +} + +static void siu_dai_spbstop(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + + siu_write32(base + SIU_SBACTIV, 0); + /* SPB stop */ + siu_write32(base + SIU_SBCTL, 0); + + port_info->stfifo = 0; +} + +/* API functions */ + +/* Playback and capture hardware properties are identical */ +static struct snd_pcm_hardware siu_dai_pcm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = SIU_BUFFER_BYTES_MAX, + .period_bytes_min = SIU_PERIOD_BYTES_MIN, + .period_bytes_max = SIU_PERIOD_BYTES_MAX, + .periods_min = SIU_PERIODS_MIN, + .periods_max = SIU_PERIODS_MAX, +}; + +static int siu_dai_info_volume(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_info *uinfo) +{ + struct siu_port *port_info = snd_kcontrol_chip(kctrl); + + dev_dbg(port_info->pcm->card->dev, "%s\n", __func__); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SIU_MAX_VOLUME; + + return 0; +} + +static int siu_dai_get_volume(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) +{ + struct siu_port *port_info = snd_kcontrol_chip(kctrl); + struct device *dev = port_info->pcm->card->dev; + u32 vol; + + dev_dbg(dev, "%s\n", __func__); + + switch (kctrl->private_value) { + case VOLUME_PLAYBACK: + /* Playback is always on port 0 */ + vol = port_info->playback.volume; + ucontrol->value.integer.value[0] = vol & 0xffff; + ucontrol->value.integer.value[1] = vol >> 16 & 0xffff; + break; + case VOLUME_CAPTURE: + /* Capture is always on port 1 */ + vol = port_info->capture.volume; + ucontrol->value.integer.value[0] = vol & 0xffff; + ucontrol->value.integer.value[1] = vol >> 16 & 0xffff; + break; + default: + dev_err(dev, "%s() invalid private_value=%ld\n", + __func__, kctrl->private_value); + return -EINVAL; + } + + return 0; +} + +static int siu_dai_put_volume(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) +{ + struct siu_port *port_info = snd_kcontrol_chip(kctrl); + struct device *dev = port_info->pcm->card->dev; + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 new_vol; + u32 cur_vol; + + dev_dbg(dev, "%s\n", __func__); + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > SIU_MAX_VOLUME || + ucontrol->value.integer.value[1] < 0 || + ucontrol->value.integer.value[1] > SIU_MAX_VOLUME) + return -EINVAL; + + new_vol = ucontrol->value.integer.value[0] | + ucontrol->value.integer.value[1] << 16; + + /* See comment above - DSP firmware implementation */ + switch (kctrl->private_value) { + case VOLUME_PLAYBACK: + /* Playback is always on port 0 */ + cur_vol = port_info->playback.volume; + siu_write32(base + SIU_SBDVCA, new_vol); + port_info->playback.volume = new_vol; + break; + case VOLUME_CAPTURE: + /* Capture is always on port 1 */ + cur_vol = port_info->capture.volume; + siu_write32(base + SIU_SBDVCB, new_vol); + port_info->capture.volume = new_vol; + break; + default: + dev_err(dev, "%s() invalid private_value=%ld\n", + __func__, kctrl->private_value); + return -EINVAL; + } + + if (cur_vol != new_vol) + return 1; + + return 0; +} + +static struct snd_kcontrol_new playback_controls = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = siu_dai_info_volume, + .get = siu_dai_get_volume, + .put = siu_dai_put_volume, + .private_value = VOLUME_PLAYBACK, +}; + +static struct snd_kcontrol_new capture_controls = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Capture Volume", + .index = 0, + .info = siu_dai_info_volume, + .get = siu_dai_get_volume, + .put = siu_dai_put_volume, + .private_value = VOLUME_CAPTURE, +}; + +int siu_init_port(int port, struct siu_port **port_info, struct snd_card *card) +{ + struct device *dev = card->dev; + struct snd_kcontrol *kctrl; + int ret; + + *port_info = kzalloc(sizeof(**port_info), GFP_KERNEL); + if (!*port_info) + return -ENOMEM; + + dev_dbg(dev, "%s: port #%d@%p\n", __func__, port, *port_info); + + (*port_info)->playback.volume = DFLT_VOLUME_LEVEL; + (*port_info)->capture.volume = DFLT_VOLUME_LEVEL; + + /* + * Add mixer support. The SPB is used to change the volume. Both + * ports use the same SPB. Therefore, we only register one + * control instance since it will be used by both channels. + * In error case we continue without controls. + */ + kctrl = snd_ctl_new1(&playback_controls, *port_info); + ret = snd_ctl_add(card, kctrl); + if (ret < 0) + dev_err(dev, + "failed to add playback controls %p port=%d err=%d\n", + kctrl, port, ret); + + kctrl = snd_ctl_new1(&capture_controls, *port_info); + ret = snd_ctl_add(card, kctrl); + if (ret < 0) + dev_err(dev, + "failed to add capture controls %p port=%d err=%d\n", + kctrl, port, ret); + + return 0; +} + +void siu_free_port(struct siu_port *port_info) +{ + kfree(port_info); +} + +static int siu_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct snd_pcm_runtime *rt = substream->runtime; + struct siu_port *port_info = siu_port_info(substream); + int ret; + + dev_dbg(substream->pcm->card->dev, "%s: port=%d@%p\n", __func__, + info->port_id, port_info); + + snd_soc_set_runtime_hwparams(substream, &siu_dai_pcm_hw); + + ret = snd_pcm_hw_constraint_integer(rt, SNDRV_PCM_HW_PARAM_PERIODS); + if (unlikely(ret < 0)) + return ret; + + siu_dai_start(port_info); + + return 0; +} + +static void siu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(substream); + + dev_dbg(substream->pcm->card->dev, "%s: port=%d@%p\n", __func__, + info->port_id, port_info); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_info->play_cap &= ~PLAYBACK_ENABLED; + else + port_info->play_cap &= ~CAPTURE_ENABLED; + + /* Stop the siu if the other stream is not using it */ + if (!port_info->play_cap) { + /* during stmread or stmwrite ? */ + BUG_ON(port_info->playback.rw_flg || port_info->capture.rw_flg); + siu_dai_spbstop(port_info); + siu_dai_stop(); + } +} + +/* PCM part of siu_dai_playback_prepare() / siu_dai_capture_prepare() */ +static int siu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct snd_pcm_runtime *rt = substream->runtime; + struct siu_port *port_info = siu_port_info(substream); + struct siu_stream *siu_stream; + int self, ret; + + dev_dbg(substream->pcm->card->dev, + "%s: port %d, active streams %lx, %d channels\n", + __func__, info->port_id, port_info->play_cap, rt->channels); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + self = PLAYBACK_ENABLED; + siu_stream = &port_info->playback; + } else { + self = CAPTURE_ENABLED; + siu_stream = &port_info->capture; + } + + /* Set up the siu if not already done */ + if (!port_info->play_cap) { + siu_stream->rw_flg = 0; /* stream-data transfer flag */ + + siu_dai_spbAselect(port_info); + siu_dai_spbBselect(port_info); + + siu_dai_open(siu_stream); + + siu_dai_pcmdatapack(siu_stream); + + ret = siu_dai_spbstart(port_info); + if (ret < 0) + goto fail; + } + + port_info->play_cap |= self; + +fail: + return ret; +} + +/* + * SIU can set bus format to I2S / PCM / SPDIF independently for playback and + * capture, however, the current API sets the bus format globally for a DAI. + */ +static int siu_dai_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 ifctl; + + dev_dbg(dai->dev, "%s: fmt 0x%x on port %d\n", + __func__, fmt, info->port_id); + + if (info->port_id < 0) + return -ENODEV; + + /* Here select between I2S / PCM / SPDIF */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ifctl = siu_flags[info->port_id].playback.i2s | + siu_flags[info->port_id].capture.i2s; + break; + case SND_SOC_DAIFMT_LEFT_J: + ifctl = siu_flags[info->port_id].playback.pcm | + siu_flags[info->port_id].capture.pcm; + break; + /* SPDIF disabled - see comment at the top */ + default: + return -EINVAL; + } + + ifctl |= ~(siu_flags[info->port_id].playback.mask | + siu_flags[info->port_id].capture.mask) & + siu_read32(base + SIU_IFCTL); + siu_write32(base + SIU_IFCTL, ifctl); + + return 0; +} + +static int siu_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct clk *siu_clk, *parent_clk; + char *siu_name, *parent_name; + int ret; + + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + + dev_dbg(dai->dev, "%s: using clock %d\n", __func__, clk_id); + + switch (clk_id) { + case SIU_CLKA_PLL: + siu_name = "siua_clk"; + parent_name = "pll_clk"; + break; + case SIU_CLKA_EXT: + siu_name = "siua_clk"; + parent_name = "siumcka_clk"; + break; + case SIU_CLKB_PLL: + siu_name = "siub_clk"; + parent_name = "pll_clk"; + break; + case SIU_CLKB_EXT: + siu_name = "siub_clk"; + parent_name = "siumckb_clk"; + break; + default: + return -EINVAL; + } + + siu_clk = clk_get(siu_i2s_dai.dev, siu_name); + if (IS_ERR(siu_clk)) + return PTR_ERR(siu_clk); + + parent_clk = clk_get(siu_i2s_dai.dev, parent_name); + if (!IS_ERR(parent_clk)) { + ret = clk_set_parent(siu_clk, parent_clk); + if (!ret) + clk_set_rate(siu_clk, freq); + clk_put(parent_clk); + } + + clk_put(siu_clk); + + return 0; +} + +static struct snd_soc_dai_ops siu_dai_ops = { + .startup = siu_dai_startup, + .shutdown = siu_dai_shutdown, + .prepare = siu_dai_prepare, + .set_sysclk = siu_dai_set_sysclk, + .set_fmt = siu_dai_set_fmt, +}; + +struct snd_soc_dai siu_i2s_dai = { + .name = "sh-siu", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_8000_48000, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_8000_48000, + }, + .ops = &siu_dai_ops, +}; +EXPORT_SYMBOL_GPL(siu_i2s_dai); + +static int __devinit siu_probe(struct platform_device *pdev) +{ + const struct firmware *fw_entry; + struct resource *res, *region; + struct siu_info *info; + int ret; + + info = kmalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev); + if (ret) + goto ereqfw; + + /* + * Loaded firmware is "const" - read only, but we have to modify it in + * snd_siu_sh7343_spbAselect() and snd_siu_sh7343_spbBselect() + */ + memcpy(&info->fw, fw_entry->data, fw_entry->size); + + release_firmware(fw_entry); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + goto egetres; + } + + region = request_mem_region(res->start, resource_size(res), + pdev->name); + if (!region) { + dev_err(&pdev->dev, "SIU region already claimed\n"); + ret = -EBUSY; + goto ereqmemreg; + } + + ret = -ENOMEM; + info->pram = ioremap(res->start, PRAM_SIZE); + if (!info->pram) + goto emappram; + info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE); + if (!info->xram) + goto emapxram; + info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE); + if (!info->yram) + goto emapyram; + info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) - + REG_OFFSET); + if (!info->reg) + goto emapreg; + + siu_i2s_dai.dev = &pdev->dev; + siu_i2s_dai.private_data = info; + + ret = snd_soc_register_dais(&siu_i2s_dai, 1); + if (ret < 0) + goto edaiinit; + + ret = snd_soc_register_platform(&siu_platform); + if (ret < 0) + goto esocregp; + + pm_runtime_enable(&pdev->dev); + + return ret; + +esocregp: + snd_soc_unregister_dais(&siu_i2s_dai, 1); +edaiinit: + iounmap(info->reg); +emapreg: + iounmap(info->yram); +emapyram: + iounmap(info->xram); +emapxram: + iounmap(info->pram); +emappram: + release_mem_region(res->start, resource_size(res)); +ereqmemreg: +egetres: +ereqfw: + kfree(info); + + return ret; +} + +static int __devexit siu_remove(struct platform_device *pdev) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct resource *res; + + pm_runtime_disable(&pdev->dev); + + snd_soc_unregister_platform(&siu_platform); + snd_soc_unregister_dais(&siu_i2s_dai, 1); + + iounmap(info->reg); + iounmap(info->yram); + iounmap(info->xram); + iounmap(info->pram); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res) + release_mem_region(res->start, resource_size(res)); + kfree(info); + + return 0; +} + +static struct platform_driver siu_driver = { + .driver = { + .name = "sh_siu", + }, + .probe = siu_probe, + .remove = __devexit_p(siu_remove), +}; + +static int __init siu_init(void) +{ + return platform_driver_register(&siu_driver); +} + +static void __exit siu_exit(void) +{ + platform_driver_unregister(&siu_driver); +} + +module_init(siu_init) +module_exit(siu_exit) + +MODULE_AUTHOR("Carlos Munoz "); +MODULE_DESCRIPTION("ALSA SoC SH7722 SIU driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c new file mode 100644 index 000000000000..c5efc30f0136 --- /dev/null +++ b/sound/soc/sh/siu_pcm.c @@ -0,0 +1,616 @@ +/* + * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2006 Carlos Munoz + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "siu.h" + +#define GET_MAX_PERIODS(buf_bytes, period_bytes) \ + ((buf_bytes) / (period_bytes)) +#define PERIOD_OFFSET(buf_addr, period_num, period_bytes) \ + ((buf_addr) + ((period_num) * (period_bytes))) + +#define RWF_STM_RD 0x01 /* Read in progress */ +#define RWF_STM_WT 0x02 /* Write in progress */ + +struct siu_port *siu_ports[SIU_PORT_NUM]; + +/* transfersize is number of u32 dma transfers per period */ +static int siu_pcm_stmwrite_stop(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->playback; + u32 stfifo; + + if (!siu_stream->rw_flg) + return -EPERM; + + /* output FIFO disable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, stfifo & ~0x0c180c18); + pr_debug("%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo & ~0x0c180c18); + + /* during stmwrite clear */ + siu_stream->rw_flg = 0; + + return 0; +} + +static int siu_pcm_stmwrite_start(struct siu_port *port_info) +{ + struct siu_stream *siu_stream = &port_info->playback; + + if (siu_stream->rw_flg) + return -EPERM; + + /* Current period in buffer */ + port_info->playback.cur_period = 0; + + /* during stmwrite flag set */ + siu_stream->rw_flg = RWF_STM_WT; + + /* DMA transfer start */ + tasklet_schedule(&siu_stream->tasklet); + + return 0; +} + +static void siu_dma_tx_complete(void *arg) +{ + struct siu_stream *siu_stream = arg; + + if (!siu_stream->rw_flg) + return; + + /* Update completed period count */ + if (++siu_stream->cur_period >= + GET_MAX_PERIODS(siu_stream->buf_bytes, + siu_stream->period_bytes)) + siu_stream->cur_period = 0; + + pr_debug("%s: done period #%d (%u/%u bytes), cookie %d\n", + __func__, siu_stream->cur_period, + siu_stream->cur_period * siu_stream->period_bytes, + siu_stream->buf_bytes, siu_stream->cookie); + + tasklet_schedule(&siu_stream->tasklet); + + /* Notify alsa: a period is done */ + snd_pcm_period_elapsed(siu_stream->substream); +} + +static int siu_pcm_wr_set(struct siu_port *port_info, + dma_addr_t buff, u32 size) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->playback; + struct snd_pcm_substream *substream = siu_stream->substream; + struct device *dev = substream->pcm->card->dev; + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + struct scatterlist sg; + u32 stfifo; + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buff)), + size, offset_in_page(buff)); + sg_dma_address(&sg) = buff; + + desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, + &sg, 1, DMA_TO_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dev, "Failed to allocate a dma descriptor\n"); + return -ENOMEM; + } + + desc->callback = siu_dma_tx_complete; + desc->callback_param = siu_stream; + cookie = desc->tx_submit(desc); + if (cookie < 0) { + dev_err(dev, "Failed to submit a dma transfer\n"); + return cookie; + } + + siu_stream->tx_desc = desc; + siu_stream->cookie = cookie; + + dma_async_issue_pending(siu_stream->chan); + + /* only output FIFO enable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, stfifo | (port_info->stfifo & 0x0c180c18)); + dev_dbg(dev, "%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo | (port_info->stfifo & 0x0c180c18)); + + return 0; +} + +static int siu_pcm_rd_set(struct siu_port *port_info, + dma_addr_t buff, size_t size) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->capture; + struct snd_pcm_substream *substream = siu_stream->substream; + struct device *dev = substream->pcm->card->dev; + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + struct scatterlist sg; + u32 stfifo; + + dev_dbg(dev, "%s: %u@%llx\n", __func__, size, (unsigned long long)buff); + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buff)), + size, offset_in_page(buff)); + sg_dma_address(&sg) = buff; + + desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, + &sg, 1, DMA_FROM_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dev, "Failed to allocate dma descriptor\n"); + return -ENOMEM; + } + + desc->callback = siu_dma_tx_complete; + desc->callback_param = siu_stream; + cookie = desc->tx_submit(desc); + if (cookie < 0) { + dev_err(dev, "Failed to submit dma descriptor\n"); + return cookie; + } + + siu_stream->tx_desc = desc; + siu_stream->cookie = cookie; + + dma_async_issue_pending(siu_stream->chan); + + /* only input FIFO enable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, siu_read32(base + SIU_STFIFO) | + (port_info->stfifo & 0x13071307)); + dev_dbg(dev, "%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo | (port_info->stfifo & 0x13071307)); + + return 0; +} + +static void siu_io_tasklet(unsigned long data) +{ + struct siu_stream *siu_stream = (struct siu_stream *)data; + struct snd_pcm_substream *substream = siu_stream->substream; + struct device *dev = substream->pcm->card->dev; + struct snd_pcm_runtime *rt = substream->runtime; + struct siu_port *port_info = siu_port_info(substream); + + dev_dbg(dev, "%s: flags %x\n", __func__, siu_stream->rw_flg); + + if (!siu_stream->rw_flg) { + dev_dbg(dev, "%s: stream inactive\n", __func__); + return; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + dma_addr_t buff; + size_t count; + u8 *virt; + + buff = (dma_addr_t)PERIOD_OFFSET(rt->dma_addr, + siu_stream->cur_period, + siu_stream->period_bytes); + virt = PERIOD_OFFSET(rt->dma_area, + siu_stream->cur_period, + siu_stream->period_bytes); + count = siu_stream->period_bytes; + + /* DMA transfer start */ + siu_pcm_rd_set(port_info, buff, count); + } else { + siu_pcm_wr_set(port_info, + (dma_addr_t)PERIOD_OFFSET(rt->dma_addr, + siu_stream->cur_period, + siu_stream->period_bytes), + siu_stream->period_bytes); + } +} + +/* Capture */ +static int siu_pcm_stmread_start(struct siu_port *port_info) +{ + struct siu_stream *siu_stream = &port_info->capture; + + if (siu_stream->xfer_cnt > 0x1000000) + return -EINVAL; + if (siu_stream->rw_flg) + return -EPERM; + + /* Current period in buffer */ + siu_stream->cur_period = 0; + + /* during stmread flag set */ + siu_stream->rw_flg = RWF_STM_RD; + + tasklet_schedule(&siu_stream->tasklet); + + return 0; +} + +static int siu_pcm_stmread_stop(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->capture; + struct device *dev = siu_stream->substream->pcm->card->dev; + u32 stfifo; + + if (!siu_stream->rw_flg) + return -EPERM; + + /* input FIFO disable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, stfifo & ~0x13071307); + dev_dbg(dev, "%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo & ~0x13071307); + + /* during stmread flag clear */ + siu_stream->rw_flg = 0; + + return 0; +} + +static int siu_pcm_hw_params(struct snd_pcm_substream *ss, + struct snd_pcm_hw_params *hw_params) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct device *dev = ss->pcm->card->dev; + int ret; + + dev_dbg(dev, "%s: port=%d\n", __func__, info->port_id); + + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(hw_params)); + if (ret < 0) + dev_err(dev, "snd_pcm_lib_malloc_pages() failed\n"); + + return ret; +} + +static int siu_pcm_hw_free(struct snd_pcm_substream *ss) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(ss); + struct device *dev = ss->pcm->card->dev; + struct siu_stream *siu_stream; + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + dev_dbg(dev, "%s: port=%d\n", __func__, info->port_id); + + return snd_pcm_lib_free_pages(ss); +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct sh_dmae_slave *param = slave; + + pr_debug("%s: slave ID %d\n", __func__, param->slave_id); + + if (unlikely(param->dma_dev != chan->device->dev)) + return false; + + chan->private = param; + return true; +} + +static int siu_pcm_open(struct snd_pcm_substream *ss) +{ + /* Playback / Capture */ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(ss); + struct siu_stream *siu_stream; + u32 port = info->port_id; + struct siu_platform *pdata = siu_i2s_dai.dev->platform_data; + struct device *dev = ss->pcm->card->dev; + dma_cap_mask_t mask; + struct sh_dmae_slave *param; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dev_dbg(dev, "%s, port=%d@%p\n", __func__, port, port_info); + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) { + siu_stream = &port_info->playback; + param = &siu_stream->param; + param->slave_id = port ? SHDMA_SLAVE_SIUB_TX : + SHDMA_SLAVE_SIUA_TX; + } else { + siu_stream = &port_info->capture; + param = &siu_stream->param; + param->slave_id = port ? SHDMA_SLAVE_SIUB_RX : + SHDMA_SLAVE_SIUA_RX; + } + + param->dma_dev = pdata->dma_dev; + /* Get DMA channel */ + siu_stream->chan = dma_request_channel(mask, filter, param); + if (!siu_stream->chan) { + dev_err(dev, "DMA channel allocation failed!\n"); + return -EBUSY; + } + + siu_stream->substream = ss; + + return 0; +} + +static int siu_pcm_close(struct snd_pcm_substream *ss) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct device *dev = ss->pcm->card->dev; + struct siu_port *port_info = siu_port_info(ss); + struct siu_stream *siu_stream; + + dev_dbg(dev, "%s: port=%d\n", __func__, info->port_id); + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + dma_release_channel(siu_stream->chan); + siu_stream->chan = NULL; + + siu_stream->substream = NULL; + + return 0; +} + +static int siu_pcm_prepare(struct snd_pcm_substream *ss) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(ss); + struct device *dev = ss->pcm->card->dev; + struct snd_pcm_runtime *rt = ss->runtime; + struct siu_stream *siu_stream; + snd_pcm_sframes_t xfer_cnt; + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + rt = siu_stream->substream->runtime; + + siu_stream->buf_bytes = snd_pcm_lib_buffer_bytes(ss); + siu_stream->period_bytes = snd_pcm_lib_period_bytes(ss); + + dev_dbg(dev, "%s: port=%d, %d channels, period=%u bytes\n", __func__, + info->port_id, rt->channels, siu_stream->period_bytes); + + /* We only support buffers that are multiples of the period */ + if (siu_stream->buf_bytes % siu_stream->period_bytes) { + dev_err(dev, "%s() - buffer=%d not multiple of period=%d\n", + __func__, siu_stream->buf_bytes, + siu_stream->period_bytes); + return -EINVAL; + } + + xfer_cnt = bytes_to_frames(rt, siu_stream->period_bytes); + if (!xfer_cnt || xfer_cnt > 0x1000000) + return -EINVAL; + + siu_stream->format = rt->format; + siu_stream->xfer_cnt = xfer_cnt; + + dev_dbg(dev, "port=%d buf=%lx buf_bytes=%d period_bytes=%d " + "format=%d channels=%d xfer_cnt=%d\n", info->port_id, + (unsigned long)rt->dma_addr, siu_stream->buf_bytes, + siu_stream->period_bytes, + siu_stream->format, rt->channels, (int)xfer_cnt); + + return 0; +} + +static int siu_pcm_trigger(struct snd_pcm_substream *ss, int cmd) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct device *dev = ss->pcm->card->dev; + struct siu_port *port_info = siu_port_info(ss); + int ret; + + dev_dbg(dev, "%s: port=%d@%p, cmd=%d\n", __func__, + info->port_id, port_info, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = siu_pcm_stmwrite_start(port_info); + else + ret = siu_pcm_stmread_start(port_info); + + if (ret < 0) + dev_warn(dev, "%s: start failed on port=%d\n", + __func__, info->port_id); + + break; + case SNDRV_PCM_TRIGGER_STOP: + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_pcm_stmwrite_stop(port_info); + else + siu_pcm_stmread_stop(port_info); + ret = 0; + + break; + default: + dev_err(dev, "%s() unsupported cmd=%d\n", __func__, cmd); + ret = -EINVAL; + } + + return ret; +} + +/* + * So far only resolution of one period is supported, subject to extending the + * dmangine API + */ +static snd_pcm_uframes_t siu_pcm_pointer_dma(struct snd_pcm_substream *ss) +{ + struct device *dev = ss->pcm->card->dev; + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_port *port_info = siu_port_info(ss); + struct snd_pcm_runtime *rt = ss->runtime; + size_t ptr; + struct siu_stream *siu_stream; + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + /* + * ptr is the offset into the buffer where the dma is currently at. We + * check if the dma buffer has just wrapped. + */ + ptr = PERIOD_OFFSET(rt->dma_addr, + siu_stream->cur_period, + siu_stream->period_bytes) - rt->dma_addr; + + dev_dbg(dev, + "%s: port=%d, events %x, FSTS %x, xferred %u/%u, cookie %d\n", + __func__, info->port_id, siu_read32(base + SIU_EVNTC), + siu_read32(base + SIU_SBFSTS), ptr, siu_stream->buf_bytes, + siu_stream->cookie); + + if (ptr >= siu_stream->buf_bytes) + ptr = 0; + + return bytes_to_frames(ss->runtime, ptr); +} + +static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + /* card->dev == socdev->dev, see snd_soc_new_pcms() */ + struct siu_info *info = siu_i2s_dai.private_data; + struct platform_device *pdev = to_platform_device(card->dev); + int ret; + int i; + + /* pdev->id selects between SIUA and SIUB */ + if (pdev->id < 0 || pdev->id >= SIU_PORT_NUM) + return -EINVAL; + + info->port_id = pdev->id; + + /* + * While the siu has 2 ports, only one port can be on at a time (only 1 + * SPB). So far all the boards using the siu had only one of the ports + * wired to a codec. To simplify things, we only register one port with + * alsa. In case both ports are needed, it should be changed here + */ + for (i = pdev->id; i < pdev->id + 1; i++) { + struct siu_port **port_info = &siu_ports[i]; + + ret = siu_init_port(i, port_info, card); + if (ret < 0) + return ret; + + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, NULL, + SIU_BUFFER_BYTES_MAX, SIU_BUFFER_BYTES_MAX); + if (ret < 0) { + dev_err(card->dev, + "snd_pcm_lib_preallocate_pages_for_all() err=%d", + ret); + goto fail; + } + + (*port_info)->pcm = pcm; + + /* IO tasklets */ + tasklet_init(&(*port_info)->playback.tasklet, siu_io_tasklet, + (unsigned long)&(*port_info)->playback); + tasklet_init(&(*port_info)->capture.tasklet, siu_io_tasklet, + (unsigned long)&(*port_info)->capture); + } + + dev_info(card->dev, "SuperH SIU driver initialized.\n"); + return 0; + +fail: + siu_free_port(siu_ports[pdev->id]); + dev_err(card->dev, "SIU: failed to initialize.\n"); + return ret; +} + +static void siu_pcm_free(struct snd_pcm *pcm) +{ + struct platform_device *pdev = to_platform_device(pcm->card->dev); + struct siu_port *port_info = siu_ports[pdev->id]; + + tasklet_kill(&port_info->capture.tasklet); + tasklet_kill(&port_info->playback.tasklet); + + siu_free_port(port_info); + snd_pcm_lib_preallocate_free_for_all(pcm); + + dev_dbg(pcm->card->dev, "%s\n", __func__); +} + +static struct snd_pcm_ops siu_pcm_ops = { + .open = siu_pcm_open, + .close = siu_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = siu_pcm_hw_params, + .hw_free = siu_pcm_hw_free, + .prepare = siu_pcm_prepare, + .trigger = siu_pcm_trigger, + .pointer = siu_pcm_pointer_dma, +}; + +struct snd_soc_platform siu_platform = { + .name = "siu-audio", + .pcm_ops = &siu_pcm_ops, + .pcm_new = siu_pcm_new, + .pcm_free = siu_pcm_free, +}; +EXPORT_SYMBOL_GPL(siu_platform); -- cgit v1.2.2 From 84549d239ab9bb2e3a85c6efcf0e6478a38b4260 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Mon, 25 Jan 2010 16:42:25 +0800 Subject: ASoC: ad1836: reset and restore clock control mode in suspend/resume entry tests show frequent suspend/resume(frequent poweroff/on ad1836 internal components) maybe make ad1836 clock mode wrong sometimes after wakeup. This patch reset/restore ad1836 clock mode while executing PM, then ad1836 can always resume to right clock status. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1836.h | 1 + 2 files changed, 33 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 2c18e3d1b71e..83add2f3afba 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,6 +223,36 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } +#ifdef CONFIG_PM +static int ad1836_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* reset clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} + +static int ad1836_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* restore clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 |= AD1836_ADC_AUX; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} +#else +#define ad1836_soc_suspend NULL +#define ad1836_soc_resume NULL +#endif + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -404,6 +434,8 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, + .suspend = ad1836_soc_suspend, + .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 7660ee6973c0..e9d90d3951c5 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -54,6 +54,7 @@ #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) +#define AD1836_ADC_AUX (0x6 << 6) #define AD1836_ADC_CTRL3 14 -- cgit v1.2.2 From cf944ee55cc318bdb1d4b2f3f5cce3257f7c07b3 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Tue, 26 Jan 2010 09:06:14 +0100 Subject: ALSA: cs46xx: Fix cpu idling with resume Make sure that capture DMA doesn't stay enabled after system resume as that potentially prevents the processor from entering deep sleep states. Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index e6b4a879ae2e..56fcf00c0e27 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3644,6 +3644,7 @@ int snd_cs46xx_resume(struct pci_dev *pci) #ifdef CONFIG_SND_CS46XX_NEW_DSP int i; #endif + unsigned int tmp; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -3685,6 +3686,15 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* + * Stop capture DMA. + */ + tmp = snd_cs46xx_peek(chip, BA1_CCTL); + chip->capt.ctl = tmp & 0x0000ffff; + snd_cs46xx_poke(chip, BA1_CCTL, tmp & 0xffff0000); + + mdelay(5); + /* reset playback/capture */ snd_cs46xx_set_play_sample_rate(chip, 8000); snd_cs46xx_set_capture_sample_rate(chip, 8000); -- cgit v1.2.2 From ccc5df058da70d1c26c72cd1c24072a89998d735 Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Tue, 26 Jan 2010 15:59:33 +0800 Subject: ALSA: hda - Add support for more the 8 streams In azx_stream_start() and azx_stream_stop(), it use azx_readb/azx_writeb to read/write SIE, it just enable/disable 8 streams. But according to the HDA spec, it support 30 streams, and the new HDA controller will support more then 8 streams. So we should use azx_readl/azx_writel to read/write SIE. Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6d331c4cf185..6eeefda63838 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -954,8 +954,8 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) azx_dev->insufficient = 1; /* enable SIE */ - azx_writeb(chip, INTCTL, - azx_readb(chip, INTCTL) | (1 << azx_dev->index)); + azx_writel(chip, INTCTL, + azx_readl(chip, INTCTL) | (1 << azx_dev->index)); /* set DMA start and interrupt mask */ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_DMA_START | SD_INT_MASK); @@ -974,8 +974,8 @@ static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) { azx_stream_clear(chip, azx_dev); /* disable SIE */ - azx_writeb(chip, INTCTL, - azx_readb(chip, INTCTL) & ~(1 << azx_dev->index)); + azx_writel(chip, INTCTL, + azx_readl(chip, INTCTL) & ~(1 << azx_dev->index)); } -- cgit v1.2.2 From e473b847424bd215b686cbc1e781e82c904ee967 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Wed, 20 Jan 2010 17:06:33 +0530 Subject: ASoC: DaVinci: Fix stream restart error Sometimes after a suspend-resume cycle, the ALSA application restarts the stream when resume fails and McASP fails to work as the clock is not enabled. This patch corrects this bug. Testes on TI DA850/OMAP-L138 EVM. Signed-off-by: Chaithrika U S Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a613bbb0bc91..ab6518d86f18 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -768,13 +768,12 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!dev->clk_active) { clk_enable(dev->clk); dev->clk_active = 1; } - /* Fall through */ - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; -- cgit v1.2.2 From e7636925789b042ff9d98c51d48392e8c5549480 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 26 Jan 2010 17:08:24 +0100 Subject: ALSA: pcm_lib - return back hw_ptr_interrupt Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela --- sound/core/oss/pcm_oss.c | 3 +-- sound/core/pcm_lib.c | 7 +++++-- sound/core/pcm_native.c | 2 ++ 3 files changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 255ad910077a..82d4e3329b3d 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -635,8 +635,7 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) static inline snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) { - snd_pcm_uframes_t ptr = runtime->status->hw_ptr; - return ptr - (ptr % runtime->period_size); + return runtime->hw_ptr_interrupt; } /* define extended formats in the recent OSS versions (if any) */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e2a817eac2a9..aa54195ef3b0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -325,8 +325,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ - delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) - + runtime->period_size; + delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) @@ -437,6 +436,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + if (in_interrupt) { + runtime->hw_ptr_interrupt = new_hw_ptr - + (new_hw_ptr % runtime->period_size); + } runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 56ec35e8510b..7a002db512b4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1252,6 +1252,8 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; + runtime->hw_ptr_interrupt = runtime->status->hw_ptr - + runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v1.2.2 From 63b62ab0d52c736b3274b294df962e0a4b7aae79 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Wed, 27 Jan 2010 11:46:17 +0800 Subject: ASoC: ad1836: use soc-cache framework for codec registers access Signed-off-by: Barry Song Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 92 ++++++++++++----------------------------------- sound/soc/soc-cache.c | 67 ++++++++++++++++++++++++++++++++++ 2 files changed, 90 insertions(+), 69 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 83add2f3afba..3c80137d5938 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -171,58 +171,6 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, return 0; } - -/* - * interface to read/write ad1836 register - */ -#define AD1836_SPI_REG_SHFT 12 -#define AD1836_SPI_READ (1 << 11) -#define AD1836_SPI_VAL_MSK 0x3FF - -/* - * write to the ad1836 register space - */ - -static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 *reg_cache = codec->reg_cache; - int ret = 0; - - if (value != reg_cache[reg]) { - unsigned short buf; - struct spi_transfer t = { - .tx_buf = &buf, - .len = 2, - }; - struct spi_message m; - - buf = (reg << AD1836_SPI_REG_SHFT) | - (value & AD1836_SPI_VAL_MSK); - spi_message_init(&m); - spi_message_add_tail(&t, &m); - ret = spi_sync(codec->control_data, &m); - if (ret == 0) - reg_cache[reg] = value; - } - - return ret; -} - -/* - * read from the ad1836 register space cache - */ -static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *reg_cache = codec->reg_cache; - - if (reg >= codec->reg_cache_size) - return -EINVAL; - - return reg_cache[reg]; -} - #ifdef CONFIG_PM static int ad1836_soc_suspend(struct platform_device *pdev, pm_message_t state) @@ -231,10 +179,10 @@ static int ad1836_soc_suspend(struct platform_device *pdev, struct snd_soc_codec *codec = socdev->card->codec; /* reset clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); } static int ad1836_soc_resume(struct platform_device *pdev) @@ -243,10 +191,10 @@ static int ad1836_soc_resume(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->card->codec; /* restore clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); adc_ctrl2 |= AD1836_ADC_AUX; - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); } #else #define ad1836_soc_suspend NULL @@ -336,32 +284,38 @@ static int ad1836_register(struct ad1836_priv *ad1836) codec->owner = THIS_MODULE; codec->dai = &ad1836_dai; codec->num_dai = 1; - codec->write = ad1836_write_reg; - codec->read = ad1836_read_reg_cache; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ad1836_dai.dev = codec->dev; ad1836_codec = codec; + ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", + ret); + kfree(ad1836); + return ret; + } + /* default setting for ad1836 */ /* de-emphasis: 48kHz, power-on dac */ - codec->write(codec, AD1836_DAC_CTRL1, 0x300); + snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); /* unmute dac channels */ - codec->write(codec, AD1836_DAC_CTRL2, 0x0); + snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); /* high-pass filter enable, power-on adc */ - codec->write(codec, AD1836_ADC_CTRL1, 0x100); + snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); /* unmute adc channles, adc aux mode */ - codec->write(codec, AD1836_ADC_CTRL2, 0x180); + snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); /* left/right diff:PGA/MUX */ - codec->write(codec, AD1836_ADC_CTRL3, 0x3A); + snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); /* volume */ - codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF); ret = snd_soc_register_codec(codec); if (ret != 0) { diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 02c235711bb8..cde7b63de113 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -15,6 +15,68 @@ #include #include +static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg << 4) | ((value >> 8) & 0x000f); + data[1] = value & 0x00ff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +#if defined(CONFIG_SPI_MASTER) +static int snd_soc_4_12_spi_write(void *control_data, const char *data, + int len) +{ + struct spi_device *spi = control_data; + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[1]; + msg[1] = data[0]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#else +#define snd_soc_4_12_spi_write NULL +#endif + static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -179,6 +241,11 @@ static struct { unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { + { + .addr_bits = 4, .data_bits = 12, + .write = snd_soc_4_12_write, .read = snd_soc_4_12_read, + .spi_write = snd_soc_4_12_spi_write, + }, { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, -- cgit v1.2.2 From 994dc4245d3f50329da4ead453a5dfcfbc716a0d Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Wed, 27 Jan 2010 11:46:18 +0800 Subject: ASoC: ad1938: use soc-cache framework for codec registers access Signed-off-by: Barry Song Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1938.c | 164 ++++++++++------------------------------------ sound/soc/soc-cache.c | 108 ++++++++++++++++++++++++++++++ 2 files changed, 144 insertions(+), 128 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 47d9ac0ec9d9..c233810d463d 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -46,6 +46,11 @@ struct ad1938_priv { u8 reg_cache[AD1938_NUM_REGS]; }; +/* ad1938 register cache & default register settings */ +static const u8 ad1938_reg[AD1938_NUM_REGS] = { + 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, +}; + static struct snd_soc_codec *ad1938_codec; struct snd_soc_codec_device soc_codec_dev_ad1938; static int ad1938_register(struct ad1938_priv *ad1938); @@ -129,10 +134,10 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; int reg; - reg = codec->read(codec, AD1938_DAC_CTRL2); + reg = snd_soc_read(codec, AD1938_DAC_CTRL2); reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & (~AD1938_DAC_MASTER_MUTE); - codec->write(codec, AD1938_DAC_CTRL2, reg); + snd_soc_write(codec, AD1938_DAC_CTRL2, reg); return 0; } @@ -141,8 +146,8 @@ static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int width) { struct snd_soc_codec *codec = dai->codec; - int dac_reg = codec->read(codec, AD1938_DAC_CTRL1); - int adc_reg = codec->read(codec, AD1938_ADC_CTRL2); + int dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1); + int adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2); dac_reg &= ~AD1938_DAC_CHAN_MASK; adc_reg &= ~AD1938_ADC_CHAN_MASK; @@ -168,8 +173,8 @@ static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return -EINVAL; } - codec->write(codec, AD1938_DAC_CTRL1, dac_reg); - codec->write(codec, AD1938_ADC_CTRL2, adc_reg); + snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg); + snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg); return 0; } @@ -180,8 +185,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; int adc_reg, dac_reg; - adc_reg = codec->read(codec, AD1938_ADC_CTRL2); - dac_reg = codec->read(codec, AD1938_DAC_CTRL1); + adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2); + dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1); /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) @@ -258,8 +263,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - codec->write(codec, AD1938_ADC_CTRL2, adc_reg); - codec->write(codec, AD1938_DAC_CTRL1, dac_reg); + snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg); + snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg); return 0; } @@ -288,116 +293,13 @@ static int ad1938_hw_params(struct snd_pcm_substream *substream, break; } - reg = codec->read(codec, AD1938_DAC_CTRL2); + reg = snd_soc_read(codec, AD1938_DAC_CTRL2); reg = (reg & (~AD1938_DAC_WORD_LEN_MASK)) | word_len; - codec->write(codec, AD1938_DAC_CTRL2, reg); + snd_soc_write(codec, AD1938_DAC_CTRL2, reg); - reg = codec->read(codec, AD1938_ADC_CTRL1); + reg = snd_soc_read(codec, AD1938_ADC_CTRL1); reg = (reg & (~AD1938_ADC_WORD_LEN_MASK)) | word_len; - codec->write(codec, AD1938_ADC_CTRL1, reg); - - return 0; -} - -/* - * interface to read/write ad1938 register - */ - -#define AD1938_SPI_ADDR 0x4 -#define AD1938_SPI_READ 0x1 -#define AD1938_SPI_BUFLEN 3 - -/* - * write to the ad1938 register space - */ - -static int ad1938_write_reg(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 *reg_cache = codec->reg_cache; - int ret = 0; - - if (value != reg_cache[reg]) { - uint8_t buf[AD1938_SPI_BUFLEN]; - struct spi_transfer t = { - .tx_buf = buf, - .len = AD1938_SPI_BUFLEN, - }; - struct spi_message m; - - buf[0] = AD1938_SPI_ADDR << 1; - buf[1] = reg; - buf[2] = value; - spi_message_init(&m); - spi_message_add_tail(&t, &m); - ret = spi_sync(codec->control_data, &m); - if (ret == 0) - reg_cache[reg] = value; - } - - return ret; -} - -/* - * read from the ad1938 register space cache - */ - -static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u8 *reg_cache = codec->reg_cache; - - if (reg >= codec->reg_cache_size) - return -EINVAL; - - return reg_cache[reg]; -} - -/* - * read from the ad1938 register space - */ - -static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, - unsigned int reg) -{ - char w_buf[AD1938_SPI_BUFLEN]; - char r_buf[AD1938_SPI_BUFLEN]; - int ret; - - struct spi_transfer t = { - .tx_buf = w_buf, - .rx_buf = r_buf, - .len = AD1938_SPI_BUFLEN, - }; - struct spi_message m; - - w_buf[0] = (AD1938_SPI_ADDR << 1) | AD1938_SPI_READ; - w_buf[1] = reg; - w_buf[2] = 0; - - spi_message_init(&m); - spi_message_add_tail(&t, &m); - ret = spi_sync(codec->control_data, &m); - if (ret == 0) - return r_buf[2]; - else - return -EIO; -} - -static int ad1938_fill_cache(struct snd_soc_codec *codec) -{ - int i; - u8 *reg_cache = codec->reg_cache; - struct spi_device *spi = codec->control_data; - - for (i = 0; i < codec->reg_cache_size; i++) { - int ret = ad1938_read_reg(codec, i); - if (ret == -EIO) { - dev_err(&spi->dev, "AD1938 SPI read failure\n"); - return ret; - } - reg_cache[i] = ret; - } + snd_soc_write(codec, AD1938_ADC_CTRL1, reg); return 0; } @@ -487,31 +389,37 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->owner = THIS_MODULE; codec->dai = &ad1938_dai; codec->num_dai = 1; - codec->write = ad1938_write_reg; - codec->read = ad1938_read_reg_cache; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ad1938_dai.dev = codec->dev; ad1938_codec = codec; + memcpy(codec->reg_cache, ad1938_reg, AD1938_NUM_REGS); + + ret = snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_SPI); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", + ret); + kfree(ad1938); + return ret; + } + /* default setting for ad1938 */ /* unmute dac channels */ - codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); + snd_soc_write(codec, AD1938_DAC_CHNL_MUTE, 0x0); /* de-emphasis: 48kHz, powedown dac */ - codec->write(codec, AD1938_DAC_CTRL2, 0x1A); + snd_soc_write(codec, AD1938_DAC_CTRL2, 0x1A); /* powerdown dac, dac in tdm mode */ - codec->write(codec, AD1938_DAC_CTRL0, 0x41); + snd_soc_write(codec, AD1938_DAC_CTRL0, 0x41); /* high-pass filter enable */ - codec->write(codec, AD1938_ADC_CTRL0, 0x3); + snd_soc_write(codec, AD1938_ADC_CTRL0, 0x3); /* sata delay=1, adc aux mode */ - codec->write(codec, AD1938_ADC_CTRL1, 0x43); + snd_soc_write(codec, AD1938_ADC_CTRL1, 0x43); /* pll input: mclki/xi */ - codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); - codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04); - - ad1938_fill_cache(codec); + snd_soc_write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); + snd_soc_write(codec, AD1938_PLL_CLK_CTRL1, 0x04); ret = snd_soc_register_codec(codec); if (ret != 0) { diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index cde7b63de113..097e33510a7a 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -233,6 +233,108 @@ static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, #define snd_soc_8_16_read_i2c NULL #endif +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + struct i2c_msg xfer[2]; + u16 reg = r; + u8 data; + int ret; + struct i2c_client *client = codec->control_data; + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 2; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 1; + xfer[1].buf = &data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return data; +} +#else +#define snd_soc_16_8_read_i2c NULL +#endif + +static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + reg &= 0xff; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[3]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg >> 8) & 0xff; + data[1] = reg & 0xff; + data[2] = value; + + reg &= 0xff; + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret == 3) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +#if defined(CONFIG_SPI_MASTER) +static int snd_soc_16_8_spi_write(void *control_data, const char *data, + int len) +{ + struct spi_device *spi = control_data; + struct spi_transfer t; + struct spi_message m; + u8 msg[3]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + msg[2] = data[2]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#else +#define snd_soc_16_8_spi_write NULL +#endif + + static struct { int addr_bits; int data_bits; @@ -260,6 +362,12 @@ static struct { .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, .i2c_read = snd_soc_8_16_read_i2c, }, + { + .addr_bits = 16, .data_bits = 8, + .write = snd_soc_16_8_write, .read = snd_soc_16_8_read, + .i2c_read = snd_soc_16_8_read_i2c, + .spi_write = snd_soc_16_8_spi_write, + }, }; /** -- cgit v1.2.2 From 7910b4a1db63fefc3d291853d33c34c5b6352e8e Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 27 Jan 2010 18:10:13 +0100 Subject: ALSA: pcm_native - fix runtime->boundary calculation The code in pcm_lib updating runtime->hw_ptr_interrupt expects that runtime->boundary is divisible with runtime->period_size. Thanks are going to Clemens Ladisch for the notice. Fix the runtime->boundary calculation using buffer_size * period_size as base and find a least common multiple for 32bit platforms when the expression might overflow. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_native.c | 39 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 36 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7a002db512b4..9cbaf90d3d88 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include @@ -366,6 +367,38 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) return usecs; } +static int calc_boundary(struct snd_pcm_runtime *runtime) +{ + u_int64_t boundary; + + boundary = (u_int64_t)runtime->buffer_size * + (u_int64_t)runtime->period_size; +#if BITS_PER_LONG < 64 + /* try to find lowest common multiple for buffer and period */ + if (boundary > LONG_MAX - runtime->buffer_size) { + u_int32_t remainder = -1; + u_int32_t divident = runtime->buffer_size; + u_int32_t divisor = runtime->period_size; + while (remainder) { + remainder = divident % divisor; + if (remainder) { + divident = divisor; + divisor = remainder; + } + } + boundary = div_u64(boundary, divisor); + if (boundary > LONG_MAX - runtime->buffer_size) + return -ERANGE; + } +#endif + if (boundary == 0) + return -ERANGE; + runtime->boundary = boundary; + while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) + runtime->boundary *= 2; + return 0; +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -441,9 +474,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->stop_threshold = runtime->buffer_size; runtime->silence_threshold = 0; runtime->silence_size = 0; - runtime->boundary = runtime->buffer_size; - while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) - runtime->boundary *= 2; + err = calc_boundary(runtime); + if (err < 0) + goto _error; snd_pcm_timer_resolution_change(substream); runtime->status->state = SNDRV_PCM_STATE_SETUP; -- cgit v1.2.2 From 8ce28d6abff34886d3797b25324c940471b99164 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jan 2010 20:26:08 +0100 Subject: ALSA: hda - Add an ASUS mobo to MSI blacklist Sid Boyce reported that his machine locks up without enable_msi=0 option. This looks like another ASUS mobo with Nvidia combo. Reported-by: Sid Boyce Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec9c348336cc..565de38a3fc7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2332,6 +2332,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) */ static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ + SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */ {} }; -- cgit v1.2.2 From fc93ea2f9315eda2ec8645c2f8bcc30f75a6b88e Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 14:59:08 +0900 Subject: ASoC: AC97: S3C: Add controller driver Add the AC97 controller driver for Samsung SoCs that have one. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 6 +- sound/soc/s3c24xx/Makefile | 3 +- sound/soc/s3c24xx/s3c-ac97.c | 518 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-ac97.h | 23 ++ 4 files changed, 548 insertions(+), 2 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c-ac97.c create mode 100644 sound/soc/s3c24xx/s3c-ac97.h (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b489f1ae103d..ad3690ec3de8 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -32,7 +32,11 @@ config SND_S3C2443_SOC_AC97 select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_BUS - + +config SND_S3C_SOC_AC97 + tristate + select SND_SOC_AC97_BUS + config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b744657733d7..b7411bd59f33 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -4,12 +4,14 @@ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o +snd-soc-s3c-ac97-objs := s3c-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o +obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o @@ -37,4 +39,3 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o - diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c new file mode 100644 index 000000000000..ee8ed9d7e703 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -0,0 +1,518 @@ +/* sound/soc/s3c24xx/s3c-ac97.c + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.c + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include + +#include + +#include +#include +#include + +#include "s3c-dma.h" +#include "s3c-ac97.h" + +#define AC_CMD_ADDR(x) (x << 16) +#define AC_CMD_DATA(x) (x & 0xffff) + +struct s3c_ac97_info { + unsigned state; + struct clk *ac97_clk; + void __iomem *regs; + struct mutex lock; + struct completion done; +}; +static struct s3c_ac97_info s3c_ac97; + +static struct s3c2410_dma_client s3c_dma_client_out = { + .name = "AC97 PCMOut" +}; + +static struct s3c2410_dma_client s3c_dma_client_in = { + .name = "AC97 PCMIn" +}; + +static struct s3c2410_dma_client s3c_dma_client_micin = { + .name = "AC97 MicIn" +}; + +static struct s3c_dma_params s3c_ac97_pcm_out = { + .client = &s3c_dma_client_out, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_pcm_in = { + .client = &s3c_dma_client_in, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_mic_in = { + .client = &s3c_dma_client_micin, + .dma_size = 4, +}; + +static void s3c_ac97_activate(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl, stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + INIT_COMPLETION(s3c_ac97.done); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to activate!"); +} + +static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + u32 ac_glbctrl, ac_codec_cmd; + u32 stat, addr, data; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to read!"); + + stat = readl(s3c_ac97.regs + S3C_AC97_STAT); + addr = (stat >> 16) & 0x7f; + data = (stat & 0xffff); + + if (addr != reg) + printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); + + mutex_unlock(&s3c_ac97.lock); + + return (unsigned short)data; +} + +static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + u32 ac_glbctrl, ac_codec_cmd; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to write!"); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + mutex_unlock(&s3c_ac97.lock); +} + +static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + writel(S3C_AC97_GLBCTRL_COLDRESET, + s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); +} + +static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + u32 stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + s3c_ac97_activate(ac97); +} + +static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) +{ + u32 ac_glbctrl, ac_glbstat; + + ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); + + if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + complete(&s3c_ac97.done); + } + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= (1<<30); /* Clear interrupt */ + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = s3c_ac97_read, + .write = s3c_ac97_write, + .warm_reset = s3c_ac97_warm_reset, + .reset = s3c_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_dai->dma_data = &s3c_ac97_pcm_out; + else + cpu_dai->dma_data = &s3c_ac97_pcm_in; + + return 0; +} + +static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + else + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + else + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + else + cpu_dai->dma_data = &s3c_ac97_mic_in; + + return 0; +} + +static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static struct snd_soc_dai_ops s3c_ac97_dai_ops = { + .hw_params = s3c_ac97_hw_params, + .trigger = s3c_ac97_trigger, +}; + +static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { + .hw_params = s3c_ac97_hw_mic_params, + .trigger = s3c_ac97_mic_trigger, +}; + +struct snd_soc_dai s3c_ac97_dai[] = { + [S3C_AC97_DAI_PCM] = { + .name = "s3c-ac97", + .id = S3C_AC97_DAI_PCM, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_dai_ops, + }, + [S3C_AC97_DAI_MIC] = { + .name = "s3c-ac97-mic", + .id = S3C_AC97_DAI_MIC, + .ac97_control = 1, + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_mic_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(s3c_ac97_dai); + +static __devinit int s3c_ac97_probe(struct platform_device *pdev) +{ + struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; + struct s3c_audio_pdata *ac97_pdata; + int ret; + + ac97_pdata = pdev->dev.platform_data; + if (!ac97_pdata || !ac97_pdata->cfg_gpio) { + dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); + return -EINVAL; + } + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); + return -ENXIO; + } + + dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); + if (!dmamic_res) { + dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq_res) { + dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); + return -ENXIO; + } + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "s3c-ac97")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + return -EBUSY; + } + + s3c_ac97_pcm_out.channel = dmatx_res->start; + s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_pcm_in.channel = dmarx_res->start; + s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_mic_in.channel = dmamic_res->start; + s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; + + init_completion(&s3c_ac97.done); + mutex_init(&s3c_ac97.lock); + + s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); + if (s3c_ac97.regs == NULL) { + dev_err(&pdev->dev, "Unable to ioremap register region\n"); + ret = -ENXIO; + goto err1; + } + + s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + if (IS_ERR(s3c_ac97.ac97_clk)) { + dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); + ret = -ENODEV; + goto err2; + } + clk_enable(s3c_ac97.ac97_clk); + + if (ac97_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err3; + } + + ret = request_irq(irq_res->start, s3c_ac97_irq, + IRQF_DISABLED, "AC97", NULL); + if (ret < 0) { + printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); + goto err4; + } + + s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; + s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; + + ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + if (ret) + goto err5; + + return 0; + +err5: + free_irq(irq_res->start, NULL); +err4: +err3: + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); +err2: + iounmap(s3c_ac97.regs); +err1: + release_mem_region(mem_res->start, resource_size(mem_res)); + + return ret; +} + +static __devexit int s3c_ac97_remove(struct platform_device *pdev) +{ + struct resource *mem_res, *irq_res; + + snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (irq_res) + free_irq(irq_res->start, NULL); + + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); + + iounmap(s3c_ac97.regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mem_res) + release_mem_region(mem_res->start, resource_size(mem_res)); + + return 0; +} + +static struct platform_driver s3c_ac97_driver = { + .probe = s3c_ac97_probe, + .remove = s3c_ac97_remove, + .driver = { + .name = "s3c-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_ac97_init(void) +{ + return platform_driver_register(&s3c_ac97_driver); +} +module_init(s3c_ac97_init); + +static void __exit s3c_ac97_exit(void) +{ + platform_driver_unregister(&s3c_ac97_driver); +} +module_exit(s3c_ac97_exit); + +MODULE_AUTHOR("Jaswinder Singh, "); +MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-ac97.h b/sound/soc/s3c24xx/s3c-ac97.h new file mode 100644 index 000000000000..278198379def --- /dev/null +++ b/sound/soc/s3c24xx/s3c-ac97.h @@ -0,0 +1,23 @@ +/* sound/soc/s3c24xx/s3c-ac97.h + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.h + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __S3C_AC97_H_ +#define __S3C_AC97_H_ + +#define S3C_AC97_DAI_PCM 0 +#define S3C_AC97_DAI_MIC 1 + +extern struct snd_soc_dai s3c_ac97_dai[]; + +#endif /* __S3C_AC97_H_ */ -- cgit v1.2.2 From ff6e64dabf66b8e4e7def21857320085fc68db6b Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 14:59:19 +0900 Subject: ASoC: AC97: SMDK: Add wm9713 machine driver This patch adds the common machine driver for SMDKs that have a WM9713 codec attched to the AC97 controller. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 8 ++++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk_wm9713.c | 97 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 107 insertions(+) create mode 100644 sound/soc/s3c24xx/smdk_wm9713.c (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index ad3690ec3de8..d1c6f9392463 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -115,3 +115,11 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_S3C24XX_SOC_I2S select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC + +config SND_SOC_SMDK_WM9713 + tristate "SoC AC97 Audio support for SMDK with WM9713" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM9713 + select SND_S3C_SOC_AC97 + help + Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b7411bd59f33..1117678ae4e1 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o +snd-soc-smdk-wm9713-objs := smdk_wm9713.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -39,3 +40,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o +obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c new file mode 100644 index 000000000000..7dd933f7cbf9 --- /dev/null +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -0,0 +1,97 @@ +/* + * smdk_wm9713.c -- SoC audio for SMDK + * + * Copyright 2010 Samsung Electronics Co. Ltd. + * Author: Jaswinder Singh Brar + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + * + */ + +#include +#include +#include + +#include "../codecs/wm9713.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk; + +/* + Playback (HeadPhone):- + Headphone Playback Switch - On + $ amixer cset numid=4 1 + + Right Headphone Out Mux - Headphone + $ amixer cset numid=92 2 + Left Headphone Out Mux - Headphone + $ amixer cset numid=93 2 + + Right HP Mixer PCM Playback Switch - On + $ amixer cset numid=75 1 + Left HP Mixer PCM Playback Switch - On + $ amixer cset numid=81 1 + + Capture (LineIn):- + Right Capture Source - Line + $ amixer cset numid=86 2 + Left Capture Source - Line + $ amixer cset numid=87 2 +*/ + +static struct snd_soc_dai_link smdk_dai = { + .name = "AC97", + .stream_name = "AC97 PCM", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], +}; + +static struct snd_soc_card smdk = { + .name = "SMDK", + .platform = &s3c24xx_soc_platform, + .dai_link = &smdk_dai, + .num_links = 1, +}; + +static struct snd_soc_device smdk_snd_ac97_devdata = { + .card = &smdk, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *smdk_snd_ac97_device; + +static int __init smdk_init(void) +{ + int ret; + + smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk_snd_ac97_device, + &smdk_snd_ac97_devdata); + smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; + + ret = platform_device_add(smdk_snd_ac97_device); + if (ret) + platform_device_put(smdk_snd_ac97_device); + + return ret; +} + +static void __exit smdk_exit(void) +{ + platform_device_unregister(smdk_snd_ac97_device); +} + +module_init(smdk_init); +module_exit(smdk_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar@samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); +MODULE_LICENSE("GPL"); -- cgit v1.2.2 From 1ec2963a8cd5fbc5f49dfa20c94229f1b46d1968 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 15:01:03 +0900 Subject: ASoC: AC97: SMDK2443: Switch to s3c-ac97.c Switch to use s3c-ac97.c AC97 controller driver. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 4 +++- sound/soc/s3c24xx/smdk2443_wm9710.c | 4 ++-- 2 files changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index d1c6f9392463..8b62798a04b8 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -75,8 +75,10 @@ config SND_S3C64XX_SOC_WM8580 config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 - select SND_S3C2443_SOC_AC97 + select S3C2410_DMA + select AC97_BUS select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 help Say Y if you want to add support for SoC audio on smdk2443 with the WM9710. diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 12b783b12fcb..362258835e8d 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -21,7 +21,7 @@ #include "../codecs/ac97.h" #include "s3c-dma.h" -#include "s3c24xx-ac97.h" +#include "s3c-ac97.h" static struct snd_soc_card smdk2443; @@ -29,7 +29,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai = &s3c2443_ac97_dai[0], + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], .codec_dai = &ac97_dai, }, }; -- cgit v1.2.2 From c67d90ffd43a6cf18def21a0de7db56504d78295 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 15:02:04 +0900 Subject: ASoC: AC97: LN2440SBC: Switch to s3c-ac97.c Switch to use s3c-ac97.c AC97 controller driver. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 7 +++---- sound/soc/s3c24xx/ln2440sbc_alc650.c | 4 ++-- 2 files changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 8b62798a04b8..69d143e3ab25 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -29,9 +29,6 @@ config SND_S3C_SOC_PCM config SND_S3C2443_SOC_AC97 tristate - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_BUS config SND_S3C_SOC_AC97 tristate @@ -86,8 +83,10 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710 config SND_S3C24XX_SOC_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C2443_SOC_AC97 + select S3C2410_DMA + select AC97_BUS select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 help Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index d00d359a03e6..ffa954fe6931 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -25,7 +25,7 @@ #include "../codecs/ac97.h" #include "s3c-dma.h" -#include "s3c24xx-ac97.h" +#include "s3c-ac97.h" static struct snd_soc_card ln2440sbc; @@ -33,7 +33,7 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai = &s3c2443_ac97_dai[0], + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], .codec_dai = &ac97_dai, }, }; -- cgit v1.2.2 From 7beba4d50d5f70c3851f608927882959d532671c Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 15:04:36 +0900 Subject: ASoC: AC97: S3C2443: Remove unused driver Since, we have generic AC97 controller driver and all the machines have moved to that, there is no need for old s3c2443-ac97.c to exist. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 3 - sound/soc/s3c24xx/Makefile | 2 - sound/soc/s3c24xx/s3c2443-ac97.c | 432 --------------------------------------- sound/soc/s3c24xx/s3c24xx-ac97.h | 25 --- 4 files changed, 462 deletions(-) delete mode 100644 sound/soc/s3c24xx/s3c2443-ac97.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-ac97.h (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 69d143e3ab25..15fe57e5a232 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -27,9 +27,6 @@ config SND_S3C64XX_SOC_I2S config SND_S3C_SOC_PCM tristate -config SND_S3C2443_SOC_AC97 - tristate - config SND_S3C_SOC_AC97 tristate select SND_SOC_AC97_BUS diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 1117678ae4e1..df071a376fa2 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -3,14 +3,12 @@ snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o -snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-ac97-objs := s3c-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o -obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c deleted file mode 100644 index 0191e3acb0b4..000000000000 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ /dev/null @@ -1,432 +0,0 @@ -/* - * s3c2443-ac97.c -- ALSA Soc Audio Layer - * - * (c) 2007 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Copyright (C) 2005, Sean Choi - * All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include -#include - -#include "s3c-dma.h" -#include "s3c24xx-ac97.h" - -struct s3c24xx_ac97_info { - void __iomem *regs; - struct clk *ac97_clk; -}; -static struct s3c24xx_ac97_info s3c24xx_ac97; - -static DECLARE_COMPLETION(ac97_completion); -static u32 codec_ready; -static DEFINE_MUTEX(ac97_mutex); - -static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - u32 ac_glbctrl; - u32 ac_codec_cmd; - u32 stat, addr, data; - - mutex_lock(&ac97_mutex); - - codec_ready = S3C_AC97_GLBSTAT_CODECREADY; - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - wait_for_completion(&ac97_completion); - - stat = readl(s3c24xx_ac97.regs + S3C_AC97_STAT); - addr = (stat >> 16) & 0x7f; - data = (stat & 0xffff); - - if (addr != reg) - printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," - " rep addr = %02x\n", reg, addr); - - mutex_unlock(&ac97_mutex); - - return (unsigned short)data; -} - -static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - u32 ac_glbctrl; - u32 ac_codec_cmd; - - mutex_lock(&ac97_mutex); - - codec_ready = S3C_AC97_GLBSTAT_CODECREADY; - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - wait_for_completion(&ac97_completion); - - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - mutex_unlock(&ac97_mutex); - -} - -static void s3c2443_ac97_warm_reset(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_WARMRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); -} - -static void s3c2443_ac97_cold_reset(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA | - S3C_AC97_GLBCTRL_PCMINTM_DMA | S3C_AC97_GLBCTRL_MICINTM_DMA; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); -} - -static irqreturn_t s3c2443_ac97_irq(int irq, void *dev_id) -{ - int status; - u32 ac_glbctrl; - - status = readl(s3c24xx_ac97.regs + S3C_AC97_GLBSTAT) & codec_ready; - - if (status) { - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - complete(&ac97_completion); - } - return IRQ_HANDLED; -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = s3c2443_ac97_read, - .write = s3c2443_ac97_write, - .warm_reset = s3c2443_ac97_warm_reset, - .reset = s3c2443_ac97_cold_reset, -}; - -static struct s3c2410_dma_client s3c2443_dma_client_out = { - .name = "AC97 PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2443_dma_client_in = { - .name = "AC97 PCM Stereo in" -}; - -static struct s3c2410_dma_client s3c2443_dma_client_micin = { - .name = "AC97 Mic Mono in" -}; - -static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = { - .client = &s3c2443_dma_client_out, - .channel = DMACH_PCM_OUT, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = { - .client = &s3c2443_dma_client_in, - .channel = DMACH_PCM_IN, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2443_ac97_mic_mono_in = { - .client = &s3c2443_dma_client_micin, - .channel = DMACH_MIC_IN, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, - .dma_size = 4, -}; - -static int s3c2443_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - int ret; - u32 ac_glbctrl; - - s3c24xx_ac97.regs = ioremap(S3C2440_PA_AC97, 0x100); - if (s3c24xx_ac97.regs == NULL) - return -ENXIO; - - s3c24xx_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); - if (s3c24xx_ac97.ac97_clk == NULL) { - printk(KERN_ERR "s3c2443-ac97 failed to get ac97_clock\n"); - iounmap(s3c24xx_ac97.regs); - return -ENODEV; - } - clk_enable(s3c24xx_ac97.ac97_clk); - - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq, - IRQF_DISABLED, "AC97", NULL); - if (ret < 0) { - printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); - clk_disable(s3c24xx_ac97.ac97_clk); - clk_put(s3c24xx_ac97.ac97_clk); - iounmap(s3c24xx_ac97.regs); - } - return ret; -} - -static void s3c2443_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - free_irq(IRQ_S3C244x_AC97, NULL); - clk_disable(s3c24xx_ac97.ac97_clk); - clk_put(s3c24xx_ac97.ac97_clk); - iounmap(s3c24xx_ac97.regs); -} - -static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; - else - cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_in; - - return 0; -} - -static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - else - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - else - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; - break; - } - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - cpu_dai->dma_data = &s3c2443_ac97_mic_mono_in; - - return 0; -} - -static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - } - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -#define s3c2443_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) - -static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = { - .hw_params = s3c2443_ac97_hw_params, - .trigger = s3c2443_ac97_trigger, -}; - -static struct snd_soc_dai_ops s3c2443_ac97_mic_dai_ops = { - .hw_params = s3c2443_ac97_hw_mic_params, - .trigger = s3c2443_ac97_mic_trigger, -}; - -struct snd_soc_dai s3c2443_ac97_dai[] = { -{ - .name = "s3c2443-ac97", - .id = 0, - .ac97_control = 1, - .probe = s3c2443_ac97_probe, - .remove = s3c2443_ac97_remove, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c2443_ac97_dai_ops, -}, -{ - .name = "pxa2xx-ac97-mic", - .id = 1, - .ac97_control = 1, - .capture = { - .stream_name = "AC97 Mic Capture", - .channels_min = 1, - .channels_max = 1, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c2443_ac97_mic_dai_ops, -}, -}; -EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int __init s3c2443_ac97_init(void) -{ - return snd_soc_register_dais(s3c2443_ac97_dai, - ARRAY_SIZE(s3c2443_ac97_dai)); -} -module_init(s3c2443_ac97_init); - -static void __exit s3c2443_ac97_exit(void) -{ - snd_soc_unregister_dais(s3c2443_ac97_dai, - ARRAY_SIZE(s3c2443_ac97_dai)); -} -module_exit(s3c2443_ac97_exit); - - -MODULE_AUTHOR("Graeme Gregory"); -MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h deleted file mode 100644 index e96f941a810b..000000000000 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ /dev/null @@ -1,25 +0,0 @@ -/* - * s3c24xx-ac97.c -- ALSA Soc Audio Layer - * - * (c) 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Revision history - * 10th Nov 2006 Initial version. - */ - -#ifndef S3C24XXAC97_H_ -#define S3C24XXAC97_H_ - -#define AC_CMD_ADDR(x) (x << 16) -#define AC_CMD_DATA(x) (x & 0xffff) - -extern struct snd_soc_dai s3c2443_ac97_dai[]; - -#endif /*S3C24XXAC97_H_*/ -- cgit v1.2.2 From 583b2be626b047eeb4f9a26721e38fe4992b2d02 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Jan 2010 20:54:13 +0000 Subject: ASoC: Note jumper settings for smdk_wm9713 driver on SMDK6410 The board supports both GPIO sets for the AC97 bus and the analogue outputs can be switched between this and the WM8580 so add some comments saying what the setup the standard kernel expects is. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/smdk_wm9713.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 7dd933f7cbf9..6fa2c9d17d7a 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -21,6 +21,12 @@ static struct snd_soc_card smdk; +/* + * Default CFG switch settings to use this driver: + * + * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + */ + /* Playback (HeadPhone):- Headphone Playback Switch - On -- cgit v1.2.2 From 0d34e91596ef537c2893a031f0483014bb82adf3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 27 Jan 2010 18:56:23 +0100 Subject: ASoC: add a WM8978 codec driver The WM8978 codec from Wolfson Microelectronics is very similar to wm8974, but is stereo and also has some differences in pin configuration and internal signal routing. This driver is based on wm8974 and takes the differences into account. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8978.c | 1124 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8978.h | 89 ++++ 4 files changed, 1219 insertions(+) create mode 100644 sound/soc/codecs/wm8978.c create mode 100644 sound/soc/codecs/wm8978.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62ff26a08a2f..0aad72fc1961 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8974 if I2C + select SND_SOC_WM8978 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C @@ -230,6 +231,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8974 tristate +config SND_SOC_WM8978 + tristate + config SND_SOC_WM8988 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ea9835412e6a..fbd290e41e9e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -44,6 +44,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8974-objs := wm8974.o +snd-soc-wm8978-objs := wm8978.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8993-objs := wm8993.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o +obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c new file mode 100644 index 000000000000..d9d4e9dd1adb --- /dev/null +++ b/sound/soc/codecs/wm8978.c @@ -0,0 +1,1124 @@ +/* + * wm8978.c -- WM8978 ALSA SoC Audio Codec driver + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2007 Carlos Munoz + * Copyright 2006-2009 Wolfson Microelectronics PLC. + * Based on wm8974 and wm8990 by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8978.h" + +static struct snd_soc_codec *wm8978_codec; + +/* wm8978 register cache. Note that register 0 is not included in the cache. */ +static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x00...0x03 */ + 0x0050, 0x0000, 0x0140, 0x0000, /* 0x04...0x07 */ + 0x0000, 0x0000, 0x0000, 0x00ff, /* 0x08...0x0b */ + 0x00ff, 0x0000, 0x0100, 0x00ff, /* 0x0c...0x0f */ + 0x00ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ + 0x002c, 0x002c, 0x002c, 0x0000, /* 0x14...0x17 */ + 0x0032, 0x0000, 0x0000, 0x0000, /* 0x18...0x1b */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x1c...0x1f */ + 0x0038, 0x000b, 0x0032, 0x0000, /* 0x20...0x23 */ + 0x0008, 0x000c, 0x0093, 0x00e9, /* 0x24...0x27 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x28...0x2b */ + 0x0033, 0x0010, 0x0010, 0x0100, /* 0x2c...0x2f */ + 0x0100, 0x0002, 0x0001, 0x0001, /* 0x30...0x33 */ + 0x0039, 0x0039, 0x0039, 0x0039, /* 0x34...0x37 */ + 0x0001, 0x0001, /* 0x38...0x3b */ +}; + +/* codec private data */ +struct wm8978_priv { + struct snd_soc_codec codec; + unsigned int f_pllout; + unsigned int f_mclk; + unsigned int f_256fs; + unsigned int f_opclk; + enum wm8978_sysclk_src sysclk; + u16 reg_cache[WM8978_CACHEREGNUM]; +}; + +static const char *wm8978_companding[] = {"Off", "NC", "u-law", "A-law"}; +static const char *wm8978_eqmode[] = {"Capture", "Playback"}; +static const char *wm8978_bw[] = {"Narrow", "Wide"}; +static const char *wm8978_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz"}; +static const char *wm8978_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz"}; +static const char *wm8978_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz"}; +static const char *wm8978_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"}; +static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"}; +static const char *wm8978_alc3[] = {"ALC", "Limiter"}; +static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"}; + +static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); +static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); +static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); +static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); +static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); +static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); +static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); +static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); + +static const struct snd_kcontrol_new wm8978_snd_controls[] = { + + SOC_SINGLE("Digital Loopback Switch", + WM8978_COMPANDING_CONTROL, 0, 1, 0), + + SOC_ENUM("ADC Companding", adc_compand), + SOC_ENUM("DAC Companding", dac_compand), + + SOC_DOUBLE("DAC Inversion Switch", WM8978_DAC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("PCM Volume", + WM8978_LEFT_DAC_DIGITAL_VOLUME, WM8978_RIGHT_DAC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", WM8978_ADC_CONTROL, 4, 7, 0), + SOC_DOUBLE("ADC Inversion Switch", WM8978_ADC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Volume", + WM8978_LEFT_ADC_DIGITAL_VOLUME, WM8978_RIGHT_ADC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_ENUM("Equaliser Function", eqmode), + SOC_ENUM("EQ1 Cut Off", eq1), + SOC_SINGLE_TLV("EQ1 Volume", WM8978_EQ1, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ2 Bandwith", eq2bw), + SOC_ENUM("EQ2 Cut Off", eq2), + SOC_SINGLE_TLV("EQ2 Volume", WM8978_EQ2, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ3 Bandwith", eq3bw), + SOC_ENUM("EQ3 Cut Off", eq3), + SOC_SINGLE_TLV("EQ3 Volume", WM8978_EQ3, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ4 Bandwith", eq4bw), + SOC_ENUM("EQ4 Cut Off", eq4), + SOC_SINGLE_TLV("EQ4 Volume", WM8978_EQ4, 0, 24, 1, eq_tlv), + + SOC_ENUM("EQ5 Cut Off", eq5), + SOC_SINGLE_TLV("EQ5 Volume", WM8978_EQ5, 0, 24, 1, eq_tlv), + + SOC_SINGLE("DAC Playback Limiter Switch", + WM8978_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Decay", + WM8978_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Playback Limiter Attack", + WM8978_DAC_LIMITER_1, 0, 15, 0), + + SOC_SINGLE("DAC Playback Limiter Threshold", + WM8978_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE("DAC Playback Limiter Boost", + WM8978_DAC_LIMITER_2, 0, 15, 0), + + SOC_ENUM("ALC Enable Switch", alc1), + SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), + + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), + + SOC_ENUM("ALC Capture Mode", alc3), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", + WM8978_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("Capture PGA ZC Switch", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + + /* OUT1 - Headphones */ + SOC_DOUBLE_R("Headphone Playback ZC Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Headphone Playback Volume", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT2 - Speakers */ + SOC_DOUBLE_R("Speaker Playback ZC Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Speaker Playback Volume", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT3/4 - Line Output */ + SOC_DOUBLE_R("Line Playback Switch", + WM8978_OUT3_MIXER_CONTROL, WM8978_OUT4_MIXER_CONTROL, 6, 1, 1), + + /* Mixer #3: Boost (Input) mixer */ + SOC_DOUBLE_R("PGA Boost (+20dB)", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 8, 1, 0), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 0, 7, 0, boost_tlv), + + /* Input PGA volume */ + SOC_DOUBLE_R_TLV("Input PGA Volume", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 0, 63, 0, inpga_tlv), + + /* Headphone */ + SOC_DOUBLE_R("Headphone Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 6, 1, 1), + + /* Speaker */ + SOC_DOUBLE_R("Speaker Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), +}; + +/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ +static const struct snd_kcontrol_new wm8978_left_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_LEFT_MIXER_CONTROL, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8978_right_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 0, 1, 0), +}; + +/* OUT3/OUT4 Mixer not implemented */ + +/* Mixer #2: Input PGA Mute */ +static const struct snd_kcontrol_new wm8978_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", WM8978_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new wm8978_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", WM8978_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8978_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 1, 0), + + /* Mixer #1: OUT1,2 */ + SOC_MIXER_ARRAY("Left Output Mixer", WM8978_POWER_MANAGEMENT_3, + 2, 0, wm8978_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", WM8978_POWER_MANAGEMENT_3, + 3, 0, wm8978_right_out_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", WM8978_POWER_MANAGEMENT_2, + 2, 0, wm8978_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", WM8978_POWER_MANAGEMENT_2, + 3, 0, wm8978_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", WM8978_LEFT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", WM8978_RIGHT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", WM8978_POWER_MANAGEMENT_2, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", WM8978_POWER_MANAGEMENT_2, + 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", WM8978_POWER_MANAGEMENT_3, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", WM8978_POWER_MANAGEMENT_3, + 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT4 VMID", WM8978_POWER_MANAGEMENT_3, + 8, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8978_POWER_MANAGEMENT_1, 4, 0), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Output mixer */ + {"Right Output Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right Output Mixer", "Aux Playback Switch", "RAUX"}, + {"Right Output Mixer", "Line Bypass Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left Output Mixer", "Aux Playback Switch", "LAUX"}, + {"Left Output Mixer", "Line Bypass Switch", "Left Boost Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, +}; + +static int wm8978_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets, + ARRAY_SIZE(wm8978_dapm_widgets)); + + /* set up the WM8978 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* PLL divisors */ +struct wm8978_pll_div { + u32 k; + u8 n; + u8 div2; +}; + +#define FIXED_PLL_SIZE (1 << 24) + +static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 k_part; + unsigned int k, n_div, n_mod; + + n_div = target / source; + if (n_div < 6) { + source >>= 1; + pll_div->div2 = 1; + n_div = target / source; + } else { + pll_div->div2 = 0; + } + + if (n_div < 6 || n_div > 12) + dev_warn(wm8978_codec->dev, + "WM8978 N value exceeds recommended range! N = %u\n", + n_div); + + pll_div->n = n_div; + n_mod = target - source * n_div; + k_part = FIXED_PLL_SIZE * (long long)n_mod + source / 2; + + do_div(k_part, source); + + k = k_part & 0xFFFFFFFF; + + pll_div->k = k; +} +/* + * Calculate internal frequencies and dividers, according to Figure 40 + * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 + */ +static int wm8978_configure_pll(struct snd_soc_codec *codec) +{ + struct wm8978_priv *wm8978 = codec->private_data; + struct wm8978_pll_div pll_div; + unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, + f_256fs = wm8978->f_256fs; + unsigned int f2, opclk_div; + + if (!f_mclk) + return -EINVAL; + + if (f_opclk) { + /* + * The user needs OPCLK. Choose OPCLKDIV to put + * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. + * f_opclk = f_mclk * prescale * R / 4 / OPCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 16 <= f_opclk < f_mclk * 13 / 4. + */ + if (16 * f_opclk < 3 * f_mclk || 4 * f_opclk >= 13 * f_mclk) + return -EINVAL; + + if (4 * f_opclk < 3 * f_mclk) + /* Have to use OPCLKDIV */ + opclk_div = (3 * f_mclk / 4 + f_opclk - 1) / f_opclk; + else + opclk_div = 1; + + dev_dbg(codec->dev, "%s: OPCLKDIV=%d\n", __func__, opclk_div); + + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 0x30, + (opclk_div - 1) << 4); + + wm8978->f_pllout = f_opclk * opclk_div; + } else if (f_256fs) { + /* + * Not using OPCLK, choose R: + * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. + * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK + * must be 3.781MHz <= f_MCLK <= 32.768MHz + */ + if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) + return -EINVAL; + + /* + * MCLKDIV will be selected in .hw_params(), just choose a + * suitable f_PLLOUT + */ + if (4 * f_256fs < 3 * f_mclk) + /* Will have to use MCLKDIV */ + wm8978->f_pllout = wm8978->f_mclk * 3 / 4; + else + wm8978->f_pllout = f_256fs; + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + } else { + return -EINVAL; + } + + f2 = wm8978->f_pllout * 4; + + dev_dbg(codec->dev, "%s: f_MCLK=%uHz, f_PLLOUT=%uHz\n", __func__, + wm8978->f_mclk, wm8978->f_pllout); + + pll_factors(&pll_div, f2, wm8978->f_mclk); + + dev_dbg(codec->dev, "%s: calculated PLL N=0x%x, K=0x%x, div2=%d\n", + __func__, pll_div.n, pll_div.k, pll_div.div2); + + /* Turn PLL off for configuration... */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + + snd_soc_write(codec, WM8978_PLL_N, (pll_div.div2 << 4) | pll_div.n); + snd_soc_write(codec, WM8978_PLL_K1, pll_div.k >> 18); + snd_soc_write(codec, WM8978_PLL_K2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8978_PLL_K3, pll_div.k & 0x1ff); + + /* ...and on again */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + if (f_opclk) + /* Output PLL (OPCLK) to GPIO1 */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 4); + + return 0; +} + +/* + * Configure WM8978 clock dividers. + */ +static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + switch (div_id) { + case WM8978_OPCLKRATE: + wm8978->f_opclk = div; + + if (wm8978->f_mclk) + ret = wm8978_configure_pll(codec); + break; + case WM8978_MCLKDIV: + if (div & ~0xe0) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); + break; + case WM8978_ADCCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); + break; + case WM8978_DACCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); + break; + case WM8978_BCLKDIV: + if (div & ~0x1c) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x1c, div); + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "%s: ID %d, value %u\n", __func__, div_id, div); + + return ret; +} + +/* + * @freq: when .set_pll() us not used, freq is codec MCLK input frequency + */ +static int wm8978_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + dev_dbg(codec->dev, "%s: ID %d, freq %u\n", __func__, clk_id, freq); + + if (freq) { + wm8978->f_mclk = freq; + + /* Even if MCLK is used for system clock, might have to drive OPCLK */ + if (wm8978->f_opclk) + ret = wm8978_configure_pll(codec); + + /* Our sysclk is fixed to 256 * fs, will configure in .hw_params() */ + + if (!ret) + wm8978->sysclk = clk_id; + } + + if (wm8978->sysclk == WM8978_PLL && (!freq || clk_id == WM8978_MCLK)) { + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + wm8978->sysclk = WM8978_MCLK; + wm8978->f_pllout = 0; + wm8978->f_opclk = 0; + } + + return ret; +} + +/* + * Set ADC and Voice DAC format. + */ +static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + /* + * BCLK polarity mask = 0x100, LRC clock polarity mask = 0x80, + * Data Format mask = 0x18: all will be calculated anew + */ + u16 iface = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x198; + u16 clk = snd_soc_read(codec, WM8978_CLOCKING); + + dev_dbg(codec->dev, "%s\n", __func__); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + clk &= ~1; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x18; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); + snd_soc_write(codec, WM8978_CLOCKING, clk); + + return 0; +} + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8978_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + /* Word length mask = 0x60 */ + u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; + /* Sampling rate mask = 0xe (for filters) */ + u16 add_ctl = snd_soc_read(codec, WM8978_ADDITIONAL_CONTROL) & ~0xe; + u16 clking = snd_soc_read(codec, WM8978_CLOCKING); + enum wm8978_sysclk_src current_clk_id = clking & 0x100 ? + WM8978_PLL : WM8978_MCLK; + unsigned int f_sel, diff, diff_best = INT_MAX; + int i, best = 0; + + if (!wm8978->f_mclk) + return -EINVAL; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_ctl |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_ctl |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_ctl |= 0x60; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case 8000: + add_ctl |= 0x5 << 1; + break; + case 11025: + add_ctl |= 0x4 << 1; + break; + case 16000: + add_ctl |= 0x3 << 1; + break; + case 22050: + add_ctl |= 0x2 << 1; + break; + case 32000: + add_ctl |= 0x1 << 1; + break; + case 44100: + case 48000: + break; + } + + /* Sampling rate is known now, can configure the MCLK divider */ + wm8978->f_256fs = params_rate(params) * 256; + + if (wm8978->sysclk == WM8978_MCLK) { + f_sel = wm8978->f_mclk; + } else { + if (!wm8978->f_pllout) { + int ret = wm8978_configure_pll(codec); + if (ret < 0) + return ret; + } + f_sel = wm8978->f_pllout; + } + + /* + * In some cases it is possible to reconfigure PLL to a higher frequency + * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or + * 512 * fs, so, we should be fine. + */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + + if (diff < diff_best) { + diff_best = diff; + best = i; + } + + if (!diff) + break; + } + + if (diff) + dev_warn(codec->dev, "Imprecise clock: %u%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best], + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); + + dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, + params_format(params), params_rate(params), best); + + /* MCLK divisor mask = 0xe0 */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5); + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface_ctl); + snd_soc_write(codec, WM8978_ADDITIONAL_CONTROL, add_ctl); + + if (wm8978->sysclk != current_clk_id) { + if (wm8978->sysclk == WM8978_PLL) + /* Run CODEC from PLL instead of MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, + 0x100, 0x100); + else + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + } + + return 0; +} + +static int wm8978_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int wm8978_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + power1 |= 1; /* VMID 75k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_STANDBY: + /* bit 3: enable bias, bit 2: enable I/O tie off buffer */ + power1 |= 0xc; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, + power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_OFF: + /* Preserve PLL - OPCLK may be used by someone */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, ~0x20, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1); + + codec->bias_level = level; + return 0; +} + +#define WM8978_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8978_dai_ops = { + .hw_params = wm8978_hw_params, + .digital_mute = wm8978_mute, + .set_fmt = wm8978_set_dai_fmt, + .set_clkdiv = wm8978_set_dai_clkdiv, + .set_sysclk = wm8978_set_dai_sysclk, +}; + +/* Also supports 12kHz */ +struct snd_soc_dai wm8978_dai = { + .name = "WM8978 HiFi", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .ops = &wm8978_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8978_dai); + +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); + /* Also switch PLL off */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); + /* Put to sleep */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); + + return 0; +} + +static int wm8978_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int i; + u16 *cache = codec->reg_cache; + + /* Wake up the codec */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { + if (i == WM8978_RESET) + continue; + if (cache[i] != wm8978_reg[i]) + snd_soc_write(codec, i, cache[i]); + } + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (wm8978->f_pllout) + /* Switch PLL on */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + return 0; +} + +static int wm8978_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8978_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8978_codec; + codec = wm8978_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8978_snd_controls, + ARRAY_SIZE(wm8978_snd_controls)); + wm8978_add_widgets(codec); + +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8978_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8978 = { + .probe = wm8978_probe, + .remove = wm8978_remove, + .suspend = wm8978_suspend, + .resume = wm8978_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8978); + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + WM8978_LEFT_DAC_DIGITAL_VOLUME, + WM8978_RIGHT_DAC_DIGITAL_VOLUME, + WM8978_LEFT_ADC_DIGITAL_VOLUME, + WM8978_RIGHT_ADC_DIGITAL_VOLUME, + WM8978_LEFT_INP_PGA_CONTROL, + WM8978_RIGHT_INP_PGA_CONTROL, + WM8978_LOUT1_HP_CONTROL, + WM8978_ROUT1_HP_CONTROL, + WM8978_LOUT2_SPK_CONTROL, + WM8978_ROUT2_SPK_CONTROL, +}; + +static __devinit int wm8978_register(struct wm8978_priv *wm8978) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8978->codec; + + if (wm8978_codec) { + dev_err(codec->dev, "Another WM8978 is registered\n"); + return -EINVAL; + } + + /* + * Set default system clock to PLL, it is more precise, this is also the + * default hardware setting + */ + wm8978->sysclk = WM8978_PLL; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8978; + codec->name = "WM8978"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8978_set_bias_level; + codec->dai = &wm8978_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8978_CACHEREGNUM; + codec->reg_cache = &wm8978->reg_cache; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + memcpy(codec->reg_cache, wm8978_reg, sizeof(wm8978_reg)); + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + + /* Reset the codec */ + ret = snd_soc_write(codec, WM8978_RESET, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8978_dai.dev = codec->dev; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8978_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8978_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8978); + return ret; +} + +static __devexit void wm8978_unregister(struct wm8978_priv *wm8978) +{ + wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8978_dai); + snd_soc_unregister_codec(&wm8978->codec); + kfree(wm8978); + wm8978_codec = NULL; +} + +static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8978_priv *wm8978; + struct snd_soc_codec *codec; + + wm8978 = kzalloc(sizeof(struct wm8978_priv), GFP_KERNEL); + if (wm8978 == NULL) + return -ENOMEM; + + codec = &wm8978->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8978); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8978_register(wm8978); +} + +static __devexit int wm8978_i2c_remove(struct i2c_client *client) +{ + struct wm8978_priv *wm8978 = i2c_get_clientdata(client); + wm8978_unregister(wm8978); + return 0; +} + +static const struct i2c_device_id wm8978_i2c_id[] = { + { "wm8978", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); + +static struct i2c_driver wm8978_i2c_driver = { + .driver = { + .name = "WM8978", + .owner = THIS_MODULE, + }, + .probe = wm8978_i2c_probe, + .remove = __devexit_p(wm8978_i2c_remove), + .id_table = wm8978_i2c_id, +}; + +static int __init wm8978_modinit(void) +{ + return i2c_add_driver(&wm8978_i2c_driver); +} +module_init(wm8978_modinit); + +static void __exit wm8978_exit(void) +{ + i2c_del_driver(&wm8978_i2c_driver); +} +module_exit(wm8978_exit); + +MODULE_DESCRIPTION("ASoC WM8978 codec driver"); +MODULE_AUTHOR("Guennadi Liakhovetski "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h new file mode 100644 index 000000000000..b58f0bf947e7 --- /dev/null +++ b/sound/soc/codecs/wm8978.h @@ -0,0 +1,89 @@ +/* + * wm8978.h -- codec driver for WM8978 + * + * Copyright 2009 Guennadi Liakhovetski + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __WM8978_H__ +#define __WM8978_H__ + +/* + * Register values. + */ +#define WM8978_RESET 0x00 +#define WM8978_POWER_MANAGEMENT_1 0x01 +#define WM8978_POWER_MANAGEMENT_2 0x02 +#define WM8978_POWER_MANAGEMENT_3 0x03 +#define WM8978_AUDIO_INTERFACE 0x04 +#define WM8978_COMPANDING_CONTROL 0x05 +#define WM8978_CLOCKING 0x06 +#define WM8978_ADDITIONAL_CONTROL 0x07 +#define WM8978_GPIO_CONTROL 0x08 +#define WM8978_JACK_DETECT_CONTROL_1 0x09 +#define WM8978_DAC_CONTROL 0x0A +#define WM8978_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8978_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8978_JACK_DETECT_CONTROL_2 0x0D +#define WM8978_ADC_CONTROL 0x0E +#define WM8978_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8978_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8978_EQ1 0x12 +#define WM8978_EQ2 0x13 +#define WM8978_EQ3 0x14 +#define WM8978_EQ4 0x15 +#define WM8978_EQ5 0x16 +#define WM8978_DAC_LIMITER_1 0x18 +#define WM8978_DAC_LIMITER_2 0x19 +#define WM8978_NOTCH_FILTER_1 0x1b +#define WM8978_NOTCH_FILTER_2 0x1c +#define WM8978_NOTCH_FILTER_3 0x1d +#define WM8978_NOTCH_FILTER_4 0x1e +#define WM8978_ALC_CONTROL_1 0x20 +#define WM8978_ALC_CONTROL_2 0x21 +#define WM8978_ALC_CONTROL_3 0x22 +#define WM8978_NOISE_GATE 0x23 +#define WM8978_PLL_N 0x24 +#define WM8978_PLL_K1 0x25 +#define WM8978_PLL_K2 0x26 +#define WM8978_PLL_K3 0x27 +#define WM8978_3D_CONTROL 0x29 +#define WM8978_BEEP_CONTROL 0x2b +#define WM8978_INPUT_CONTROL 0x2c +#define WM8978_LEFT_INP_PGA_CONTROL 0x2d +#define WM8978_RIGHT_INP_PGA_CONTROL 0x2e +#define WM8978_LEFT_ADC_BOOST_CONTROL 0x2f +#define WM8978_RIGHT_ADC_BOOST_CONTROL 0x30 +#define WM8978_OUTPUT_CONTROL 0x31 +#define WM8978_LEFT_MIXER_CONTROL 0x32 +#define WM8978_RIGHT_MIXER_CONTROL 0x33 +#define WM8978_LOUT1_HP_CONTROL 0x34 +#define WM8978_ROUT1_HP_CONTROL 0x35 +#define WM8978_LOUT2_SPK_CONTROL 0x36 +#define WM8978_ROUT2_SPK_CONTROL 0x37 +#define WM8978_OUT3_MIXER_CONTROL 0x38 +#define WM8978_OUT4_MIXER_CONTROL 0x39 + +#define WM8978_CACHEREGNUM 58 + +/* Clock divider Id's */ +enum wm8978_clk_id { + WM8978_OPCLKRATE, + WM8978_MCLKDIV, + WM8978_ADCCLK, + WM8978_DACCLK, + WM8978_BCLKDIV, +}; + +enum wm8978_sysclk_src { + WM8978_PLL, + WM8978_MCLK +}; + +extern struct snd_soc_dai wm8978_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8978; + +#endif /* __WM8978_H__ */ -- cgit v1.2.2 From b09f3e78ee7bb69171411b75bd9e771fc7f24749 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jan 2010 00:01:53 +0100 Subject: ALSA: hda - Allow override more fields via patch loader Allow the override of vendor-id, subsystem-id, revision-id and chip name via patch loading. Updated the document, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 53 +++++++++++++++++++++++++++++++++-------------- 1 file changed, 38 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index b36919c0d363..a1fc83753cc6 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -625,6 +625,10 @@ enum { LINE_MODE_PINCFG, LINE_MODE_VERB, LINE_MODE_HINT, + LINE_MODE_VENDOR_ID, + LINE_MODE_SUBSYSTEM_ID, + LINE_MODE_REVISION_ID, + LINE_MODE_CHIP_NAME, NUM_LINE_MODES, }; @@ -654,53 +658,71 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, } /* parse the contents after the other command tags, [pincfg], [verb], - * [hint] and [model] + * [vendor_id], [subsystem_id], [revision_id], [chip_name], [hint] and [model] * just pass to the sysfs helper (only when any codec was specified) */ static void parse_pincfg_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; parse_user_pin_configs(*codecp, buf); } static void parse_verb_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; parse_init_verbs(*codecp, buf); } static void parse_hint_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; parse_hints(*codecp, buf); } static void parse_model_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; kfree((*codecp)->modelname); (*codecp)->modelname = kstrdup(buf, GFP_KERNEL); } +static void parse_chip_name_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + kfree((*codecp)->chip_name); + (*codecp)->chip_name = kstrdup(buf, GFP_KERNEL); +} + +#define DEFINE_PARSE_ID_MODE(name) \ +static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ + struct hda_codec **codecp) \ +{ \ + unsigned long val; \ + if (!strict_strtoul(buf, 0, &val)) \ + (*codecp)->name = val; \ +} + +DEFINE_PARSE_ID_MODE(vendor_id); +DEFINE_PARSE_ID_MODE(subsystem_id); +DEFINE_PARSE_ID_MODE(revision_id); + + struct hda_patch_item { const char *tag; void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc); + int need_codec; }; static struct hda_patch_item patch_items[NUM_LINE_MODES] = { - [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode }, - [LINE_MODE_MODEL] = { "[model]", parse_model_mode }, - [LINE_MODE_VERB] = { "[verb]", parse_verb_mode }, - [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode }, - [LINE_MODE_HINT] = { "[hint]", parse_hint_mode }, + [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode, 0 }, + [LINE_MODE_MODEL] = { "[model]", parse_model_mode, 1 }, + [LINE_MODE_VERB] = { "[verb]", parse_verb_mode, 1 }, + [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode, 1 }, + [LINE_MODE_HINT] = { "[hint]", parse_hint_mode, 1 }, + [LINE_MODE_VENDOR_ID] = { "[vendor_id]", parse_vendor_id_mode, 1 }, + [LINE_MODE_SUBSYSTEM_ID] = { "[subsystem_id]", parse_subsystem_id_mode, 1 }, + [LINE_MODE_REVISION_ID] = { "[revision_id]", parse_revision_id_mode, 1 }, + [LINE_MODE_CHIP_NAME] = { "[chip_name]", parse_chip_name_mode, 1 }, }; /* check the line starting with '[' -- change the parser mode accodingly */ @@ -783,7 +805,8 @@ int snd_hda_load_patch(struct hda_bus *bus, const char *patch) continue; if (*buf == '[') line_mode = parse_line_mode(buf, bus); - else if (patch_items[line_mode].parser) + else if (patch_items[line_mode].parser && + (codec || !patch_items[line_mode].need_codec)) patch_items[line_mode].parser(buf, bus, &codec); } release_firmware(fw); -- cgit v1.2.2 From 8fc176d5abb2d92c52df859faac7974b4a1585c1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Jan 2010 13:46:16 +0900 Subject: ASoC: fsi: Add spin lock operation for accessing shared area fsi_master_xxx function should be protected by spin lock, because it are used from both FSI-A and FSI-B. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 29 ++++++++++++++++++++++++++--- 1 file changed, 26 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5f9f2693f4eb..ebf358808db1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -110,6 +110,7 @@ struct fsi_master { struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; + spinlock_t lock; }; /************************************************************************ @@ -168,30 +169,51 @@ static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_write((u32)(master->base + reg), data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_write((u32)(master->base + reg), data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) { + u32 ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return 0; - return __fsi_reg_read((u32)(master->base + reg)); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_read((u32)(master->base + reg)); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static int fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } /************************************************************************ @@ -929,6 +951,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsib.base = master->base + 0x40; master->fsib.master = master; + spin_lock_init(&master->lock); pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); -- cgit v1.2.2 From c812459396733b42655e0d656763af02e06f97ed Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 28 Jan 2010 15:57:04 +0200 Subject: ASoC: TWL4030: Modify codec default settings Change the legacy default register configuration, which left some internal components on. Now we have either DAPM, or other ways to control these bits, so there is no need to enable them by default. The affected parts: Disable ADCL and ADCR Disable ARXL2 and ARXR2 analog PGA (playback) Disable APLL by default Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 74f0d65f0784..e0106a5fd40b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -64,12 +64,12 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ - 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ @@ -99,7 +99,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ - 0x16, /* REG_APLL_CTL (0x3A) */ + 0x06, /* REG_APLL_CTL (0x3A) */ 0x00, /* REG_DTMF_CTL (0x3B) */ 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ -- cgit v1.2.2 From fb58a2ff300cb3fd6077484ca7d8c6e6f13a0350 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Jan 2010 10:22:45 +0000 Subject: ASoC: Remove version display from WM9713 The version isn't being updated or used, the kernel revision tracking is enough. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9713.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index c58aab375edb..96e46d9a0171 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -28,8 +28,6 @@ #include "wm9713.h" -#define WM9713_VERSION "0.15" - struct wm9713_priv { u32 pll_in; /* PLL input frequency */ }; @@ -1186,8 +1184,6 @@ static int wm9713_soc_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0, reg; - printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); - socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (socdev->card->codec == NULL) -- cgit v1.2.2 From e03a8d2cf663429e2480a8db78b132ee300f79af Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Jan 2010 12:36:07 +0000 Subject: ASoC: Add TLV information and additional volumes to WM9713 Also renames a few things to make volumes and switches match up in alsamixer. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9713.c | 60 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 96e46d9a0171..ceb86b4ddb25 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -113,15 +114,27 @@ SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ }; +static const DECLARE_TLV_DB_SCALE(out_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(misc_tlv, -1500, 300, 0); +static unsigned int mic_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { -SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), +SOC_DOUBLE_TLV("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1, out_tlv), SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), -SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), +SOC_DOUBLE_TLV("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1, + out_tlv), SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1), -SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), -SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), -SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), -SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), +SOC_DOUBLE_TLV("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1, main_tlv), +SOC_DOUBLE_TLV("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic 1 Preamp Volume", AC97_3D_CONTROL, 10, 3, 0, mic_tlv), +SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 12, 3, 0, mic_tlv), SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), @@ -131,7 +144,7 @@ SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0), SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), -SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), +SOC_SINGLE_TLV("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1, misc_tlv), SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), @@ -152,28 +165,43 @@ SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), -SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), +SOC_SINGLE_TLV("Out4 Playback Volume", AC97_MASTER_MONO, 8, 31, 1, out_tlv), SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), -SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), +SOC_SINGLE_TLV("Out3 Playback Volume", AC97_MASTER_MONO, 0, 31, 1, out_tlv), -SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), +SOC_SINGLE_TLV("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1, main_tlv), SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), +SOC_SINGLE_TLV("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1, out_tlv), -SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), -SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), -SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), +SOC_SINGLE_TLV("Headphone Mixer Beep Playback Volume", AC97_AUX, 12, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Speaker Mixer Beep Playback Volume", AC97_AUX, 8, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Mono Mixer Beep Playback Volume", AC97_AUX, 4, 7, 1, misc_tlv), -SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), +SOC_SINGLE_TLV("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1, + misc_tlv), SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), +SOC_SINGLE_TLV("Headphone Mixer Aux Playback Volume", AC97_REC_SEL, 12, 7, 1, + misc_tlv), + +SOC_SINGLE_TLV("Speaker Mixer Voice Playback Volume", AC97_PCM, 8, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Speaker Mixer Aux Playback Volume", AC97_REC_SEL, 8, 7, 1, + misc_tlv), + +SOC_SINGLE_TLV("Mono Mixer Voice Playback Volume", AC97_PCM, 4, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Mono Mixer Aux Playback Volume", AC97_REC_SEL, 4, 7, 1, + misc_tlv), + SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), -SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), SOC_ENUM("Bass Control", wm9713_enum[16]), SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), -- cgit v1.2.2 From 2718625fba1e07bf2ce8a752036658737c1f76a7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Jan 2010 12:36:29 +0000 Subject: ASoC: Set codec->dev for AC97 devices Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9085b40fa04b..ca89c782132d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1368,6 +1368,7 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, codec->ac97->bus->ops = ops; codec->ac97->num = num; + codec->dev = &codec->ac97->dev; mutex_unlock(&codec->mutex); return 0; } -- cgit v1.2.2 From 7b36ea967cc5b5088a57fe225f1f72a3c160058b Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Thu, 28 Jan 2010 16:13:07 +0800 Subject: ALSA: hda - Change the AZX_MAX_PCMS to 10 In hda_codec.c, it has define "[HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 },", it support up to device 9 for HDMI. But in hda_intel.c, it only define AZX_MAX_PCMS as 8. So if it have 4 hdmi codecs, when run azx_attach_pcm_stream(), it will show error "Invalid PCM device number 8", and "... number 9", and return "-EINVAL". We should change the AZX_MAX_PCMS to 10. Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6eeefda63838..170126c28abd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -261,7 +261,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) /* max number of PCM devics per card */ -#define AZX_MAX_PCMS 8 +#define AZX_MAX_PCMS 10 /* RIRB int mask: overrun[2], response[0] */ #define RIRB_INT_RESPONSE 0x01 -- cgit v1.2.2 From c89362225152fc6f2247f65371bfe3ccced3203b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jan 2010 17:08:53 +0100 Subject: ALSA: hda - Define max number of PCM devices in hda_codec.h Define the constant rather in the common header file. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_intel.c | 10 ++++------ 3 files changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 26ceace88c96..98767df4f03a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3275,6 +3275,8 @@ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { /* * get the empty PCM device number to assign + * + * note the max device number is limited by HDA_MAX_PCMS, currently 10 */ static int get_empty_pcm_device(struct hda_bus *bus, int type) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0c8f05cc56be..b75da47571e6 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -527,6 +527,9 @@ enum { /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f +/* max number of PCM devics per card */ +#define HDA_MAX_PCMS 10 + /* * generic arrays */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170126c28abd..12230a2ed4f1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -260,8 +260,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_MAX_FRAG 32 /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) -/* max number of PCM devics per card */ -#define AZX_MAX_PCMS 10 /* RIRB int mask: overrun[2], response[0] */ #define RIRB_INT_RESPONSE 0x01 @@ -409,7 +407,7 @@ struct azx { struct azx_dev *azx_dev; /* PCM */ - struct snd_pcm *pcm[AZX_MAX_PCMS]; + struct snd_pcm *pcm[HDA_MAX_PCMS]; /* HD codec */ unsigned short codec_mask; @@ -1336,7 +1334,7 @@ static void azx_bus_reset(struct hda_bus *bus) if (chip->initialized) { int i; - for (i = 0; i < AZX_MAX_PCMS; i++) + for (i = 0; i < HDA_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus); snd_hda_resume(chip->bus); @@ -1966,7 +1964,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, int pcm_dev = cpcm->device; int s, err; - if (pcm_dev >= AZX_MAX_PCMS) { + if (pcm_dev >= HDA_MAX_PCMS) { snd_printk(KERN_ERR SFX "Invalid PCM device number %d\n", pcm_dev); return -EINVAL; @@ -2122,7 +2120,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_clear_irq_pending(chip); - for (i = 0; i < AZX_MAX_PCMS; i++) + for (i = 0; i < HDA_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) snd_hda_suspend(chip->bus); -- cgit v1.2.2 From 30ed7ed11cb88fd56d821a67b9aab1e0d50fb626 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jan 2010 17:11:45 +0100 Subject: ALSA: hda - Fix index of HP Compaq F700 mic amp The amp used for the mic input on HP Compaq F700 with Cxt5051 codec has no multiple inputs, thus its index should be 0 instead of 1. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9077e4174ee6..745e35992144 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1832,7 +1832,7 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { static struct hda_verb cxt5051_f700_init_verbs[] = { /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, -- cgit v1.2.2 From e108c7b79e91b45a3f04762c44fd404a5d9be069 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Thu, 28 Jan 2010 19:21:07 +0100 Subject: ALSA: hda - Add mute LED check for HP laptops with IDT 92HD83xxx codec This patch adds HP mute LED support for IDT 92HD81/3 family of the codecs. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dbffb5b5c69d..cb9802f4b063 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5332,6 +5332,11 @@ again: if (spec->board_config == STAC_92HD83XXX_HP) spec->gpio_led = 0x01; + if (find_mute_led_gpio(codec)) + snd_printd("mute LED gpio %d polarity %d\n", + spec->gpio_led, + spec->gpio_led_polarity); + #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { spec->gpio_mask |= spec->gpio_led; -- cgit v1.2.2 From 6016a363f6b56b46b24655bcfc0499b715851cf3 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Thu, 28 Jan 2010 14:06:53 -0700 Subject: of: unify phandle name in struct device_node In struct device_node, the phandle is named 'linux_phandle' for PowerPC and MicroBlaze, and 'node' for SPARC. There is no good reason for the difference, it is just an artifact of the code diverging over a couple of years. This patch renames both to simply .phandle. Note: the .node also existed in PowerPC/MicroBlaze, but the only user seems to be arch/powerpc/platforms/powermac/pfunc_core.c. It doesn't look like the assignment between .linux_phandle and .node is significantly different enough to warrant the separate code paths unless ibm,phandle properties actually appear in Apple device trees. I think it is safe to eliminate the old .node property and use phandle everywhere. Signed-off-by: Grant Likely Acked-by: David S. Miller Tested-by: Wolfram Sang Acked-by: Benjamin Herrenschmidt --- sound/aoa/fabrics/layout.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 586965f9605f..7a437da05646 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -768,7 +768,7 @@ static int check_codec(struct aoa_codec *codec, "required property %s not present\n", propname); return -ENODEV; } - if (*ref != codec->node->linux_phandle) { + if (*ref != codec->node->phandle) { printk(KERN_INFO "snd-aoa-fabric-layout: " "%s doesn't match!\n", propname); return -ENODEV; -- cgit v1.2.2 From 36706005d90642bccabfaacbb24d135155e984a8 Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Fri, 29 Jan 2010 12:05:51 +0100 Subject: ALSA: hda - Add support for IDT 92HD88 family codecs Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index cb9802f4b063..9694675f0b9e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -568,6 +568,11 @@ static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0f, 0x10, 0x11, 0x1f, 0x20, }; +static hda_nid_t stac92hd88xxx_pin_nids[10] = { + 0x0a, 0x0b, 0x0c, 0x0d, + 0x0f, 0x11, 0x1f, 0x20, +}; + #define STAC92HD71BXX_NUM_PINS 13 static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x00, @@ -2873,6 +2878,13 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) conn_len = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); + /* 92HD88: trace back up the link of nids to find the DAC */ + while (conn_len == 1 && (get_wcaps_type(get_wcaps(codec, conn[0])) + != AC_WID_AUD_OUT)) { + nid = conn[0]; + conn_len = snd_hda_get_connections(codec, nid, conn, + HDA_MAX_CONNECTIONS); + } for (j = 0; j < conn_len; j++) { wcaps = get_wcaps(codec, conn[j]); wtype = get_wcaps_type(wcaps); @@ -5318,6 +5330,16 @@ again: stac92hd83xxx_brd_tbl[spec->board_config]); switch (codec->vendor_id) { + case 0x111d7666: + case 0x111d7667: + case 0x111d7668: + case 0x111d7669: + spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); + spec->pin_nids = stac92hd88xxx_pin_nids; + spec->mono_nid = 0; + spec->digbeep_nid = 0; + spec->num_pwrs = 0; + break; case 0x111d7604: case 0x111d7605: case 0x111d76d5: @@ -6243,6 +6265,10 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7667, .name = "92HD88B1", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7668, .name = "92HD88B2", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7669, .name = "92HD88B4", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, -- cgit v1.2.2 From 3e59aaa7ae9de49af8810102f12857860d5bd0ed Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 29 Jan 2010 13:58:55 +0530 Subject: ASoC: AIC23: Fixing writes to non-existing registers in resume function Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23 register in resume function because of which register writes happen on some non-existing registers. Signed-off-by: Anuj Aggarwal Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a9dc5fb54774..da589d8664d0 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < TLV320AIC23_RESET; reg++) { + for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- cgit v1.2.2 From 5bbd4953a4fb5d8d597b4a53b8da97eee320b634 Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 29 Jan 2010 15:49:22 +0530 Subject: ASoC: AM3517: ASoC driver not getting compiled Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the Makefile. Whereas the config option defined in Kconfig is SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517 was not getting compiled. Signed-off-by: Anuj Aggarwal Signed-off-by: Mark Brown --- sound/soc/omap/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c523f4..19283e5edfbf 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- cgit v1.2.2 From 9e9d04c05fd01018da35fa1daa9bda844cac6162 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Fri, 29 Jan 2010 10:57:07 +0900 Subject: ASoC: AC97: SMDK-WM9713: Convert notes from cset to sset It's more robust when references are provided in control names rather than numid. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/smdk_wm9713.c | 23 +++++++---------------- 1 file changed, 7 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 6fa2c9d17d7a..24fd39f38ccb 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -29,24 +29,15 @@ static struct snd_soc_card smdk; /* Playback (HeadPhone):- - Headphone Playback Switch - On - $ amixer cset numid=4 1 - - Right Headphone Out Mux - Headphone - $ amixer cset numid=92 2 - Left Headphone Out Mux - Headphone - $ amixer cset numid=93 2 - - Right HP Mixer PCM Playback Switch - On - $ amixer cset numid=75 1 - Left HP Mixer PCM Playback Switch - On - $ amixer cset numid=81 1 + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute Capture (LineIn):- - Right Capture Source - Line - $ amixer cset numid=86 2 - Left Capture Source - Line - $ amixer cset numid=87 2 + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' */ static struct snd_soc_dai_link smdk_dai = { -- cgit v1.2.2 From 9f5b64b767203131a7a3a280859e70d4413c9672 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 27 Jan 2010 12:15:00 +0100 Subject: ASoC: add support for the sh7722 Migo-R board Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978 codec, recording via external microphone and playback via headphones are implemented. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 8 ++ sound/soc/sh/Makefile | 2 + sound/soc/sh/migor.c | 222 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 232 insertions(+) create mode 100644 sound/soc/sh/migor.c (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 3f1cd5503342..a86696bbe179 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -61,4 +61,12 @@ config SND_FSI_DA7210 This option enables generic sound support for the FSI - DA7210 unit +config SND_SIU_MIGOR + tristate "SIU sound support on Migo-R" + depends on SH_MIGOR + select SND_SOC_SH4_SIU + select SND_SOC_WM8978 + help + This option enables sound support for the SH7722 Migo-R board + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 5a97d2539d84..8a5a19293bda 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -16,7 +16,9 @@ obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o snd-soc-fsi-da7210-objs := fsi-da7210.o +snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o +obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c new file mode 100644 index 000000000000..3ccd9b393312 --- /dev/null +++ b/sound/soc/sh/migor.c @@ -0,0 +1,222 @@ +/* + * ALSA SoC driver for Migo-R + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include + +#include + +#include +#include +#include +#include + +#include "../codecs/wm8978.h" +#include "siu.h" + +/* Default 8000Hz sampling frequency */ +static unsigned long codec_freq = 8000 * 512; + +static unsigned int use_count; + +/* External clock, sourced from the codec at the SIUMCKB pin */ +static unsigned long siumckb_recalc(struct clk *clk) +{ + return codec_freq; +} + +static struct clk_ops siumckb_clk_ops = { + .recalc = siumckb_recalc, +}; + +static struct clk siumckb_clk = { + .name = "siumckb_clk", + .id = -1, + .ops = &siumckb_clk_ops, + .rate = 0, /* initialised at run-time */ +}; + +static int migor_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + unsigned int rate = params_rate(params); + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 13000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(rtd->dai->cpu_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + codec_freq = rate * 512; + /* + * This propagates the parent frequency change to children and + * recalculates the frequency table + */ + clk_set_rate(&siumckb_clk, codec_freq); + dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); + + ret = snd_soc_dai_set_sysclk(rtd->dai->cpu_dai, SIU_CLKB_EXT, + codec_freq / 2, SND_SOC_CLOCK_IN); + + if (!ret) + use_count++; + + return ret; +} + +static int migor_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + if (use_count) { + use_count--; + + if (!use_count) + snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 0, + SND_SOC_CLOCK_IN); + } else { + dev_dbg(codec_dai->dev, "Unbalanced hw_free!\n"); + } + + return 0; +} + +static struct snd_soc_ops migor_dai_ops = { + .hw_params = migor_hw_params, + .hw_free = migor_hw_free, +}; + +static const struct snd_soc_dapm_widget migor_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Onboard Microphone", NULL), + SND_SOC_DAPM_MIC("External Microphone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone output connected to LHP/RHP, enable OUT4 for VMID */ + { "Headphone", NULL, "OUT4 VMID" }, + { "OUT4 VMID", NULL, "LHP" }, + { "OUT4 VMID", NULL, "RHP" }, + + /* On-board microphone */ + { "RMICN", NULL, "Mic Bias" }, + { "RMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Onboard Microphone" }, + + /* External microphone */ + { "LMICN", NULL, "Mic Bias" }, + { "LMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "External Microphone" }, +}; + +static int migor_dai_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, migor_dapm_widgets, + ARRAY_SIZE(migor_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* migor digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link migor_dai = { + .name = "wm8978", + .stream_name = "WM8978", + .cpu_dai = &siu_i2s_dai, + .codec_dai = &wm8978_dai, + .ops = &migor_dai_ops, + .init = migor_dai_init, +}; + +/* migor audio machine driver */ +static struct snd_soc_card snd_soc_migor = { + .name = "Migo-R", + .platform = &siu_platform, + .dai_link = &migor_dai, + .num_links = 1, +}; + +/* migor audio subsystem */ +static struct snd_soc_device migor_snd_devdata = { + .card = &snd_soc_migor, + .codec_dev = &soc_codec_dev_wm8978, +}; + +static struct platform_device *migor_snd_device; + +static int __init migor_init(void) +{ + int ret; + + ret = clk_register(&siumckb_clk); + if (ret < 0) + return ret; + + /* Port number used on this machine: port B */ + migor_snd_device = platform_device_alloc("soc-audio", 1); + if (!migor_snd_device) { + ret = -ENOMEM; + goto epdevalloc; + } + + platform_set_drvdata(migor_snd_device, &migor_snd_devdata); + + migor_snd_devdata.dev = &migor_snd_device->dev; + + ret = platform_device_add(migor_snd_device); + if (ret) + goto epdevadd; + + return 0; + +epdevadd: + platform_device_put(migor_snd_device); +epdevalloc: + clk_unregister(&siumckb_clk); + return ret; +} + +static void __exit migor_exit(void) +{ + clk_unregister(&siumckb_clk); + platform_device_unregister(migor_snd_device); +} + +module_init(migor_init); +module_exit(migor_exit); + +MODULE_AUTHOR("Guennadi Liakhovetski "); +MODULE_DESCRIPTION("ALSA SoC Migor"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.2 From 640b796f2ca88113bf2fefd380bc807092ce6fa1 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 28 Jan 2010 16:28:55 +0100 Subject: ASoC: remove bogus SLEEP mode from wm8978 driver Tests showed, that bit 6 of the WM8978_POWER_MANAGEMENT_2 register of wm8978 affects codec clocks. Being useless for suspend / resume, it cannot be used in bias-level control either. Remove this bit handling. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index d9d4e9dd1adb..8dcebaa8604a 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -873,8 +873,6 @@ static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); - /* Put to sleep */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); return 0; } @@ -887,9 +885,6 @@ static int wm8978_resume(struct platform_device *pdev) int i; u16 *cache = codec->reg_cache; - /* Wake up the codec */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { if (i == WM8978_RESET) -- cgit v1.2.2 From b2c3e923110f6ca60ccb30cf4a6bda5211454c4f Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 29 Jan 2010 15:31:06 +0100 Subject: ASoC: clean up wm8974 and wm8978 clock divider handling wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 12 ++++-------- sound/soc/codecs/wm8974.h | 12 +----------- sound/soc/codecs/wm8978.c | 19 ++++--------------- sound/soc/codecs/wm8978.h | 3 --- sound/soc/sh/migor.c | 4 ---- 5 files changed, 9 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751da8c9..ee637af4737a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling Switch", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de9562d4d2..896a7f0f3fc4 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa8604a..ec2624b4c370 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf947e7..56ec83270917 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b393312..b823a5c9b9bc 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- cgit v1.2.2 From a75d7a4cf50d1cee14d8c9aaa2b971232d10f2c1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Feb 2010 13:29:50 +0100 Subject: sound: control: actually allow TLV command access Creating a control with TLV_COMMAND access was not possible because snd_ctl_new1() forgot to include it in the mask of allowable access bits. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/core/control.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 268ab7471224..6a4764dcb180 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -237,8 +237,9 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| + SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); kctl.info = ncontrol->info; kctl.get = ncontrol->get; kctl.put = ncontrol->put; -- cgit v1.2.2 From 6123637fafbf445cc9ce5774dc9516da0b2daa88 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Feb 2010 13:30:56 +0100 Subject: sound: control: fix minimum TLV length Allow TLV blocks that do not have any values; the smallest possible TLV is an empty container or one where the information is only in the tag. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 6a4764dcb180..439ce64f9d82 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1100,7 +1100,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, if (copy_from_user(&tlv, _tlv, sizeof(tlv))) return -EFAULT; - if (tlv.length < sizeof(unsigned int) * 3) + if (tlv.length < sizeof(unsigned int) * 2) return -EINVAL; down_read(&card->controls_rwsem); kctl = snd_ctl_find_numid(card, tlv.numid); -- cgit v1.2.2 From b0580913797034a1001e867b8b492c75226bf77e Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 29 Jan 2010 14:51:26 +0100 Subject: ASoC: improve MCLKDIV calculation in wm8978, when OPCLK is not used In case, if OPCLK is not used, and PLL is used for driving the codec, the choice of PLL output frequency could result in a needlessly imprecise system clock frequency. Use an iterative process to select a precise configuration. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 115 +++++++++++++++++++++++++++++++--------------- 1 file changed, 78 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index ec2624b4c370..28bb59ea6ea1 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -58,6 +58,7 @@ struct wm8978_priv { unsigned int f_mclk; unsigned int f_256fs; unsigned int f_opclk; + int mclk_idx; enum wm8978_sysclk_src sysclk; u16 reg_cache[WM8978_CACHEREGNUM]; }; @@ -402,6 +403,35 @@ static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, pll_div->k = k; } + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * find index >= idx, such that, for a given f_out, + * 3 * f_mclk / 4 <= f_PLLOUT < 13 * f_mclk / 4 + * f_out can be f_256fs or f_opclk, currently only used for f_256fs. Can be + * generalised for f_opclk with suitable coefficient arrays, but currently + * the OPCLK divisor is calculated directly, not iteratively. + */ +static int wm8978_enum_mclk(unsigned int f_out, unsigned int f_mclk, + unsigned int *f_pllout) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + unsigned int f_pllout_x4 = 4 * f_out * mclk_numerator[i] / + mclk_denominator[i]; + if (3 * f_mclk <= f_pllout_x4 && f_pllout_x4 < 13 * f_mclk) { + *f_pllout = f_pllout_x4 / 4; + return i; + } + } + + return -EINVAL; +} + /* * Calculate internal frequencies and dividers, according to Figure 40 * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 @@ -412,12 +442,16 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) struct wm8978_pll_div pll_div; unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, f_256fs = wm8978->f_256fs; - unsigned int f2, opclk_div; + unsigned int f2; if (!f_mclk) return -EINVAL; if (f_opclk) { + unsigned int opclk_div; + /* Cannot set up MCLK divider now, do later */ + wm8978->mclk_idx = -1; + /* * The user needs OPCLK. Choose OPCLKDIV to put * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. @@ -444,7 +478,7 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) wm8978->f_pllout = f_opclk * opclk_div; } else if (f_256fs) { /* - * Not using OPCLK, choose R: + * Not using OPCLK, but PLL is used for the codec, choose R: * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where * prescale = 1, or prescale = 2. Prescale is calculated inside @@ -453,18 +487,11 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK * must be 3.781MHz <= f_MCLK <= 32.768MHz */ - if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) - return -EINVAL; + int idx = wm8978_enum_mclk(f_256fs, f_mclk, &wm8978->f_pllout); + if (idx < 0) + return idx; - /* - * MCLKDIV will be selected in .hw_params(), just choose a - * suitable f_PLLOUT - */ - if (4 * f_256fs < 3 * f_mclk) - /* Will have to use MCLKDIV */ - wm8978->f_pllout = wm8978->f_mclk * 3 / 4; - else - wm8978->f_pllout = f_256fs; + wm8978->mclk_idx = idx; /* GPIO1 into default mode as input - before configuring PLL */ snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); @@ -515,6 +542,20 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, wm8978->f_opclk = div; if (wm8978->f_mclk) + /* + * We know the MCLK frequency, the user has requested + * OPCLK, configure the PLL based on that and start it + * and OPCLK immediately. We will configure PLL to match + * user-requested OPCLK frquency as good as possible. + * In fact, it is likely, that matching the sampling + * rate, when it becomes known, is more important, and + * we will not be reconfiguring PLL then, because we + * must not interrupt OPCLK. But it should be fine, + * because typically the user will request OPCLK to run + * at 256fs or 512fs, and for these cases we will also + * find an exact MCLK divider configuration - it will + * be equal to or double the OPCLK divisor. + */ ret = wm8978_configure_pll(codec); break; case WM8978_BCLKDIV: @@ -640,10 +681,6 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -/* MCLK dividers */ -static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; -static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; - /* * Set PCM DAI bit size and sample rate. */ @@ -709,9 +746,11 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->f_256fs = params_rate(params) * 256; if (wm8978->sysclk == WM8978_MCLK) { + wm8978->mclk_idx = -1; f_sel = wm8978->f_mclk; } else { if (!wm8978->f_pllout) { + /* We only enter here, if OPCLK is not used */ int ret = wm8978_configure_pll(codec); if (ret < 0) return ret; @@ -719,32 +758,34 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, f_sel = wm8978->f_pllout; } - /* - * In some cases it is possible to reconfigure PLL to a higher frequency - * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or - * 512 * fs, so, we should be fine. - */ - if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) - return -EINVAL; + if (wm8978->mclk_idx < 0) { + /* Either MCLK is used directly, or OPCLK is used */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; - for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { - diff = abs(wm8978->f_256fs * 3 - - f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); - if (diff < diff_best) { - diff_best = diff; - best = i; - } + if (diff < diff_best) { + diff_best = diff; + best = i; + } - if (!diff) - break; + if (!diff) + break; + } + } else { + /* OPCLK not used, codec driven by PLL */ + best = wm8978->mclk_idx; + diff = 0; } if (diff) - dev_warn(codec->dev, "Imprecise clock: %u%s\n", - f_sel * mclk_denominator[best] / mclk_numerator[best], - wm8978->sysclk == WM8978_MCLK ? - ", consider using PLL" : ""); + dev_warn(codec->dev, "Imprecise sampling rate: %uHz%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best] / 256, + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, params_format(params), params_rate(params), best); -- cgit v1.2.2 From 2f1ff6614cb5938e5c5760358752d92deb67fb63 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sun, 31 Jan 2010 12:02:12 -0800 Subject: ASoC: Fix continuation line formats String constants that are continued on subsequent lines with \ are not good. Signed-off-by: Joe Perches Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 8 ++------ sound/soc/blackfin/bf5xx-i2s-pcm.c | 3 +-- sound/soc/blackfin/bf5xx-tdm-pcm.c | 3 +-- 3 files changed, 4 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index cf0dfb7ca221..67cbfe7283da 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -349,9 +349,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \ size, &sport_handle->tx_dma_phy, GFP_KERNEL); if (!sport_handle->tx_dma_buf) { - pr_err("Failed to allocate memory for tx dma \ - buf - Please increase uncached DMA \ - memory region\n"); + pr_err("Failed to allocate memory for tx dma buf - Please increase uncached DMA memory region\n"); return -ENOMEM; } else memset(sport_handle->tx_dma_buf, 0, size); @@ -362,9 +360,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) sport_handle->rx_dma_buf = dma_alloc_coherent(NULL, \ size, &sport_handle->rx_dma_phy, GFP_KERNEL); if (!sport_handle->rx_dma_buf) { - pr_err("Failed to allocate memory for rx dma \ - buf - Please increase uncached DMA \ - memory region\n"); + pr_err("Failed to allocate memory for rx dma buf - Please increase uncached DMA memory region\n"); return -ENOMEM; } else memset(sport_handle->rx_dma_buf, 0, size); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 62fbb8459569..c6c6a4a7d948 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -207,8 +207,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) { - pr_err("Failed to allocate dma memory \ - Please increase uncached DMA memory region\n"); + pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); return -ENOMEM; } buf->bytes = size; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index a8c73cbbd685..5e03bb2f3cd7 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -244,8 +244,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_coherent(pcm->card->dev, size * 4, &buf->addr, GFP_KERNEL); if (!buf->area) { - pr_err("Failed to allocate dma memory \ - Please increase uncached DMA memory region\n"); + pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); return -ENOMEM; } buf->bytes = size; -- cgit v1.2.2 From 3ed7074c4cc0de5ba77e180e5d96c23ef96859f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Jan 2010 17:39:45 +0000 Subject: ASoC: Improved wm_hubs headphone handling Perform DC servo offset calibration using a series update sequence rather than startup update sequence, tuning the configuration of the WM8993 DC servo to make best use of this. Also introduce currently unused data allowing us to correct for any systematic errors in the DC servo calibration results and an alternative startup path for the headphone output which performs better with some chip revisions. The alternative setup sequence is enabled for WM8993. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 8 +++ sound/soc/codecs/wm_hubs.c | 142 ++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm_hubs.h | 6 ++ 3 files changed, 130 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 828d8174d5b7..bacfc2f20d70 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -213,6 +213,7 @@ static struct { }; struct wm8993_priv { + struct wm_hubs_data hubs_data; u16 reg_cache[WM8993_REGISTER_COUNT]; struct wm8993_platform_data pdata; struct snd_soc_codec codec; @@ -997,6 +998,11 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Tune DC servo configuration */ + snd_soc_write(codec, 0x44, 3); + snd_soc_write(codec, 0x56, 3); + snd_soc_write(codec, 0x44, 0); + /* Bring up VMID with fast soft start */ snd_soc_update_bits(codec, WM8993_ANTIPOP2, WM8993_STARTUP_BIAS_ENA | @@ -1591,6 +1597,8 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, codec->num_dai = 1; codec->private_data = wm8993; + wm8993->hubs_data.hp_startup_mode = 1; + memcpy(wm8993->reg_cache, wm8993_reg_defaults, sizeof(wm8993->reg_cache)); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a67319d9ca7e..0ad9f5d536c6 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -68,24 +68,77 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) int count = 0; dev_dbg(codec->dev, "Waiting for DC servo...\n"); + do { count++; msleep(1); reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); - dev_dbg(codec->dev, "DC servo status: %x\n", reg); - } while ((reg & WM8993_DCS_CAL_COMPLETE_MASK) - != WM8993_DCS_CAL_COMPLETE_MASK && count < 1000); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & WM8993_DCS_DATAPATH_BUSY); - if ((reg & WM8993_DCS_CAL_COMPLETE_MASK) - != WM8993_DCS_CAL_COMPLETE_MASK) + if (reg & WM8993_DCS_DATAPATH_BUSY) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); } +/* + * Startup calibration of the DC servo + */ +static void calibrate_dc_servo(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = codec->private_data; + u16 reg, dcs_cfg; + + /* Set for 32 series updates */ + snd_soc_update_bits(codec, WM8993_DC_SERVO_1, + WM8993_DCS_SERIES_NO_01_MASK, + 32 << WM8993_DCS_SERIES_NO_01_SHIFT); + + /* Enable the DC servo. Write all bits to avoid triggering startup + * or write calibration. + */ + snd_soc_update_bits(codec, WM8993_DC_SERVO_0, + 0xFFFF, + WM8993_DCS_ENA_CHAN_0 | + WM8993_DCS_ENA_CHAN_1 | + WM8993_DCS_TRIG_SERIES_1 | + WM8993_DCS_TRIG_SERIES_0); + + wait_for_dc_servo(codec); + + /* Apply correction to DC servo result */ + if (hubs->dcs_codes) { + dev_dbg(codec->dev, "Applying %d code DC servo correction\n", + hubs->dcs_codes); + + /* HPOUT1L */ + reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) & + WM8993_DCS_INTEG_CHAN_0_MASK;; + reg += hubs->dcs_codes; + dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + + /* HPOUT1R */ + reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & + WM8993_DCS_INTEG_CHAN_1_MASK; + reg += hubs->dcs_codes; + dcs_cfg |= reg; + + /* Do it */ + snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); + snd_soc_update_bits(codec, WM8993_DC_SERVO_0, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1); + + wait_for_dc_servo(codec); + } +} + /* * Update the DC servo calibration on gain changes */ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int ret; @@ -251,6 +304,47 @@ SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1, line_tlv), }; +static int hp_supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = codec->private_data; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (hubs->hp_startup_mode) { + case 0: + break; + case 1: + /* Enable the headphone amp */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_HPOUT1L_ENA | + WM8993_HPOUT1R_ENA, + WM8993_HPOUT1L_ENA | + WM8993_HPOUT1R_ENA); + + /* Enable the second stage */ + snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0, + WM8993_HPOUT1L_DLY | + WM8993_HPOUT1R_DLY, + WM8993_HPOUT1L_DLY | + WM8993_HPOUT1R_DLY); + break; + default: + dev_err(codec->dev, "Unknown HP startup mode %d\n", + hubs->hp_startup_mode); + break; + } + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, + WM8993_CP_ENA, 0); + break; + } + + return 0; +} + static int hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -271,14 +365,11 @@ static int hp_event(struct snd_soc_dapm_widget *w, reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY; snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); - /* Start the DC servo */ - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - 0xFFFF, - WM8993_DCS_ENA_CHAN_0 | - WM8993_DCS_ENA_CHAN_1 | - WM8993_DCS_TRIG_STARTUP_1 | - WM8993_DCS_TRIG_STARTUP_0); - wait_for_dc_servo(codec); + /* Smallest supported update interval */ + snd_soc_update_bits(codec, WM8993_DC_SERVO_1, + WM8993_DCS_TIMER_PERIOD_01_MASK, 1); + + calibrate_dc_servo(codec); reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT | WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT; @@ -286,23 +377,19 @@ static int hp_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - reg &= ~(WM8993_HPOUT1L_RMV_SHORT | - WM8993_HPOUT1L_DLY | - WM8993_HPOUT1L_OUTP | - WM8993_HPOUT1R_RMV_SHORT | - WM8993_HPOUT1R_DLY | - WM8993_HPOUT1R_OUTP); + snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0, + WM8993_HPOUT1L_DLY | + WM8993_HPOUT1R_DLY | + WM8993_HPOUT1L_RMV_SHORT | + WM8993_HPOUT1R_RMV_SHORT, 0); - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - 0xffff, 0); + snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0, + WM8993_HPOUT1L_OUTP | + WM8993_HPOUT1R_OUTP, 0); - snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA, 0); - - snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, - WM8993_CP_ENA, 0); break; } @@ -473,6 +560,8 @@ SND_SOC_DAPM_MIXER("Right Output Mixer", WM8993_POWER_MANAGEMENT_3, 4, 0, SND_SOC_DAPM_PGA("Left Output PGA", WM8993_POWER_MANAGEMENT_3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Headphone Supply", SND_SOC_NOPM, 0, 0, hp_supply_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0, hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -626,6 +715,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, { "Headphone PGA", NULL, "CLK_SYS" }, + { "Headphone PGA", NULL, "Headphone Supply" }, { "HPOUT1L", NULL, "Headphone PGA" }, { "HPOUT1R", NULL, "Headphone PGA" }, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 36d3fba1de8b..420104fe9c90 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -18,6 +18,12 @@ struct snd_soc_codec; extern const unsigned int wm_hubs_spkmix_tlv[]; +/* This *must* be the first element of the codec->private_data struct */ +struct wm_hubs_data { + int dcs_codes; + int hp_startup_mode; +}; + extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int); extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, -- cgit v1.2.2 From be587ef4f20cb5a0e42264909fa702a24081a160 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 18:31:06 +0000 Subject: ASoC: Activate DCS correction for WM8993 Use a two code correction for optimal performance. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index bacfc2f20d70..61239e0e9556 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1,7 +1,7 @@ /* * wm8993.c -- WM8993 ALSA SoC audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009, 2010 Wolfson Microelectronics plc * * Author: Mark Brown * @@ -1598,6 +1598,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, codec->private_data = wm8993; wm8993->hubs_data.hp_startup_mode = 1; + wm8993->hubs_data.dcs_codes = -2; memcpy(wm8993->reg_cache, wm8993_reg_defaults, sizeof(wm8993->reg_cache)); -- cgit v1.2.2 From 9e6e96a197a03752d39a63e4f83e0b707ccedad7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Jan 2010 17:47:12 +0000 Subject: ASoC: Add WM8994 CODEC driver The WM8994 is a highly integrated ultra-low power hi-fi audio subsystem designed for smartphones and other portable devices rich in multimedia features. It provides advanced digital mixing facilities enabling low power high quality interconnection of CPU, baseband and other audio sources through flexible digital and analogue routing, and integrates a class W headphone driver and stereo class D speaker drivers. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8994.c | 3870 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.h | 26 + 4 files changed, 3902 insertions(+) create mode 100644 sound/soc/codecs/wm8994.c create mode 100644 sound/soc/codecs/wm8994.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0aad72fc1961..6b8a10120f9c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C + select SND_SOC_WM8994 if I2C select SND_SOC_WM9081 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS @@ -243,6 +244,9 @@ config SND_SOC_WM8990 config SND_SOC_WM8993 tristate +config SND_SOC_WM8994 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fbd290e41e9e..209dd6c7c254 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-wm8978-objs := wm8978.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8993-objs := wm8993.o +snd-soc-wm8994-objs := wm8994.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o @@ -108,6 +109,7 @@ obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o +obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c new file mode 100644 index 000000000000..5dd4b299f69e --- /dev/null +++ b/sound/soc/codecs/wm8994.c @@ -0,0 +1,3870 @@ +/* + * wm8994.c -- WM8994 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "wm8994.h" +#include "wm_hubs.h" + +static struct snd_soc_codec *wm8994_codec; +struct snd_soc_codec_device soc_codec_dev_wm8994; + +struct fll_config { + int src; + int in; + int out; +}; + +#define WM8994_NUM_DRC 3 +#define WM8994_NUM_EQ 3 + +static int wm8994_drc_base[] = { + WM8994_AIF1_DRC1_1, + WM8994_AIF1_DRC2_1, + WM8994_AIF2_DRC_1, +}; + +static int wm8994_retune_mobile_base[] = { + WM8994_AIF1_DAC1_EQ_GAINS_1, + WM8994_AIF1_DAC2_EQ_GAINS_1, + WM8994_AIF2_EQ_GAINS_1, +}; + +#define WM8994_REG_CACHE_SIZE 0x621 + +/* codec private data */ +struct wm8994_priv { + struct wm_hubs_data hubs; + struct snd_soc_codec codec; + u16 reg_cache[WM8994_REG_CACHE_SIZE + 1]; + int sysclk[2]; + int sysclk_rate[2]; + int mclk[2]; + int aifclk[2]; + struct fll_config fll[2], fll_suspend[2]; + + int dac_rates[2]; + int lrclk_shared[2]; + + /* Platform dependant DRC configuration */ + const char **drc_texts; + int drc_cfg[WM8994_NUM_DRC]; + struct soc_enum drc_enum; + + /* Platform dependant ReTune mobile configuration */ + int num_retune_mobile_texts; + const char **retune_mobile_texts; + int retune_mobile_cfg[WM8994_NUM_EQ]; + struct soc_enum retune_mobile_enum; + + struct wm8994_pdata *pdata; +}; + +static struct { + unsigned short readable; /* Mask of readable bits */ + unsigned short writable; /* Mask of writable bits */ + unsigned short vol; /* Mask of volatile bits */ +} access_masks[] = { + { 0xFFFF, 0xFFFF, 0x0000 }, /* R0 - Software Reset */ + { 0x3B37, 0x3B37, 0x0000 }, /* R1 - Power Management (1) */ + { 0x6BF0, 0x6BF0, 0x0000 }, /* R2 - Power Management (2) */ + { 0x3FF0, 0x3FF0, 0x0000 }, /* R3 - Power Management (3) */ + { 0x3F3F, 0x3F3F, 0x0000 }, /* R4 - Power Management (4) */ + { 0x3F0F, 0x3F0F, 0x0000 }, /* R5 - Power Management (5) */ + { 0x003F, 0x003F, 0x0000 }, /* R6 - Power Management (6) */ + { 0x0000, 0x0000, 0x0000 }, /* R7 */ + { 0x0000, 0x0000, 0x0000 }, /* R8 */ + { 0x0000, 0x0000, 0x0000 }, /* R9 */ + { 0x0000, 0x0000, 0x0000 }, /* R10 */ + { 0x0000, 0x0000, 0x0000 }, /* R11 */ + { 0x0000, 0x0000, 0x0000 }, /* R12 */ + { 0x0000, 0x0000, 0x0000 }, /* R13 */ + { 0x0000, 0x0000, 0x0000 }, /* R14 */ + { 0x0000, 0x0000, 0x0000 }, /* R15 */ + { 0x0000, 0x0000, 0x0000 }, /* R16 */ + { 0x0000, 0x0000, 0x0000 }, /* R17 */ + { 0x0000, 0x0000, 0x0000 }, /* R18 */ + { 0x0000, 0x0000, 0x0000 }, /* R19 */ + { 0x0000, 0x0000, 0x0000 }, /* R20 */ + { 0x01C0, 0x01C0, 0x0000 }, /* R21 - Input Mixer (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R22 */ + { 0x0000, 0x0000, 0x0000 }, /* R23 */ + { 0x00DF, 0x01DF, 0x0000 }, /* R24 - Left Line Input 1&2 Volume */ + { 0x00DF, 0x01DF, 0x0000 }, /* R25 - Left Line Input 3&4 Volume */ + { 0x00DF, 0x01DF, 0x0000 }, /* R26 - Right Line Input 1&2 Volume */ + { 0x00DF, 0x01DF, 0x0000 }, /* R27 - Right Line Input 3&4 Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R28 - Left Output Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R29 - Right Output Volume */ + { 0x0077, 0x0077, 0x0000 }, /* R30 - Line Outputs Volume */ + { 0x0030, 0x0030, 0x0000 }, /* R31 - HPOUT2 Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R32 - Left OPGA Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R33 - Right OPGA Volume */ + { 0x007F, 0x007F, 0x0000 }, /* R34 - SPKMIXL Attenuation */ + { 0x017F, 0x017F, 0x0000 }, /* R35 - SPKMIXR Attenuation */ + { 0x003F, 0x003F, 0x0000 }, /* R36 - SPKOUT Mixers */ + { 0x003F, 0x003F, 0x0000 }, /* R37 - ClassD */ + { 0x00FF, 0x01FF, 0x0000 }, /* R38 - Speaker Volume Left */ + { 0x00FF, 0x01FF, 0x0000 }, /* R39 - Speaker Volume Right */ + { 0x00FF, 0x00FF, 0x0000 }, /* R40 - Input Mixer (2) */ + { 0x01B7, 0x01B7, 0x0000 }, /* R41 - Input Mixer (3) */ + { 0x01B7, 0x01B7, 0x0000 }, /* R42 - Input Mixer (4) */ + { 0x01C7, 0x01C7, 0x0000 }, /* R43 - Input Mixer (5) */ + { 0x01C7, 0x01C7, 0x0000 }, /* R44 - Input Mixer (6) */ + { 0x01FF, 0x01FF, 0x0000 }, /* R45 - Output Mixer (1) */ + { 0x01FF, 0x01FF, 0x0000 }, /* R46 - Output Mixer (2) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R47 - Output Mixer (3) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R48 - Output Mixer (4) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R49 - Output Mixer (5) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R50 - Output Mixer (6) */ + { 0x0038, 0x0038, 0x0000 }, /* R51 - HPOUT2 Mixer */ + { 0x0077, 0x0077, 0x0000 }, /* R52 - Line Mixer (1) */ + { 0x0077, 0x0077, 0x0000 }, /* R53 - Line Mixer (2) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R54 - Speaker Mixer */ + { 0x00C1, 0x00C1, 0x0000 }, /* R55 - Additional Control */ + { 0x00F0, 0x00F0, 0x0000 }, /* R56 - AntiPOP (1) */ + { 0x01EF, 0x01EF, 0x0000 }, /* R57 - AntiPOP (2) */ + { 0x00FF, 0x00FF, 0x0000 }, /* R58 - MICBIAS */ + { 0x000F, 0x000F, 0x0000 }, /* R59 - LDO 1 */ + { 0x0007, 0x0007, 0x0000 }, /* R60 - LDO 2 */ + { 0x0000, 0x0000, 0x0000 }, /* R61 */ + { 0x0000, 0x0000, 0x0000 }, /* R62 */ + { 0x0000, 0x0000, 0x0000 }, /* R63 */ + { 0x0000, 0x0000, 0x0000 }, /* R64 */ + { 0x0000, 0x0000, 0x0000 }, /* R65 */ + { 0x0000, 0x0000, 0x0000 }, /* R66 */ + { 0x0000, 0x0000, 0x0000 }, /* R67 */ + { 0x0000, 0x0000, 0x0000 }, /* R68 */ + { 0x0000, 0x0000, 0x0000 }, /* R69 */ + { 0x0000, 0x0000, 0x0000 }, /* R70 */ + { 0x0000, 0x0000, 0x0000 }, /* R71 */ + { 0x0000, 0x0000, 0x0000 }, /* R72 */ + { 0x0000, 0x0000, 0x0000 }, /* R73 */ + { 0x0000, 0x0000, 0x0000 }, /* R74 */ + { 0x0000, 0x0000, 0x0000 }, /* R75 */ + { 0x8000, 0x8000, 0x0000 }, /* R76 - Charge Pump (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R77 */ + { 0x0000, 0x0000, 0x0000 }, /* R78 */ + { 0x0000, 0x0000, 0x0000 }, /* R79 */ + { 0x0000, 0x0000, 0x0000 }, /* R80 */ + { 0x0301, 0x0301, 0x0000 }, /* R81 - Class W (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R82 */ + { 0x0000, 0x0000, 0x0000 }, /* R83 */ + { 0x333F, 0x333F, 0x0000 }, /* R84 - DC Servo (1) */ + { 0x0FEF, 0x0FEF, 0x0000 }, /* R85 - DC Servo (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R86 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R87 - DC Servo (4) */ + { 0x0333, 0x0000, 0x0000 }, /* R88 - DC Servo Readback */ + { 0x0000, 0x0000, 0x0000 }, /* R89 */ + { 0x0000, 0x0000, 0x0000 }, /* R90 */ + { 0x0000, 0x0000, 0x0000 }, /* R91 */ + { 0x0000, 0x0000, 0x0000 }, /* R92 */ + { 0x0000, 0x0000, 0x0000 }, /* R93 */ + { 0x0000, 0x0000, 0x0000 }, /* R94 */ + { 0x0000, 0x0000, 0x0000 }, /* R95 */ + { 0x00EE, 0x00EE, 0x0000 }, /* R96 - Analogue HP (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R97 */ + { 0x0000, 0x0000, 0x0000 }, /* R98 */ + { 0x0000, 0x0000, 0x0000 }, /* R99 */ + { 0x0000, 0x0000, 0x0000 }, /* R100 */ + { 0x0000, 0x0000, 0x0000 }, /* R101 */ + { 0x0000, 0x0000, 0x0000 }, /* R102 */ + { 0x0000, 0x0000, 0x0000 }, /* R103 */ + { 0x0000, 0x0000, 0x0000 }, /* R104 */ + { 0x0000, 0x0000, 0x0000 }, /* R105 */ + { 0x0000, 0x0000, 0x0000 }, /* R106 */ + { 0x0000, 0x0000, 0x0000 }, /* R107 */ + { 0x0000, 0x0000, 0x0000 }, /* R108 */ + { 0x0000, 0x0000, 0x0000 }, /* R109 */ + { 0x0000, 0x0000, 0x0000 }, /* R110 */ + { 0x0000, 0x0000, 0x0000 }, /* R111 */ + { 0x0000, 0x0000, 0x0000 }, /* R112 */ + { 0x0000, 0x0000, 0x0000 }, /* R113 */ + { 0x0000, 0x0000, 0x0000 }, /* R114 */ + { 0x0000, 0x0000, 0x0000 }, /* R115 */ + { 0x0000, 0x0000, 0x0000 }, /* R116 */ + { 0x0000, 0x0000, 0x0000 }, /* R117 */ + { 0x0000, 0x0000, 0x0000 }, /* R118 */ + { 0x0000, 0x0000, 0x0000 }, /* R119 */ + { 0x0000, 0x0000, 0x0000 }, /* R120 */ + { 0x0000, 0x0000, 0x0000 }, /* R121 */ + { 0x0000, 0x0000, 0x0000 }, /* R122 */ + { 0x0000, 0x0000, 0x0000 }, /* R123 */ + { 0x0000, 0x0000, 0x0000 }, /* R124 */ + { 0x0000, 0x0000, 0x0000 }, /* R125 */ + { 0x0000, 0x0000, 0x0000 }, /* R126 */ + { 0x0000, 0x0000, 0x0000 }, /* R127 */ + { 0x0000, 0x0000, 0x0000 }, /* R128 */ + { 0x0000, 0x0000, 0x0000 }, /* R129 */ + { 0x0000, 0x0000, 0x0000 }, /* R130 */ + { 0x0000, 0x0000, 0x0000 }, /* R131 */ + { 0x0000, 0x0000, 0x0000 }, /* R132 */ + { 0x0000, 0x0000, 0x0000 }, /* R133 */ + { 0x0000, 0x0000, 0x0000 }, /* R134 */ + { 0x0000, 0x0000, 0x0000 }, /* R135 */ + { 0x0000, 0x0000, 0x0000 }, /* R136 */ + { 0x0000, 0x0000, 0x0000 }, /* R137 */ + { 0x0000, 0x0000, 0x0000 }, /* R138 */ + { 0x0000, 0x0000, 0x0000 }, /* R139 */ + { 0x0000, 0x0000, 0x0000 }, /* R140 */ + { 0x0000, 0x0000, 0x0000 }, /* R141 */ + { 0x0000, 0x0000, 0x0000 }, /* R142 */ + { 0x0000, 0x0000, 0x0000 }, /* R143 */ + { 0x0000, 0x0000, 0x0000 }, /* R144 */ + { 0x0000, 0x0000, 0x0000 }, /* R145 */ + { 0x0000, 0x0000, 0x0000 }, /* R146 */ + { 0x0000, 0x0000, 0x0000 }, /* R147 */ + { 0x0000, 0x0000, 0x0000 }, /* R148 */ + { 0x0000, 0x0000, 0x0000 }, /* R149 */ + { 0x0000, 0x0000, 0x0000 }, /* R150 */ + { 0x0000, 0x0000, 0x0000 }, /* R151 */ + { 0x0000, 0x0000, 0x0000 }, /* R152 */ + { 0x0000, 0x0000, 0x0000 }, /* R153 */ + { 0x0000, 0x0000, 0x0000 }, /* R154 */ + { 0x0000, 0x0000, 0x0000 }, /* R155 */ + { 0x0000, 0x0000, 0x0000 }, /* R156 */ + { 0x0000, 0x0000, 0x0000 }, /* R157 */ + { 0x0000, 0x0000, 0x0000 }, /* R158 */ + { 0x0000, 0x0000, 0x0000 }, /* R159 */ + { 0x0000, 0x0000, 0x0000 }, /* R160 */ + { 0x0000, 0x0000, 0x0000 }, /* R161 */ + { 0x0000, 0x0000, 0x0000 }, /* R162 */ + { 0x0000, 0x0000, 0x0000 }, /* R163 */ + { 0x0000, 0x0000, 0x0000 }, /* R164 */ + { 0x0000, 0x0000, 0x0000 }, /* R165 */ + { 0x0000, 0x0000, 0x0000 }, /* R166 */ + { 0x0000, 0x0000, 0x0000 }, /* R167 */ + { 0x0000, 0x0000, 0x0000 }, /* R168 */ + { 0x0000, 0x0000, 0x0000 }, /* R169 */ + { 0x0000, 0x0000, 0x0000 }, /* R170 */ + { 0x0000, 0x0000, 0x0000 }, /* R171 */ + { 0x0000, 0x0000, 0x0000 }, /* R172 */ + { 0x0000, 0x0000, 0x0000 }, /* R173 */ + { 0x0000, 0x0000, 0x0000 }, /* R174 */ + { 0x0000, 0x0000, 0x0000 }, /* R175 */ + { 0x0000, 0x0000, 0x0000 }, /* R176 */ + { 0x0000, 0x0000, 0x0000 }, /* R177 */ + { 0x0000, 0x0000, 0x0000 }, /* R178 */ + { 0x0000, 0x0000, 0x0000 }, /* R179 */ + { 0x0000, 0x0000, 0x0000 }, /* R180 */ + { 0x0000, 0x0000, 0x0000 }, /* R181 */ + { 0x0000, 0x0000, 0x0000 }, /* R182 */ + { 0x0000, 0x0000, 0x0000 }, /* R183 */ + { 0x0000, 0x0000, 0x0000 }, /* R184 */ + { 0x0000, 0x0000, 0x0000 }, /* R185 */ + { 0x0000, 0x0000, 0x0000 }, /* R186 */ + { 0x0000, 0x0000, 0x0000 }, /* R187 */ + { 0x0000, 0x0000, 0x0000 }, /* R188 */ + { 0x0000, 0x0000, 0x0000 }, /* R189 */ + { 0x0000, 0x0000, 0x0000 }, /* R190 */ + { 0x0000, 0x0000, 0x0000 }, /* R191 */ + { 0x0000, 0x0000, 0x0000 }, /* R192 */ + { 0x0000, 0x0000, 0x0000 }, /* R193 */ + { 0x0000, 0x0000, 0x0000 }, /* R194 */ + { 0x0000, 0x0000, 0x0000 }, /* R195 */ + { 0x0000, 0x0000, 0x0000 }, /* R196 */ + { 0x0000, 0x0000, 0x0000 }, /* R197 */ + { 0x0000, 0x0000, 0x0000 }, /* R198 */ + { 0x0000, 0x0000, 0x0000 }, /* R199 */ + { 0x0000, 0x0000, 0x0000 }, /* R200 */ + { 0x0000, 0x0000, 0x0000 }, /* R201 */ + { 0x0000, 0x0000, 0x0000 }, /* R202 */ + { 0x0000, 0x0000, 0x0000 }, /* R203 */ + { 0x0000, 0x0000, 0x0000 }, /* R204 */ + { 0x0000, 0x0000, 0x0000 }, /* R205 */ + { 0x0000, 0x0000, 0x0000 }, /* R206 */ + { 0x0000, 0x0000, 0x0000 }, /* R207 */ + { 0x0000, 0x0000, 0x0000 }, /* R208 */ + { 0x0000, 0x0000, 0x0000 }, /* R209 */ + { 0x0000, 0x0000, 0x0000 }, /* R210 */ + { 0x0000, 0x0000, 0x0000 }, /* R211 */ + { 0x0000, 0x0000, 0x0000 }, /* R212 */ + { 0x0000, 0x0000, 0x0000 }, /* R213 */ + { 0x0000, 0x0000, 0x0000 }, /* R214 */ + { 0x0000, 0x0000, 0x0000 }, /* R215 */ + { 0x0000, 0x0000, 0x0000 }, /* R216 */ + { 0x0000, 0x0000, 0x0000 }, /* R217 */ + { 0x0000, 0x0000, 0x0000 }, /* R218 */ + { 0x0000, 0x0000, 0x0000 }, /* R219 */ + { 0x0000, 0x0000, 0x0000 }, /* R220 */ + { 0x0000, 0x0000, 0x0000 }, /* R221 */ + { 0x0000, 0x0000, 0x0000 }, /* R222 */ + { 0x0000, 0x0000, 0x0000 }, /* R223 */ + { 0x0000, 0x0000, 0x0000 }, /* R224 */ + { 0x0000, 0x0000, 0x0000 }, /* R225 */ + { 0x0000, 0x0000, 0x0000 }, /* R226 */ + { 0x0000, 0x0000, 0x0000 }, /* R227 */ + { 0x0000, 0x0000, 0x0000 }, /* R228 */ + { 0x0000, 0x0000, 0x0000 }, /* R229 */ + { 0x0000, 0x0000, 0x0000 }, /* R230 */ + { 0x0000, 0x0000, 0x0000 }, /* R231 */ + { 0x0000, 0x0000, 0x0000 }, /* R232 */ + { 0x0000, 0x0000, 0x0000 }, /* R233 */ + { 0x0000, 0x0000, 0x0000 }, /* R234 */ + { 0x0000, 0x0000, 0x0000 }, /* R235 */ + { 0x0000, 0x0000, 0x0000 }, /* R236 */ + { 0x0000, 0x0000, 0x0000 }, /* R237 */ + { 0x0000, 0x0000, 0x0000 }, /* R238 */ + { 0x0000, 0x0000, 0x0000 }, /* R239 */ + { 0x0000, 0x0000, 0x0000 }, /* R240 */ + { 0x0000, 0x0000, 0x0000 }, /* R241 */ + { 0x0000, 0x0000, 0x0000 }, /* R242 */ + { 0x0000, 0x0000, 0x0000 }, /* R243 */ + { 0x0000, 0x0000, 0x0000 }, /* R244 */ + { 0x0000, 0x0000, 0x0000 }, /* R245 */ + { 0x0000, 0x0000, 0x0000 }, /* R246 */ + { 0x0000, 0x0000, 0x0000 }, /* R247 */ + { 0x0000, 0x0000, 0x0000 }, /* R248 */ + { 0x0000, 0x0000, 0x0000 }, /* R249 */ + { 0x0000, 0x0000, 0x0000 }, /* R250 */ + { 0x0000, 0x0000, 0x0000 }, /* R251 */ + { 0x0000, 0x0000, 0x0000 }, /* R252 */ + { 0x0000, 0x0000, 0x0000 }, /* R253 */ + { 0x0000, 0x0000, 0x0000 }, /* R254 */ + { 0x0000, 0x0000, 0x0000 }, /* R255 */ + { 0x000F, 0x0000, 0x0000 }, /* R256 - Chip Revision */ + { 0x0074, 0x0074, 0x0000 }, /* R257 - Control Interface */ + { 0x0000, 0x0000, 0x0000 }, /* R258 */ + { 0x0000, 0x0000, 0x0000 }, /* R259 */ + { 0x0000, 0x0000, 0x0000 }, /* R260 */ + { 0x0000, 0x0000, 0x0000 }, /* R261 */ + { 0x0000, 0x0000, 0x0000 }, /* R262 */ + { 0x0000, 0x0000, 0x0000 }, /* R263 */ + { 0x0000, 0x0000, 0x0000 }, /* R264 */ + { 0x0000, 0x0000, 0x0000 }, /* R265 */ + { 0x0000, 0x0000, 0x0000 }, /* R266 */ + { 0x0000, 0x0000, 0x0000 }, /* R267 */ + { 0x0000, 0x0000, 0x0000 }, /* R268 */ + { 0x0000, 0x0000, 0x0000 }, /* R269 */ + { 0x0000, 0x0000, 0x0000 }, /* R270 */ + { 0x0000, 0x0000, 0x0000 }, /* R271 */ + { 0x807F, 0x837F, 0x0000 }, /* R272 - Write Sequencer Ctrl (1) */ + { 0x017F, 0x0000, 0x0000 }, /* R273 - Write Sequencer Ctrl (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R274 */ + { 0x0000, 0x0000, 0x0000 }, /* R275 */ + { 0x0000, 0x0000, 0x0000 }, /* R276 */ + { 0x0000, 0x0000, 0x0000 }, /* R277 */ + { 0x0000, 0x0000, 0x0000 }, /* R278 */ + { 0x0000, 0x0000, 0x0000 }, /* R279 */ + { 0x0000, 0x0000, 0x0000 }, /* R280 */ + { 0x0000, 0x0000, 0x0000 }, /* R281 */ + { 0x0000, 0x0000, 0x0000 }, /* R282 */ + { 0x0000, 0x0000, 0x0000 }, /* R283 */ + { 0x0000, 0x0000, 0x0000 }, /* R284 */ + { 0x0000, 0x0000, 0x0000 }, /* R285 */ + { 0x0000, 0x0000, 0x0000 }, /* R286 */ + { 0x0000, 0x0000, 0x0000 }, /* R287 */ + { 0x0000, 0x0000, 0x0000 }, /* R288 */ + { 0x0000, 0x0000, 0x0000 }, /* R289 */ + { 0x0000, 0x0000, 0x0000 }, /* R290 */ + { 0x0000, 0x0000, 0x0000 }, /* R291 */ + { 0x0000, 0x0000, 0x0000 }, /* R292 */ + { 0x0000, 0x0000, 0x0000 }, /* R293 */ + { 0x0000, 0x0000, 0x0000 }, /* R294 */ + { 0x0000, 0x0000, 0x0000 }, /* R295 */ + { 0x0000, 0x0000, 0x0000 }, /* R296 */ + { 0x0000, 0x0000, 0x0000 }, /* R297 */ + { 0x0000, 0x0000, 0x0000 }, /* R298 */ + { 0x0000, 0x0000, 0x0000 }, /* R299 */ + { 0x0000, 0x0000, 0x0000 }, /* R300 */ + { 0x0000, 0x0000, 0x0000 }, /* R301 */ + { 0x0000, 0x0000, 0x0000 }, /* R302 */ + { 0x0000, 0x0000, 0x0000 }, /* R303 */ + { 0x0000, 0x0000, 0x0000 }, /* R304 */ + { 0x0000, 0x0000, 0x0000 }, /* R305 */ + { 0x0000, 0x0000, 0x0000 }, /* R306 */ + { 0x0000, 0x0000, 0x0000 }, /* R307 */ + { 0x0000, 0x0000, 0x0000 }, /* R308 */ + { 0x0000, 0x0000, 0x0000 }, /* R309 */ + { 0x0000, 0x0000, 0x0000 }, /* R310 */ + { 0x0000, 0x0000, 0x0000 }, /* R311 */ + { 0x0000, 0x0000, 0x0000 }, /* R312 */ + { 0x0000, 0x0000, 0x0000 }, /* R313 */ + { 0x0000, 0x0000, 0x0000 }, /* R314 */ + { 0x0000, 0x0000, 0x0000 }, /* R315 */ + { 0x0000, 0x0000, 0x0000 }, /* R316 */ + { 0x0000, 0x0000, 0x0000 }, /* R317 */ + { 0x0000, 0x0000, 0x0000 }, /* R318 */ + { 0x0000, 0x0000, 0x0000 }, /* R319 */ + { 0x0000, 0x0000, 0x0000 }, /* R320 */ + { 0x0000, 0x0000, 0x0000 }, /* R321 */ + { 0x0000, 0x0000, 0x0000 }, /* R322 */ + { 0x0000, 0x0000, 0x0000 }, /* R323 */ + { 0x0000, 0x0000, 0x0000 }, /* R324 */ + { 0x0000, 0x0000, 0x0000 }, /* R325 */ + { 0x0000, 0x0000, 0x0000 }, /* R326 */ + { 0x0000, 0x0000, 0x0000 }, /* R327 */ + { 0x0000, 0x0000, 0x0000 }, /* R328 */ + { 0x0000, 0x0000, 0x0000 }, /* R329 */ + { 0x0000, 0x0000, 0x0000 }, /* R330 */ + { 0x0000, 0x0000, 0x0000 }, /* R331 */ + { 0x0000, 0x0000, 0x0000 }, /* R332 */ + { 0x0000, 0x0000, 0x0000 }, /* R333 */ + { 0x0000, 0x0000, 0x0000 }, /* R334 */ + { 0x0000, 0x0000, 0x0000 }, /* R335 */ + { 0x0000, 0x0000, 0x0000 }, /* R336 */ + { 0x0000, 0x0000, 0x0000 }, /* R337 */ + { 0x0000, 0x0000, 0x0000 }, /* R338 */ + { 0x0000, 0x0000, 0x0000 }, /* R339 */ + { 0x0000, 0x0000, 0x0000 }, /* R340 */ + { 0x0000, 0x0000, 0x0000 }, /* R341 */ + { 0x0000, 0x0000, 0x0000 }, /* R342 */ + { 0x0000, 0x0000, 0x0000 }, /* R343 */ + { 0x0000, 0x0000, 0x0000 }, /* R344 */ + { 0x0000, 0x0000, 0x0000 }, /* R345 */ + { 0x0000, 0x0000, 0x0000 }, /* R346 */ + { 0x0000, 0x0000, 0x0000 }, /* R347 */ + { 0x0000, 0x0000, 0x0000 }, /* R348 */ + { 0x0000, 0x0000, 0x0000 }, /* R349 */ + { 0x0000, 0x0000, 0x0000 }, /* R350 */ + { 0x0000, 0x0000, 0x0000 }, /* R351 */ + { 0x0000, 0x0000, 0x0000 }, /* R352 */ + { 0x0000, 0x0000, 0x0000 }, /* R353 */ + { 0x0000, 0x0000, 0x0000 }, /* R354 */ + { 0x0000, 0x0000, 0x0000 }, /* R355 */ + { 0x0000, 0x0000, 0x0000 }, /* R356 */ + { 0x0000, 0x0000, 0x0000 }, /* R357 */ + { 0x0000, 0x0000, 0x0000 }, /* R358 */ + { 0x0000, 0x0000, 0x0000 }, /* R359 */ + { 0x0000, 0x0000, 0x0000 }, /* R360 */ + { 0x0000, 0x0000, 0x0000 }, /* R361 */ + { 0x0000, 0x0000, 0x0000 }, /* R362 */ + { 0x0000, 0x0000, 0x0000 }, /* R363 */ + { 0x0000, 0x0000, 0x0000 }, /* R364 */ + { 0x0000, 0x0000, 0x0000 }, /* R365 */ + { 0x0000, 0x0000, 0x0000 }, /* R366 */ + { 0x0000, 0x0000, 0x0000 }, /* R367 */ + { 0x0000, 0x0000, 0x0000 }, /* R368 */ + { 0x0000, 0x0000, 0x0000 }, /* R369 */ + { 0x0000, 0x0000, 0x0000 }, /* R370 */ + { 0x0000, 0x0000, 0x0000 }, /* R371 */ + { 0x0000, 0x0000, 0x0000 }, /* R372 */ + { 0x0000, 0x0000, 0x0000 }, /* R373 */ + { 0x0000, 0x0000, 0x0000 }, /* R374 */ + { 0x0000, 0x0000, 0x0000 }, /* R375 */ + { 0x0000, 0x0000, 0x0000 }, /* R376 */ + { 0x0000, 0x0000, 0x0000 }, /* R377 */ + { 0x0000, 0x0000, 0x0000 }, /* R378 */ + { 0x0000, 0x0000, 0x0000 }, /* R379 */ + { 0x0000, 0x0000, 0x0000 }, /* R380 */ + { 0x0000, 0x0000, 0x0000 }, /* R381 */ + { 0x0000, 0x0000, 0x0000 }, /* R382 */ + { 0x0000, 0x0000, 0x0000 }, /* R383 */ + { 0x0000, 0x0000, 0x0000 }, /* R384 */ + { 0x0000, 0x0000, 0x0000 }, /* R385 */ + { 0x0000, 0x0000, 0x0000 }, /* R386 */ + { 0x0000, 0x0000, 0x0000 }, /* R387 */ + { 0x0000, 0x0000, 0x0000 }, /* R388 */ + { 0x0000, 0x0000, 0x0000 }, /* R389 */ + { 0x0000, 0x0000, 0x0000 }, /* R390 */ + { 0x0000, 0x0000, 0x0000 }, /* R391 */ + { 0x0000, 0x0000, 0x0000 }, /* R392 */ + { 0x0000, 0x0000, 0x0000 }, /* R393 */ + { 0x0000, 0x0000, 0x0000 }, /* R394 */ + { 0x0000, 0x0000, 0x0000 }, /* R395 */ + { 0x0000, 0x0000, 0x0000 }, /* R396 */ + { 0x0000, 0x0000, 0x0000 }, /* R397 */ + { 0x0000, 0x0000, 0x0000 }, /* R398 */ + { 0x0000, 0x0000, 0x0000 }, /* R399 */ + { 0x0000, 0x0000, 0x0000 }, /* R400 */ + { 0x0000, 0x0000, 0x0000 }, /* R401 */ + { 0x0000, 0x0000, 0x0000 }, /* R402 */ + { 0x0000, 0x0000, 0x0000 }, /* R403 */ + { 0x0000, 0x0000, 0x0000 }, /* R404 */ + { 0x0000, 0x0000, 0x0000 }, /* R405 */ + { 0x0000, 0x0000, 0x0000 }, /* R406 */ + { 0x0000, 0x0000, 0x0000 }, /* R407 */ + { 0x0000, 0x0000, 0x0000 }, /* R408 */ + { 0x0000, 0x0000, 0x0000 }, /* R409 */ + { 0x0000, 0x0000, 0x0000 }, /* R410 */ + { 0x0000, 0x0000, 0x0000 }, /* R411 */ + { 0x0000, 0x0000, 0x0000 }, /* R412 */ + { 0x0000, 0x0000, 0x0000 }, /* R413 */ + { 0x0000, 0x0000, 0x0000 }, /* R414 */ + { 0x0000, 0x0000, 0x0000 }, /* R415 */ + { 0x0000, 0x0000, 0x0000 }, /* R416 */ + { 0x0000, 0x0000, 0x0000 }, /* R417 */ + { 0x0000, 0x0000, 0x0000 }, /* R418 */ + { 0x0000, 0x0000, 0x0000 }, /* R419 */ + { 0x0000, 0x0000, 0x0000 }, /* R420 */ + { 0x0000, 0x0000, 0x0000 }, /* R421 */ + { 0x0000, 0x0000, 0x0000 }, /* R422 */ + { 0x0000, 0x0000, 0x0000 }, /* R423 */ + { 0x0000, 0x0000, 0x0000 }, /* R424 */ + { 0x0000, 0x0000, 0x0000 }, /* R425 */ + { 0x0000, 0x0000, 0x0000 }, /* R426 */ + { 0x0000, 0x0000, 0x0000 }, /* R427 */ + { 0x0000, 0x0000, 0x0000 }, /* R428 */ + { 0x0000, 0x0000, 0x0000 }, /* R429 */ + { 0x0000, 0x0000, 0x0000 }, /* R430 */ + { 0x0000, 0x0000, 0x0000 }, /* R431 */ + { 0x0000, 0x0000, 0x0000 }, /* R432 */ + { 0x0000, 0x0000, 0x0000 }, /* R433 */ + { 0x0000, 0x0000, 0x0000 }, /* R434 */ + { 0x0000, 0x0000, 0x0000 }, /* R435 */ + { 0x0000, 0x0000, 0x0000 }, /* R436 */ + { 0x0000, 0x0000, 0x0000 }, /* R437 */ + { 0x0000, 0x0000, 0x0000 }, /* R438 */ + { 0x0000, 0x0000, 0x0000 }, /* R439 */ + { 0x0000, 0x0000, 0x0000 }, /* R440 */ + { 0x0000, 0x0000, 0x0000 }, /* R441 */ + { 0x0000, 0x0000, 0x0000 }, /* R442 */ + { 0x0000, 0x0000, 0x0000 }, /* R443 */ + { 0x0000, 0x0000, 0x0000 }, /* R444 */ + { 0x0000, 0x0000, 0x0000 }, /* R445 */ + { 0x0000, 0x0000, 0x0000 }, /* R446 */ + { 0x0000, 0x0000, 0x0000 }, /* R447 */ + { 0x0000, 0x0000, 0x0000 }, /* R448 */ + { 0x0000, 0x0000, 0x0000 }, /* R449 */ + { 0x0000, 0x0000, 0x0000 }, /* R450 */ + { 0x0000, 0x0000, 0x0000 }, /* R451 */ + { 0x0000, 0x0000, 0x0000 }, /* R452 */ + { 0x0000, 0x0000, 0x0000 }, /* R453 */ + { 0x0000, 0x0000, 0x0000 }, /* R454 */ + { 0x0000, 0x0000, 0x0000 }, /* R455 */ + { 0x0000, 0x0000, 0x0000 }, /* R456 */ + { 0x0000, 0x0000, 0x0000 }, /* R457 */ + { 0x0000, 0x0000, 0x0000 }, /* R458 */ + { 0x0000, 0x0000, 0x0000 }, /* R459 */ + { 0x0000, 0x0000, 0x0000 }, /* R460 */ + { 0x0000, 0x0000, 0x0000 }, /* R461 */ + { 0x0000, 0x0000, 0x0000 }, /* R462 */ + { 0x0000, 0x0000, 0x0000 }, /* R463 */ + { 0x0000, 0x0000, 0x0000 }, /* R464 */ + { 0x0000, 0x0000, 0x0000 }, /* R465 */ + { 0x0000, 0x0000, 0x0000 }, /* R466 */ + { 0x0000, 0x0000, 0x0000 }, /* R467 */ + { 0x0000, 0x0000, 0x0000 }, /* R468 */ + { 0x0000, 0x0000, 0x0000 }, /* R469 */ + { 0x0000, 0x0000, 0x0000 }, /* R470 */ + { 0x0000, 0x0000, 0x0000 }, /* R471 */ + { 0x0000, 0x0000, 0x0000 }, /* R472 */ + { 0x0000, 0x0000, 0x0000 }, /* R473 */ + { 0x0000, 0x0000, 0x0000 }, /* R474 */ + { 0x0000, 0x0000, 0x0000 }, /* R475 */ + { 0x0000, 0x0000, 0x0000 }, /* R476 */ + { 0x0000, 0x0000, 0x0000 }, /* R477 */ + { 0x0000, 0x0000, 0x0000 }, /* R478 */ + { 0x0000, 0x0000, 0x0000 }, /* R479 */ + { 0x0000, 0x0000, 0x0000 }, /* R480 */ + { 0x0000, 0x0000, 0x0000 }, /* R481 */ + { 0x0000, 0x0000, 0x0000 }, /* R482 */ + { 0x0000, 0x0000, 0x0000 }, /* R483 */ + { 0x0000, 0x0000, 0x0000 }, /* R484 */ + { 0x0000, 0x0000, 0x0000 }, /* R485 */ + { 0x0000, 0x0000, 0x0000 }, /* R486 */ + { 0x0000, 0x0000, 0x0000 }, /* R487 */ + { 0x0000, 0x0000, 0x0000 }, /* R488 */ + { 0x0000, 0x0000, 0x0000 }, /* R489 */ + { 0x0000, 0x0000, 0x0000 }, /* R490 */ + { 0x0000, 0x0000, 0x0000 }, /* R491 */ + { 0x0000, 0x0000, 0x0000 }, /* R492 */ + { 0x0000, 0x0000, 0x0000 }, /* R493 */ + { 0x0000, 0x0000, 0x0000 }, /* R494 */ + { 0x0000, 0x0000, 0x0000 }, /* R495 */ + { 0x0000, 0x0000, 0x0000 }, /* R496 */ + { 0x0000, 0x0000, 0x0000 }, /* R497 */ + { 0x0000, 0x0000, 0x0000 }, /* R498 */ + { 0x0000, 0x0000, 0x0000 }, /* R499 */ + { 0x0000, 0x0000, 0x0000 }, /* R500 */ + { 0x0000, 0x0000, 0x0000 }, /* R501 */ + { 0x0000, 0x0000, 0x0000 }, /* R502 */ + { 0x0000, 0x0000, 0x0000 }, /* R503 */ + { 0x0000, 0x0000, 0x0000 }, /* R504 */ + { 0x0000, 0x0000, 0x0000 }, /* R505 */ + { 0x0000, 0x0000, 0x0000 }, /* R506 */ + { 0x0000, 0x0000, 0x0000 }, /* R507 */ + { 0x0000, 0x0000, 0x0000 }, /* R508 */ + { 0x0000, 0x0000, 0x0000 }, /* R509 */ + { 0x0000, 0x0000, 0x0000 }, /* R510 */ + { 0x0000, 0x0000, 0x0000 }, /* R511 */ + { 0x001F, 0x001F, 0x0000 }, /* R512 - AIF1 Clocking (1) */ + { 0x003F, 0x003F, 0x0000 }, /* R513 - AIF1 Clocking (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R514 */ + { 0x0000, 0x0000, 0x0000 }, /* R515 */ + { 0x001F, 0x001F, 0x0000 }, /* R516 - AIF2 Clocking (1) */ + { 0x003F, 0x003F, 0x0000 }, /* R517 - AIF2 Clocking (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R518 */ + { 0x0000, 0x0000, 0x0000 }, /* R519 */ + { 0x001F, 0x001F, 0x0000 }, /* R520 - Clocking (1) */ + { 0x0777, 0x0777, 0x0000 }, /* R521 - Clocking (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R522 */ + { 0x0000, 0x0000, 0x0000 }, /* R523 */ + { 0x0000, 0x0000, 0x0000 }, /* R524 */ + { 0x0000, 0x0000, 0x0000 }, /* R525 */ + { 0x0000, 0x0000, 0x0000 }, /* R526 */ + { 0x0000, 0x0000, 0x0000 }, /* R527 */ + { 0x00FF, 0x00FF, 0x0000 }, /* R528 - AIF1 Rate */ + { 0x00FF, 0x00FF, 0x0000 }, /* R529 - AIF2 Rate */ + { 0x000F, 0x0000, 0x0000 }, /* R530 - Rate Status */ + { 0x0000, 0x0000, 0x0000 }, /* R531 */ + { 0x0000, 0x0000, 0x0000 }, /* R532 */ + { 0x0000, 0x0000, 0x0000 }, /* R533 */ + { 0x0000, 0x0000, 0x0000 }, /* R534 */ + { 0x0000, 0x0000, 0x0000 }, /* R535 */ + { 0x0000, 0x0000, 0x0000 }, /* R536 */ + { 0x0000, 0x0000, 0x0000 }, /* R537 */ + { 0x0000, 0x0000, 0x0000 }, /* R538 */ + { 0x0000, 0x0000, 0x0000 }, /* R539 */ + { 0x0000, 0x0000, 0x0000 }, /* R540 */ + { 0x0000, 0x0000, 0x0000 }, /* R541 */ + { 0x0000, 0x0000, 0x0000 }, /* R542 */ + { 0x0000, 0x0000, 0x0000 }, /* R543 */ + { 0x0007, 0x0007, 0x0000 }, /* R544 - FLL1 Control (1) */ + { 0x3F77, 0x3F77, 0x0000 }, /* R545 - FLL1 Control (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R546 - FLL1 Control (3) */ + { 0x7FEF, 0x7FEF, 0x0000 }, /* R547 - FLL1 Control (4) */ + { 0x1FDB, 0x1FDB, 0x0000 }, /* R548 - FLL1 Control (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R549 */ + { 0x0000, 0x0000, 0x0000 }, /* R550 */ + { 0x0000, 0x0000, 0x0000 }, /* R551 */ + { 0x0000, 0x0000, 0x0000 }, /* R552 */ + { 0x0000, 0x0000, 0x0000 }, /* R553 */ + { 0x0000, 0x0000, 0x0000 }, /* R554 */ + { 0x0000, 0x0000, 0x0000 }, /* R555 */ + { 0x0000, 0x0000, 0x0000 }, /* R556 */ + { 0x0000, 0x0000, 0x0000 }, /* R557 */ + { 0x0000, 0x0000, 0x0000 }, /* R558 */ + { 0x0000, 0x0000, 0x0000 }, /* R559 */ + { 0x0000, 0x0000, 0x0000 }, /* R560 */ + { 0x0000, 0x0000, 0x0000 }, /* R561 */ + { 0x0000, 0x0000, 0x0000 }, /* R562 */ + { 0x0000, 0x0000, 0x0000 }, /* R563 */ + { 0x0000, 0x0000, 0x0000 }, /* R564 */ + { 0x0000, 0x0000, 0x0000 }, /* R565 */ + { 0x0000, 0x0000, 0x0000 }, /* R566 */ + { 0x0000, 0x0000, 0x0000 }, /* R567 */ + { 0x0000, 0x0000, 0x0000 }, /* R568 */ + { 0x0000, 0x0000, 0x0000 }, /* R569 */ + { 0x0000, 0x0000, 0x0000 }, /* R570 */ + { 0x0000, 0x0000, 0x0000 }, /* R571 */ + { 0x0000, 0x0000, 0x0000 }, /* R572 */ + { 0x0000, 0x0000, 0x0000 }, /* R573 */ + { 0x0000, 0x0000, 0x0000 }, /* R574 */ + { 0x0000, 0x0000, 0x0000 }, /* R575 */ + { 0x0007, 0x0007, 0x0000 }, /* R576 - FLL2 Control (1) */ + { 0x3F77, 0x3F77, 0x0000 }, /* R577 - FLL2 Control (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R578 - FLL2 Control (3) */ + { 0x7FEF, 0x7FEF, 0x0000 }, /* R579 - FLL2 Control (4) */ + { 0x1FDB, 0x1FDB, 0x0000 }, /* R580 - FLL2 Control (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R581 */ + { 0x0000, 0x0000, 0x0000 }, /* R582 */ + { 0x0000, 0x0000, 0x0000 }, /* R583 */ + { 0x0000, 0x0000, 0x0000 }, /* R584 */ + { 0x0000, 0x0000, 0x0000 }, /* R585 */ + { 0x0000, 0x0000, 0x0000 }, /* R586 */ + { 0x0000, 0x0000, 0x0000 }, /* R587 */ + { 0x0000, 0x0000, 0x0000 }, /* R588 */ + { 0x0000, 0x0000, 0x0000 }, /* R589 */ + { 0x0000, 0x0000, 0x0000 }, /* R590 */ + { 0x0000, 0x0000, 0x0000 }, /* R591 */ + { 0x0000, 0x0000, 0x0000 }, /* R592 */ + { 0x0000, 0x0000, 0x0000 }, /* R593 */ + { 0x0000, 0x0000, 0x0000 }, /* R594 */ + { 0x0000, 0x0000, 0x0000 }, /* R595 */ + { 0x0000, 0x0000, 0x0000 }, /* R596 */ + { 0x0000, 0x0000, 0x0000 }, /* R597 */ + { 0x0000, 0x0000, 0x0000 }, /* R598 */ + { 0x0000, 0x0000, 0x0000 }, /* R599 */ + { 0x0000, 0x0000, 0x0000 }, /* R600 */ + { 0x0000, 0x0000, 0x0000 }, /* R601 */ + { 0x0000, 0x0000, 0x0000 }, /* R602 */ + { 0x0000, 0x0000, 0x0000 }, /* R603 */ + { 0x0000, 0x0000, 0x0000 }, /* R604 */ + { 0x0000, 0x0000, 0x0000 }, /* R605 */ + { 0x0000, 0x0000, 0x0000 }, /* R606 */ + { 0x0000, 0x0000, 0x0000 }, /* R607 */ + { 0x0000, 0x0000, 0x0000 }, /* R608 */ + { 0x0000, 0x0000, 0x0000 }, /* R609 */ + { 0x0000, 0x0000, 0x0000 }, /* R610 */ + { 0x0000, 0x0000, 0x0000 }, /* R611 */ + { 0x0000, 0x0000, 0x0000 }, /* R612 */ + { 0x0000, 0x0000, 0x0000 }, /* R613 */ + { 0x0000, 0x0000, 0x0000 }, /* R614 */ + { 0x0000, 0x0000, 0x0000 }, /* R615 */ + { 0x0000, 0x0000, 0x0000 }, /* R616 */ + { 0x0000, 0x0000, 0x0000 }, /* R617 */ + { 0x0000, 0x0000, 0x0000 }, /* R618 */ + { 0x0000, 0x0000, 0x0000 }, /* R619 */ + { 0x0000, 0x0000, 0x0000 }, /* R620 */ + { 0x0000, 0x0000, 0x0000 }, /* R621 */ + { 0x0000, 0x0000, 0x0000 }, /* R622 */ + { 0x0000, 0x0000, 0x0000 }, /* R623 */ + { 0x0000, 0x0000, 0x0000 }, /* R624 */ + { 0x0000, 0x0000, 0x0000 }, /* R625 */ + { 0x0000, 0x0000, 0x0000 }, /* R626 */ + { 0x0000, 0x0000, 0x0000 }, /* R627 */ + { 0x0000, 0x0000, 0x0000 }, /* R628 */ + { 0x0000, 0x0000, 0x0000 }, /* R629 */ + { 0x0000, 0x0000, 0x0000 }, /* R630 */ + { 0x0000, 0x0000, 0x0000 }, /* R631 */ + { 0x0000, 0x0000, 0x0000 }, /* R632 */ + { 0x0000, 0x0000, 0x0000 }, /* R633 */ + { 0x0000, 0x0000, 0x0000 }, /* R634 */ + { 0x0000, 0x0000, 0x0000 }, /* R635 */ + { 0x0000, 0x0000, 0x0000 }, /* R636 */ + { 0x0000, 0x0000, 0x0000 }, /* R637 */ + { 0x0000, 0x0000, 0x0000 }, /* R638 */ + { 0x0000, 0x0000, 0x0000 }, /* R639 */ + { 0x0000, 0x0000, 0x0000 }, /* R640 */ + { 0x0000, 0x0000, 0x0000 }, /* R641 */ + { 0x0000, 0x0000, 0x0000 }, /* R642 */ + { 0x0000, 0x0000, 0x0000 }, /* R643 */ + { 0x0000, 0x0000, 0x0000 }, /* R644 */ + { 0x0000, 0x0000, 0x0000 }, /* R645 */ + { 0x0000, 0x0000, 0x0000 }, /* R646 */ + { 0x0000, 0x0000, 0x0000 }, /* R647 */ + { 0x0000, 0x0000, 0x0000 }, /* R648 */ + { 0x0000, 0x0000, 0x0000 }, /* R649 */ + { 0x0000, 0x0000, 0x0000 }, /* R650 */ + { 0x0000, 0x0000, 0x0000 }, /* R651 */ + { 0x0000, 0x0000, 0x0000 }, /* R652 */ + { 0x0000, 0x0000, 0x0000 }, /* R653 */ + { 0x0000, 0x0000, 0x0000 }, /* R654 */ + { 0x0000, 0x0000, 0x0000 }, /* R655 */ + { 0x0000, 0x0000, 0x0000 }, /* R656 */ + { 0x0000, 0x0000, 0x0000 }, /* R657 */ + { 0x0000, 0x0000, 0x0000 }, /* R658 */ + { 0x0000, 0x0000, 0x0000 }, /* R659 */ + { 0x0000, 0x0000, 0x0000 }, /* R660 */ + { 0x0000, 0x0000, 0x0000 }, /* R661 */ + { 0x0000, 0x0000, 0x0000 }, /* R662 */ + { 0x0000, 0x0000, 0x0000 }, /* R663 */ + { 0x0000, 0x0000, 0x0000 }, /* R664 */ + { 0x0000, 0x0000, 0x0000 }, /* R665 */ + { 0x0000, 0x0000, 0x0000 }, /* R666 */ + { 0x0000, 0x0000, 0x0000 }, /* R667 */ + { 0x0000, 0x0000, 0x0000 }, /* R668 */ + { 0x0000, 0x0000, 0x0000 }, /* R669 */ + { 0x0000, 0x0000, 0x0000 }, /* R670 */ + { 0x0000, 0x0000, 0x0000 }, /* R671 */ + { 0x0000, 0x0000, 0x0000 }, /* R672 */ + { 0x0000, 0x0000, 0x0000 }, /* R673 */ + { 0x0000, 0x0000, 0x0000 }, /* R674 */ + { 0x0000, 0x0000, 0x0000 }, /* R675 */ + { 0x0000, 0x0000, 0x0000 }, /* R676 */ + { 0x0000, 0x0000, 0x0000 }, /* R677 */ + { 0x0000, 0x0000, 0x0000 }, /* R678 */ + { 0x0000, 0x0000, 0x0000 }, /* R679 */ + { 0x0000, 0x0000, 0x0000 }, /* R680 */ + { 0x0000, 0x0000, 0x0000 }, /* R681 */ + { 0x0000, 0x0000, 0x0000 }, /* R682 */ + { 0x0000, 0x0000, 0x0000 }, /* R683 */ + { 0x0000, 0x0000, 0x0000 }, /* R684 */ + { 0x0000, 0x0000, 0x0000 }, /* R685 */ + { 0x0000, 0x0000, 0x0000 }, /* R686 */ + { 0x0000, 0x0000, 0x0000 }, /* R687 */ + { 0x0000, 0x0000, 0x0000 }, /* R688 */ + { 0x0000, 0x0000, 0x0000 }, /* R689 */ + { 0x0000, 0x0000, 0x0000 }, /* R690 */ + { 0x0000, 0x0000, 0x0000 }, /* R691 */ + { 0x0000, 0x0000, 0x0000 }, /* R692 */ + { 0x0000, 0x0000, 0x0000 }, /* R693 */ + { 0x0000, 0x0000, 0x0000 }, /* R694 */ + { 0x0000, 0x0000, 0x0000 }, /* R695 */ + { 0x0000, 0x0000, 0x0000 }, /* R696 */ + { 0x0000, 0x0000, 0x0000 }, /* R697 */ + { 0x0000, 0x0000, 0x0000 }, /* R698 */ + { 0x0000, 0x0000, 0x0000 }, /* R699 */ + { 0x0000, 0x0000, 0x0000 }, /* R700 */ + { 0x0000, 0x0000, 0x0000 }, /* R701 */ + { 0x0000, 0x0000, 0x0000 }, /* R702 */ + { 0x0000, 0x0000, 0x0000 }, /* R703 */ + { 0x0000, 0x0000, 0x0000 }, /* R704 */ + { 0x0000, 0x0000, 0x0000 }, /* R705 */ + { 0x0000, 0x0000, 0x0000 }, /* R706 */ + { 0x0000, 0x0000, 0x0000 }, /* R707 */ + { 0x0000, 0x0000, 0x0000 }, /* R708 */ + { 0x0000, 0x0000, 0x0000 }, /* R709 */ + { 0x0000, 0x0000, 0x0000 }, /* R710 */ + { 0x0000, 0x0000, 0x0000 }, /* R711 */ + { 0x0000, 0x0000, 0x0000 }, /* R712 */ + { 0x0000, 0x0000, 0x0000 }, /* R713 */ + { 0x0000, 0x0000, 0x0000 }, /* R714 */ + { 0x0000, 0x0000, 0x0000 }, /* R715 */ + { 0x0000, 0x0000, 0x0000 }, /* R716 */ + { 0x0000, 0x0000, 0x0000 }, /* R717 */ + { 0x0000, 0x0000, 0x0000 }, /* R718 */ + { 0x0000, 0x0000, 0x0000 }, /* R719 */ + { 0x0000, 0x0000, 0x0000 }, /* R720 */ + { 0x0000, 0x0000, 0x0000 }, /* R721 */ + { 0x0000, 0x0000, 0x0000 }, /* R722 */ + { 0x0000, 0x0000, 0x0000 }, /* R723 */ + { 0x0000, 0x0000, 0x0000 }, /* R724 */ + { 0x0000, 0x0000, 0x0000 }, /* R725 */ + { 0x0000, 0x0000, 0x0000 }, /* R726 */ + { 0x0000, 0x0000, 0x0000 }, /* R727 */ + { 0x0000, 0x0000, 0x0000 }, /* R728 */ + { 0x0000, 0x0000, 0x0000 }, /* R729 */ + { 0x0000, 0x0000, 0x0000 }, /* R730 */ + { 0x0000, 0x0000, 0x0000 }, /* R731 */ + { 0x0000, 0x0000, 0x0000 }, /* R732 */ + { 0x0000, 0x0000, 0x0000 }, /* R733 */ + { 0x0000, 0x0000, 0x0000 }, /* R734 */ + { 0x0000, 0x0000, 0x0000 }, /* R735 */ + { 0x0000, 0x0000, 0x0000 }, /* R736 */ + { 0x0000, 0x0000, 0x0000 }, /* R737 */ + { 0x0000, 0x0000, 0x0000 }, /* R738 */ + { 0x0000, 0x0000, 0x0000 }, /* R739 */ + { 0x0000, 0x0000, 0x0000 }, /* R740 */ + { 0x0000, 0x0000, 0x0000 }, /* R741 */ + { 0x0000, 0x0000, 0x0000 }, /* R742 */ + { 0x0000, 0x0000, 0x0000 }, /* R743 */ + { 0x0000, 0x0000, 0x0000 }, /* R744 */ + { 0x0000, 0x0000, 0x0000 }, /* R745 */ + { 0x0000, 0x0000, 0x0000 }, /* R746 */ + { 0x0000, 0x0000, 0x0000 }, /* R747 */ + { 0x0000, 0x0000, 0x0000 }, /* R748 */ + { 0x0000, 0x0000, 0x0000 }, /* R749 */ + { 0x0000, 0x0000, 0x0000 }, /* R750 */ + { 0x0000, 0x0000, 0x0000 }, /* R751 */ + { 0x0000, 0x0000, 0x0000 }, /* R752 */ + { 0x0000, 0x0000, 0x0000 }, /* R753 */ + { 0x0000, 0x0000, 0x0000 }, /* R754 */ + { 0x0000, 0x0000, 0x0000 }, /* R755 */ + { 0x0000, 0x0000, 0x0000 }, /* R756 */ + { 0x0000, 0x0000, 0x0000 }, /* R757 */ + { 0x0000, 0x0000, 0x0000 }, /* R758 */ + { 0x0000, 0x0000, 0x0000 }, /* R759 */ + { 0x0000, 0x0000, 0x0000 }, /* R760 */ + { 0x0000, 0x0000, 0x0000 }, /* R761 */ + { 0x0000, 0x0000, 0x0000 }, /* R762 */ + { 0x0000, 0x0000, 0x0000 }, /* R763 */ + { 0x0000, 0x0000, 0x0000 }, /* R764 */ + { 0x0000, 0x0000, 0x0000 }, /* R765 */ + { 0x0000, 0x0000, 0x0000 }, /* R766 */ + { 0x0000, 0x0000, 0x0000 }, /* R767 */ + { 0xE1F8, 0xE1F8, 0x0000 }, /* R768 - AIF1 Control (1) */ + { 0xCD1F, 0xCD1F, 0x0000 }, /* R769 - AIF1 Control (2) */ + { 0xF000, 0xF000, 0x0000 }, /* R770 - AIF1 Master/Slave */ + { 0x01F0, 0x01F0, 0x0000 }, /* R771 - AIF1 BCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R772 - AIF1ADC LRCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R773 - AIF1DAC LRCLK */ + { 0x0003, 0x0003, 0x0000 }, /* R774 - AIF1DAC Data */ + { 0x0003, 0x0003, 0x0000 }, /* R775 - AIF1ADC Data */ + { 0x0000, 0x0000, 0x0000 }, /* R776 */ + { 0x0000, 0x0000, 0x0000 }, /* R777 */ + { 0x0000, 0x0000, 0x0000 }, /* R778 */ + { 0x0000, 0x0000, 0x0000 }, /* R779 */ + { 0x0000, 0x0000, 0x0000 }, /* R780 */ + { 0x0000, 0x0000, 0x0000 }, /* R781 */ + { 0x0000, 0x0000, 0x0000 }, /* R782 */ + { 0x0000, 0x0000, 0x0000 }, /* R783 */ + { 0xF1F8, 0xF1F8, 0x0000 }, /* R784 - AIF2 Control (1) */ + { 0xFD1F, 0xFD1F, 0x0000 }, /* R785 - AIF2 Control (2) */ + { 0xF000, 0xF000, 0x0000 }, /* R786 - AIF2 Master/Slave */ + { 0x01F0, 0x01F0, 0x0000 }, /* R787 - AIF2 BCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R788 - AIF2ADC LRCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R789 - AIF2DAC LRCLK */ + { 0x0003, 0x0003, 0x0000 }, /* R790 - AIF2DAC Data */ + { 0x0003, 0x0003, 0x0000 }, /* R791 - AIF2ADC Data */ + { 0x0000, 0x0000, 0x0000 }, /* R792 */ + { 0x0000, 0x0000, 0x0000 }, /* R793 */ + { 0x0000, 0x0000, 0x0000 }, /* R794 */ + { 0x0000, 0x0000, 0x0000 }, /* R795 */ + { 0x0000, 0x0000, 0x0000 }, /* R796 */ + { 0x0000, 0x0000, 0x0000 }, /* R797 */ + { 0x0000, 0x0000, 0x0000 }, /* R798 */ + { 0x0000, 0x0000, 0x0000 }, /* R799 */ + { 0x0000, 0x0000, 0x0000 }, /* R800 */ + { 0x0000, 0x0000, 0x0000 }, /* R801 */ + { 0x0000, 0x0000, 0x0000 }, /* R802 */ + { 0x0000, 0x0000, 0x0000 }, /* R803 */ + { 0x0000, 0x0000, 0x0000 }, /* R804 */ + { 0x0000, 0x0000, 0x0000 }, /* R805 */ + { 0x0000, 0x0000, 0x0000 }, /* R806 */ + { 0x0000, 0x0000, 0x0000 }, /* R807 */ + { 0x0000, 0x0000, 0x0000 }, /* R808 */ + { 0x0000, 0x0000, 0x0000 }, /* R809 */ + { 0x0000, 0x0000, 0x0000 }, /* R810 */ + { 0x0000, 0x0000, 0x0000 }, /* R811 */ + { 0x0000, 0x0000, 0x0000 }, /* R812 */ + { 0x0000, 0x0000, 0x0000 }, /* R813 */ + { 0x0000, 0x0000, 0x0000 }, /* R814 */ + { 0x0000, 0x0000, 0x0000 }, /* R815 */ + { 0x0000, 0x0000, 0x0000 }, /* R816 */ + { 0x0000, 0x0000, 0x0000 }, /* R817 */ + { 0x0000, 0x0000, 0x0000 }, /* R818 */ + { 0x0000, 0x0000, 0x0000 }, /* R819 */ + { 0x0000, 0x0000, 0x0000 }, /* R820 */ + { 0x0000, 0x0000, 0x0000 }, /* R821 */ + { 0x0000, 0x0000, 0x0000 }, /* R822 */ + { 0x0000, 0x0000, 0x0000 }, /* R823 */ + { 0x0000, 0x0000, 0x0000 }, /* R824 */ + { 0x0000, 0x0000, 0x0000 }, /* R825 */ + { 0x0000, 0x0000, 0x0000 }, /* R826 */ + { 0x0000, 0x0000, 0x0000 }, /* R827 */ + { 0x0000, 0x0000, 0x0000 }, /* R828 */ + { 0x0000, 0x0000, 0x0000 }, /* R829 */ + { 0x0000, 0x0000, 0x0000 }, /* R830 */ + { 0x0000, 0x0000, 0x0000 }, /* R831 */ + { 0x0000, 0x0000, 0x0000 }, /* R832 */ + { 0x0000, 0x0000, 0x0000 }, /* R833 */ + { 0x0000, 0x0000, 0x0000 }, /* R834 */ + { 0x0000, 0x0000, 0x0000 }, /* R835 */ + { 0x0000, 0x0000, 0x0000 }, /* R836 */ + { 0x0000, 0x0000, 0x0000 }, /* R837 */ + { 0x0000, 0x0000, 0x0000 }, /* R838 */ + { 0x0000, 0x0000, 0x0000 }, /* R839 */ + { 0x0000, 0x0000, 0x0000 }, /* R840 */ + { 0x0000, 0x0000, 0x0000 }, /* R841 */ + { 0x0000, 0x0000, 0x0000 }, /* R842 */ + { 0x0000, 0x0000, 0x0000 }, /* R843 */ + { 0x0000, 0x0000, 0x0000 }, /* R844 */ + { 0x0000, 0x0000, 0x0000 }, /* R845 */ + { 0x0000, 0x0000, 0x0000 }, /* R846 */ + { 0x0000, 0x0000, 0x0000 }, /* R847 */ + { 0x0000, 0x0000, 0x0000 }, /* R848 */ + { 0x0000, 0x0000, 0x0000 }, /* R849 */ + { 0x0000, 0x0000, 0x0000 }, /* R850 */ + { 0x0000, 0x0000, 0x0000 }, /* R851 */ + { 0x0000, 0x0000, 0x0000 }, /* R852 */ + { 0x0000, 0x0000, 0x0000 }, /* R853 */ + { 0x0000, 0x0000, 0x0000 }, /* R854 */ + { 0x0000, 0x0000, 0x0000 }, /* R855 */ + { 0x0000, 0x0000, 0x0000 }, /* R856 */ + { 0x0000, 0x0000, 0x0000 }, /* R857 */ + { 0x0000, 0x0000, 0x0000 }, /* R858 */ + { 0x0000, 0x0000, 0x0000 }, /* R859 */ + { 0x0000, 0x0000, 0x0000 }, /* R860 */ + { 0x0000, 0x0000, 0x0000 }, /* R861 */ + { 0x0000, 0x0000, 0x0000 }, /* R862 */ + { 0x0000, 0x0000, 0x0000 }, /* R863 */ + { 0x0000, 0x0000, 0x0000 }, /* R864 */ + { 0x0000, 0x0000, 0x0000 }, /* R865 */ + { 0x0000, 0x0000, 0x0000 }, /* R866 */ + { 0x0000, 0x0000, 0x0000 }, /* R867 */ + { 0x0000, 0x0000, 0x0000 }, /* R868 */ + { 0x0000, 0x0000, 0x0000 }, /* R869 */ + { 0x0000, 0x0000, 0x0000 }, /* R870 */ + { 0x0000, 0x0000, 0x0000 }, /* R871 */ + { 0x0000, 0x0000, 0x0000 }, /* R872 */ + { 0x0000, 0x0000, 0x0000 }, /* R873 */ + { 0x0000, 0x0000, 0x0000 }, /* R874 */ + { 0x0000, 0x0000, 0x0000 }, /* R875 */ + { 0x0000, 0x0000, 0x0000 }, /* R876 */ + { 0x0000, 0x0000, 0x0000 }, /* R877 */ + { 0x0000, 0x0000, 0x0000 }, /* R878 */ + { 0x0000, 0x0000, 0x0000 }, /* R879 */ + { 0x0000, 0x0000, 0x0000 }, /* R880 */ + { 0x0000, 0x0000, 0x0000 }, /* R881 */ + { 0x0000, 0x0000, 0x0000 }, /* R882 */ + { 0x0000, 0x0000, 0x0000 }, /* R883 */ + { 0x0000, 0x0000, 0x0000 }, /* R884 */ + { 0x0000, 0x0000, 0x0000 }, /* R885 */ + { 0x0000, 0x0000, 0x0000 }, /* R886 */ + { 0x0000, 0x0000, 0x0000 }, /* R887 */ + { 0x0000, 0x0000, 0x0000 }, /* R888 */ + { 0x0000, 0x0000, 0x0000 }, /* R889 */ + { 0x0000, 0x0000, 0x0000 }, /* R890 */ + { 0x0000, 0x0000, 0x0000 }, /* R891 */ + { 0x0000, 0x0000, 0x0000 }, /* R892 */ + { 0x0000, 0x0000, 0x0000 }, /* R893 */ + { 0x0000, 0x0000, 0x0000 }, /* R894 */ + { 0x0000, 0x0000, 0x0000 }, /* R895 */ + { 0x0000, 0x0000, 0x0000 }, /* R896 */ + { 0x0000, 0x0000, 0x0000 }, /* R897 */ + { 0x0000, 0x0000, 0x0000 }, /* R898 */ + { 0x0000, 0x0000, 0x0000 }, /* R899 */ + { 0x0000, 0x0000, 0x0000 }, /* R900 */ + { 0x0000, 0x0000, 0x0000 }, /* R901 */ + { 0x0000, 0x0000, 0x0000 }, /* R902 */ + { 0x0000, 0x0000, 0x0000 }, /* R903 */ + { 0x0000, 0x0000, 0x0000 }, /* R904 */ + { 0x0000, 0x0000, 0x0000 }, /* R905 */ + { 0x0000, 0x0000, 0x0000 }, /* R906 */ + { 0x0000, 0x0000, 0x0000 }, /* R907 */ + { 0x0000, 0x0000, 0x0000 }, /* R908 */ + { 0x0000, 0x0000, 0x0000 }, /* R909 */ + { 0x0000, 0x0000, 0x0000 }, /* R910 */ + { 0x0000, 0x0000, 0x0000 }, /* R911 */ + { 0x0000, 0x0000, 0x0000 }, /* R912 */ + { 0x0000, 0x0000, 0x0000 }, /* R913 */ + { 0x0000, 0x0000, 0x0000 }, /* R914 */ + { 0x0000, 0x0000, 0x0000 }, /* R915 */ + { 0x0000, 0x0000, 0x0000 }, /* R916 */ + { 0x0000, 0x0000, 0x0000 }, /* R917 */ + { 0x0000, 0x0000, 0x0000 }, /* R918 */ + { 0x0000, 0x0000, 0x0000 }, /* R919 */ + { 0x0000, 0x0000, 0x0000 }, /* R920 */ + { 0x0000, 0x0000, 0x0000 }, /* R921 */ + { 0x0000, 0x0000, 0x0000 }, /* R922 */ + { 0x0000, 0x0000, 0x0000 }, /* R923 */ + { 0x0000, 0x0000, 0x0000 }, /* R924 */ + { 0x0000, 0x0000, 0x0000 }, /* R925 */ + { 0x0000, 0x0000, 0x0000 }, /* R926 */ + { 0x0000, 0x0000, 0x0000 }, /* R927 */ + { 0x0000, 0x0000, 0x0000 }, /* R928 */ + { 0x0000, 0x0000, 0x0000 }, /* R929 */ + { 0x0000, 0x0000, 0x0000 }, /* R930 */ + { 0x0000, 0x0000, 0x0000 }, /* R931 */ + { 0x0000, 0x0000, 0x0000 }, /* R932 */ + { 0x0000, 0x0000, 0x0000 }, /* R933 */ + { 0x0000, 0x0000, 0x0000 }, /* R934 */ + { 0x0000, 0x0000, 0x0000 }, /* R935 */ + { 0x0000, 0x0000, 0x0000 }, /* R936 */ + { 0x0000, 0x0000, 0x0000 }, /* R937 */ + { 0x0000, 0x0000, 0x0000 }, /* R938 */ + { 0x0000, 0x0000, 0x0000 }, /* R939 */ + { 0x0000, 0x0000, 0x0000 }, /* R940 */ + { 0x0000, 0x0000, 0x0000 }, /* R941 */ + { 0x0000, 0x0000, 0x0000 }, /* R942 */ + { 0x0000, 0x0000, 0x0000 }, /* R943 */ + { 0x0000, 0x0000, 0x0000 }, /* R944 */ + { 0x0000, 0x0000, 0x0000 }, /* R945 */ + { 0x0000, 0x0000, 0x0000 }, /* R946 */ + { 0x0000, 0x0000, 0x0000 }, /* R947 */ + { 0x0000, 0x0000, 0x0000 }, /* R948 */ + { 0x0000, 0x0000, 0x0000 }, /* R949 */ + { 0x0000, 0x0000, 0x0000 }, /* R950 */ + { 0x0000, 0x0000, 0x0000 }, /* R951 */ + { 0x0000, 0x0000, 0x0000 }, /* R952 */ + { 0x0000, 0x0000, 0x0000 }, /* R953 */ + { 0x0000, 0x0000, 0x0000 }, /* R954 */ + { 0x0000, 0x0000, 0x0000 }, /* R955 */ + { 0x0000, 0x0000, 0x0000 }, /* R956 */ + { 0x0000, 0x0000, 0x0000 }, /* R957 */ + { 0x0000, 0x0000, 0x0000 }, /* R958 */ + { 0x0000, 0x0000, 0x0000 }, /* R959 */ + { 0x0000, 0x0000, 0x0000 }, /* R960 */ + { 0x0000, 0x0000, 0x0000 }, /* R961 */ + { 0x0000, 0x0000, 0x0000 }, /* R962 */ + { 0x0000, 0x0000, 0x0000 }, /* R963 */ + { 0x0000, 0x0000, 0x0000 }, /* R964 */ + { 0x0000, 0x0000, 0x0000 }, /* R965 */ + { 0x0000, 0x0000, 0x0000 }, /* R966 */ + { 0x0000, 0x0000, 0x0000 }, /* R967 */ + { 0x0000, 0x0000, 0x0000 }, /* R968 */ + { 0x0000, 0x0000, 0x0000 }, /* R969 */ + { 0x0000, 0x0000, 0x0000 }, /* R970 */ + { 0x0000, 0x0000, 0x0000 }, /* R971 */ + { 0x0000, 0x0000, 0x0000 }, /* R972 */ + { 0x0000, 0x0000, 0x0000 }, /* R973 */ + { 0x0000, 0x0000, 0x0000 }, /* R974 */ + { 0x0000, 0x0000, 0x0000 }, /* R975 */ + { 0x0000, 0x0000, 0x0000 }, /* R976 */ + { 0x0000, 0x0000, 0x0000 }, /* R977 */ + { 0x0000, 0x0000, 0x0000 }, /* R978 */ + { 0x0000, 0x0000, 0x0000 }, /* R979 */ + { 0x0000, 0x0000, 0x0000 }, /* R980 */ + { 0x0000, 0x0000, 0x0000 }, /* R981 */ + { 0x0000, 0x0000, 0x0000 }, /* R982 */ + { 0x0000, 0x0000, 0x0000 }, /* R983 */ + { 0x0000, 0x0000, 0x0000 }, /* R984 */ + { 0x0000, 0x0000, 0x0000 }, /* R985 */ + { 0x0000, 0x0000, 0x0000 }, /* R986 */ + { 0x0000, 0x0000, 0x0000 }, /* R987 */ + { 0x0000, 0x0000, 0x0000 }, /* R988 */ + { 0x0000, 0x0000, 0x0000 }, /* R989 */ + { 0x0000, 0x0000, 0x0000 }, /* R990 */ + { 0x0000, 0x0000, 0x0000 }, /* R991 */ + { 0x0000, 0x0000, 0x0000 }, /* R992 */ + { 0x0000, 0x0000, 0x0000 }, /* R993 */ + { 0x0000, 0x0000, 0x0000 }, /* R994 */ + { 0x0000, 0x0000, 0x0000 }, /* R995 */ + { 0x0000, 0x0000, 0x0000 }, /* R996 */ + { 0x0000, 0x0000, 0x0000 }, /* R997 */ + { 0x0000, 0x0000, 0x0000 }, /* R998 */ + { 0x0000, 0x0000, 0x0000 }, /* R999 */ + { 0x0000, 0x0000, 0x0000 }, /* R1000 */ + { 0x0000, 0x0000, 0x0000 }, /* R1001 */ + { 0x0000, 0x0000, 0x0000 }, /* R1002 */ + { 0x0000, 0x0000, 0x0000 }, /* R1003 */ + { 0x0000, 0x0000, 0x0000 }, /* R1004 */ + { 0x0000, 0x0000, 0x0000 }, /* R1005 */ + { 0x0000, 0x0000, 0x0000 }, /* R1006 */ + { 0x0000, 0x0000, 0x0000 }, /* R1007 */ + { 0x0000, 0x0000, 0x0000 }, /* R1008 */ + { 0x0000, 0x0000, 0x0000 }, /* R1009 */ + { 0x0000, 0x0000, 0x0000 }, /* R1010 */ + { 0x0000, 0x0000, 0x0000 }, /* R1011 */ + { 0x0000, 0x0000, 0x0000 }, /* R1012 */ + { 0x0000, 0x0000, 0x0000 }, /* R1013 */ + { 0x0000, 0x0000, 0x0000 }, /* R1014 */ + { 0x0000, 0x0000, 0x0000 }, /* R1015 */ + { 0x0000, 0x0000, 0x0000 }, /* R1016 */ + { 0x0000, 0x0000, 0x0000 }, /* R1017 */ + { 0x0000, 0x0000, 0x0000 }, /* R1018 */ + { 0x0000, 0x0000, 0x0000 }, /* R1019 */ + { 0x0000, 0x0000, 0x0000 }, /* R1020 */ + { 0x0000, 0x0000, 0x0000 }, /* R1021 */ + { 0x0000, 0x0000, 0x0000 }, /* R1022 */ + { 0x0000, 0x0000, 0x0000 }, /* R1023 */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1024 - AIF1 ADC1 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1025 - AIF1 ADC1 Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1026 - AIF1 DAC1 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1027 - AIF1 DAC1 Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1028 - AIF1 ADC2 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1029 - AIF1 ADC2 Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1030 - AIF1 DAC2 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1031 - AIF1 DAC2 Right Volume */ + { 0x0000, 0x0000, 0x0000 }, /* R1032 */ + { 0x0000, 0x0000, 0x0000 }, /* R1033 */ + { 0x0000, 0x0000, 0x0000 }, /* R1034 */ + { 0x0000, 0x0000, 0x0000 }, /* R1035 */ + { 0x0000, 0x0000, 0x0000 }, /* R1036 */ + { 0x0000, 0x0000, 0x0000 }, /* R1037 */ + { 0x0000, 0x0000, 0x0000 }, /* R1038 */ + { 0x0000, 0x0000, 0x0000 }, /* R1039 */ + { 0xF800, 0xF800, 0x0000 }, /* R1040 - AIF1 ADC1 Filters */ + { 0x7800, 0x7800, 0x0000 }, /* R1041 - AIF1 ADC2 Filters */ + { 0x0000, 0x0000, 0x0000 }, /* R1042 */ + { 0x0000, 0x0000, 0x0000 }, /* R1043 */ + { 0x0000, 0x0000, 0x0000 }, /* R1044 */ + { 0x0000, 0x0000, 0x0000 }, /* R1045 */ + { 0x0000, 0x0000, 0x0000 }, /* R1046 */ + { 0x0000, 0x0000, 0x0000 }, /* R1047 */ + { 0x0000, 0x0000, 0x0000 }, /* R1048 */ + { 0x0000, 0x0000, 0x0000 }, /* R1049 */ + { 0x0000, 0x0000, 0x0000 }, /* R1050 */ + { 0x0000, 0x0000, 0x0000 }, /* R1051 */ + { 0x0000, 0x0000, 0x0000 }, /* R1052 */ + { 0x0000, 0x0000, 0x0000 }, /* R1053 */ + { 0x0000, 0x0000, 0x0000 }, /* R1054 */ + { 0x0000, 0x0000, 0x0000 }, /* R1055 */ + { 0x02B6, 0x02B6, 0x0000 }, /* R1056 - AIF1 DAC1 Filters (1) */ + { 0x3F00, 0x3F00, 0x0000 }, /* R1057 - AIF1 DAC1 Filters (2) */ + { 0x02B6, 0x02B6, 0x0000 }, /* R1058 - AIF1 DAC2 Filters (1) */ + { 0x3F00, 0x3F00, 0x0000 }, /* R1059 - AIF1 DAC2 Filters (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R1060 */ + { 0x0000, 0x0000, 0x0000 }, /* R1061 */ + { 0x0000, 0x0000, 0x0000 }, /* R1062 */ + { 0x0000, 0x0000, 0x0000 }, /* R1063 */ + { 0x0000, 0x0000, 0x0000 }, /* R1064 */ + { 0x0000, 0x0000, 0x0000 }, /* R1065 */ + { 0x0000, 0x0000, 0x0000 }, /* R1066 */ + { 0x0000, 0x0000, 0x0000 }, /* R1067 */ + { 0x0000, 0x0000, 0x0000 }, /* R1068 */ + { 0x0000, 0x0000, 0x0000 }, /* R1069 */ + { 0x0000, 0x0000, 0x0000 }, /* R1070 */ + { 0x0000, 0x0000, 0x0000 }, /* R1071 */ + { 0x0000, 0x0000, 0x0000 }, /* R1072 */ + { 0x0000, 0x0000, 0x0000 }, /* R1073 */ + { 0x0000, 0x0000, 0x0000 }, /* R1074 */ + { 0x0000, 0x0000, 0x0000 }, /* R1075 */ + { 0x0000, 0x0000, 0x0000 }, /* R1076 */ + { 0x0000, 0x0000, 0x0000 }, /* R1077 */ + { 0x0000, 0x0000, 0x0000 }, /* R1078 */ + { 0x0000, 0x0000, 0x0000 }, /* R1079 */ + { 0x0000, 0x0000, 0x0000 }, /* R1080 */ + { 0x0000, 0x0000, 0x0000 }, /* R1081 */ + { 0x0000, 0x0000, 0x0000 }, /* R1082 */ + { 0x0000, 0x0000, 0x0000 }, /* R1083 */ + { 0x0000, 0x0000, 0x0000 }, /* R1084 */ + { 0x0000, 0x0000, 0x0000 }, /* R1085 */ + { 0x0000, 0x0000, 0x0000 }, /* R1086 */ + { 0x0000, 0x0000, 0x0000 }, /* R1087 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1088 - AIF1 DRC1 (1) */ + { 0x1FFF, 0x1FFF, 0x0000 }, /* R1089 - AIF1 DRC1 (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1090 - AIF1 DRC1 (3) */ + { 0x07FF, 0x07FF, 0x0000 }, /* R1091 - AIF1 DRC1 (4) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R1092 - AIF1 DRC1 (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R1093 */ + { 0x0000, 0x0000, 0x0000 }, /* R1094 */ + { 0x0000, 0x0000, 0x0000 }, /* R1095 */ + { 0x0000, 0x0000, 0x0000 }, /* R1096 */ + { 0x0000, 0x0000, 0x0000 }, /* R1097 */ + { 0x0000, 0x0000, 0x0000 }, /* R1098 */ + { 0x0000, 0x0000, 0x0000 }, /* R1099 */ + { 0x0000, 0x0000, 0x0000 }, /* R1100 */ + { 0x0000, 0x0000, 0x0000 }, /* R1101 */ + { 0x0000, 0x0000, 0x0000 }, /* R1102 */ + { 0x0000, 0x0000, 0x0000 }, /* R1103 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1104 - AIF1 DRC2 (1) */ + { 0x1FFF, 0x1FFF, 0x0000 }, /* R1105 - AIF1 DRC2 (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1106 - AIF1 DRC2 (3) */ + { 0x07FF, 0x07FF, 0x0000 }, /* R1107 - AIF1 DRC2 (4) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R1108 - AIF1 DRC2 (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R1109 */ + { 0x0000, 0x0000, 0x0000 }, /* R1110 */ + { 0x0000, 0x0000, 0x0000 }, /* R1111 */ + { 0x0000, 0x0000, 0x0000 }, /* R1112 */ + { 0x0000, 0x0000, 0x0000 }, /* R1113 */ + { 0x0000, 0x0000, 0x0000 }, /* R1114 */ + { 0x0000, 0x0000, 0x0000 }, /* R1115 */ + { 0x0000, 0x0000, 0x0000 }, /* R1116 */ + { 0x0000, 0x0000, 0x0000 }, /* R1117 */ + { 0x0000, 0x0000, 0x0000 }, /* R1118 */ + { 0x0000, 0x0000, 0x0000 }, /* R1119 */ + { 0x0000, 0x0000, 0x0000 }, /* R1120 */ + { 0x0000, 0x0000, 0x0000 }, /* R1121 */ + { 0x0000, 0x0000, 0x0000 }, /* R1122 */ + { 0x0000, 0x0000, 0x0000 }, /* R1123 */ + { 0x0000, 0x0000, 0x0000 }, /* R1124 */ + { 0x0000, 0x0000, 0x0000 }, /* R1125 */ + { 0x0000, 0x0000, 0x0000 }, /* R1126 */ + { 0x0000, 0x0000, 0x0000 }, /* R1127 */ + { 0x0000, 0x0000, 0x0000 }, /* R1128 */ + { 0x0000, 0x0000, 0x0000 }, /* R1129 */ + { 0x0000, 0x0000, 0x0000 }, /* R1130 */ + { 0x0000, 0x0000, 0x0000 }, /* R1131 */ + { 0x0000, 0x0000, 0x0000 }, /* R1132 */ + { 0x0000, 0x0000, 0x0000 }, /* R1133 */ + { 0x0000, 0x0000, 0x0000 }, /* R1134 */ + { 0x0000, 0x0000, 0x0000 }, /* R1135 */ + { 0x0000, 0x0000, 0x0000 }, /* R1136 */ + { 0x0000, 0x0000, 0x0000 }, /* R1137 */ + { 0x0000, 0x0000, 0x0000 }, /* R1138 */ + { 0x0000, 0x0000, 0x0000 }, /* R1139 */ + { 0x0000, 0x0000, 0x0000 }, /* R1140 */ + { 0x0000, 0x0000, 0x0000 }, /* R1141 */ + { 0x0000, 0x0000, 0x0000 }, /* R1142 */ + { 0x0000, 0x0000, 0x0000 }, /* R1143 */ + { 0x0000, 0x0000, 0x0000 }, /* R1144 */ + { 0x0000, 0x0000, 0x0000 }, /* R1145 */ + { 0x0000, 0x0000, 0x0000 }, /* R1146 */ + { 0x0000, 0x0000, 0x0000 }, /* R1147 */ + { 0x0000, 0x0000, 0x0000 }, /* R1148 */ + { 0x0000, 0x0000, 0x0000 }, /* R1149 */ + { 0x0000, 0x0000, 0x0000 }, /* R1150 */ + { 0x0000, 0x0000, 0x0000 }, /* R1151 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1152 - AIF1 DAC1 EQ Gains (1) */ + { 0xFFC0, 0xFFC0, 0x0000 }, /* R1153 - AIF1 DAC1 EQ Gains (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1154 - AIF1 DAC1 EQ Band 1 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1155 - AIF1 DAC1 EQ Band 1 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1156 - AIF1 DAC1 EQ Band 1 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1157 - AIF1 DAC1 EQ Band 2 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1158 - AIF1 DAC1 EQ Band 2 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1159 - AIF1 DAC1 EQ Band 2 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1160 - AIF1 DAC1 EQ Band 2 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1161 - AIF1 DAC1 EQ Band 3 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1162 - AIF1 DAC1 EQ Band 3 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1163 - AIF1 DAC1 EQ Band 3 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1164 - AIF1 DAC1 EQ Band 3 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1165 - AIF1 DAC1 EQ Band 4 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1166 - AIF1 DAC1 EQ Band 4 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1167 - AIF1 DAC1 EQ Band 4 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1168 - AIF1 DAC1 EQ Band 4 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1169 - AIF1 DAC1 EQ Band 5 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1170 - AIF1 DAC1 EQ Band 5 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1171 - AIF1 DAC1 EQ Band 5 PG */ + { 0x0000, 0x0000, 0x0000 }, /* R1172 */ + { 0x0000, 0x0000, 0x0000 }, /* R1173 */ + { 0x0000, 0x0000, 0x0000 }, /* R1174 */ + { 0x0000, 0x0000, 0x0000 }, /* R1175 */ + { 0x0000, 0x0000, 0x0000 }, /* R1176 */ + { 0x0000, 0x0000, 0x0000 }, /* R1177 */ + { 0x0000, 0x0000, 0x0000 }, /* R1178 */ + { 0x0000, 0x0000, 0x0000 }, /* R1179 */ + { 0x0000, 0x0000, 0x0000 }, /* R1180 */ + { 0x0000, 0x0000, 0x0000 }, /* R1181 */ + { 0x0000, 0x0000, 0x0000 }, /* R1182 */ + { 0x0000, 0x0000, 0x0000 }, /* R1183 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1184 - AIF1 DAC2 EQ Gains (1) */ + { 0xFFC0, 0xFFC0, 0x0000 }, /* R1185 - AIF1 DAC2 EQ Gains (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1186 - AIF1 DAC2 EQ Band 1 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1187 - AIF1 DAC2 EQ Band 1 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1188 - AIF1 DAC2 EQ Band 1 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1189 - AIF1 DAC2 EQ Band 2 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1190 - AIF1 DAC2 EQ Band 2 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1191 - AIF1 DAC2 EQ Band 2 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1192 - AIF1 DAC2 EQ Band 2 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1193 - AIF1 DAC2 EQ Band 3 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1194 - AIF1 DAC2 EQ Band 3 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1195 - AIF1 DAC2 EQ Band 3 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1196 - AIF1 DAC2 EQ Band 3 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1197 - AIF1 DAC2 EQ Band 4 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1198 - AIF1 DAC2 EQ Band 4 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1199 - AIF1 DAC2 EQ Band 4 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1200 - AIF1 DAC2 EQ Band 4 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1201 - AIF1 DAC2 EQ Band 5 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1202 - AIF1 DAC2 EQ Band 5 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1203 - AIF1 DAC2 EQ Band 5 PG */ + { 0x0000, 0x0000, 0x0000 }, /* R1204 */ + { 0x0000, 0x0000, 0x0000 }, /* R1205 */ + { 0x0000, 0x0000, 0x0000 }, /* R1206 */ + { 0x0000, 0x0000, 0x0000 }, /* R1207 */ + { 0x0000, 0x0000, 0x0000 }, /* R1208 */ + { 0x0000, 0x0000, 0x0000 }, /* R1209 */ + { 0x0000, 0x0000, 0x0000 }, /* R1210 */ + { 0x0000, 0x0000, 0x0000 }, /* R1211 */ + { 0x0000, 0x0000, 0x0000 }, /* R1212 */ + { 0x0000, 0x0000, 0x0000 }, /* R1213 */ + { 0x0000, 0x0000, 0x0000 }, /* R1214 */ + { 0x0000, 0x0000, 0x0000 }, /* R1215 */ + { 0x0000, 0x0000, 0x0000 }, /* R1216 */ + { 0x0000, 0x0000, 0x0000 }, /* R1217 */ + { 0x0000, 0x0000, 0x0000 }, /* R1218 */ + { 0x0000, 0x0000, 0x0000 }, /* R1219 */ + { 0x0000, 0x0000, 0x0000 }, /* R1220 */ + { 0x0000, 0x0000, 0x0000 }, /* R1221 */ + { 0x0000, 0x0000, 0x0000 }, /* R1222 */ + { 0x0000, 0x0000, 0x0000 }, /* R1223 */ + { 0x0000, 0x0000, 0x0000 }, /* R1224 */ + { 0x0000, 0x0000, 0x0000 }, /* R1225 */ + { 0x0000, 0x0000, 0x0000 }, /* R1226 */ + { 0x0000, 0x0000, 0x0000 }, /* R1227 */ + { 0x0000, 0x0000, 0x0000 }, /* R1228 */ + { 0x0000, 0x0000, 0x0000 }, /* R1229 */ + { 0x0000, 0x0000, 0x0000 }, /* R1230 */ + { 0x0000, 0x0000, 0x0000 }, /* R1231 */ + { 0x0000, 0x0000, 0x0000 }, /* R1232 */ + { 0x0000, 0x0000, 0x0000 }, /* R1233 */ + { 0x0000, 0x0000, 0x0000 }, /* R1234 */ + { 0x0000, 0x0000, 0x0000 }, /* R1235 */ + { 0x0000, 0x0000, 0x0000 }, /* R1236 */ + { 0x0000, 0x0000, 0x0000 }, /* R1237 */ + { 0x0000, 0x0000, 0x0000 }, /* R1238 */ + { 0x0000, 0x0000, 0x0000 }, /* R1239 */ + { 0x0000, 0x0000, 0x0000 }, /* R1240 */ + { 0x0000, 0x0000, 0x0000 }, /* R1241 */ + { 0x0000, 0x0000, 0x0000 }, /* R1242 */ + { 0x0000, 0x0000, 0x0000 }, /* R1243 */ + { 0x0000, 0x0000, 0x0000 }, /* R1244 */ + { 0x0000, 0x0000, 0x0000 }, /* R1245 */ + { 0x0000, 0x0000, 0x0000 }, /* R1246 */ + { 0x0000, 0x0000, 0x0000 }, /* R1247 */ + { 0x0000, 0x0000, 0x0000 }, /* R1248 */ + { 0x0000, 0x0000, 0x0000 }, /* R1249 */ + { 0x0000, 0x0000, 0x0000 }, /* R1250 */ + { 0x0000, 0x0000, 0x0000 }, /* R1251 */ + { 0x0000, 0x0000, 0x0000 }, /* R1252 */ + { 0x0000, 0x0000, 0x0000 }, /* R1253 */ + { 0x0000, 0x0000, 0x0000 }, /* R1254 */ + { 0x0000, 0x0000, 0x0000 }, /* R1255 */ + { 0x0000, 0x0000, 0x0000 }, /* R1256 */ + { 0x0000, 0x0000, 0x0000 }, /* R1257 */ + { 0x0000, 0x0000, 0x0000 }, /* R1258 */ + { 0x0000, 0x0000, 0x0000 }, /* R1259 */ + { 0x0000, 0x0000, 0x0000 }, /* R1260 */ + { 0x0000, 0x0000, 0x0000 }, /* R1261 */ + { 0x0000, 0x0000, 0x0000 }, /* R1262 */ + { 0x0000, 0x0000, 0x0000 }, /* R1263 */ + { 0x0000, 0x0000, 0x0000 }, /* R1264 */ + { 0x0000, 0x0000, 0x0000 }, /* R1265 */ + { 0x0000, 0x0000, 0x0000 }, /* R1266 */ + { 0x0000, 0x0000, 0x0000 }, /* R1267 */ + { 0x0000, 0x0000, 0x0000 }, /* R1268 */ + { 0x0000, 0x0000, 0x0000 }, /* R1269 */ + { 0x0000, 0x0000, 0x0000 }, /* R1270 */ + { 0x0000, 0x0000, 0x0000 }, /* R1271 */ + { 0x0000, 0x0000, 0x0000 }, /* R1272 */ + { 0x0000, 0x0000, 0x0000 }, /* R1273 */ + { 0x0000, 0x0000, 0x0000 }, /* R1274 */ + { 0x0000, 0x0000, 0x0000 }, /* R1275 */ + { 0x0000, 0x0000, 0x0000 }, /* R1276 */ + { 0x0000, 0x0000, 0x0000 }, /* R1277 */ + { 0x0000, 0x0000, 0x0000 }, /* R1278 */ + { 0x0000, 0x0000, 0x0000 }, /* R1279 */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1280 - AIF2 ADC Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1281 - AIF2 ADC Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1282 - AIF2 DAC Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1283 - AIF2 DAC Right Volume */ + { 0x0000, 0x0000, 0x0000 }, /* R1284 */ + { 0x0000, 0x0000, 0x0000 }, /* R1285 */ + { 0x0000, 0x0000, 0x0000 }, /* R1286 */ + { 0x0000, 0x0000, 0x0000 }, /* R1287 */ + { 0x0000, 0x0000, 0x0000 }, /* R1288 */ + { 0x0000, 0x0000, 0x0000 }, /* R1289 */ + { 0x0000, 0x0000, 0x0000 }, /* R1290 */ + { 0x0000, 0x0000, 0x0000 }, /* R1291 */ + { 0x0000, 0x0000, 0x0000 }, /* R1292 */ + { 0x0000, 0x0000, 0x0000 }, /* R1293 */ + { 0x0000, 0x0000, 0x0000 }, /* R1294 */ + { 0x0000, 0x0000, 0x0000 }, /* R1295 */ + { 0xF800, 0xF800, 0x0000 }, /* R1296 - AIF2 ADC Filters */ + { 0x0000, 0x0000, 0x0000 }, /* R1297 */ + { 0x0000, 0x0000, 0x0000 }, /* R1298 */ + { 0x0000, 0x0000, 0x0000 }, /* R1299 */ + { 0x0000, 0x0000, 0x0000 }, /* R1300 */ + { 0x0000, 0x0000, 0x0000 }, /* R1301 */ + { 0x0000, 0x0000, 0x0000 }, /* R1302 */ + { 0x0000, 0x0000, 0x0000 }, /* R1303 */ + { 0x0000, 0x0000, 0x0000 }, /* R1304 */ + { 0x0000, 0x0000, 0x0000 }, /* R1305 */ + { 0x0000, 0x0000, 0x0000 }, /* R1306 */ + { 0x0000, 0x0000, 0x0000 }, /* R1307 */ + { 0x0000, 0x0000, 0x0000 }, /* R1308 */ + { 0x0000, 0x0000, 0x0000 }, /* R1309 */ + { 0x0000, 0x0000, 0x0000 }, /* R1310 */ + { 0x0000, 0x0000, 0x0000 }, /* R1311 */ + { 0x02B6, 0x02B6, 0x0000 }, /* R1312 - AIF2 DAC Filters (1) */ + { 0x3F00, 0x3F00, 0x0000 }, /* R1313 - AIF2 DAC Filters (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R1314 */ + { 0x0000, 0x0000, 0x0000 }, /* R1315 */ + { 0x0000, 0x0000, 0x0000 }, /* R1316 */ + { 0x0000, 0x0000, 0x0000 }, /* R1317 */ + { 0x0000, 0x0000, 0x0000 }, /* R1318 */ + { 0x0000, 0x0000, 0x0000 }, /* R1319 */ + { 0x0000, 0x0000, 0x0000 }, /* R1320 */ + { 0x0000, 0x0000, 0x0000 }, /* R1321 */ + { 0x0000, 0x0000, 0x0000 }, /* R1322 */ + { 0x0000, 0x0000, 0x0000 }, /* R1323 */ + { 0x0000, 0x0000, 0x0000 }, /* R1324 */ + { 0x0000, 0x0000, 0x0000 }, /* R1325 */ + { 0x0000, 0x0000, 0x0000 }, /* R1326 */ + { 0x0000, 0x0000, 0x0000 }, /* R1327 */ + { 0x0000, 0x0000, 0x0000 }, /* R1328 */ + { 0x0000, 0x0000, 0x0000 }, /* R1329 */ + { 0x0000, 0x0000, 0x0000 }, /* R1330 */ + { 0x0000, 0x0000, 0x0000 }, /* R1331 */ + { 0x0000, 0x0000, 0x0000 }, /* R1332 */ + { 0x0000, 0x0000, 0x0000 }, /* R1333 */ + { 0x0000, 0x0000, 0x0000 }, /* R1334 */ + { 0x0000, 0x0000, 0x0000 }, /* R1335 */ + { 0x0000, 0x0000, 0x0000 }, /* R1336 */ + { 0x0000, 0x0000, 0x0000 }, /* R1337 */ + { 0x0000, 0x0000, 0x0000 }, /* R1338 */ + { 0x0000, 0x0000, 0x0000 }, /* R1339 */ + { 0x0000, 0x0000, 0x0000 }, /* R1340 */ + { 0x0000, 0x0000, 0x0000 }, /* R1341 */ + { 0x0000, 0x0000, 0x0000 }, /* R1342 */ + { 0x0000, 0x0000, 0x0000 }, /* R1343 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1344 - AIF2 DRC (1) */ + { 0x1FFF, 0x1FFF, 0x0000 }, /* R1345 - AIF2 DRC (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1346 - AIF2 DRC (3) */ + { 0x07FF, 0x07FF, 0x0000 }, /* R1347 - AIF2 DRC (4) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R1348 - AIF2 DRC (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R1349 */ + { 0x0000, 0x0000, 0x0000 }, /* R1350 */ + { 0x0000, 0x0000, 0x0000 }, /* R1351 */ + { 0x0000, 0x0000, 0x0000 }, /* R1352 */ + { 0x0000, 0x0000, 0x0000 }, /* R1353 */ + { 0x0000, 0x0000, 0x0000 }, /* R1354 */ + { 0x0000, 0x0000, 0x0000 }, /* R1355 */ + { 0x0000, 0x0000, 0x0000 }, /* R1356 */ + { 0x0000, 0x0000, 0x0000 }, /* R1357 */ + { 0x0000, 0x0000, 0x0000 }, /* R1358 */ + { 0x0000, 0x0000, 0x0000 }, /* R1359 */ + { 0x0000, 0x0000, 0x0000 }, /* R1360 */ + { 0x0000, 0x0000, 0x0000 }, /* R1361 */ + { 0x0000, 0x0000, 0x0000 }, /* R1362 */ + { 0x0000, 0x0000, 0x0000 }, /* R1363 */ + { 0x0000, 0x0000, 0x0000 }, /* R1364 */ + { 0x0000, 0x0000, 0x0000 }, /* R1365 */ + { 0x0000, 0x0000, 0x0000 }, /* R1366 */ + { 0x0000, 0x0000, 0x0000 }, /* R1367 */ + { 0x0000, 0x0000, 0x0000 }, /* R1368 */ + { 0x0000, 0x0000, 0x0000 }, /* R1369 */ + { 0x0000, 0x0000, 0x0000 }, /* R1370 */ + { 0x0000, 0x0000, 0x0000 }, /* R1371 */ + { 0x0000, 0x0000, 0x0000 }, /* R1372 */ + { 0x0000, 0x0000, 0x0000 }, /* R1373 */ + { 0x0000, 0x0000, 0x0000 }, /* R1374 */ + { 0x0000, 0x0000, 0x0000 }, /* R1375 */ + { 0x0000, 0x0000, 0x0000 }, /* R1376 */ + { 0x0000, 0x0000, 0x0000 }, /* R1377 */ + { 0x0000, 0x0000, 0x0000 }, /* R1378 */ + { 0x0000, 0x0000, 0x0000 }, /* R1379 */ + { 0x0000, 0x0000, 0x0000 }, /* R1380 */ + { 0x0000, 0x0000, 0x0000 }, /* R1381 */ + { 0x0000, 0x0000, 0x0000 }, /* R1382 */ + { 0x0000, 0x0000, 0x0000 }, /* R1383 */ + { 0x0000, 0x0000, 0x0000 }, /* R1384 */ + { 0x0000, 0x0000, 0x0000 }, /* R1385 */ + { 0x0000, 0x0000, 0x0000 }, /* R1386 */ + { 0x0000, 0x0000, 0x0000 }, /* R1387 */ + { 0x0000, 0x0000, 0x0000 }, /* R1388 */ + { 0x0000, 0x0000, 0x0000 }, /* R1389 */ + { 0x0000, 0x0000, 0x0000 }, /* R1390 */ + { 0x0000, 0x0000, 0x0000 }, /* R1391 */ + { 0x0000, 0x0000, 0x0000 }, /* R1392 */ + { 0x0000, 0x0000, 0x0000 }, /* R1393 */ + { 0x0000, 0x0000, 0x0000 }, /* R1394 */ + { 0x0000, 0x0000, 0x0000 }, /* R1395 */ + { 0x0000, 0x0000, 0x0000 }, /* R1396 */ + { 0x0000, 0x0000, 0x0000 }, /* R1397 */ + { 0x0000, 0x0000, 0x0000 }, /* R1398 */ + { 0x0000, 0x0000, 0x0000 }, /* R1399 */ + { 0x0000, 0x0000, 0x0000 }, /* R1400 */ + { 0x0000, 0x0000, 0x0000 }, /* R1401 */ + { 0x0000, 0x0000, 0x0000 }, /* R1402 */ + { 0x0000, 0x0000, 0x0000 }, /* R1403 */ + { 0x0000, 0x0000, 0x0000 }, /* R1404 */ + { 0x0000, 0x0000, 0x0000 }, /* R1405 */ + { 0x0000, 0x0000, 0x0000 }, /* R1406 */ + { 0x0000, 0x0000, 0x0000 }, /* R1407 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1408 - AIF2 EQ Gains (1) */ + { 0xFFC0, 0xFFC0, 0x0000 }, /* R1409 - AIF2 EQ Gains (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1410 - AIF2 EQ Band 1 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1411 - AIF2 EQ Band 1 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1412 - AIF2 EQ Band 1 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1413 - AIF2 EQ Band 2 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1414 - AIF2 EQ Band 2 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1415 - AIF2 EQ Band 2 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1416 - AIF2 EQ Band 2 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1417 - AIF2 EQ Band 3 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1418 - AIF2 EQ Band 3 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1419 - AIF2 EQ Band 3 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1420 - AIF2 EQ Band 3 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1421 - AIF2 EQ Band 4 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1422 - AIF2 EQ Band 4 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1423 - AIF2 EQ Band 4 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1424 - AIF2 EQ Band 4 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1425 - AIF2 EQ Band 5 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1426 - AIF2 EQ Band 5 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1427 - AIF2 EQ Band 5 PG */ + { 0x0000, 0x0000, 0x0000 }, /* R1428 */ + { 0x0000, 0x0000, 0x0000 }, /* R1429 */ + { 0x0000, 0x0000, 0x0000 }, /* R1430 */ + { 0x0000, 0x0000, 0x0000 }, /* R1431 */ + { 0x0000, 0x0000, 0x0000 }, /* R1432 */ + { 0x0000, 0x0000, 0x0000 }, /* R1433 */ + { 0x0000, 0x0000, 0x0000 }, /* R1434 */ + { 0x0000, 0x0000, 0x0000 }, /* R1435 */ + { 0x0000, 0x0000, 0x0000 }, /* R1436 */ + { 0x0000, 0x0000, 0x0000 }, /* R1437 */ + { 0x0000, 0x0000, 0x0000 }, /* R1438 */ + { 0x0000, 0x0000, 0x0000 }, /* R1439 */ + { 0x0000, 0x0000, 0x0000 }, /* R1440 */ + { 0x0000, 0x0000, 0x0000 }, /* R1441 */ + { 0x0000, 0x0000, 0x0000 }, /* R1442 */ + { 0x0000, 0x0000, 0x0000 }, /* R1443 */ + { 0x0000, 0x0000, 0x0000 }, /* R1444 */ + { 0x0000, 0x0000, 0x0000 }, /* R1445 */ + { 0x0000, 0x0000, 0x0000 }, /* R1446 */ + { 0x0000, 0x0000, 0x0000 }, /* R1447 */ + { 0x0000, 0x0000, 0x0000 }, /* R1448 */ + { 0x0000, 0x0000, 0x0000 }, /* R1449 */ + { 0x0000, 0x0000, 0x0000 }, /* R1450 */ + { 0x0000, 0x0000, 0x0000 }, /* R1451 */ + { 0x0000, 0x0000, 0x0000 }, /* R1452 */ + { 0x0000, 0x0000, 0x0000 }, /* R1453 */ + { 0x0000, 0x0000, 0x0000 }, /* R1454 */ + { 0x0000, 0x0000, 0x0000 }, /* R1455 */ + { 0x0000, 0x0000, 0x0000 }, /* R1456 */ + { 0x0000, 0x0000, 0x0000 }, /* R1457 */ + { 0x0000, 0x0000, 0x0000 }, /* R1458 */ + { 0x0000, 0x0000, 0x0000 }, /* R1459 */ + { 0x0000, 0x0000, 0x0000 }, /* R1460 */ + { 0x0000, 0x0000, 0x0000 }, /* R1461 */ + { 0x0000, 0x0000, 0x0000 }, /* R1462 */ + { 0x0000, 0x0000, 0x0000 }, /* R1463 */ + { 0x0000, 0x0000, 0x0000 }, /* R1464 */ + { 0x0000, 0x0000, 0x0000 }, /* R1465 */ + { 0x0000, 0x0000, 0x0000 }, /* R1466 */ + { 0x0000, 0x0000, 0x0000 }, /* R1467 */ + { 0x0000, 0x0000, 0x0000 }, /* R1468 */ + { 0x0000, 0x0000, 0x0000 }, /* R1469 */ + { 0x0000, 0x0000, 0x0000 }, /* R1470 */ + { 0x0000, 0x0000, 0x0000 }, /* R1471 */ + { 0x0000, 0x0000, 0x0000 }, /* R1472 */ + { 0x0000, 0x0000, 0x0000 }, /* R1473 */ + { 0x0000, 0x0000, 0x0000 }, /* R1474 */ + { 0x0000, 0x0000, 0x0000 }, /* R1475 */ + { 0x0000, 0x0000, 0x0000 }, /* R1476 */ + { 0x0000, 0x0000, 0x0000 }, /* R1477 */ + { 0x0000, 0x0000, 0x0000 }, /* R1478 */ + { 0x0000, 0x0000, 0x0000 }, /* R1479 */ + { 0x0000, 0x0000, 0x0000 }, /* R1480 */ + { 0x0000, 0x0000, 0x0000 }, /* R1481 */ + { 0x0000, 0x0000, 0x0000 }, /* R1482 */ + { 0x0000, 0x0000, 0x0000 }, /* R1483 */ + { 0x0000, 0x0000, 0x0000 }, /* R1484 */ + { 0x0000, 0x0000, 0x0000 }, /* R1485 */ + { 0x0000, 0x0000, 0x0000 }, /* R1486 */ + { 0x0000, 0x0000, 0x0000 }, /* R1487 */ + { 0x0000, 0x0000, 0x0000 }, /* R1488 */ + { 0x0000, 0x0000, 0x0000 }, /* R1489 */ + { 0x0000, 0x0000, 0x0000 }, /* R1490 */ + { 0x0000, 0x0000, 0x0000 }, /* R1491 */ + { 0x0000, 0x0000, 0x0000 }, /* R1492 */ + { 0x0000, 0x0000, 0x0000 }, /* R1493 */ + { 0x0000, 0x0000, 0x0000 }, /* R1494 */ + { 0x0000, 0x0000, 0x0000 }, /* R1495 */ + { 0x0000, 0x0000, 0x0000 }, /* R1496 */ + { 0x0000, 0x0000, 0x0000 }, /* R1497 */ + { 0x0000, 0x0000, 0x0000 }, /* R1498 */ + { 0x0000, 0x0000, 0x0000 }, /* R1499 */ + { 0x0000, 0x0000, 0x0000 }, /* R1500 */ + { 0x0000, 0x0000, 0x0000 }, /* R1501 */ + { 0x0000, 0x0000, 0x0000 }, /* R1502 */ + { 0x0000, 0x0000, 0x0000 }, /* R1503 */ + { 0x0000, 0x0000, 0x0000 }, /* R1504 */ + { 0x0000, 0x0000, 0x0000 }, /* R1505 */ + { 0x0000, 0x0000, 0x0000 }, /* R1506 */ + { 0x0000, 0x0000, 0x0000 }, /* R1507 */ + { 0x0000, 0x0000, 0x0000 }, /* R1508 */ + { 0x0000, 0x0000, 0x0000 }, /* R1509 */ + { 0x0000, 0x0000, 0x0000 }, /* R1510 */ + { 0x0000, 0x0000, 0x0000 }, /* R1511 */ + { 0x0000, 0x0000, 0x0000 }, /* R1512 */ + { 0x0000, 0x0000, 0x0000 }, /* R1513 */ + { 0x0000, 0x0000, 0x0000 }, /* R1514 */ + { 0x0000, 0x0000, 0x0000 }, /* R1515 */ + { 0x0000, 0x0000, 0x0000 }, /* R1516 */ + { 0x0000, 0x0000, 0x0000 }, /* R1517 */ + { 0x0000, 0x0000, 0x0000 }, /* R1518 */ + { 0x0000, 0x0000, 0x0000 }, /* R1519 */ + { 0x0000, 0x0000, 0x0000 }, /* R1520 */ + { 0x0000, 0x0000, 0x0000 }, /* R1521 */ + { 0x0000, 0x0000, 0x0000 }, /* R1522 */ + { 0x0000, 0x0000, 0x0000 }, /* R1523 */ + { 0x0000, 0x0000, 0x0000 }, /* R1524 */ + { 0x0000, 0x0000, 0x0000 }, /* R1525 */ + { 0x0000, 0x0000, 0x0000 }, /* R1526 */ + { 0x0000, 0x0000, 0x0000 }, /* R1527 */ + { 0x0000, 0x0000, 0x0000 }, /* R1528 */ + { 0x0000, 0x0000, 0x0000 }, /* R1529 */ + { 0x0000, 0x0000, 0x0000 }, /* R1530 */ + { 0x0000, 0x0000, 0x0000 }, /* R1531 */ + { 0x0000, 0x0000, 0x0000 }, /* R1532 */ + { 0x0000, 0x0000, 0x0000 }, /* R1533 */ + { 0x0000, 0x0000, 0x0000 }, /* R1534 */ + { 0x0000, 0x0000, 0x0000 }, /* R1535 */ + { 0x01EF, 0x01EF, 0x0000 }, /* R1536 - DAC1 Mixer Volumes */ + { 0x0037, 0x0037, 0x0000 }, /* R1537 - DAC1 Left Mixer Routing */ + { 0x0037, 0x0037, 0x0000 }, /* R1538 - DAC1 Right Mixer Routing */ + { 0x01EF, 0x01EF, 0x0000 }, /* R1539 - DAC2 Mixer Volumes */ + { 0x0037, 0x0037, 0x0000 }, /* R1540 - DAC2 Left Mixer Routing */ + { 0x0037, 0x0037, 0x0000 }, /* R1541 - DAC2 Right Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1542 - AIF1 ADC1 Left Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1543 - AIF1 ADC1 Right Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1544 - AIF1 ADC2 Left Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1545 - AIF1 ADC2 Right mixer Routing */ + { 0x0000, 0x0000, 0x0000 }, /* R1546 */ + { 0x0000, 0x0000, 0x0000 }, /* R1547 */ + { 0x0000, 0x0000, 0x0000 }, /* R1548 */ + { 0x0000, 0x0000, 0x0000 }, /* R1549 */ + { 0x0000, 0x0000, 0x0000 }, /* R1550 */ + { 0x0000, 0x0000, 0x0000 }, /* R1551 */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1552 - DAC1 Left Volume */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1553 - DAC1 Right Volume */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1554 - DAC2 Left Volume */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1555 - DAC2 Right Volume */ + { 0x0003, 0x0003, 0x0000 }, /* R1556 - DAC Softmute */ + { 0x0000, 0x0000, 0x0000 }, /* R1557 */ + { 0x0000, 0x0000, 0x0000 }, /* R1558 */ + { 0x0000, 0x0000, 0x0000 }, /* R1559 */ + { 0x0000, 0x0000, 0x0000 }, /* R1560 */ + { 0x0000, 0x0000, 0x0000 }, /* R1561 */ + { 0x0000, 0x0000, 0x0000 }, /* R1562 */ + { 0x0000, 0x0000, 0x0000 }, /* R1563 */ + { 0x0000, 0x0000, 0x0000 }, /* R1564 */ + { 0x0000, 0x0000, 0x0000 }, /* R1565 */ + { 0x0000, 0x0000, 0x0000 }, /* R1566 */ + { 0x0000, 0x0000, 0x0000 }, /* R1567 */ + { 0x0003, 0x0003, 0x0000 }, /* R1568 - Oversampling */ + { 0x03C3, 0x03C3, 0x0000 }, /* R1569 - Sidetone */ +}; + +static int wm8994_readable(unsigned int reg) +{ + if (reg >= ARRAY_SIZE(access_masks)) + return 0; + return access_masks[reg].readable != 0; +} + +static int wm8994_volatile(unsigned int reg) +{ + if (reg >= WM8994_REG_CACHE_SIZE) + return 1; + + switch (reg) { + case WM8994_SOFTWARE_RESET: + case WM8994_CHIP_REVISION: + case WM8994_DC_SERVO_1: + case WM8994_DC_SERVO_READBACK: + case WM8994_RATE_STATUS: + case WM8994_LDO_1: + case WM8994_LDO_2: + return 1; + default: + return 0; + } +} + +static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct wm8994_priv *wm8994 = codec->private_data; + + BUG_ON(reg > WM8994_MAX_REGISTER); + + if (!wm8994_volatile(reg)) + wm8994->reg_cache[reg] = value; + + return wm8994_reg_write(codec->control_data, reg, value); +} + +static unsigned int wm8994_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *reg_cache = codec->reg_cache; + + BUG_ON(reg > WM8994_MAX_REGISTER); + + if (wm8994_volatile(reg)) + return wm8994_reg_read(codec->control_data, reg); + else + return reg_cache[reg]; +} + +static int configure_aif_clock(struct snd_soc_codec *codec, int aif) +{ + struct wm8994_priv *wm8994 = codec->private_data; + int rate; + int reg1 = 0; + int offset; + + if (aif) + offset = 4; + else + offset = 0; + + switch (wm8994->sysclk[aif]) { + case WM8994_SYSCLK_MCLK1: + rate = wm8994->mclk[0]; + break; + + case WM8994_SYSCLK_MCLK2: + reg1 |= 0x8; + rate = wm8994->mclk[1]; + break; + + case WM8994_SYSCLK_FLL1: + reg1 |= 0x10; + rate = wm8994->fll[0].out; + break; + + case WM8994_SYSCLK_FLL2: + reg1 |= 0x18; + rate = wm8994->fll[1].out; + break; + + default: + return -EINVAL; + } + + if (rate >= 13500000) { + rate /= 2; + reg1 |= WM8994_AIF1CLK_DIV; + + dev_dbg(codec->dev, "Dividing AIF%d clock to %dHz\n", + aif + 1, rate); + } + wm8994->aifclk[aif] = rate; + + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset, + WM8994_AIF1CLK_SRC_MASK | WM8994_AIF1CLK_DIV, + reg1); + + return 0; +} + +static int configure_clock(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = codec->private_data; + int old, new; + + /* Bring up the AIF clocks first */ + configure_aif_clock(codec, 0); + configure_aif_clock(codec, 1); + + /* Then switch CLK_SYS over to the higher of them; a change + * can only happen as a result of a clocking change which can + * only be made outside of DAPM so we can safely redo the + * clocking. + */ + + /* If they're equal it doesn't matter which is used */ + if (wm8994->aifclk[0] == wm8994->aifclk[1]) + return 0; + + if (wm8994->aifclk[0] < wm8994->aifclk[1]) + new = WM8994_SYSCLK_SRC; + else + new = 0; + + old = snd_soc_read(codec, WM8994_CLOCKING_1) & WM8994_SYSCLK_SRC; + + /* If there's no change then we're done. */ + if (old == new) + return 0; + + snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int check_clk_sys(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, WM8994_CLOCKING_1); + const char *clk; + + /* Check what we're currently using for CLK_SYS */ + if (reg & WM8994_SYSCLK_SRC) + clk = "AIF2CLK"; + else + clk = "AIF1CLK"; + + return strcmp(source->name, clk) == 0; +} + +static const char *sidetone_hpf_text[] = { + "2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz" +}; + +static const struct soc_enum sidetone_hpf = + SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text); + +static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); +static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); + +#define WM8994_DRC_SWITCH(xname, reg, shift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ + .put = wm8994_put_drc_sw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, 1, 0) } + +static int wm8994_put_drc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int mask, ret; + + /* Can't enable both ADC and DAC paths simultaneously */ + if (mc->shift == WM8994_AIF1DAC1_DRC_ENA_SHIFT) + mask = WM8994_AIF1ADC1L_DRC_ENA_MASK | + WM8994_AIF1ADC1R_DRC_ENA_MASK; + else + mask = WM8994_AIF1DAC1_DRC_ENA_MASK; + + ret = snd_soc_read(codec, mc->reg); + if (ret < 0) + return ret; + if (ret & mask) + return -EINVAL; + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + + + +static void wm8994_set_drc(struct snd_soc_codec *codec, int drc) +{ + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int base = wm8994_drc_base[drc]; + int cfg = wm8994->drc_cfg[drc]; + int save, i; + + /* Save any enables; the configuration should clear them. */ + save = snd_soc_read(codec, base); + save &= WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA; + + for (i = 0; i < WM8994_DRC_REGS; i++) + snd_soc_update_bits(codec, base + i, 0xffff, + pdata->drc_cfgs[cfg].regs[i]); + + snd_soc_update_bits(codec, base, WM8994_AIF1DAC1_DRC_ENA | + WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA, save); +} + +/* Icky as hell but saves code duplication */ +static int wm8994_get_drc(const char *name) +{ + if (strcmp(name, "AIF1DRC1 Mode") == 0) + return 0; + if (strcmp(name, "AIF1DRC2 Mode") == 0) + return 1; + if (strcmp(name, "AIF2DRC Mode") == 0) + return 2; + return -EINVAL; +} + +static int wm8994_put_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int drc = wm8994_get_drc(kcontrol->id.name); + int value = ucontrol->value.integer.value[0]; + + if (drc < 0) + return drc; + + if (value >= pdata->num_drc_cfgs) + return -EINVAL; + + wm8994->drc_cfg[drc] = value; + + wm8994_set_drc(codec, drc); + + return 0; +} + +static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + int drc = wm8994_get_drc(kcontrol->id.name); + + ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc]; + + return 0; +} + +static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block) +{ + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int base = wm8994_retune_mobile_base[block]; + int iface, best, best_val, save, i, cfg; + + if (!pdata || !wm8994->num_retune_mobile_texts) + return; + + switch (block) { + case 0: + case 1: + iface = 0; + break; + case 2: + iface = 1; + break; + default: + return; + } + + /* Find the version of the currently selected configuration + * with the nearest sample rate. */ + cfg = wm8994->retune_mobile_cfg[block]; + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8994->retune_mobile_texts[cfg]) == 0 && + abs(pdata->retune_mobile_cfgs[i].rate + - wm8994->dac_rates[iface]) < best_val) { + best = i; + best_val = abs(pdata->retune_mobile_cfgs[i].rate + - wm8994->dac_rates[iface]); + } + } + + dev_dbg(codec->dev, "ReTune Mobile %d %s/%dHz for %dHz sample rate\n", + block, + pdata->retune_mobile_cfgs[best].name, + pdata->retune_mobile_cfgs[best].rate, + wm8994->dac_rates[iface]); + + /* The EQ will be disabled while reconfiguring it, remember the + * current configuration. + */ + save = snd_soc_read(codec, base); + save &= WM8994_AIF1DAC1_EQ_ENA; + + for (i = 0; i < WM8994_EQ_REGS; i++) + snd_soc_update_bits(codec, base + i, 0xffff, + pdata->retune_mobile_cfgs[best].regs[i]); + + snd_soc_update_bits(codec, base, WM8994_AIF1DAC1_EQ_ENA, save); +} + +/* Icky as hell but saves code duplication */ +static int wm8994_get_retune_mobile_block(const char *name) +{ + if (strcmp(name, "AIF1.1 EQ Mode") == 0) + return 0; + if (strcmp(name, "AIF1.2 EQ Mode") == 0) + return 1; + if (strcmp(name, "AIF2 EQ Mode") == 0) + return 2; + return -EINVAL; +} + +static int wm8994_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int block = wm8994_get_retune_mobile_block(kcontrol->id.name); + int value = ucontrol->value.integer.value[0]; + + if (block < 0) + return block; + + if (value >= pdata->num_retune_mobile_cfgs) + return -EINVAL; + + wm8994->retune_mobile_cfg[block] = value; + + wm8994_set_retune_mobile(codec, block); + + return 0; +} + +static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + int block = wm8994_get_retune_mobile_block(kcontrol->id.name); + + ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; + + return 0; +} + +static const struct snd_kcontrol_new wm8994_snd_controls[] = { +SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, + WM8994_AIF1_ADC1_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1_ADC2_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, + WM8994_AIF2_ADC_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), + +SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, + WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME, + WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv), + +SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv), +SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv), + +SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0), +SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0), +SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0), + +WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2), +WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1), +WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0), + +WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2), +WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1), +WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0), + +WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2), +WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1), +WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0), + +SOC_SINGLE_TLV("DAC1 Right Sidetone Volume", WM8994_DAC1_MIXER_VOLUMES, + 5, 12, 0, st_tlv), +SOC_SINGLE_TLV("DAC1 Left Sidetone Volume", WM8994_DAC1_MIXER_VOLUMES, + 0, 12, 0, st_tlv), +SOC_SINGLE_TLV("DAC2 Right Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES, + 5, 12, 0, st_tlv), +SOC_SINGLE_TLV("DAC2 Left Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES, + 0, 12, 0, st_tlv), +SOC_ENUM("Sidetone HPF Mux", sidetone_hpf), +SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0), + +SOC_DOUBLE_R_TLV("DAC1 Volume", WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R("DAC1 Switch", WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_RIGHT_VOLUME, 9, 1, 1), + +SOC_DOUBLE_R_TLV("DAC2 Volume", WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R("DAC2 Switch", WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_RIGHT_VOLUME, 9, 1, 1), + +SOC_SINGLE_TLV("SPKL DAC2 Volume", WM8994_SPKMIXL_ATTENUATION, + 6, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKL DAC1 Volume", WM8994_SPKMIXL_ATTENUATION, + 2, 1, 1, wm_hubs_spkmix_tlv), + +SOC_SINGLE_TLV("SPKR DAC2 Volume", WM8994_SPKMIXR_ATTENUATION, + 6, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKR DAC1 Volume", WM8994_SPKMIXR_ATTENUATION, + 2, 1, 1, wm_hubs_spkmix_tlv), + +SOC_SINGLE_TLV("AIF1DAC1 3D Stereo Volume", WM8994_AIF1_DAC1_FILTERS_2, + 10, 15, 0, wm8994_3d_tlv), +SOC_SINGLE("AIF1DAC1 3D Stereo Switch", WM8994_AIF1_DAC2_FILTERS_2, + 8, 1, 0), +SOC_SINGLE_TLV("AIF1DAC2 3D Stereo Volume", WM8994_AIF1_DAC2_FILTERS_2, + 10, 15, 0, wm8994_3d_tlv), +SOC_SINGLE("AIF1DAC2 3D Stereo Switch", WM8994_AIF1_DAC2_FILTERS_2, + 8, 1, 0), +SOC_SINGLE_TLV("AIF2DAC 3D Stereo Volume", WM8994_AIF1_DAC1_FILTERS_2, + 10, 15, 0, wm8994_3d_tlv), +SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF1_DAC2_FILTERS_2, + 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8994_eq_controls[] = { +SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ2 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ3 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ4 Volume", WM8994_AIF1_DAC1_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ5 Volume", WM8994_AIF1_DAC1_EQ_GAINS_2, 6, 31, 0, + eq_tlv), + +SOC_SINGLE_TLV("AIF1DAC2 EQ1 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ2 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ3 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ4 Volume", WM8994_AIF1_DAC2_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ5 Volume", WM8994_AIF1_DAC2_EQ_GAINS_2, 6, 31, 0, + eq_tlv), + +SOC_SINGLE_TLV("AIF2 EQ1 Volume", WM8994_AIF2_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ2 Volume", WM8994_AIF2_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ3 Volume", WM8994_AIF2_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ4 Volume", WM8994_AIF2_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, + eq_tlv), +}; + +static int clk_sys_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return configure_clock(codec); + + case SND_SOC_DAPM_POST_PMD: + configure_clock(codec); + break; + } + + return 0; +} + +static void wm8994_update_class_w(struct snd_soc_codec *codec) +{ + int enable = 1; + int source = 0; /* GCC flow analysis can't track enable */ + int reg, reg_r; + + /* Only support direct DAC->headphone paths */ + reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1); + if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) { + dev_dbg(codec->dev, "HPL connected to output mixer\n"); + enable = 0; + } + + reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2); + if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) { + dev_dbg(codec->dev, "HPR connected to output mixer\n"); + enable = 0; + } + + /* We also need the same setting for L/R and only one path */ + reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING); + switch (reg) { + case WM8994_AIF2DACL_TO_DAC1L: + dev_dbg(codec->dev, "Class W source AIF2DAC\n"); + source = 2 << WM8994_CP_DYN_SRC_SEL_SHIFT; + break; + case WM8994_AIF1DAC2L_TO_DAC1L: + dev_dbg(codec->dev, "Class W source AIF1DAC2\n"); + source = 1 << WM8994_CP_DYN_SRC_SEL_SHIFT; + break; + case WM8994_AIF1DAC1L_TO_DAC1L: + dev_dbg(codec->dev, "Class W source AIF1DAC1\n"); + source = 0 << WM8994_CP_DYN_SRC_SEL_SHIFT; + break; + default: + dev_dbg(codec->dev, "DAC mixer setting: %x\n", reg); + enable = 0; + break; + } + + reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING); + if (reg_r != reg) { + dev_dbg(codec->dev, "Left and right DAC mixers different\n"); + enable = 0; + } + + if (enable) { + dev_dbg(codec->dev, "Class W enabled\n"); + snd_soc_update_bits(codec, WM8994_CLASS_W_1, + WM8994_CP_DYN_PWR | + WM8994_CP_DYN_SRC_SEL_MASK, + source | WM8994_CP_DYN_PWR); + + } else { + dev_dbg(codec->dev, "Class W disabled\n"); + snd_soc_update_bits(codec, WM8994_CLASS_W_1, + WM8994_CP_DYN_PWR, 0); + } +} + +static const char *hp_mux_text[] = { + "Mixer", + "DAC", +}; + +#define WM8994_HP_ENUM(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = wm8994_put_hp_enum, \ + .private_value = (unsigned long)&xenum } + +static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = w->codec; + int ret; + + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + wm8994_update_class_w(codec); + + return ret; +} + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_1, 8, 2, hp_mux_text); + +static const struct snd_kcontrol_new hpl_mux = + WM8994_HP_ENUM("Left Headphone Mux", hpl_enum); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_2, 8, 2, hp_mux_text); + +static const struct snd_kcontrol_new hpr_mux = + WM8994_HP_ENUM("Right Headphone Mux", hpr_enum); + +static const char *adc_mux_text[] = { + "ADC", + "DMIC", +}; + +static const struct soc_enum adc_enum = + SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text); + +static const struct snd_kcontrol_new adcl_mux = + SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); + +static const struct snd_kcontrol_new adcr_mux = + SOC_DAPM_ENUM_VIRT("ADCR Mux", adc_enum); + +static const struct snd_kcontrol_new left_speaker_mixer[] = { +SOC_DAPM_SINGLE("DAC2 Switch", WM8994_SPEAKER_MIXER, 9, 1, 0), +SOC_DAPM_SINGLE("Input Switch", WM8994_SPEAKER_MIXER, 7, 1, 0), +SOC_DAPM_SINGLE("IN1LP Switch", WM8994_SPEAKER_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8994_SPEAKER_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 1, 1, 0), +}; + +static const struct snd_kcontrol_new right_speaker_mixer[] = { +SOC_DAPM_SINGLE("DAC2 Switch", WM8994_SPEAKER_MIXER, 8, 1, 0), +SOC_DAPM_SINGLE("Input Switch", WM8994_SPEAKER_MIXER, 6, 1, 0), +SOC_DAPM_SINGLE("IN1RP Switch", WM8994_SPEAKER_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8994_SPEAKER_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 0, 1, 0), +}; + +/* Debugging; dump chip status after DAPM transitions */ +static int post_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + dev_dbg(codec->dev, "SRC status: %x\n", + snd_soc_read(codec, + WM8994_RATE_STATUS)); + return 0; +} + +static const struct snd_kcontrol_new aif1adc1l_mix[] = { +SOC_DAPM_SINGLE("ADC/DMIC Switch", WM8994_AIF1_ADC1_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif1adc1r_mix[] = { +SOC_DAPM_SINGLE("ADC/DMIC Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif2dac2l_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 2, 1, 0), +SOC_DAPM_SINGLE("AIF1.2 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif2dac2r_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 2, 1, 0), +SOC_DAPM_SINGLE("AIF1.2 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + +#define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = wm8994_put_class_w, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = w->codec; + int ret; + + ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + wm8994_update_class_w(codec); + + return ret; +} + +static const struct snd_kcontrol_new dac1l_mix[] = { +WM8994_CLASS_W_SWITCH("Right Sidetone Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 5, 1, 0), +WM8994_CLASS_W_SWITCH("Left Sidetone Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 4, 1, 0), +WM8994_CLASS_W_SWITCH("AIF2 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 2, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.2 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 1, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.1 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new dac1r_mix[] = { +WM8994_CLASS_W_SWITCH("Right Sidetone Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 5, 1, 0), +WM8994_CLASS_W_SWITCH("Left Sidetone Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 4, 1, 0), +WM8994_CLASS_W_SWITCH("AIF2 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 2, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.2 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 1, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.1 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const char *sidetone_text[] = { + "ADC/DMIC1", "DMIC2", +}; + +static const struct soc_enum sidetone1_enum = + SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text); + +static const struct snd_kcontrol_new sidetone1_mux = + SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum); + +static const struct soc_enum sidetone2_enum = + SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text); + +static const struct snd_kcontrol_new sidetone2_mux = + SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum); + +static const char *aif1dac_text[] = { + "AIF1DACDAT", "AIF3DACDAT", +}; + +static const struct soc_enum aif1dac_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); + +static const struct snd_kcontrol_new aif1dac_mux = + SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum); + +static const char *aif2dac_text[] = { + "AIF2DACDAT", "AIF3DACDAT", +}; + +static const struct soc_enum aif2dac_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text); + +static const struct snd_kcontrol_new aif2dac_mux = + SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum); + +static const char *aif2adc_text[] = { + "AIF2ADCDAT", "AIF3DACDAT", +}; + +static const struct soc_enum aif2adc_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text); + +static const struct snd_kcontrol_new aif2adc_mux = + SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum); + +static const char *aif3adc_text[] = { + "AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", +}; + +static const struct soc_enum aif3adc_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text); + +static const struct snd_kcontrol_new aif3adc_mux = + SOC_DAPM_ENUM("AIF3ADC Mux", aif3adc_enum); + +static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("DMIC1DAT"), +SND_SOC_DAPM_INPUT("DMIC2DAT"), + +SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 9, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 8, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC1L", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 9, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC1R", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 8, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 11, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 10, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC2L", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 11, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC2R", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 10, 0), + +SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, + aif1adc1l_mix, ARRAY_SIZE(aif1adc1l_mix)), +SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0, + aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)), + +SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0, + aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)), +SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0, + aif2dac2r_mix, ARRAY_SIZE(aif2dac2r_mix)), + +SND_SOC_DAPM_MUX("Left Sidetone", SND_SOC_NOPM, 0, 0, &sidetone1_mux), +SND_SOC_DAPM_MUX("Right Sidetone", SND_SOC_NOPM, 0, 0, &sidetone2_mux), + +SND_SOC_DAPM_MIXER("DAC1L Mixer", SND_SOC_NOPM, 0, 0, + dac1l_mix, ARRAY_SIZE(dac1l_mix)), +SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, + dac1r_mix, ARRAY_SIZE(dac1r_mix)), + +SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0, + WM8994_POWER_MANAGEMENT_4, 13, 0), +SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0, + WM8994_POWER_MANAGEMENT_4, 12, 0), +SND_SOC_DAPM_AIF_IN("AIF2DACL", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 13, 0), +SND_SOC_DAPM_AIF_IN("AIF2DACR", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 12, 0), + +SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), +SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux), +SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux), +SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &aif3adc_mux), + +SND_SOC_DAPM_AIF_IN("AIF3DACDAT", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0), + +SND_SOC_DAPM_ADC("DMIC2L", NULL, WM8994_POWER_MANAGEMENT_4, 5, 0), +SND_SOC_DAPM_ADC("DMIC2R", NULL, WM8994_POWER_MANAGEMENT_4, 4, 0), +SND_SOC_DAPM_ADC("DMIC1L", NULL, WM8994_POWER_MANAGEMENT_4, 3, 0), +SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), + +/* Power is done with the muxes since the ADC power also controls the + * downsampling chain, the chip will automatically manage the analogue + * specific portions. + */ +SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), +SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), + +SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0), +SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), +SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), +SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), + +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), + +SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), +SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), + +SND_SOC_DAPM_POST("Debug log", post_ev), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + { "CLK_SYS", NULL, "AIF1CLK", check_clk_sys }, + { "CLK_SYS", NULL, "AIF2CLK", check_clk_sys }, + + { "DSP1CLK", NULL, "CLK_SYS" }, + { "DSP2CLK", NULL, "CLK_SYS" }, + { "DSPINTCLK", NULL, "CLK_SYS" }, + + { "AIF1ADC1L", NULL, "AIF1CLK" }, + { "AIF1ADC1L", NULL, "DSP1CLK" }, + { "AIF1ADC1R", NULL, "AIF1CLK" }, + { "AIF1ADC1R", NULL, "DSP1CLK" }, + { "AIF1ADC1R", NULL, "DSPINTCLK" }, + + { "AIF1DAC1L", NULL, "AIF1CLK" }, + { "AIF1DAC1L", NULL, "DSP1CLK" }, + { "AIF1DAC1R", NULL, "AIF1CLK" }, + { "AIF1DAC1R", NULL, "DSP1CLK" }, + { "AIF1DAC1R", NULL, "DSPINTCLK" }, + + { "AIF1ADC2L", NULL, "AIF1CLK" }, + { "AIF1ADC2L", NULL, "DSP1CLK" }, + { "AIF1ADC2R", NULL, "AIF1CLK" }, + { "AIF1ADC2R", NULL, "DSP1CLK" }, + { "AIF1ADC2R", NULL, "DSPINTCLK" }, + + { "AIF1DAC2L", NULL, "AIF1CLK" }, + { "AIF1DAC2L", NULL, "DSP1CLK" }, + { "AIF1DAC2R", NULL, "AIF1CLK" }, + { "AIF1DAC2R", NULL, "DSP1CLK" }, + { "AIF1DAC2R", NULL, "DSPINTCLK" }, + + { "AIF2ADCL", NULL, "AIF2CLK" }, + { "AIF2ADCL", NULL, "DSP2CLK" }, + { "AIF2ADCR", NULL, "AIF2CLK" }, + { "AIF2ADCR", NULL, "DSP2CLK" }, + { "AIF2ADCR", NULL, "DSPINTCLK" }, + + { "AIF2DACL", NULL, "AIF2CLK" }, + { "AIF2DACL", NULL, "DSP2CLK" }, + { "AIF2DACR", NULL, "AIF2CLK" }, + { "AIF2DACR", NULL, "DSP2CLK" }, + { "AIF2DACR", NULL, "DSPINTCLK" }, + + { "DMIC1L", NULL, "DMIC1DAT" }, + { "DMIC1L", NULL, "CLK_SYS" }, + { "DMIC1R", NULL, "DMIC1DAT" }, + { "DMIC1R", NULL, "CLK_SYS" }, + { "DMIC2L", NULL, "DMIC2DAT" }, + { "DMIC2L", NULL, "CLK_SYS" }, + { "DMIC2R", NULL, "DMIC2DAT" }, + { "DMIC2R", NULL, "CLK_SYS" }, + + { "ADCL", NULL, "AIF1CLK" }, + { "ADCL", NULL, "DSP1CLK" }, + { "ADCL", NULL, "DSPINTCLK" }, + + { "ADCR", NULL, "AIF1CLK" }, + { "ADCR", NULL, "DSP1CLK" }, + { "ADCR", NULL, "DSPINTCLK" }, + + { "ADCL Mux", "ADC", "ADCL" }, + { "ADCL Mux", "DMIC", "DMIC1L" }, + { "ADCR Mux", "ADC", "ADCR" }, + { "ADCR Mux", "DMIC", "DMIC1R" }, + + { "DAC1L", NULL, "AIF1CLK" }, + { "DAC1L", NULL, "DSP1CLK" }, + { "DAC1L", NULL, "DSPINTCLK" }, + + { "DAC1R", NULL, "AIF1CLK" }, + { "DAC1R", NULL, "DSP1CLK" }, + { "DAC1R", NULL, "DSPINTCLK" }, + + { "DAC2L", NULL, "AIF2CLK" }, + { "DAC2L", NULL, "DSP2CLK" }, + { "DAC2L", NULL, "DSPINTCLK" }, + + { "DAC2R", NULL, "AIF2DACR" }, + { "DAC2R", NULL, "AIF2CLK" }, + { "DAC2R", NULL, "DSP2CLK" }, + { "DAC2R", NULL, "DSPINTCLK" }, + + { "TOCLK", NULL, "CLK_SYS" }, + + /* AIF1 outputs */ + { "AIF1ADC1L", NULL, "AIF1ADC1L Mixer" }, + { "AIF1ADC1L Mixer", "ADC/DMIC Switch", "ADCL Mux" }, + { "AIF1ADC1L Mixer", "AIF2 Switch", "AIF2DACL" }, + + { "AIF1ADC1R", NULL, "AIF1ADC1R Mixer" }, + { "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" }, + { "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" }, + + /* Pin level routing for AIF3 */ + { "AIF1DAC1L", NULL, "AIF1DAC Mux" }, + { "AIF1DAC1R", NULL, "AIF1DAC Mux" }, + { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, + { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, + + { "AIF2DACL", NULL, "AIF2DAC Mux" }, + { "AIF2DACR", NULL, "AIF2DAC Mux" }, + + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, + { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, + + /* DAC1 inputs */ + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, + { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, + { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, + { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, + { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, + { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, + { "DAC1R Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "DAC1R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + /* DAC2/AIF2 outputs */ + { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, + { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, + { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, + { "AIF2DAC2L Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, + { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, + { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, + { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, + { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, + + /* AIF3 output */ + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1L" }, + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1R" }, + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2L" }, + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2R" }, + { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, + { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + + /* Sidetone */ + { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, + { "Left Sidetone", "DMIC2", "DMIC2L" }, + { "Right Sidetone", "ADC/DMIC1", "ADCR Mux" }, + { "Right Sidetone", "DMIC2", "DMIC2R" }, + + /* Output stages */ + { "Left Output Mixer", "DAC Switch", "DAC1L" }, + { "Right Output Mixer", "DAC Switch", "DAC1R" }, + + { "SPKL", "DAC1 Switch", "DAC1L" }, + { "SPKL", "DAC2 Switch", "DAC2L" }, + + { "SPKR", "DAC1 Switch", "DAC1R" }, + { "SPKR", "DAC2 Switch", "DAC2R" }, + + { "Left Headphone Mux", "DAC", "DAC1L" }, + { "Right Headphone Mux", "DAC", "DAC1R" }, +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +struct fll_div { + u16 outdiv; + u16 n; + u16 k; + u16 clk_ref_div; + u16 fll_fratio; +}; + +static int wm8994_get_fll_config(struct fll_div *fll, + int freq_in, int freq_out) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + pr_debug("FLL input=%dHz, output=%dHz\n", freq_in, freq_out); + + /* Scale the input frequency down to <= 13.5MHz */ + fll->clk_ref_div = 0; + while (freq_in > 13500000) { + fll->clk_ref_div++; + freq_in /= 2; + + if (fll->clk_ref_div > 3) + return -EINVAL; + } + pr_debug("CLK_REF_DIV=%d, Fref=%dHz\n", fll->clk_ref_div, freq_in); + + /* Scale the output to give 90MHz<=Fvco<=100MHz */ + fll->outdiv = 3; + while (freq_out * (fll->outdiv + 1) < 90000000) { + fll->outdiv++; + if (fll->outdiv > 63) + return -EINVAL; + } + freq_out *= fll->outdiv + 1; + pr_debug("OUTDIV=%d, Fvco=%dHz\n", fll->outdiv, freq_out); + + if (freq_in > 1000000) { + fll->fll_fratio = 0; + } else { + fll->fll_fratio = 3; + freq_in *= 8; + } + pr_debug("FLL_FRATIO=%d, Fref=%dHz\n", fll->fll_fratio, freq_in); + + /* Now, calculate N.K */ + Ndiv = freq_out / freq_in; + + fll->n = Ndiv; + Nmod = freq_out % freq_in; + pr_debug("Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, freq_in); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll->k = K / 10; + + pr_debug("N=%x K=%x\n", fll->n, fll->k); + + return 0; +} + +static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8994_priv *wm8994 = codec->private_data; + int reg_offset, ret; + struct fll_div fll; + u16 reg, aif1, aif2; + + aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) + & WM8994_AIF1CLK_ENA; + + aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) + & WM8994_AIF2CLK_ENA; + + switch (id) { + case WM8994_FLL1: + reg_offset = 0; + id = 0; + break; + case WM8994_FLL2: + reg_offset = 0x20; + id = 1; + break; + default: + return -EINVAL; + } + + /* Are we changing anything? */ + if (wm8994->fll[id].src == src && + wm8994->fll[id].in == freq_in && wm8994->fll[id].out == freq_out) + return 0; + + /* If we're stopping the FLL redo the old config - no + * registers will actually be written but we avoid GCC flow + * analysis bugs spewing warnings. + */ + if (freq_out) + ret = wm8994_get_fll_config(&fll, freq_in, freq_out); + else + ret = wm8994_get_fll_config(&fll, wm8994->fll[id].in, + wm8994->fll[id].out); + if (ret < 0) + return ret; + + /* Gate the AIF clocks while we reclock */ + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA, 0); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA, 0); + + /* We always need to disable the FLL while reconfiguring */ + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, + WM8994_FLL1_ENA, 0); + + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | + (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, + WM8994_FLL1_OUTDIV_MASK | + WM8994_FLL1_FRATIO_MASK, reg); + + snd_soc_write(codec, WM8994_FLL1_CONTROL_3 + reg_offset, fll.k); + + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset, + WM8994_FLL1_N_MASK, + fll.n << WM8994_FLL1_N_SHIFT); + + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8994_FLL1_REFCLK_DIV_MASK, + fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT); + + /* Enable (with fractional mode if required) */ + if (freq_out) { + if (fll.k) + reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC; + else + reg = WM8994_FLL1_ENA; + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, + WM8994_FLL1_ENA | WM8994_FLL1_FRAC, + reg); + } + + wm8994->fll[id].in = freq_in; + wm8994->fll[id].out = freq_out; + + /* Enable any gated AIF clocks */ + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA, aif1); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA, aif2); + + configure_clock(codec); + + return 0; +} + +static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8994_priv *wm8994 = codec->private_data; + + switch (dai->id) { + case 1: + case 2: + break; + + default: + /* AIF3 shares clocking with AIF1/2 */ + return -EINVAL; + } + + switch (clk_id) { + case WM8994_SYSCLK_MCLK1: + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK1; + wm8994->mclk[0] = freq; + dev_dbg(dai->dev, "AIF%d using MCLK1 at %uHz\n", + dai->id, freq); + break; + + case WM8994_SYSCLK_MCLK2: + /* TODO: Set GPIO AF */ + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK2; + wm8994->mclk[1] = freq; + dev_dbg(dai->dev, "AIF%d using MCLK2 at %uHz\n", + dai->id, freq); + break; + + case WM8994_SYSCLK_FLL1: + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_FLL1; + dev_dbg(dai->dev, "AIF%d using FLL1\n", dai->id); + break; + + case WM8994_SYSCLK_FLL2: + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_FLL2; + dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id); + break; + + default: + return -EINVAL; + } + + configure_clock(codec); + + return 0; +} + +static int wm8994_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_VMID_SEL_MASK, 0x2); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Tweak DC servo configuration for improved + * performance. */ + snd_soc_write(codec, 0x102, 0x3); + snd_soc_write(codec, 0x56, 0x3); + snd_soc_write(codec, 0x102, 0); + + /* Discharge LINEOUT1 & 2 */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); + + /* Startup bias, VMID ramp & buffer */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (0x11 << WM8994_VMID_RAMP_SHIFT)); + + /* Main bias enable, VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, + WM8994_BIAS_ENA | 0x2); + + msleep(20); + } + + /* VMID=2x500k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_VMID_SEL_MASK, 0x4); + + break; + + case SND_SOC_BIAS_OFF: + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); + + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0); + + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); + + msleep(5); + + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + + break; + } + codec->bias_level = level; + return 0; +} + +static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int ms_reg; + int aif1_reg; + int ms = 0; + int aif1 = 0; + + switch (dai->id) { + case 1: + ms_reg = WM8994_AIF1_MASTER_SLAVE; + aif1_reg = WM8994_AIF1_CONTROL_1; + break; + case 2: + ms_reg = WM8994_AIF2_MASTER_SLAVE; + aif1_reg = WM8994_AIF2_CONTROL_1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + ms = WM8994_AIF1_MSTR; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif1 |= WM8994_AIF1_LRCLK_INV; + case SND_SOC_DAIFMT_DSP_A: + aif1 |= 0x18; + break; + case SND_SOC_DAIFMT_I2S: + aif1 |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif1 |= 0x8; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8994_AIF1_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8994_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif1 |= WM8994_AIF1_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, aif1_reg, + WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV | + WM8994_AIF1_FMT_MASK, + aif1); + snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, + ms); + + return 0; +} + +static struct { + int val, rate; +} srs[] = { + { 0, 8000 }, + { 1, 11025 }, + { 2, 12000 }, + { 3, 16000 }, + { 4, 22050 }, + { 5, 24000 }, + { 6, 32000 }, + { 7, 44100 }, + { 8, 48000 }, + { 9, 88200 }, + { 10, 96000 }, +}; + +static int fs_ratios[] = { + 64, 128, 192, 256, 348, 512, 768, 1024, 1408, 1536 +}; + +static int bclk_divs[] = { + 10, 15, 20, 30, 40, 50, 60, 80, 110, 120, 160, 220, 240, 320, 440, 480, + 640, 880, 960, 1280, 1760, 1920 +}; + +static int wm8994_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8994_priv *wm8994 = codec->private_data; + int aif1_reg; + int bclk_reg; + int lrclk_reg; + int rate_reg; + int aif1 = 0; + int bclk = 0; + int lrclk = 0; + int rate_val = 0; + int id = dai->id - 1; + + int i, cur_val, best_val, bclk_rate, best; + + switch (dai->id) { + case 1: + aif1_reg = WM8994_AIF1_CONTROL_1; + bclk_reg = WM8994_AIF1_BCLK; + rate_reg = WM8994_AIF1_RATE; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + wm8994->lrclk_shared[0]) + lrclk_reg = WM8994_AIF1DAC_LRCLK; + else + lrclk_reg = WM8994_AIF1ADC_LRCLK; + break; + case 2: + aif1_reg = WM8994_AIF2_CONTROL_1; + bclk_reg = WM8994_AIF2_BCLK; + rate_reg = WM8994_AIF2_RATE; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + wm8994->lrclk_shared[1]) + lrclk_reg = WM8994_AIF2DAC_LRCLK; + else + lrclk_reg = WM8994_AIF2ADC_LRCLK; + break; + default: + return -EINVAL; + } + + bclk_rate = params_rate(params) * 2; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + bclk_rate *= 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + bclk_rate *= 20; + aif1 |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bclk_rate *= 24; + aif1 |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bclk_rate *= 32; + aif1 |= 0x60; + break; + default: + return -EINVAL; + } + + /* Try to find an appropriate sample rate; look for an exact match. */ + for (i = 0; i < ARRAY_SIZE(srs); i++) + if (srs[i].rate == params_rate(params)) + break; + if (i == ARRAY_SIZE(srs)) + return -EINVAL; + rate_val |= srs[i].val << WM8994_AIF1_SR_SHIFT; + + dev_dbg(dai->dev, "Sample rate is %dHz\n", srs[i].rate); + dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n", + dai->id, wm8994->aifclk[id], bclk_rate); + + if (wm8994->aifclk[id] == 0) { + dev_err(dai->dev, "AIF%dCLK not configured\n", dai->id); + return -EINVAL; + } + + /* AIFCLK/fs ratio; look for a close match in either direction */ + best = 0; + best_val = abs((fs_ratios[0] * params_rate(params)) + - wm8994->aifclk[id]); + for (i = 1; i < ARRAY_SIZE(fs_ratios); i++) { + cur_val = abs((fs_ratios[i] * params_rate(params)) + - wm8994->aifclk[id]); + if (cur_val >= best_val) + continue; + best = i; + best_val = cur_val; + } + dev_dbg(dai->dev, "Selected AIF%dCLK/fs = %d\n", + dai->id, fs_ratios[best]); + rate_val |= best; + + /* We may not get quite the right frequency if using + * approximate clocks so look for the closest match that is + * higher than the target (we need to ensure that there enough + * BCLKs to clock out the samples). + */ + best = 0; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + if (bclk_divs[i] < 0) + continue; + cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) + - bclk_rate * 10; + if (cur_val < 0) /* BCLK table is sorted */ + break; + best = i; + } + bclk_rate = wm8994->aifclk[id] / bclk_divs[best]; + dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", + bclk_divs[best], bclk_rate); + bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; + + lrclk = bclk_rate / params_rate(params); + dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", + lrclk, bclk_rate / lrclk); + + snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); + snd_soc_update_bits(codec, bclk_reg, WM8994_AIF1_BCLK_DIV_MASK, bclk); + snd_soc_update_bits(codec, lrclk_reg, WM8994_AIF1DAC_RATE_MASK, + lrclk); + snd_soc_update_bits(codec, rate_reg, WM8994_AIF1_SR_MASK | + WM8994_AIF1CLK_RATE_MASK, rate_val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (dai->id) { + case 1: + wm8994->dac_rates[0] = params_rate(params); + wm8994_set_retune_mobile(codec, 0); + wm8994_set_retune_mobile(codec, 1); + break; + case 2: + wm8994->dac_rates[1] = params_rate(params); + wm8994_set_retune_mobile(codec, 2); + break; + } + } + + return 0; +} + +static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int mute_reg; + int reg; + + switch (codec_dai->id) { + case 1: + mute_reg = WM8994_AIF1_DAC1_FILTERS_1; + break; + case 2: + mute_reg = WM8994_AIF2_DAC_FILTERS_1; + break; + default: + return -EINVAL; + } + + if (mute) + reg = WM8994_AIF1DAC1_MUTE; + else + reg = 0; + + snd_soc_update_bits(codec, mute_reg, WM8994_AIF1DAC1_MUTE, reg); + + return 0; +} + +#define WM8994_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { + .set_sysclk = wm8994_set_dai_sysclk, + .set_fmt = wm8994_set_dai_fmt, + .hw_params = wm8994_hw_params, + .digital_mute = wm8994_aif_mute, + .set_pll = wm8994_set_fll, +}; + +static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { + .set_sysclk = wm8994_set_dai_sysclk, + .set_fmt = wm8994_set_dai_fmt, + .hw_params = wm8994_hw_params, + .digital_mute = wm8994_aif_mute, + .set_pll = wm8994_set_fll, +}; + +struct snd_soc_dai wm8994_dai[] = { + { + .name = "WM8994 AIF1", + .id = 1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .ops = &wm8994_aif1_dai_ops, + }, + { + .name = "WM8994 AIF2", + .id = 2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .ops = &wm8994_aif2_dai_ops, + }, + { + .name = "WM8994 AIF3", + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .playback = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + } +}; +EXPORT_SYMBOL_GPL(wm8994_dai); + +#ifdef CONFIG_PM +static int wm8994_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8994_priv *wm8994 = codec->private_data; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { + memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], + sizeof(struct fll_config)); + ret = wm8994_set_fll(&codec->dai[0], i + 1, 0, 0, 0); + if (ret < 0) + dev_warn(codec->dev, "Failed to stop FLL%d: %d\n", + i + 1, ret); + } + + wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8994_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8994_priv *wm8994 = codec->private_data; + u16 *reg_cache = codec->reg_cache; + int i, ret; + + /* Restore the registers */ + for (i = 1; i < ARRAY_SIZE(wm8994->reg_cache); i++) { + switch (i) { + case WM8994_LDO_1: + case WM8994_LDO_2: + case WM8994_SOFTWARE_RESET: + /* Handled by other MFD drivers */ + continue; + default: + break; + } + + if (!access_masks[i].writable) + continue; + + wm8994_reg_write(codec->control_data, i, reg_cache[i]); + } + + wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { + ret = wm8994_set_fll(&codec->dai[0], i + 1, + wm8994->fll_suspend[i].src, + wm8994->fll_suspend[i].in, + wm8994->fll_suspend[i].out); + if (ret < 0) + dev_warn(codec->dev, "Failed to restore FLL%d: %d\n", + i + 1, ret); + } + + return 0; +} +#else +#define wm8994_suspend NULL +#define wm8994_resume NULL +#endif + +static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) +{ + struct snd_soc_codec *codec = &wm8994->codec; + struct wm8994_pdata *pdata = wm8994->pdata; + struct snd_kcontrol_new controls[] = { + SOC_ENUM_EXT("AIF1.1 EQ Mode", + wm8994->retune_mobile_enum, + wm8994_get_retune_mobile_enum, + wm8994_put_retune_mobile_enum), + SOC_ENUM_EXT("AIF1.2 EQ Mode", + wm8994->retune_mobile_enum, + wm8994_get_retune_mobile_enum, + wm8994_put_retune_mobile_enum), + SOC_ENUM_EXT("AIF2 EQ Mode", + wm8994->retune_mobile_enum, + wm8994_get_retune_mobile_enum, + wm8994_put_retune_mobile_enum), + }; + int ret, i, j; + const char **t; + + /* We need an array of texts for the enum API but the number + * of texts is likely to be less than the number of + * configurations due to the sample rate dependency of the + * configurations. */ + wm8994->num_retune_mobile_texts = 0; + wm8994->retune_mobile_texts = NULL; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + for (j = 0; j < wm8994->num_retune_mobile_texts; j++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8994->retune_mobile_texts[j]) == 0) + break; + } + + if (j != wm8994->num_retune_mobile_texts) + continue; + + /* Expand the array... */ + t = krealloc(wm8994->retune_mobile_texts, + sizeof(char *) * + (wm8994->num_retune_mobile_texts + 1), + GFP_KERNEL); + if (t == NULL) + continue; + + /* ...store the new entry... */ + t[wm8994->num_retune_mobile_texts] = + pdata->retune_mobile_cfgs[i].name; + + /* ...and remember the new version. */ + wm8994->num_retune_mobile_texts++; + wm8994->retune_mobile_texts = t; + } + + dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", + wm8994->num_retune_mobile_texts); + + wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; + wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; + + ret = snd_soc_add_controls(&wm8994->codec, controls, + ARRAY_SIZE(controls)); + if (ret != 0) + dev_err(wm8994->codec.dev, + "Failed to add ReTune Mobile controls: %d\n", ret); +} + +static void wm8994_handle_pdata(struct wm8994_priv *wm8994) +{ + struct snd_soc_codec *codec = &wm8994->codec; + struct wm8994_pdata *pdata = wm8994->pdata; + int ret, i; + + if (!pdata) + return; + + wm_hubs_handle_analogue_pdata(codec, pdata->lineout1_diff, + pdata->lineout2_diff, + pdata->lineout1fb, + pdata->lineout2fb, + pdata->jd_scthr, + pdata->jd_thr, + pdata->micbias1_lvl, + pdata->micbias2_lvl); + + dev_dbg(codec->dev, "%d DRC configurations\n", pdata->num_drc_cfgs); + + if (pdata->num_drc_cfgs) { + struct snd_kcontrol_new controls[] = { + SOC_ENUM_EXT("AIF1DRC1 Mode", wm8994->drc_enum, + wm8994_get_drc_enum, wm8994_put_drc_enum), + SOC_ENUM_EXT("AIF1DRC2 Mode", wm8994->drc_enum, + wm8994_get_drc_enum, wm8994_put_drc_enum), + SOC_ENUM_EXT("AIF2DRC Mode", wm8994->drc_enum, + wm8994_get_drc_enum, wm8994_put_drc_enum), + }; + + /* We need an array of texts for the enum API */ + wm8994->drc_texts = kmalloc(sizeof(char *) + * pdata->num_drc_cfgs, GFP_KERNEL); + if (!wm8994->drc_texts) { + dev_err(wm8994->codec.dev, + "Failed to allocate %d DRC config texts\n", + pdata->num_drc_cfgs); + return; + } + + for (i = 0; i < pdata->num_drc_cfgs; i++) + wm8994->drc_texts[i] = pdata->drc_cfgs[i].name; + + wm8994->drc_enum.max = pdata->num_drc_cfgs; + wm8994->drc_enum.texts = wm8994->drc_texts; + + ret = snd_soc_add_controls(&wm8994->codec, controls, + ARRAY_SIZE(controls)); + if (ret != 0) + dev_err(wm8994->codec.dev, + "Failed to add DRC mode controls: %d\n", ret); + + for (i = 0; i < WM8994_NUM_DRC; i++) + wm8994_set_drc(codec, i); + } + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", + pdata->num_retune_mobile_cfgs); + + if (pdata->num_retune_mobile_cfgs) + wm8994_handle_retune_mobile_pdata(wm8994); + else + snd_soc_add_controls(&wm8994->codec, wm8994_eq_controls, + ARRAY_SIZE(wm8994_eq_controls)); +} + +static int wm8994_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8994_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8994_codec; + codec = wm8994_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + return ret; + } + + wm8994_handle_pdata(codec->private_data); + + wm_hubs_add_analogue_controls(codec); + snd_soc_add_controls(codec, wm8994_snd_controls, + ARRAY_SIZE(wm8994_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8994_dapm_widgets, + ARRAY_SIZE(wm8994_dapm_widgets)); + wm_hubs_add_analogue_routes(codec, 0, 0); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +static int wm8994_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8994 = { + .probe = wm8994_probe, + .remove = wm8994_remove, + .suspend = wm8994_suspend, + .resume = wm8994_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8994); + +static int wm8994_codec_probe(struct platform_device *pdev) +{ + int ret; + struct wm8994_priv *wm8994; + struct snd_soc_codec *codec; + int i; + u16 rev; + + if (wm8994_codec) { + dev_err(&pdev->dev, "Another WM8994 is registered\n"); + return -EINVAL; + } + + wm8994 = kzalloc(sizeof(struct wm8994_priv), GFP_KERNEL); + if (!wm8994) { + dev_err(&pdev->dev, "Failed to allocate private data\n"); + return -ENOMEM; + } + + codec = &wm8994->codec; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8994; + codec->control_data = dev_get_drvdata(pdev->dev.parent); + codec->name = "WM8994"; + codec->owner = THIS_MODULE; + codec->read = wm8994_read; + codec->write = wm8994_write; + codec->readable_register = wm8994_readable; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8994_set_bias_level; + codec->dai = &wm8994_dai[0]; + codec->num_dai = 3; + codec->reg_cache_size = WM8994_MAX_REGISTER; + codec->reg_cache = &wm8994->reg_cache; + codec->dev = &pdev->dev; + + wm8994->pdata = pdev->dev.parent->platform_data; + + /* Fill the cache with physical values we inherited; don't reset */ + ret = wm8994_bulk_read(codec->control_data, 0, + ARRAY_SIZE(wm8994->reg_cache) - 1, + codec->reg_cache); + if (ret < 0) { + dev_err(codec->dev, "Failed to fill register cache: %d\n", + ret); + goto err; + } + + /* Clear the cached values for unreadable/volatile registers to + * avoid potential confusion. + */ + for (i = 0; i < ARRAY_SIZE(wm8994->reg_cache); i++) + if (wm8994_volatile(i) || !wm8994_readable(i)) + wm8994->reg_cache[i] = 0; + + /* Set revision-specific configuration */ + rev = snd_soc_read(codec, WM8994_CHIP_REVISION); + switch (rev) { + case 2: + case 3: + wm8994->hubs.dcs_codes = -5; + wm8994->hubs.hp_startup_mode = 1; + break; + default: + break; + } + + + /* Remember if AIFnLRCLK is configured as a GPIO. This should be + * configured on init - if a system wants to do this dynamically + * at runtime we can deal with that then. + */ + ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_1); + if (ret < 0) { + dev_err(codec->dev, "Failed to read GPIO1 state: %d\n", ret); + goto err; + } + if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { + wm8994->lrclk_shared[0] = 1; + wm8994_dai[0].symmetric_rates = 1; + } else { + wm8994->lrclk_shared[0] = 0; + } + + ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_6); + if (ret < 0) { + dev_err(codec->dev, "Failed to read GPIO6 state: %d\n", ret); + goto err; + } + if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { + wm8994->lrclk_shared[1] = 1; + wm8994_dai[1].symmetric_rates = 1; + } else { + wm8994->lrclk_shared[1] = 0; + } + + for (i = 0; i < ARRAY_SIZE(wm8994_dai); i++) + wm8994_dai[i].dev = codec->dev; + + wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8994_codec = codec; + + /* Latch volume updates (right only; we always do left then right). */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME, + WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME, + WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME, + WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME, + WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME, + WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME, + WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME, + WM8994_DAC1_VU, WM8994_DAC1_VU); + snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME, + WM8994_DAC2_VU, WM8994_DAC2_VU); + + /* Set the low bit of the 3D stereo depth so TLV matches */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_FILTERS_2, + 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, + 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_FILTERS_2, + 1 << WM8994_AIF1DAC2_3D_GAIN_SHIFT, + 1 << WM8994_AIF1DAC2_3D_GAIN_SHIFT); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_FILTERS_2, + 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, + 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); + + wm8994_update_class_w(codec); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(wm8994_dai, ARRAY_SIZE(wm8994_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + platform_set_drvdata(pdev, wm8994); + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8994); + return ret; +} + +static int __devexit wm8994_codec_remove(struct platform_device *pdev) +{ + struct wm8994_priv *wm8994 = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = &wm8994->codec; + + wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(wm8994_dai, ARRAY_SIZE(wm8994_dai)); + snd_soc_unregister_codec(&wm8994->codec); + kfree(wm8994); + wm8994_codec = NULL; + + return 0; +} + +static struct platform_driver wm8994_codec_driver = { + .driver = { + .name = "wm8994-codec", + .owner = THIS_MODULE, + }, + .probe = wm8994_codec_probe, + .remove = __devexit_p(wm8994_codec_remove), +}; + +static __init int wm8994_init(void) +{ + return platform_driver_register(&wm8994_codec_driver); +} +module_init(wm8994_init); + +static __exit void wm8994_exit(void) +{ + platform_driver_unregister(&wm8994_codec_driver); +} +module_exit(wm8994_exit); + + +MODULE_DESCRIPTION("ASoC WM8994 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8994-codec"); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h new file mode 100644 index 000000000000..0a5e1424dea0 --- /dev/null +++ b/sound/soc/codecs/wm8994.h @@ -0,0 +1,26 @@ +/* + * wm8994.h -- WM8994 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8994_H +#define _WM8994_H + +#include + +extern struct snd_soc_codec_device soc_codec_dev_wm8994; +extern struct snd_soc_dai wm8994_dai[]; + +/* Sources for AIF1/2 SYSCLK - use with set_dai_sysclk() */ +#define WM8994_SYSCLK_MCLK1 1 +#define WM8994_SYSCLK_MCLK2 2 +#define WM8994_SYSCLK_FLL1 3 +#define WM8994_SYSCLK_FLL2 4 + +#define WM8994_FLL1 1 +#define WM8994_FLL2 2 + +#endif -- cgit v1.2.2 From c85a400499093b2025238413198e48e4d825723e Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Mon, 1 Feb 2010 16:17:01 -0200 Subject: ALSA: trivial: sound seq ioctl dbg: print hexadecimal value padded with 0s Instead of padding with blanks and printing "number=0x a", print "number=0x0a". Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 8ca2be339f3b..48eca9ff9ee7 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2190,7 +2190,7 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, if (p->cmd == cmd) return p->func(client, arg); } - snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%2x)\n", + snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", cmd, _IOC_TYPE(cmd), _IOC_NR(cmd)); return -ENOTTY; } -- cgit v1.2.2 From fead215d1c0a385fc27a1fa96b7abbc4d66fb4c6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Feb 2010 10:06:55 +0000 Subject: ASoC: Fix WM8994 dependency The dependency on MFD_WM8994 rather than I2C went awry. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6b8a10120f9c..5ab59219a8de 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C - select SND_SOC_WM8994 if I2C + select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM9081 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS -- cgit v1.2.2 From 07cd8ada1aba5556b0d5d2264ce0f40d1ff1d131 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Tue, 2 Feb 2010 18:53:19 +0900 Subject: ASoC: Fix BCLK calculation of WM8994 This fixes BCLK calculation and removes unnecessary check code. Signed-off-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5dd4b299f69e..29f3771c33a4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3267,15 +3267,12 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, */ best = 0; for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { - if (bclk_divs[i] < 0) - continue; - cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - - bclk_rate * 10; + cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - bclk_rate; if (cur_val < 0) /* BCLK table is sorted */ break; best = i; } - bclk_rate = wm8994->aifclk[id] / bclk_divs[best]; + bclk_rate = wm8994->aifclk[id] * 10 / bclk_divs[best]; dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", bclk_divs[best], bclk_rate); bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; -- cgit v1.2.2 From 59cdd9bc057a54384a7838231dd2672a89dff2ac Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 1 Feb 2010 23:22:16 -0800 Subject: ASoC: Fix continuation line formats String constants that are continued on subsequent lines with \ are not good. Signed-off-by: Joe Perches Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index 9e61a7c2d9ac..a98f40c3cd29 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -229,8 +229,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, spin_unlock_irqrestore(&pcm->lock, flags); - dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs \ - SCLK_DIV=%d SYNC_DIV=%d\n", + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", clk_get_rate(clk), pcm->sclk_per_fs, sclk_div, sync_div); -- cgit v1.2.2 From 026384d614b827f368834860c9b623ce08439b7e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 2 Feb 2010 18:45:27 +0800 Subject: ASoC: fix PXA SSP port resume Unconditionally save the register states when suspending and restore them again at resume time. Register contents were not preserved over suspend, and hence the driver takes false assumptions about them. The clock must be enabled to access the register block. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3bd7712f029b..e69397f40f72 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -135,10 +135,11 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_priv *priv = cpu_dai->private_data; if (!cpu_dai->active) - return 0; + clk_enable(priv->dev.ssp->clk); ssp_save_state(&priv->dev, &priv->state); clk_disable(priv->dev.ssp->clk); + return 0; } @@ -146,12 +147,13 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) { struct ssp_priv *priv = cpu_dai->private_data; - if (!cpu_dai->active) - return 0; - clk_enable(priv->dev.ssp->clk); ssp_restore_state(&priv->dev, &priv->state); - ssp_enable(&priv->dev); + + if (cpu_dai->active) + ssp_enable(&priv->dev); + else + clk_disable(priv->dev.ssp->clk); return 0; } -- cgit v1.2.2 From d5e1ca05f758fec2845a97fd7aa1eeca91c51a21 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 2 Feb 2010 17:48:51 +0100 Subject: ALSA: dummy driver - add model parameter This is a cleanup for the dummy driver. The model kernel module parameter is introduced to select the soundcard emulation. Signed-off-by: Jaroslav Kysela --- sound/drivers/dummy.c | 290 +++++++++++++++++++++++++++++++------------------- 1 file changed, 180 insertions(+), 110 deletions(-) (limited to 'sound') diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 252e04ce602f..7f41990ed68b 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -45,109 +45,23 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}"); #define MAX_PCM_SUBSTREAMS 128 #define MAX_MIDI_DEVICES 2 -#if 0 /* emu10k1 emulation */ -#define MAX_BUFFER_SIZE (128 * 1024) -static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) -{ - int err; - err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (err < 0) - return err; - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); - if (err < 0) - return err; - return 0; -} -#define add_playback_constraints emu10k1_playback_constraints -#endif - -#if 0 /* RME9652 emulation */ -#define MAX_BUFFER_SIZE (26 * 64 * 1024) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE -#define USE_CHANNELS_MIN 26 -#define USE_CHANNELS_MAX 26 -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 2 -#endif - -#if 0 /* ICE1712 emulation */ -#define MAX_BUFFER_SIZE (256 * 1024) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE -#define USE_CHANNELS_MIN 10 -#define USE_CHANNELS_MAX 10 -#define USE_PERIODS_MIN 1 -#define USE_PERIODS_MAX 1024 -#endif - -#if 0 /* UDA1341 emulation */ -#define MAX_BUFFER_SIZE (16380) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 255 -#endif - -#if 0 /* simple AC97 bridge (intel8x0) with 48kHz AC97 only codec */ -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_RATE SNDRV_PCM_RATE_48000 -#define USE_RATE_MIN 48000 -#define USE_RATE_MAX 48000 -#endif - -#if 0 /* CA0106 */ -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_RATE (SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_192000) -#define USE_RATE_MIN 48000 -#define USE_RATE_MAX 192000 -#define MAX_BUFFER_SIZE ((65536-64)*8) -#define MAX_PERIOD_SIZE (65536-64) -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 8 -#endif - - /* defaults */ -#ifndef MAX_BUFFER_SIZE #define MAX_BUFFER_SIZE (64*1024) -#endif -#ifndef MAX_PERIOD_SIZE +#define MIN_PERIOD_SIZE 64 #define MAX_PERIOD_SIZE MAX_BUFFER_SIZE -#endif -#ifndef USE_FORMATS #define USE_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) -#endif -#ifndef USE_RATE #define USE_RATE SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000 #define USE_RATE_MIN 5500 #define USE_RATE_MAX 48000 -#endif -#ifndef USE_CHANNELS_MIN #define USE_CHANNELS_MIN 1 -#endif -#ifndef USE_CHANNELS_MAX #define USE_CHANNELS_MAX 2 -#endif -#ifndef USE_PERIODS_MIN #define USE_PERIODS_MIN 1 -#endif -#ifndef USE_PERIODS_MAX #define USE_PERIODS_MAX 1024 -#endif -#ifndef add_playback_constraints -#define add_playback_constraints(x) 0 -#endif -#ifndef add_capture_constraints -#define add_capture_constraints(x) 0 -#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; +static char *model[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = NULL}; static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; @@ -162,6 +76,8 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for dummy soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable this dummy soundcard."); +module_param_array(model, charp, NULL, 0444); +MODULE_PARM_DESC(model, "Soundcard model."); module_param_array(pcm_devs, int, NULL, 0444); MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver."); module_param_array(pcm_substreams, int, NULL, 0444); @@ -193,15 +109,120 @@ struct dummy_timer_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *); }; +struct dummy_model { + const char *name; + int (*playback_constraints)(struct snd_pcm_runtime *runtime); + int (*capture_constraints)(struct snd_pcm_runtime *runtime); + u64 formats; + size_t buffer_bytes_max; + size_t period_bytes_min; + size_t period_bytes_max; + unsigned int periods_min; + unsigned int periods_max; + unsigned int rates; + unsigned int rate_min; + unsigned int rate_max; + unsigned int channels_min; + unsigned int channels_max; +}; + struct snd_dummy { struct snd_card *card; + struct dummy_model *model; struct snd_pcm *pcm; + struct snd_pcm_hardware pcm_hw; spinlock_t mixer_lock; int mixer_volume[MIXER_ADDR_LAST+1][2]; int capture_source[MIXER_ADDR_LAST+1][2]; const struct dummy_timer_ops *timer_ops; }; +/* + * card models + */ + +static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) +{ + int err; + err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); + if (err < 0) + return err; + return 0; +} + +struct dummy_model model_emu10k1 = { + .name = "emu10k1", + .playback_constraints = emu10k1_playback_constraints, + .buffer_bytes_max = 128 * 1024, +}; + +struct dummy_model model_rme9652 = { + .name = "rme9652", + .buffer_bytes_max = 26 * 64 * 1024, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 26, + .channels_max = 26, + .periods_min = 2, + .periods_max = 2, +}; + +struct dummy_model model_ice1712 = { + .name = "ice1712", + .buffer_bytes_max = 256 * 1024, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 10, + .channels_max = 10, + .periods_min = 1, + .periods_max = 1024, +}; + +struct dummy_model model_uda1341 = { + .name = "uda1341", + .buffer_bytes_max = 16380, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, + .periods_min = 2, + .periods_max = 255, +}; + +struct dummy_model model_ac97 = { + .name = "ac97", + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, +}; + +struct dummy_model model_ca0106 = { + .name = "ca0106", + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .buffer_bytes_max = ((65536-64)*8), + .period_bytes_max = (65536-64), + .periods_min = 2, + .periods_max = 8, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_192000, + .rate_min = 48000, + .rate_max = 192000, +}; + +struct dummy_model *dummy_models[] = { + &model_emu10k1, + &model_rme9652, + &model_ice1712, + &model_uda1341, + &model_ac97, + &model_ca0106, + NULL +}; + /* * system timer interface */ @@ -509,7 +530,7 @@ static struct snd_pcm_hardware dummy_pcm_hardware = { .channels_min = USE_CHANNELS_MIN, .channels_max = USE_CHANNELS_MAX, .buffer_bytes_max = MAX_BUFFER_SIZE, - .period_bytes_min = 64, + .period_bytes_min = MIN_PERIOD_SIZE, .period_bytes_max = MAX_PERIOD_SIZE, .periods_min = USE_PERIODS_MIN, .periods_max = USE_PERIODS_MAX, @@ -538,6 +559,7 @@ static int dummy_pcm_hw_free(struct snd_pcm_substream *substream) static int dummy_pcm_open(struct snd_pcm_substream *substream) { struct snd_dummy *dummy = snd_pcm_substream_chip(substream); + struct dummy_model *model = dummy->model; struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -551,7 +573,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) if (err < 0) return err; - runtime->hw = dummy_pcm_hardware; + runtime->hw = dummy->pcm_hw; if (substream->pcm->device & 1) { runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED; runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED; @@ -560,10 +582,16 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = add_playback_constraints(substream->runtime); - else - err = add_capture_constraints(substream->runtime); + if (model == NULL) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (model->playback_constraints) + err = model->playback_constraints(substream->runtime); + } else { + if (model->capture_constraints) + err = model->capture_constraints(substream->runtime); + } if (err < 0) { dummy->timer_ops->free(substream); return err; @@ -823,17 +851,19 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) /* * proc interface */ -static void print_formats(struct snd_info_buffer *buffer) +static void print_formats(struct snd_dummy *dummy, + struct snd_info_buffer *buffer) { int i; for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { - if (dummy_pcm_hardware.formats & (1ULL << i)) + if (dummy->pcm_hw.formats & (1ULL << i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } } -static void print_rates(struct snd_info_buffer *buffer) +static void print_rates(struct snd_dummy *dummy, + struct snd_info_buffer *buffer) { static int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, @@ -841,19 +871,19 @@ static void print_rates(struct snd_info_buffer *buffer) }; int i; - if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_CONTINUOUS) + if (dummy->pcm_hw.rates & SNDRV_PCM_RATE_CONTINUOUS) snd_iprintf(buffer, " continuous"); - if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_KNOT) + if (dummy->pcm_hw.rates & SNDRV_PCM_RATE_KNOT) snd_iprintf(buffer, " knot"); for (i = 0; i < ARRAY_SIZE(rates); i++) - if (dummy_pcm_hardware.rates & (1 << i)) + if (dummy->pcm_hw.rates & (1 << i)) snd_iprintf(buffer, " %d", rates[i]); } -#define get_dummy_int_ptr(ofs) \ - (unsigned int *)((char *)&dummy_pcm_hardware + (ofs)) -#define get_dummy_ll_ptr(ofs) \ - (unsigned long long *)((char *)&dummy_pcm_hardware + (ofs)) +#define get_dummy_int_ptr(dummy, ofs) \ + (unsigned int *)((char *)&((dummy)->pcm_hw) + (ofs)) +#define get_dummy_ll_ptr(dummy, ofs) \ + (unsigned long long *)((char *)&((dummy)->pcm_hw) + (ofs)) struct dummy_hw_field { const char *name; @@ -884,20 +914,21 @@ static struct dummy_hw_field fields[] = { static void dummy_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { + struct snd_dummy *dummy = entry->private_data; int i; for (i = 0; i < ARRAY_SIZE(fields); i++) { snd_iprintf(buffer, "%s ", fields[i].name); if (fields[i].size == sizeof(int)) snd_iprintf(buffer, fields[i].format, - *get_dummy_int_ptr(fields[i].offset)); + *get_dummy_int_ptr(dummy, fields[i].offset)); else snd_iprintf(buffer, fields[i].format, - *get_dummy_ll_ptr(fields[i].offset)); + *get_dummy_ll_ptr(dummy, fields[i].offset)); if (!strcmp(fields[i].name, "formats")) - print_formats(buffer); + print_formats(dummy, buffer); else if (!strcmp(fields[i].name, "rates")) - print_rates(buffer); + print_rates(dummy, buffer); snd_iprintf(buffer, "\n"); } } @@ -905,6 +936,7 @@ static void dummy_proc_read(struct snd_info_entry *entry, static void dummy_proc_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { + struct snd_dummy *dummy = entry->private_data; char line[64]; while (!snd_info_get_line(buffer, line, sizeof(line))) { @@ -924,9 +956,9 @@ static void dummy_proc_write(struct snd_info_entry *entry, if (strict_strtoull(item, 0, &val)) continue; if (fields[i].size == sizeof(int)) - *get_dummy_int_ptr(fields[i].offset) = val; + *get_dummy_int_ptr(dummy, fields[i].offset) = val; else - *get_dummy_ll_ptr(fields[i].offset) = val; + *get_dummy_ll_ptr(dummy, fields[i].offset) = val; } } @@ -938,6 +970,7 @@ static void __devinit dummy_proc_init(struct snd_dummy *chip) snd_info_set_text_ops(entry, chip, dummy_proc_read); entry->c.text.write = dummy_proc_write; entry->mode |= S_IWUSR; + entry->private_data = chip; } } #else @@ -948,6 +981,7 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) { struct snd_card *card; struct snd_dummy *dummy; + struct dummy_model *m = NULL, **mdl; int idx, err; int dev = devptr->id; @@ -957,6 +991,15 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) return err; dummy = card->private_data; dummy->card = card; + for (mdl = dummy_models; *mdl && model[dev]; mdl++) { + if (strcmp(model[dev], (*mdl)->name) == 0) { + printk(KERN_INFO + "snd-dummy: Using model '%s' for card %i\n", + (*mdl)->name, card->number); + m = dummy->model = *mdl; + break; + } + } for (idx = 0; idx < MAX_PCM_DEVICES && idx < pcm_devs[dev]; idx++) { if (pcm_substreams[dev] < 1) pcm_substreams[dev] = 1; @@ -966,6 +1009,33 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) if (err < 0) goto __nodev; } + + dummy->pcm_hw = dummy_pcm_hardware; + if (m) { + if (m->formats) + dummy->pcm_hw.formats = m->formats; + if (m->buffer_bytes_max) + dummy->pcm_hw.buffer_bytes_max = m->buffer_bytes_max; + if (m->period_bytes_min) + dummy->pcm_hw.period_bytes_min = m->period_bytes_min; + if (m->period_bytes_max) + dummy->pcm_hw.period_bytes_max = m->period_bytes_max; + if (m->periods_min) + dummy->pcm_hw.periods_min = m->periods_min; + if (m->periods_max) + dummy->pcm_hw.periods_max = m->periods_max; + if (m->rates) + dummy->pcm_hw.rates = m->rates; + if (m->rate_min) + dummy->pcm_hw.rate_min = m->rate_min; + if (m->rate_max) + dummy->pcm_hw.rate_max = m->rate_max; + if (m->channels_min) + dummy->pcm_hw.channels_min = m->channels_min; + if (m->channels_max) + dummy->pcm_hw.channels_max = m->channels_max; + } + err = snd_card_dummy_new_mixer(dummy); if (err < 0) goto __nodev; -- cgit v1.2.2 From 0f69d9782c6e6a7b0e60113a850845bc642c3f4e Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 3 Feb 2010 17:37:23 +0100 Subject: ASoC: fix compilation breakage in sound/soc/sh/fsi.c ctrl_outl() has become void at some point, which breaks compilation of fsi.c. Make writing functions void, as their output is anyway not evaluated, and use __raw_writel and __raw_readl instead of deprecated ctrl_outl and ctrl_inl respectively. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 40 +++++++++++++++++----------------------- 1 file changed, 17 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ebf358808db1..3c36d24a6c20 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -120,35 +120,35 @@ struct fsi_master { ************************************************************************/ -static int __fsi_reg_write(u32 reg, u32 data) +static void __fsi_reg_write(u32 reg, u32 data) { /* valid data area is 24bit */ data &= 0x00ffffff; - return ctrl_outl(data, reg); + __raw_writel(data, reg); } static u32 __fsi_reg_read(u32 reg) { - return ctrl_inl(reg); + return __raw_readl(reg); } -static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) { u32 val = __fsi_reg_read(reg); val &= ~mask; val |= data & mask; - return __fsi_reg_write(reg, val); + __fsi_reg_write(reg, val); } -static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +static void fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_write((u32)(fsi->base + reg), data); + __fsi_reg_write((u32)(fsi->base + reg), data); } static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) @@ -159,28 +159,25 @@ static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) return __fsi_reg_read((u32)(fsi->base + reg)); } -static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +static void fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); + __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } -static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) +static void fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_write((u32)(master->base + reg), data); + __fsi_reg_write((u32)(master->base + reg), data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) @@ -199,21 +196,18 @@ static u32 fsi_master_read(struct fsi_master *master, u32 reg) return ret; } -static int fsi_master_mask_set(struct fsi_master *master, +static void fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + __fsi_reg_mask_set((u32)(master->base + reg), mask, data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } /************************************************************************ -- cgit v1.2.2 From 8c961bcca1d10be4f2c06375eb561679167653a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 18:46:10 +0000 Subject: ASoC: Allow CODECs to ask soc-cache to suppress physical writes Currently the soc-cache code will always write to the device, meaning that we need the device to be powered and active at pretty much all times the system is active. Allowing cache only writes lays some groundwork for future enhancements to allow devices to be put into a full off state when the audio subsystem is idle. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 097e33510a7a..84b6916db87d 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -38,6 +38,10 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -100,6 +104,10 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -153,6 +161,9 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -181,6 +192,9 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, if (!snd_soc_codec_volatile_register(codec, reg)) reg_cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 3) == 3) return 0; else @@ -193,10 +207,14 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; if (reg >= codec->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) + snd_soc_codec_volatile_register(codec, reg)) { + if (codec->cache_only) + return -EINVAL; + return codec->hw_read(codec, reg); - else + } else { return cache[reg]; + } } #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) @@ -294,6 +312,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 3); if (ret == 3) return 0; -- cgit v1.2.2 From a9694faa287888b4fb10849649b6c94d0a1c9940 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Thu, 4 Feb 2010 08:58:23 +0100 Subject: ALSA: hda - Adding support for another IDT 92HD83XXX codec Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9694675f0b9e..693dd14d9ec1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5341,6 +5341,7 @@ again: spec->num_pwrs = 0; break; case 0x111d7604: + case 0x111d76d4: case 0x111d7605: case 0x111d76d5: if (spec->board_config == STAC_92HD83XXX_PWR_REF) @@ -6263,6 +6264,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d4, .name = "92HD83C1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx}, -- cgit v1.2.2 From 04b5efe5fa7f71c37b938053666fac317b67c636 Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Thu, 4 Feb 2010 10:28:02 +0100 Subject: ALSA: hda - Fix docking output for IDT 92HD8xx codecs This patch fixes docking output support for IDT 92HD81/83/88 family codecs. Typically one of ports 0xE or 0xF is used for docking output, while only port 0xF is common on all the three codec families. We don't want the pin to select the analog mixer here. Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 30 +++++++++++++----------------- 1 file changed, 13 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 693dd14d9ec1..834c5980fe5d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5291,7 +5291,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; int num_dacs; - hda_nid_t nid; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -5387,24 +5386,21 @@ again: return err; } - switch (spec->board_config) { - case STAC_DELL_S14: - nid = 0xf; - break; - default: - nid = 0xe; - break; - } - - num_dacs = snd_hda_get_connections(codec, nid, + /* docking output support */ + num_dacs = snd_hda_get_connections(codec, 0xF, conn, STAC92HD83_DAC_COUNT + 1) - 1; - if (num_dacs < 0) - num_dacs = STAC92HD83_DAC_COUNT; - - /* set port X to select the last DAC - */ - snd_hda_codec_write_cache(codec, nid, 0, + /* skip non-DAC connections */ + while (num_dacs >= 0 && + (get_wcaps_type(get_wcaps(codec, conn[num_dacs])) + != AC_WID_AUD_OUT)) + num_dacs--; + /* set port E and F to select the last DAC */ + if (num_dacs >= 0) { + snd_hda_codec_write_cache(codec, 0xE, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + snd_hda_codec_write_cache(codec, 0xF, 0, AC_VERB_SET_CONNECT_SEL, num_dacs); + } codec->proc_widget_hook = stac92hd_proc_hook; -- cgit v1.2.2 From a3032b47c46920ed3f2fd58e64f484e3dab49f23 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 18:48:03 +0000 Subject: ASoC: Add a cache_sync bit to the CODEC structure Add a bit to the CODEC structure indicating if a cache sync is required. By default this will be set if a cache only write is done to a soc-cache register cache. This allows us to avoid syncing the cache back after using cache only writes if there were no changes. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 84b6916db87d..5869dc3be781 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -39,8 +39,10 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) @@ -105,8 +107,10 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) @@ -161,8 +165,10 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; @@ -192,8 +198,10 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, if (!snd_soc_codec_volatile_register(codec, reg)) reg_cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } if (codec->hw_write(codec->control_data, data, 3) == 3) return 0; @@ -313,8 +321,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } ret = codec->hw_write(codec->control_data, data, 3); if (ret == 3) -- cgit v1.2.2 From 3bf6e4217e3c69438f6dc41a009664107eb27ab1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 19:05:09 +0000 Subject: ASoC: Convert WM8993 to use shared cache I/O code Saves a little bit of code duplication. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 152 +++++++++++++--------------------------------- 1 file changed, 43 insertions(+), 109 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 61239e0e9556..3c9336cd4eeb 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -231,34 +231,6 @@ struct wm8993_priv { int fll_src; }; -static unsigned int wm8993_read_hw(struct snd_soc_codec *codec, u8 reg) -{ - struct i2c_msg xfer[2]; - u16 data; - int ret; - struct i2c_client *i2c = codec->control_data; - - /* Write register */ - xfer[0].addr = i2c->addr; - xfer[0].flags = 0; - xfer[0].len = 1; - xfer[0].buf = ® - - /* Read data */ - xfer[1].addr = i2c->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 2; - xfer[1].buf = (u8 *)&data; - - ret = i2c_transfer(i2c->adapter, xfer, 2); - if (ret != 2) { - dev_err(codec->dev, "Failed to read 0x%x: %d\n", reg, ret); - return 0; - } - - return (data >> 8) | ((data & 0xff) << 8); -} - static int wm8993_volatile(unsigned int reg) { switch (reg) { @@ -273,48 +245,6 @@ static int wm8993_volatile(unsigned int reg) } } -static unsigned int wm8993_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *reg_cache = codec->reg_cache; - - BUG_ON(reg > WM8993_MAX_REGISTER); - - if (wm8993_volatile(reg)) - return wm8993_read_hw(codec, reg); - else - return reg_cache[reg]; -} - -static int wm8993_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 *reg_cache = codec->reg_cache; - u8 data[3]; - int ret; - - BUG_ON(reg > WM8993_MAX_REGISTER); - - /* data is - * D15..D9 WM8993 register offset - * D8...D0 register data - */ - data[0] = reg; - data[1] = value >> 8; - data[2] = value & 0x00ff; - - if (!wm8993_volatile(reg)) - reg_cache[reg] = value; - - ret = codec->hw_write(codec->control_data, data, 3); - - if (ret == 3) - return 0; - if (ret < 0) - return ret; - return -EIO; -} - struct _fll_div { u16 fll_fratio; u16 fll_outdiv; @@ -443,9 +373,9 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, wm8993->fll_fref = 0; wm8993->fll_fout = 0; - reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1); + reg1 = snd_soc_read(codec, WM8993_FLL_CONTROL_1); reg1 &= ~WM8993_FLL_ENA; - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1); return 0; } @@ -454,7 +384,7 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, if (ret != 0) return ret; - reg5 = wm8993_read(codec, WM8993_FLL_CONTROL_5); + reg5 = snd_soc_read(codec, WM8993_FLL_CONTROL_5); reg5 &= ~WM8993_FLL_CLK_SRC_MASK; switch (fll_id) { @@ -476,33 +406,33 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, /* Any FLL configuration change requires that the FLL be * disabled first. */ - reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1); + reg1 = snd_soc_read(codec, WM8993_FLL_CONTROL_1); reg1 &= ~WM8993_FLL_ENA; - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1); /* Apply the configuration */ if (fll_div.k) reg1 |= WM8993_FLL_FRAC_MASK; else reg1 &= ~WM8993_FLL_FRAC_MASK; - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1); - wm8993_write(codec, WM8993_FLL_CONTROL_2, - (fll_div.fll_outdiv << WM8993_FLL_OUTDIV_SHIFT) | - (fll_div.fll_fratio << WM8993_FLL_FRATIO_SHIFT)); - wm8993_write(codec, WM8993_FLL_CONTROL_3, fll_div.k); + snd_soc_write(codec, WM8993_FLL_CONTROL_2, + (fll_div.fll_outdiv << WM8993_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio << WM8993_FLL_FRATIO_SHIFT)); + snd_soc_write(codec, WM8993_FLL_CONTROL_3, fll_div.k); - reg4 = wm8993_read(codec, WM8993_FLL_CONTROL_4); + reg4 = snd_soc_read(codec, WM8993_FLL_CONTROL_4); reg4 &= ~WM8993_FLL_N_MASK; reg4 |= fll_div.n << WM8993_FLL_N_SHIFT; - wm8993_write(codec, WM8993_FLL_CONTROL_4, reg4); + snd_soc_write(codec, WM8993_FLL_CONTROL_4, reg4); reg5 &= ~WM8993_FLL_CLK_REF_DIV_MASK; reg5 |= fll_div.fll_clk_ref_div << WM8993_FLL_CLK_REF_DIV_SHIFT; - wm8993_write(codec, WM8993_FLL_CONTROL_5, reg5); + snd_soc_write(codec, WM8993_FLL_CONTROL_5, reg5); /* Enable the FLL */ - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); @@ -523,7 +453,7 @@ static int configure_clock(struct snd_soc_codec *codec) case WM8993_SYSCLK_MCLK: dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8993->mclk_rate); - reg = wm8993_read(codec, WM8993_CLOCKING_2); + reg = snd_soc_read(codec, WM8993_CLOCKING_2); reg &= ~(WM8993_MCLK_DIV | WM8993_SYSCLK_SRC); if (wm8993->mclk_rate > 13500000) { reg |= WM8993_MCLK_DIV; @@ -532,14 +462,14 @@ static int configure_clock(struct snd_soc_codec *codec) reg &= ~WM8993_MCLK_DIV; wm8993->sysclk_rate = wm8993->mclk_rate; } - wm8993_write(codec, WM8993_CLOCKING_2, reg); + snd_soc_write(codec, WM8993_CLOCKING_2, reg); break; case WM8993_SYSCLK_FLL: dev_dbg(codec->dev, "Using %dHz FLL clock\n", wm8993->fll_fout); - reg = wm8993_read(codec, WM8993_CLOCKING_2); + reg = snd_soc_read(codec, WM8993_CLOCKING_2); reg |= WM8993_SYSCLK_SRC; if (wm8993->fll_fout > 13500000) { reg |= WM8993_MCLK_DIV; @@ -548,7 +478,7 @@ static int configure_clock(struct snd_soc_codec *codec) reg &= ~WM8993_MCLK_DIV; wm8993->sysclk_rate = wm8993->fll_fout; } - wm8993_write(codec, WM8993_CLOCKING_2, reg); + snd_soc_write(codec, WM8993_CLOCKING_2, reg); break; default: @@ -1083,8 +1013,8 @@ static int wm8993_set_dai_fmt(struct snd_soc_dai *dai, { struct snd_soc_codec *codec = dai->codec; struct wm8993_priv *wm8993 = codec->private_data; - unsigned int aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1); - unsigned int aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4); + unsigned int aif1 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_1); + unsigned int aif4 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_4); aif1 &= ~(WM8993_BCLK_DIR | WM8993_AIF_BCLK_INV | WM8993_AIF_LRCLK_INV | WM8993_AIF_FMT_MASK); @@ -1167,8 +1097,8 @@ static int wm8993_set_dai_fmt(struct snd_soc_dai *dai, return -EINVAL; } - wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); - wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); return 0; } @@ -1182,16 +1112,16 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, int ret, i, best, best_val, cur_val; unsigned int clocking1, clocking3, aif1, aif4; - clocking1 = wm8993_read(codec, WM8993_CLOCKING_1); + clocking1 = snd_soc_read(codec, WM8993_CLOCKING_1); clocking1 &= ~WM8993_BCLK_DIV_MASK; - clocking3 = wm8993_read(codec, WM8993_CLOCKING_3); + clocking3 = snd_soc_read(codec, WM8993_CLOCKING_3); clocking3 &= ~(WM8993_CLK_SYS_RATE_MASK | WM8993_SAMPLE_RATE_MASK); - aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1); + aif1 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_1); aif1 &= ~WM8993_AIF_WL_MASK; - aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4); + aif4 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_4); aif4 &= ~WM8993_LRCLK_RATE_MASK; /* What BCLK do we need? */ @@ -1284,14 +1214,14 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm8993->bclk / wm8993->fs); aif4 |= wm8993->bclk / wm8993->fs; - wm8993_write(codec, WM8993_CLOCKING_1, clocking1); - wm8993_write(codec, WM8993_CLOCKING_3, clocking3); - wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); - wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); + snd_soc_write(codec, WM8993_CLOCKING_1, clocking1); + snd_soc_write(codec, WM8993_CLOCKING_3, clocking3); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); /* ReTune Mobile? */ if (wm8993->pdata.num_retune_configs) { - u16 eq1 = wm8993_read(codec, WM8993_EQ1); + u16 eq1 = snd_soc_read(codec, WM8993_EQ1); struct wm8993_retune_mobile_setting *s; best = 0; @@ -1314,7 +1244,7 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, 0); for (i = 1; i < ARRAY_SIZE(s->config); i++) - wm8993_write(codec, WM8993_EQ1 + i, s->config[i]); + snd_soc_write(codec, WM8993_EQ1 + i, s->config[i]); snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, eq1); } @@ -1327,14 +1257,14 @@ static int wm8993_digital_mute(struct snd_soc_dai *codec_dai, int mute) struct snd_soc_codec *codec = codec_dai->codec; unsigned int reg; - reg = wm8993_read(codec, WM8993_DAC_CTRL); + reg = snd_soc_read(codec, WM8993_DAC_CTRL); if (mute) reg |= WM8993_DAC_MUTE; else reg &= ~WM8993_DAC_MUTE; - wm8993_write(codec, WM8993_DAC_CTRL, reg); + snd_soc_write(codec, WM8993_DAC_CTRL, reg); return 0; } @@ -1586,9 +1516,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, INIT_LIST_HEAD(&codec->dapm_paths); codec->name = "WM8993"; - codec->read = wm8993_read; - codec->write = wm8993_write; - codec->hw_write = (hw_write_t)i2c_master_send; + codec->volatile_register = wm8993_volatile; codec->reg_cache = wm8993->reg_cache; codec->reg_cache_size = ARRAY_SIZE(wm8993->reg_cache); codec->bias_level = SND_SOC_BIAS_OFF; @@ -1603,20 +1531,26 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, memcpy(wm8993->reg_cache, wm8993_reg_defaults, sizeof(wm8993->reg_cache)); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + i2c_set_clientdata(i2c, wm8993); codec->control_data = i2c; wm8993_codec = codec; codec->dev = &i2c->dev; - val = wm8993_read_hw(codec, WM8993_SOFTWARE_RESET); + val = snd_soc_read(codec, WM8993_SOFTWARE_RESET); if (val != wm8993_reg_defaults[WM8993_SOFTWARE_RESET]) { dev_err(codec->dev, "Invalid ID register value %x\n", val); ret = -EINVAL; goto err; } - ret = wm8993_write(codec, WM8993_SOFTWARE_RESET, 0xffff); + ret = snd_soc_write(codec, WM8993_SOFTWARE_RESET, 0xffff); if (ret != 0) goto err; -- cgit v1.2.2 From b37e399bfc7fcb5b523e3e2e74686c8cc95c0cba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 11:51:42 +0000 Subject: ASoC: Initial WM8993 regulator API hookup At the minute the regulators are simply enabled for the entire lifetime of the device. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 41 ++++++++++++++++++++++++++++++++++++++--- 1 file changed, 38 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 3c9336cd4eeb..e97b3f45b24b 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -29,6 +30,16 @@ #include "wm8993.h" #include "wm_hubs.h" +#define WM8993_NUM_SUPPLIES 6 +static const char *wm8993_supply_names[WM8993_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD", + "AVDD1", + "AVDD2", + "CPVDD", + "SPKVDD", +}; + static u16 wm8993_reg_defaults[WM8993_REGISTER_COUNT] = { 0x8993, /* R0 - Software Reset */ 0x0000, /* R1 - Power Management (1) */ @@ -215,6 +226,7 @@ static struct { struct wm8993_priv { struct wm_hubs_data hubs_data; u16 reg_cache[WM8993_REGISTER_COUNT]; + struct regulator_bulk_data supplies[WM8993_NUM_SUPPLIES]; struct wm8993_platform_data pdata; struct snd_soc_codec codec; int master; @@ -1496,6 +1508,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, struct snd_soc_codec *codec; unsigned int val; int ret; + int i; if (wm8993_codec) { dev_err(&i2c->dev, "A WM8993 is already registered\n"); @@ -1543,16 +1556,33 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, codec->dev = &i2c->dev; + for (i = 0; i < ARRAY_SIZE(wm8993->supplies); i++) + wm8993->supplies[i].supply = wm8993_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + val = snd_soc_read(codec, WM8993_SOFTWARE_RESET); if (val != wm8993_reg_defaults[WM8993_SOFTWARE_RESET]) { dev_err(codec->dev, "Invalid ID register value %x\n", val); ret = -EINVAL; - goto err; + goto err_enable; } ret = snd_soc_write(codec, WM8993_SOFTWARE_RESET, 0xffff); if (ret != 0) - goto err; + goto err_enable; /* By default we're using the output mixers */ wm8993->class_w_users = 2; @@ -1582,7 +1612,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret != 0) - goto err; + goto err_enable; wm8993_dai.dev = codec->dev; @@ -1596,6 +1626,10 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, err_bias: wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); err: wm8993_codec = NULL; kfree(wm8993); @@ -1610,6 +1644,7 @@ static int wm8993_i2c_remove(struct i2c_client *client) snd_soc_unregister_dai(&wm8993_dai); wm8993_set_bias_level(&wm8993->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); kfree(wm8993); return 0; -- cgit v1.2.2 From cf56f62746c3e2f70bfad3d6fd051427a0022368 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 17:55:55 +0000 Subject: ASoC: Disable WM8993 regulators when turning bias off While the regulators are disabled we cache all register writes. Currently we assume that the regulator disable actually takes effect, after the merge with the regulator tree in 2.6.34 the regulator API will be able to notify us if the power is actually removed (due to constraints or regulator sharing it may not be). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 54 +++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 45 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index e97b3f45b24b..bf022f68b84f 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -923,10 +923,33 @@ static const struct snd_soc_dapm_route routes[] = { { "Right Headphone Mux", "DAC", "DACR" }, }; +static void wm8993_cache_restore(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i; + + if (!codec->cache_sync) + return; + + /* Reenable hardware writes */ + codec->cache_only = 0; + + /* Restore the register settings */ + for (i = 1; i < WM8993_MAX_REGISTER; i++) { + if (cache[i] == wm8993_reg_defaults[i]) + continue; + snd_soc_write(codec, i, cache[i]); + } + + /* We're in sync again */ + codec->cache_sync = 0; +} + static int wm8993_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8993_priv *wm8993 = codec->private_data; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -940,6 +963,13 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) + return ret; + + wm8993_cache_restore(codec); + /* Tune DC servo configuration */ snd_soc_write(codec, 0x44, 3); snd_soc_write(codec, 0x56, 3); @@ -992,6 +1022,18 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, WM8993_VMID_SEL_MASK | WM8993_BIAS_ENA, 0); + +#ifdef CONFIG_REGULATOR + /* Post 2.6.34 we will be able to get a callback when + * the regulators are disabled which we can use but + * for now just assume that the power will be cut if + * the regulator API is in use. + */ + codec->cache_sync = 1; +#endif + + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); break; } @@ -1460,15 +1502,7 @@ static int wm8993_resume(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; struct wm8993_priv *wm8993 = codec->private_data; - u16 *cache = wm8993->reg_cache; - int i, ret; - - /* Restore the register settings */ - for (i = 1; i < WM8993_MAX_REGISTER; i++) { - if (cache[i] == wm8993_reg_defaults[i]) - continue; - snd_soc_write(codec, i, cache[i]); - } + int ret; wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1584,6 +1618,8 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_enable; + codec->cache_only = 1; + /* By default we're using the output mixers */ wm8993->class_w_users = 2; -- cgit v1.2.2 From c133421800d9d1dfec0c98de6c9da0a7a99e0573 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Jan 2010 22:37:11 +0000 Subject: ASoC: Add support for BIAS_OFF when idle to WM8904 As well as disabling the biases of the CODEC the drop into BIAS_OFF will also disable all the regulators powering the CODEC, allowing even greater power savings on appropriately configured systems. Since the regulator API does not currently provide notification when regulators are disabled we assume that this always happens when we stop using the regulators. Once 2.6.34 is merged this code can be optimised to only sync the cache when power was actually removed. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8904.c | 52 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 39 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 992a7f23df5c..dc782c43a7cb 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2033,11 +2033,37 @@ static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static void wm8904_sync_cache(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + /* Sync back cached values if they're different from the + * hardware default. + */ + for (i = 1; i < ARRAY_SIZE(wm8904->reg_cache); i++) { + if (!wm8904_access[i].writable) + continue; + + if (wm8904->reg_cache[i] == wm8904_reg[i]) + continue; + + snd_soc_write(codec, i, wm8904->reg_cache[i]); + } + + codec->cache_sync = 0; +} + static int wm8904_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8904_priv *wm8904 = codec->private_data; - int ret, i; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -2065,18 +2091,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } - /* Sync back cached values if they're - * different from the hardware default. - */ - for (i = 1; i < ARRAY_SIZE(wm8904->reg_cache); i++) { - if (!wm8904_access[i].writable) - continue; - - if (wm8904->reg_cache[i] == wm8904_reg[i]) - continue; - - snd_soc_write(codec, i, wm8904->reg_cache[i]); - } + wm8904_sync_cache(codec); /* Enable bias */ snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, @@ -2112,6 +2127,15 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); +#ifdef CONFIG_REGULATOR + /* Post 2.6.34 we will be able to get a callback when + * the regulators are disabled which we can use but + * for now just assume that the power will be cut if + * the regulator API is in use. + */ + codec->cache_sync = 1; +#endif + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); break; @@ -2365,6 +2389,8 @@ static int wm8904_register(struct wm8904_priv *wm8904, codec->reg_cache_size = WM8904_MAX_REGISTER; codec->reg_cache = &wm8904->reg_cache; codec->volatile_register = wm8904_volatile_register; + codec->cache_sync = 1; + codec->idle_bias_off = 1; memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); -- cgit v1.2.2 From e4bc669610d75106a00b0f96f2410ac5898ef1ca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 19:51:33 +0000 Subject: ASoC: Optimise WM8904 output stage power control Handle the output PGAs as part of the output powerup since they can never be powered separately and reorder things so that we remove the output shorts after both line and headphone outputs have been brought up, minimising the opportunity for any issues. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8904.c | 34 +++++++++++++++++++++++++++------- 1 file changed, 27 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index dc782c43a7cb..80dd8df0b864 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -979,6 +979,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, int dcs_l, dcs_r; int dcs_l_reg, dcs_r_reg; int timeout; + int pwr_reg; /* This code is shared between HP and LINEOUT; we do all our * power management in stereo pairs to avoid latency issues so @@ -988,6 +989,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, switch (reg) { case WM8904_ANALOGUE_HP_0: + pwr_reg = WM8904_POWER_MANAGEMENT_2; dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1; dcs_r_reg = WM8904_DC_SERVO_8; dcs_l_reg = WM8904_DC_SERVO_9; @@ -995,6 +997,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_r = 1; break; case WM8904_ANALOGUE_LINEOUT_0: + pwr_reg = WM8904_POWER_MANAGEMENT_3; dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3; dcs_r_reg = WM8904_DC_SERVO_6; dcs_l_reg = WM8904_DC_SERVO_7; @@ -1007,12 +1010,18 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, } switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: + /* Power on the PGAs */ + snd_soc_update_bits(codec, pwr_reg, + WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA, + WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA); + /* Power on the amplifier */ snd_soc_update_bits(codec, reg, WM8904_HPL_ENA | WM8904_HPR_ENA, WM8904_HPL_ENA | WM8904_HPR_ENA); + /* Enable the first stage */ snd_soc_update_bits(codec, reg, WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY, @@ -1064,7 +1073,9 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, reg, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP); + break; + case SND_SOC_DAPM_POST_PMU: /* Unshort the output itself */ snd_soc_update_bits(codec, reg, WM8904_HPL_RMV_SHORT | @@ -1079,7 +1090,9 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, reg, WM8904_HPL_RMV_SHORT | WM8904_HPR_RMV_SHORT, 0); + break; + case SND_SOC_DAPM_POST_PMD: /* Cache the DC servo configuration; this will be * invalidated if we change the configuration. */ wm8904->dcs_state[dcs_l] = snd_soc_read(codec, dcs_l_reg); @@ -1094,6 +1107,11 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY | WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, 0); + + /* PGAs too */ + snd_soc_update_bits(codec, pwr_reg, + WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA, + 0); break; } @@ -1212,18 +1230,20 @@ SND_SOC_DAPM_DAC("DACR", NULL, WM8904_POWER_MANAGEMENT_6, 2, 0), SND_SOC_DAPM_SUPPLY("Charge pump", WM8904_CHARGE_PUMP_0, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("HPL PGA", WM8904_POWER_MANAGEMENT_2, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA("HPR PGA", WM8904_POWER_MANAGEMENT_2, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("HPL PGA", SND_SOC_NOPM, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("HPR PGA", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_PGA("LINEL PGA", WM8904_POWER_MANAGEMENT_3, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA("LINER PGA", WM8904_POWER_MANAGEMENT_3, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINEL PGA", SND_SOC_NOPM, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINER PGA", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM, WM8904_ANALOGUE_HP_0, 0, NULL, 0, out_pga_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA_E("Line Output", SND_SOC_NOPM, WM8904_ANALOGUE_LINEOUT_0, 0, NULL, 0, out_pga_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), -- cgit v1.2.2 From 8c1264740e7c9688c5d11b96d26e4393618ef60e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 19:33:49 +0000 Subject: ASoC: Add WM8912 DAC support The WM8912 is a DAC only device register compatible with the WM8904 CODEC with ADC portions omitted. Support it within the WM8904 driver based on the configured I2C device name. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8904.c | 90 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 72 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 80dd8df0b864..593e47d0e0eb 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -33,6 +33,11 @@ static struct snd_soc_codec *wm8904_codec; struct snd_soc_codec_device soc_codec_dev_wm8904; +enum wm8904_type { + WM8904, + WM8912, +}; + #define WM8904_NUM_DCS_CHANNELS 4 #define WM8904_NUM_SUPPLIES 5 @@ -49,6 +54,8 @@ struct wm8904_priv { struct snd_soc_codec codec; u16 reg_cache[WM8904_MAX_REGISTER + 1]; + enum wm8904_type devtype; + struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES]; struct wm8904_pdata *pdata; @@ -1411,30 +1418,62 @@ static const struct snd_soc_dapm_route wm8904_intercon[] = { { "LINER PGA", NULL, "LINER Mux" }, }; +static const struct snd_soc_dapm_route wm8912_intercon[] = { + { "HPL PGA", NULL, "DACL" }, + { "HPR PGA", NULL, "DACR" }, + + { "LINEL PGA", NULL, "DACL" }, + { "LINER PGA", NULL, "DACR" }, +}; + static int wm8904_add_widgets(struct snd_soc_codec *codec) { - snd_soc_add_controls(codec, wm8904_adc_snd_controls, - ARRAY_SIZE(wm8904_adc_snd_controls)); - snd_soc_add_controls(codec, wm8904_dac_snd_controls, - ARRAY_SIZE(wm8904_dac_snd_controls)); - snd_soc_add_controls(codec, wm8904_snd_controls, - ARRAY_SIZE(wm8904_snd_controls)); + struct wm8904_priv *wm8904 = codec->private_data; snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets, ARRAY_SIZE(wm8904_core_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets, - ARRAY_SIZE(wm8904_adc_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, - ARRAY_SIZE(wm8904_dac_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets, - ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(codec, core_intercon, ARRAY_SIZE(core_intercon)); - snd_soc_dapm_add_routes(codec, adc_intercon, ARRAY_SIZE(adc_intercon)); - snd_soc_dapm_add_routes(codec, dac_intercon, ARRAY_SIZE(dac_intercon)); - snd_soc_dapm_add_routes(codec, wm8904_intercon, - ARRAY_SIZE(wm8904_intercon)); + + switch (wm8904->devtype) { + case WM8904: + snd_soc_add_controls(codec, wm8904_adc_snd_controls, + ARRAY_SIZE(wm8904_adc_snd_controls)); + snd_soc_add_controls(codec, wm8904_dac_snd_controls, + ARRAY_SIZE(wm8904_dac_snd_controls)); + snd_soc_add_controls(codec, wm8904_snd_controls, + ARRAY_SIZE(wm8904_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets, + ARRAY_SIZE(wm8904_adc_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + ARRAY_SIZE(wm8904_dac_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets, + ARRAY_SIZE(wm8904_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, core_intercon, + ARRAY_SIZE(core_intercon)); + snd_soc_dapm_add_routes(codec, adc_intercon, + ARRAY_SIZE(adc_intercon)); + snd_soc_dapm_add_routes(codec, dac_intercon, + ARRAY_SIZE(dac_intercon)); + snd_soc_dapm_add_routes(codec, wm8904_intercon, + ARRAY_SIZE(wm8904_intercon)); + break; + + case WM8912: + snd_soc_add_controls(codec, wm8904_dac_snd_controls, + ARRAY_SIZE(wm8904_dac_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + ARRAY_SIZE(wm8904_dac_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, dac_intercon, + ARRAY_SIZE(dac_intercon)); + snd_soc_dapm_add_routes(codec, wm8912_intercon, + ARRAY_SIZE(wm8912_intercon)); + break; + } snd_soc_dapm_new_widgets(codec); return 0; @@ -2412,6 +2451,18 @@ static int wm8904_register(struct wm8904_priv *wm8904, codec->cache_sync = 1; codec->idle_bias_off = 1; + switch (wm8904->devtype) { + case WM8904: + break; + case WM8912: + memset(&wm8904_dai.capture, 0, sizeof(wm8904_dai.capture)); + break; + default: + dev_err(codec->dev, "Unknown device type %d\n", + wm8904->devtype); + return -EINVAL; + } + memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); @@ -2542,6 +2593,8 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, codec = &wm8904->codec; codec->hw_write = (hw_write_t)i2c_master_send; + wm8904->devtype = id->driver_data; + i2c_set_clientdata(i2c, wm8904); codec->control_data = i2c; wm8904->pdata = i2c->dev.platform_data; @@ -2559,7 +2612,8 @@ static __devexit int wm8904_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8904_i2c_id[] = { - { "wm8904", 0 }, + { "wm8904", WM8904 }, + { "wm8912", WM8912 }, { } }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); -- cgit v1.2.2 From cb67286d6629ecb5bfc44071d664cf1cbd01a350 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 4 Feb 2010 09:10:10 +0200 Subject: ASoC: TWL4030: Module unloading fix The module unloading path had several problems: - it freed up the private structure twice - it freed up the codec structure, which was allocated as part of the private structure - it did not freed up the reg_cache - it did not unregistered the dais and the codec Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e0106a5fd40b..b32aeb38e3a6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2152,8 +2152,6 @@ static int twl4030_soc_remove(struct platform_device *pdev) twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - kfree(codec->private_data); - kfree(codec); return 0; } @@ -2237,6 +2235,9 @@ static int __devexit twl4030_codec_remove(struct platform_device *pdev) { struct twl4030_priv *twl4030 = platform_get_drvdata(pdev); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + snd_soc_unregister_codec(&twl4030->codec); + kfree(twl4030->codec.reg_cache); kfree(twl4030); twl4030_codec = NULL; -- cgit v1.2.2 From 88102f3f841b680412714d0b0b7da33c2a00c1f9 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 4 Feb 2010 14:12:58 +0100 Subject: ALSA: hda - Remove superfluous init verb entries for ALC88[235] The default values are no need to be set in init_verbs. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 75 +++++++------------------------------------ 1 file changed, 12 insertions(+), 63 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b2f543d3b833..40ebf2746bb1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7332,29 +7332,18 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = { static struct hda_verb alc882_base_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* mute analog input loopbacks */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Front Pin: output 0 (0x0c) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -7391,14 +7380,8 @@ static struct hda_verb alc882_base_init_verbs[] = { /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* Input mixer3 */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* ADC2: mute amp left and right */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -7442,26 +7425,17 @@ static struct hda_verb alc_hp15_unsol_verbs[] = { static struct hda_verb alc885_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* mute analog input loopbacks */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Front HP Pin: output 0 (0x0c) */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -7495,17 +7469,11 @@ static struct hda_verb alc885_init_verbs[] = { /* Mixer elements: 0x18, , 0x1a, 0x1b */ /* Input mixer1 */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* ADC2: mute amp left and right */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* ADC3: mute amp left and right */ @@ -7991,18 +7959,6 @@ static struct hda_verb alc883_auto_init_verbs[] = { {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* * Set up output mixers (0x0c - 0x0f) */ @@ -8027,16 +7983,9 @@ static struct hda_verb alc883_auto_init_verbs[] = { /* FIXME: use matrix-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } }; -- cgit v1.2.2 From 84898e87cc0fff976202d5b91656f2db949fc2dd Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 4 Feb 2010 14:16:14 +0100 Subject: ALSA: hda - Add ALC269VB support - Add new models ALC269VB_AMIC ALC269VB_DMIC - Add alc269vb_laptop_dmic_setup The record source index Dmic is 0x6 for ALC269VB. - Change eeepc words for ALC269 - Modify init_verb tables of patch_alc269 patch_alc662 patch_alc882 - Modify common patch for ALC270 ALC269VB ALC275 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 346 ++++++++++++++++++++++++++++++------------ 1 file changed, 246 insertions(+), 100 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 40ebf2746bb1..826ecdbdd2bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -131,8 +131,10 @@ enum { enum { ALC269_BASIC, ALC269_QUANTA_FL1, - ALC269_ASUS_AMIC, - ALC269_ASUS_DMIC, + ALC269_AMIC, + ALC269_DMIC, + ALC269VB_AMIC, + ALC269VB_DMIC, ALC269_FUJITSU, ALC269_LIFEBOOK, ALC269_AUTO, @@ -13182,6 +13184,15 @@ static hda_nid_t alc269_capsrc_nids[1] = { 0x23, }; +static hda_nid_t alc269vb_adc_nids[1] = { + /* ADC1 */ + 0x09, +}; + +static hda_nid_t alc269vb_capsrc_nids[1] = { + 0x22, +}; + /* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24), * not a mux! */ @@ -13250,7 +13261,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -static struct snd_kcontrol_new alc269_eeepc_mixer[] = { +static struct snd_kcontrol_new alc269_laptop_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -13258,16 +13269,47 @@ static struct snd_kcontrol_new alc269_eeepc_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc269vb_laptop_mixer[] = { + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + { } /* end */ +}; + /* capture mixer elements */ -static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { +static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("IntMic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("IntMic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), { } /* end */ }; /* FSC amilo */ -#define alc269_fujitsu_mixer alc269_eeepc_mixer +#define alc269_fujitsu_mixer alc269_laptop_mixer static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -13410,7 +13452,7 @@ static void alc269_lifebook_init_hook(struct hda_codec *codec) alc269_lifebook_mic_autoswitch(codec); } -static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { +static struct hda_verb alc269_laptop_dmic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -13421,7 +13463,7 @@ static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { {} }; -static struct hda_verb alc269_eeepc_amic_init_verbs[] = { +static struct hda_verb alc269_laptop_amic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -13431,6 +13473,28 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = { {} }; +static struct hda_verb alc269vb_laptop_dmic_init_verbs[] = { + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x06}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc269vb_laptop_amic_init_verbs[] = { + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* toggle speaker-output according to the hp-jack state */ static void alc269_speaker_automute(struct hda_codec *codec) { @@ -13448,7 +13512,7 @@ static void alc269_speaker_automute(struct hda_codec *codec) } /* unsolicited event for HP jack sensing */ -static void alc269_eeepc_unsol_event(struct hda_codec *codec, +static void alc269_laptop_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { @@ -13461,7 +13525,7 @@ static void alc269_eeepc_unsol_event(struct hda_codec *codec, } } -static void alc269_eeepc_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->ext_mic.pin = 0x18; @@ -13471,7 +13535,17 @@ static void alc269_eeepc_dmic_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc269_eeepc_amic_setup(struct hda_codec *codec) +static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 6; + spec->auto_mic = 1; +} + +static void alc269_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->ext_mic.pin = 0x18; @@ -13481,7 +13555,7 @@ static void alc269_eeepc_amic_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc269_eeepc_inithook(struct hda_codec *codec) +static void alc269_laptop_inithook(struct hda_codec *codec) { alc269_speaker_automute(codec); alc_mic_automute(codec); @@ -13494,22 +13568,10 @@ static struct hda_verb alc269_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the - * analog-loopback mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* - * Set up output mixers (0x0c - 0x0e) + * Set up output mixers (0x02 - 0x03) */ /* set vol=0 to output mixers */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -13534,26 +13596,57 @@ static struct hda_verb alc269_init_verbs[] = { {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* FIXME: use Mux-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* FIXME: use matrix-type input source selection */ + /* set EAPD */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + +static struct hda_verb alc269vb_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* + * Set up output mixers (0x02 - 0x03) + */ + /* set vol=0 to output mixers */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* FIXME: use Mux-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x00}, /* set EAPD */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -13601,6 +13694,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; + hda_nid_t real_capsrc_nids; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc269_ignore); @@ -13622,11 +13716,20 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - add_verb(spec, alc269_init_verbs); + if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { + add_verb(spec, alc269vb_init_verbs); + real_capsrc_nids = alc269vb_capsrc_nids[0]; + alc_ssid_check(codec, 0x21, 0x1b, 0x14); + } else { + add_verb(spec, alc269_init_verbs); + real_capsrc_nids = alc269_capsrc_nids[0]; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + } + spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; /* set default input source */ - snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], + snd_hda_codec_write_cache(codec, real_capsrc_nids, 0, AC_VERB_SET_CONNECT_SEL, spec->input_mux->items[0].index); @@ -13637,8 +13740,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(codec); - alc_ssid_check(codec, 0x15, 0x1b, 0x14); - return 1; } @@ -13664,8 +13765,8 @@ static void alc269_auto_init(struct hda_codec *codec) static const char *alc269_models[ALC269_MODEL_LAST] = { [ALC269_BASIC] = "basic", [ALC269_QUANTA_FL1] = "quanta", - [ALC269_ASUS_AMIC] = "asus-amic", - [ALC269_ASUS_DMIC] = "asus-dmic", + [ALC269_AMIC] = "laptop-amic", + [ALC269_DMIC] = "laptop-dmic", [ALC269_FUJITSU] = "fujitsu", [ALC269_LIFEBOOK] = "lifebook", [ALC269_AUTO] = "auto", @@ -13674,41 +13775,49 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { static struct snd_pci_quirk alc269_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1), SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", - ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80JT", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_DMIC), - SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_AMIC), + ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC), + SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901", - ALC269_ASUS_DMIC), + ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", - ALC269_ASUS_DMIC), - SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_ASUS_DMIC), - SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_ASUS_DMIC), + ALC269_DMIC), + SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), + SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), {} @@ -13738,47 +13847,75 @@ static struct alc_config_preset alc269_presets[] = { .setup = alc269_quanta_fl1_setup, .init_hook = alc269_quanta_fl1_init_hook, }, - [ALC269_ASUS_AMIC] = { - .mixers = { alc269_eeepc_mixer }, - .cap_mixer = alc269_epc_capture_mixer, + [ALC269_AMIC] = { + .mixers = { alc269_laptop_mixer }, + .cap_mixer = alc269_laptop_analog_capture_mixer, .init_verbs = { alc269_init_verbs, - alc269_eeepc_amic_init_verbs }, + alc269_laptop_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_eeepc_unsol_event, - .setup = alc269_eeepc_amic_setup, - .init_hook = alc269_eeepc_inithook, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_amic_setup, + .init_hook = alc269_laptop_inithook, }, - [ALC269_ASUS_DMIC] = { - .mixers = { alc269_eeepc_mixer }, - .cap_mixer = alc269_epc_capture_mixer, + [ALC269_DMIC] = { + .mixers = { alc269_laptop_mixer }, + .cap_mixer = alc269_laptop_digital_capture_mixer, .init_verbs = { alc269_init_verbs, - alc269_eeepc_dmic_init_verbs }, + alc269_laptop_dmic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_eeepc_unsol_event, - .setup = alc269_eeepc_dmic_setup, - .init_hook = alc269_eeepc_inithook, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_dmic_setup, + .init_hook = alc269_laptop_inithook, + }, + [ALC269VB_AMIC] = { + .mixers = { alc269vb_laptop_mixer }, + .cap_mixer = alc269vb_laptop_analog_capture_mixer, + .init_verbs = { alc269vb_init_verbs, + alc269vb_laptop_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_amic_setup, + .init_hook = alc269_laptop_inithook, + }, + [ALC269VB_DMIC] = { + .mixers = { alc269vb_laptop_mixer }, + .cap_mixer = alc269vb_laptop_digital_capture_mixer, + .init_verbs = { alc269vb_init_verbs, + alc269vb_laptop_dmic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269vb_laptop_dmic_setup, + .init_hook = alc269_laptop_inithook, }, [ALC269_FUJITSU] = { .mixers = { alc269_fujitsu_mixer }, - .cap_mixer = alc269_epc_capture_mixer, + .cap_mixer = alc269_laptop_digital_capture_mixer, .init_verbs = { alc269_init_verbs, - alc269_eeepc_dmic_init_verbs }, + alc269_laptop_dmic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_eeepc_unsol_event, - .setup = alc269_eeepc_dmic_setup, - .init_hook = alc269_eeepc_inithook, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_dmic_setup, + .init_hook = alc269_laptop_inithook, }, [ALC269_LIFEBOOK] = { .mixers = { alc269_lifebook_mixer }, @@ -13799,6 +13936,7 @@ static int patch_alc269(struct hda_codec *codec) struct alc_spec *spec; int board_config; int err; + int is_alc269vb = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -13815,6 +13953,7 @@ static int patch_alc269(struct hda_codec *codec) alc_free(codec); return -ENOMEM; } + is_alc269vb = 1; } board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, @@ -13850,7 +13989,7 @@ static int patch_alc269(struct hda_codec *codec) if (board_config != ALC269_AUTO) setup_preset(codec, &alc269_presets[board_config]); - if (codec->subsystem_id == 0x17aa3bf8) { + if (board_config == ALC269_QUANTA_FL1) { /* Due to a hardware problem on Lenovo Ideadpad, we need to * fix the sample rate of analog I/O to 44.1kHz */ @@ -13863,9 +14002,16 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; - spec->adc_nids = alc269_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->capsrc_nids = alc269_capsrc_nids; + if (!is_alc269vb) { + spec->adc_nids = alc269_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->capsrc_nids = alc269_capsrc_nids; + } else { + spec->adc_nids = alc269vb_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); + spec->capsrc_nids = alc269vb_capsrc_nids; + } + if (!spec->cap_mixer) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); -- cgit v1.2.2 From cec27c891b805b2ab2302f9fcbdacb6f179ac0d4 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 4 Feb 2010 14:18:18 +0100 Subject: ALSA: hda - Add support of ALC665 - Add support for ALC665 - Add more ASUS model - Modify common patch for ALC272 ALC273 ALC661 ALC662 ALC663 ALC665 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 112 +++++++++++++++++------------------------- 1 file changed, 44 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 826ecdbdd2bb..82772f0ab3e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16597,13 +16597,6 @@ static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -16653,6 +16646,28 @@ static struct hda_verb alc662_init_verbs[] = { { } }; +static struct hda_verb alc663_init_verbs[] = { + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } +}; + +static struct hda_verb alc272_init_verbs[] = { + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } +}; + static struct hda_verb alc662_sue_init_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, @@ -16672,61 +16687,6 @@ static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = { {} }; -/* - * generic initialization of ADC, input mixers and output mixers - */ -static struct hda_verb alc662_auto_init_verbs[] = { - /* - * Unmute ADC and set the default input to mic-in - */ - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for front - * panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* - * Set up output mixers (0x0c - 0x0f) - */ - /* set vol=0 to output mixers */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { } -}; - -/* additional verbs for ALC663 */ -static struct hda_verb alc663_auto_init_verbs[] = { - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } -}; - static struct hda_verb alc663_m51va_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -17477,6 +17437,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1), SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), @@ -17512,6 +17473,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), @@ -18157,9 +18119,13 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - add_verb(spec, alc662_auto_init_verbs); - if (codec->vendor_id == 0x10ec0663) - add_verb(spec, alc663_auto_init_verbs); + add_verb(spec, alc662_init_verbs); + if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || + codec->vendor_id == 0x10ec0665) + add_verb(spec, alc663_init_verbs); + + if (codec->vendor_id == 0x10ec0272) + add_verb(spec, alc272_init_verbs); err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -18251,11 +18217,20 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(codec); - if (codec->vendor_id == 0x10ec0662) + + switch (codec->vendor_id) { + case 0x10ec0662: set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - else + break; + case 0x10ec0272: + case 0x10ec0663: + case 0x10ec0665: set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); - + break; + case 0x10ec0273: + set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); + break; + } spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; @@ -18305,6 +18280,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, + { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, -- cgit v1.2.2 From 21956b61f594f7924d98240da74bc81c28601fa9 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 2 Feb 2010 19:58:25 +0100 Subject: ALSA: ctxfi - fix PTP address initialization After hours of debugging, I finally found the reason why some source and runtime combination does not work. The PTP (page table pages) address must be aligned. I am not sure how much, but alignment to PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines to ensure proper virtual -> physical address translation. Cc: Signed-off-by: Jaroslav Kysela --- sound/pci/ctxfi/ctatc.c | 15 ++------------- sound/pci/ctxfi/ctvmem.c | 38 ++++++++++++++++++-------------------- sound/pci/ctxfi/ctvmem.h | 8 +++++--- 3 files changed, 25 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index cb65bd0dd35b..459c1f62783b 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -166,18 +166,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index) { - struct ct_vm *vm; - void *kvirt_addr; - unsigned long phys_addr; - - vm = atc->vm; - kvirt_addr = vm->get_ptp_virt(vm, index); - if (kvirt_addr == NULL) - phys_addr = (~0UL); - else - phys_addr = virt_to_phys(kvirt_addr); - - return phys_addr; + return atc->vm->get_ptp_phys(atc->vm, index); } static unsigned int convert_format(snd_pcm_format_t snd_format) @@ -1669,7 +1658,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, } /* Set up device virtual memory management object */ - err = ct_vm_create(&atc->vm); + err = ct_vm_create(&atc->vm, pci); if (err < 0) goto error1; diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 6b78752e9503..65da6e466f80 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -138,7 +138,7 @@ ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size) return NULL; } - ptp = vm->ptp[0]; + ptp = (unsigned long *)vm->ptp[0].area; pte_start = (block->addr >> CT_PAGE_SHIFT); pages = block->size >> CT_PAGE_SHIFT; for (i = 0; i < pages; i++) { @@ -158,25 +158,25 @@ static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block) } /* * - * return the host (kmalloced) addr of the @index-th device - * page talbe page on success, or NULL on failure. - * The first returned NULL indicates the termination. + * return the host physical addr of the @index-th device + * page table page on success, or ~0UL on failure. + * The first returned ~0UL indicates the termination. * */ -static void * -ct_get_ptp_virt(struct ct_vm *vm, int index) +static dma_addr_t +ct_get_ptp_phys(struct ct_vm *vm, int index) { - void *addr; + dma_addr_t addr; - addr = (index >= CT_PTP_NUM) ? NULL : vm->ptp[index]; + addr = (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr; return addr; } -int ct_vm_create(struct ct_vm **rvm) +int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci) { struct ct_vm *vm; struct ct_vm_block *block; - int i; + int i, err = 0; *rvm = NULL; @@ -188,23 +188,21 @@ int ct_vm_create(struct ct_vm **rvm) /* Allocate page table pages */ for (i = 0; i < CT_PTP_NUM; i++) { - vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!vm->ptp[i]) + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(pci), + PAGE_SIZE, &vm->ptp[i]); + if (err < 0) break; } - if (!i) { + if (err < 0) { /* no page table pages are allocated */ - kfree(vm); + ct_vm_destroy(vm); return -ENOMEM; } vm->size = CT_ADDRS_PER_PAGE * i; - /* Initialise remaining ptps */ - for (; i < CT_PTP_NUM; i++) - vm->ptp[i] = NULL; - vm->map = ct_vm_map; vm->unmap = ct_vm_unmap; - vm->get_ptp_virt = ct_get_ptp_virt; + vm->get_ptp_phys = ct_get_ptp_phys; INIT_LIST_HEAD(&vm->unused); INIT_LIST_HEAD(&vm->used); block = kzalloc(sizeof(*block), GFP_KERNEL); @@ -242,7 +240,7 @@ void ct_vm_destroy(struct ct_vm *vm) /* free allocated page table pages */ for (i = 0; i < CT_PTP_NUM; i++) - kfree(vm->ptp[i]); + snd_dma_free_pages(&vm->ptp[i]); vm->size = 0; diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index 01e4fd0386a3..b23adfca4de6 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -22,6 +22,8 @@ #include #include +#include +#include /* The chip can handle the page table of 4k pages * (emu20k1 can handle even 8k pages, but we don't use it right now) @@ -41,7 +43,7 @@ struct snd_pcm_substream; /* Virtual memory management object for card device */ struct ct_vm { - void *ptp[CT_PTP_NUM]; /* Device page table pages */ + struct snd_dma_buffer ptp[CT_PTP_NUM]; /* Device page table pages */ unsigned int size; /* Available addr space in bytes */ struct list_head unused; /* List of unused blocks */ struct list_head used; /* List of used blocks */ @@ -52,10 +54,10 @@ struct ct_vm { int size); /* Unmap device logical addr area. */ void (*unmap)(struct ct_vm *, struct ct_vm_block *block); - void *(*get_ptp_virt)(struct ct_vm *vm, int index); + dma_addr_t (*get_ptp_phys)(struct ct_vm *vm, int index); }; -int ct_vm_create(struct ct_vm **rvm); +int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci); void ct_vm_destroy(struct ct_vm *vm); #endif /* CTVMEM_H */ -- cgit v1.2.2 From 350a514787a4516746f738f69bff6aa0d4ac70e9 Mon Sep 17 00:00:00 2001 From: Sebastien Alaiwan Date: Fri, 5 Feb 2010 08:58:20 +0100 Subject: ALSA: ice1712: fix: lock samplerate when samplerate locking is enabled I found that the sampling rate locking setting of the ice1712 sound driver was only half-respected : when the driver was locked to, let's say, 44100Hz, and a usermode app was requesting 48000Hz playback, the request was succesful although the soundcard would continue to run at 44100Hz. Here's a patch that will make those requests to fail. Signed-off-by: Sebastien Alaiwan Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index c7cff6f8168a..fb61943fc4dc 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1180,6 +1180,10 @@ static int snd_ice1712_playback_pro_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (is_pro_rate_locked(ice)) { + runtime->hw.rate_min = PRO_RATE_DEFAULT; + runtime->hw.rate_max = PRO_RATE_DEFAULT; + } if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); @@ -1197,6 +1201,11 @@ static int snd_ice1712_capture_pro_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (is_pro_rate_locked(ice)) { + runtime->hw.rate_min = PRO_RATE_DEFAULT; + runtime->hw.rate_max = PRO_RATE_DEFAULT; + } + return 0; } -- cgit v1.2.2 From 1eb6dc7dabcb4aa762d96f4f6978f3ef86321d68 Mon Sep 17 00:00:00 2001 From: Maxim Levitsky Date: Thu, 4 Feb 2010 22:21:47 +0200 Subject: ALSA: hda - Delay switching to polling mode if an interrupt was missing My sound codec seems sometimes (very rarely) to omit interrupts (ALC268) However, interrupt mode still works. Thus if we get timeout, poll the codec once. If we get 3 such polls in a row, then switch to polling mode. This patch is maybe an bandaid, but this might be a workaround for hardware bug. Signed-off-by: Maxim Levitsky Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 565de38a3fc7..d853e2c33bb7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -426,6 +426,7 @@ struct azx { /* flags */ int position_fix; + int poll_count; unsigned int running :1; unsigned int initialized :1; unsigned int single_cmd :1; @@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = { #define get_azx_dev(substream) (substream->runtime->private_data) static int azx_acquire_irq(struct azx *chip, int do_disconnect); - +static int azx_send_cmd(struct hda_bus *bus, unsigned int val); /* * Interface for HD codec */ @@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, { struct azx *chip = bus->private_data; unsigned long timeout; + int do_poll = 0; again: timeout = jiffies + msecs_to_jiffies(1000); for (;;) { - if (chip->polling_mode) { + if (chip->polling_mode || do_poll) { spin_lock_irq(&chip->reg_lock); azx_update_rirb(chip); spin_unlock_irq(&chip->reg_lock); @@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->rirb.cmds[addr]) { smp_rmb(); bus->rirb_error = 0; + + if (!do_poll) + chip->poll_count = 0; return chip->rirb.res[addr]; /* the last value */ } if (time_after(jiffies, timeout)) @@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } + if (!chip->polling_mode && chip->poll_count < 2) { + snd_printdd(SFX "azx_get_response timeout, " + "polling the codec once: last cmd=0x%08x\n", + chip->last_cmd[addr]); + do_poll = 1; + chip->poll_count++; + goto again; + } + + if (!chip->polling_mode) { snd_printk(KERN_WARNING SFX "azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", -- cgit v1.2.2 From 9492837a6f54b069e13e40e3c89898bb8837a386 Mon Sep 17 00:00:00 2001 From: Maxim Levitsky Date: Thu, 4 Feb 2010 22:26:37 +0200 Subject: ALSA: cosmetic: make hda intel interrupt name consistent with others This renames the interrupt name in /proc/interrupt. HDA Intel -> hda_intel This also eliminates space from the name, probably helping some parsers. Don't think anybody depends on this name in userspace Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d853e2c33bb7..b8faa6dc5abe 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2058,7 +2058,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) { if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, - "HDA Intel", chip)) { + "hda_intel", chip)) { printk(KERN_ERR "hda-intel: unable to grab IRQ %d, " "disabling device\n", chip->pci->irq); if (do_disconnect) -- cgit v1.2.2 From 9d4c7464458770d309169f7a7ce1ea6f8a4a7de5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 5 Feb 2010 10:19:41 +0100 Subject: ALSA: ice1724 - aureon - fix wm8770 volume offset The volume register is from 0..0x7f and 0..0x1a range is mute. Also, fix mute combining in wm_vol_put(). The wrong behaviour was noticed by Peter Christensen. Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/aureon.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 765d7bd4c3d4..9e66f6d306f8 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -703,11 +703,13 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho { unsigned char nvol; - if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) { nvol = 0; - else + } else { nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) / WM_VOL_MAX; + nvol += 0x1b; + } wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x180 | nvol); @@ -778,7 +780,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ for (ch = 0; ch < 2; ch++) { unsigned int vol = ucontrol->value.integer.value[ch]; if (vol > WM_VOL_MAX) - continue; + vol = WM_VOL_MAX; vol |= spec->master[ch] & WM_VOL_MUTE; if (vol != spec->master[ch]) { int dac; @@ -834,8 +836,8 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * for (i = 0; i < voices; i++) { unsigned int vol = ucontrol->value.integer.value[i]; if (vol > WM_VOL_MAX) - continue; - vol |= spec->vol[ofs+i]; + vol = WM_VOL_MAX; + vol |= spec->vol[ofs+i] & WM_VOL_MUTE; if (vol != spec->vol[ofs+i]) { spec->vol[ofs+i] = vol; idx = WM_DAC_ATTEN + ofs + i; -- cgit v1.2.2 From 3b9447fb7fa1829731290e64ef928d4f6461310a Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Fri, 5 Feb 2010 00:55:33 +0200 Subject: ASoC: pandora: Add APLL supply to fix audio output Pandora's external DAC is using 256*Fs output from the TWL4030 codec, and TWL4030 needs to have APLL enabled for it's 256*Fs output to function. Signed-off-by: Grazvydas Ignotas Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 71b2c161158d..68980c19a3bc 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { }; static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"PCM DAC", NULL, "APLL Enable"}, {"Headphone Amplifier", NULL, "PCM DAC"}, {"Line Out", NULL, "PCM DAC"}, {"Headphone Jack", NULL, "Headphone Amplifier"}, -- cgit v1.2.2 From c50749de02f272be6e09b9016e13a17307d29066 Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Fri, 5 Feb 2010 16:29:53 +0200 Subject: ASoC: pandora: Add DAC regulator support Pandora's external DAC is connected to VSIM TWL4030 supply, so let's start switching it too to save more power. Also DAC got it's own DAPM handler. Signed-off-by: Grazvydas Ignotas Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 42 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 37 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 68980c19a3bc..de10f76baded 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -40,6 +41,8 @@ #define PREFIX "ASoC omap3pandora: " +static struct regulator *omap3pandora_dac_reg; + static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, unsigned int fmt) { @@ -106,21 +109,37 @@ static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_CBS_CFS); } -static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, +static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { + /* + * The PCM1773 DAC datasheet requires 1ms delay between switching + * VCC power on/off and /PD pin high/low + */ if (SND_SOC_DAPM_EVENT_ON(event)) { + regulator_enable(omap3pandora_dac_reg); + mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); } else { - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); - mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + mdelay(1); + regulator_disable(omap3pandora_dac_reg); } return 0; } +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + else + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + + return 0; +} + /* * Audio paths on Pandora board: * @@ -130,7 +149,9 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM, + 0, 0, omap3pandora_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -306,8 +327,18 @@ static int __init omap3pandora_soc_init(void) goto fail2; } + omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc"); + if (IS_ERR(omap3pandora_dac_reg)) { + pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n", + dev_name(&omap3pandora_snd_device->dev), + PTR_ERR(omap3pandora_dac_reg)); + goto fail3; + } + return 0; +fail3: + platform_device_del(omap3pandora_snd_device); fail2: platform_device_put(omap3pandora_snd_device); fail1: @@ -320,6 +351,7 @@ module_init(omap3pandora_soc_init); static void __exit omap3pandora_soc_exit(void) { + regulator_put(omap3pandora_dac_reg); platform_device_unregister(omap3pandora_snd_device); gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); -- cgit v1.2.2 From 07f804495cb08c8fdf16eee8f7d90edce4a3c9c5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2010 15:06:13 +0100 Subject: ALSA: hda - Detect HP mute-LED GPIO setup from GPIO counts The GPIO pin number for the mute LED control on HP laptops can be determined more easily by checking the number of available GPIO pins of the codec chip. On a small package with up to 3 GPIOs, GPIO 0 is used while GPIO 3 is used for others. This fixes the missing mute GPIO for some HP laptops with new codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 21 ++++++++------------- 1 file changed, 8 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 834c5980fe5d..39961879c414 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4754,19 +4754,14 @@ static int hp_blike_system(u32 subsystem_id); static void set_hp_led_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - switch (codec->vendor_id) { - case 0x111d7608: - /* GPIO 0 */ - spec->gpio_led = 0x01; - break; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - /* GPIO 3 */ - spec->gpio_led = 0x08; - break; - } + unsigned int gpio; + + gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + gpio &= AC_GPIO_IO_COUNT; + if (gpio > 3) + spec->gpio_led = 0x08; /* GPIO 3 */ + else + spec->gpio_led = 0x01; /* GPIO 0 */ } /* -- cgit v1.2.2 From c21bd0254371c207636e84c9e033d13a6fe48d43 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2010 15:16:08 +0100 Subject: ALSA: hda - Merge HP mute-LED status callback on both IDT 92HD7x and 8x codecs Merge the mute-LED status callback function for both IDT 92HD7x and 8x codecs to one function. Also it's changed to check all DACs, and called in the initialization to sync with the current status. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 57 ++++++++++++++++++++---------------------- 1 file changed, 27 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 39961879c414..ea254235470d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4363,6 +4363,12 @@ static int stac92xx_init(struct hda_codec *codec) if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) stac_issue_unsol_event(codec, nid); } + +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* sync mute LED */ + if (spec->gpio_led && codec->patch_ops.check_power_status) + codec->patch_ops.check_power_status(codec, 0x01); +#endif if (spec->dac_list) stac92xx_power_down(codec); return 0; @@ -4909,6 +4915,11 @@ static int stac92xx_resume(struct hda_codec *codec) stac_issue_unsol_event(codec, spec->autocfg.line_out_pins[0]); } +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* sync mute LED */ + if (spec->gpio_led && codec->patch_ops.check_power_status) + codec->patch_ops.check_power_status(codec, 0x01); +#endif return 0; } @@ -4928,43 +4939,29 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + int i, muted = 1; - if (nid == 0x10) { - if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & - HDA_AMP_MUTE) - spec->gpio_data &= ~spec->gpio_led; /* orange */ - else - spec->gpio_data |= spec->gpio_led; /* white */ - - if (!spec->gpio_led_polarity) { - /* LED state is inverted on these systems */ - spec->gpio_data ^= spec->gpio_led; + for (i = 0; i < spec->multiout.num_dacs; i++) { + nid = spec->multiout.dac_nids[i]; + if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE)) { + muted = 0; /* something heard */ + break; } - - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, - spec->gpio_data); } + if (muted) + spec->gpio_data &= ~spec->gpio_led; /* orange */ + else + spec->gpio_data |= spec->gpio_led; /* white */ - return 0; -} - -static int idt92hd83xxx_hp_check_power_status(struct hda_codec *codec, - hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; + if (!spec->gpio_led_polarity) { + /* LED state is inverted on these systems */ + spec->gpio_data ^= spec->gpio_led; + } - if (nid != 0x13) - return 0; - if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data |= spec->gpio_led; /* mute LED on */ - else - spec->gpio_data &= ~spec->gpio_led; /* mute LED off */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); - return 0; } - #endif static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) @@ -5361,7 +5358,7 @@ again: spec->gpio_data |= spec->gpio_led; /* register check_power_status callback. */ codec->patch_ops.check_power_status = - idt92hd83xxx_hp_check_power_status; + stac92xx_hp_check_power_status; } #endif -- cgit v1.2.2 From b99a776d0b17ae0f3a54e86009887a00ac4889d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2010 15:21:09 +0100 Subject: ALSA: hda - Remove static gpio_led setup via model We have now a better mute-LED GPIO detection, and no need to assign the values statically per model option. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea254235470d..ec0637e7d488 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5343,9 +5343,6 @@ again: codec->patch_ops = stac92xx_patch_ops; - if (spec->board_config == STAC_92HD83XXX_HP) - spec->gpio_led = 0x01; - if (find_mute_led_gpio(codec)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, @@ -5673,7 +5670,6 @@ again: */ spec->num_smuxes = 1; spec->num_dmuxes = 1; - spec->gpio_led = 0x01; /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); @@ -5688,8 +5684,6 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; - /* orange/white mute led on GPIO3, orange=0, white=1 */ - spec->gpio_led = 0x08; break; } -- cgit v1.2.2 From dce17d4ff366230aeeaaf42512bba3711243cf1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Feb 2010 09:25:26 +0100 Subject: ALSA: hda - Fix default polarity of mute-LED GPIO on 92HD83x/88x codecs The previous commit caused a regression on HP laptops with 92HD83x/88x codecs. The default polarity of mute-LED GPIO is inverted on these devices. Reference: Novell bnc#578190 https://bugzilla.novell.com/show_bug.cgi?id=578190 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ec0637e7d488..8c416bb18a57 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4790,7 +4790,7 @@ static void set_hp_led_gpio(struct hda_codec *codec) * Need more information on whether it is true across the entire series. * -- kunal */ -static int find_mute_led_gpio(struct hda_codec *codec) +static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity) { struct sigmatel_spec *spec = codec->spec; const struct dmi_device *dev = NULL; @@ -4817,7 +4817,7 @@ static int find_mute_led_gpio(struct hda_codec *codec) */ if (!hp_blike_system(codec->subsystem_id)) { set_hp_led_gpio(codec); - spec->gpio_led_polarity = 1; + spec->gpio_led_polarity = default_polarity; return 1; } } @@ -5343,7 +5343,7 @@ again: codec->patch_ops = stac92xx_patch_ops; - if (find_mute_led_gpio(codec)) + if (find_mute_led_gpio(codec, 0)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); @@ -5705,7 +5705,7 @@ again: } } - if (find_mute_led_gpio(codec)) + if (find_mute_led_gpio(codec, 1)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); -- cgit v1.2.2 From cebe41d4b8f8092359de31e241815fcb4b4dc0be Mon Sep 17 00:00:00 2001 From: Alexey Dobriyan Date: Sat, 6 Feb 2010 00:21:03 +0200 Subject: sound: use DEFINE_PCI_DEVICE_TABLE Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to .devinit.rodata section, so they can be discarded in some cases, and make them const. Signed-off-by: Alexey Dobriyan Signed-off-by: Takashi Iwai --- sound/oss/kahlua.c | 2 +- sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 2 +- sound/pci/als4000.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au8810.c | 2 +- sound/pci/au88x0/au8820.c | 2 +- sound/pci/au88x0/au8830.c | 2 +- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/cmipci.c | 4 ++-- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs5530.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/xfi.c | 2 +- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigodjx.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/indigoiox.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 2 +- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/hifier.c | 2 +- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- sound/pci/pcxhr/pcxhr.c | 2 +- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sis7019.c | 2 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222.c | 2 +- sound/pci/ymfpci/ymfpci.c | 2 +- 67 files changed, 70 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 89466b056be7..24d152ccf80d 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -198,7 +198,7 @@ MODULE_LICENSE("GPL"); * 5530 only. The 5510/5520 decode is different. */ -static struct pci_device_id id_tbl[] = { +static DEFINE_PCI_DEVICE_TABLE(id_tbl) = { { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 8f5098f92c37..4382d0fa6b9a 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1048,7 +1048,7 @@ snd_ad1889_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_device_id snd_ad1889_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) }, { 0, }, }; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index aaf4da68969c..5c6e322a48f0 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -275,7 +275,7 @@ struct snd_ali { #endif }; -static struct pci_device_id snd_ali_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ali_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, {0, } }; diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 3aa35af7ca91..d7653cb7ac60 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -145,7 +145,7 @@ struct snd_als300_substream_data { int block_counter_register; }; -static struct pci_device_id snd_als300_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als300_ids) = { { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 }, { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS }, { 0, } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 3dbacde1a5af..d75cf7b06426 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -117,7 +117,7 @@ struct snd_card_als4000 { #endif }; -static struct pci_device_id snd_als4000_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als4000_ids) = { { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */ { 0, } }; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752dff2a44..81e2bfc11257 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -286,7 +286,7 @@ struct atiixp { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index e7e147bf8eb2..91d7036b6411 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -261,7 +261,7 @@ struct atiixp_modem { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index c0e8c6b295cb..aa51cc7771dd 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,6 +1,6 @@ #include "au8810.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index a6527330df58..2f321e7306cd 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,6 +1,6 @@ #include "au8820.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index 6c702ad4352a..279b78f06d22 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,6 +1,6 @@ #include "au8830.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 4d34bb0d99d3..67921f93a41e 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -164,7 +164,7 @@ MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); -static struct pci_device_id snd_aw2_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 69867ace7860..4679ed83a43b 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -350,7 +350,7 @@ struct snd_azf3328 { #endif }; -static const struct pci_device_id snd_azf3328_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_azf3328_ids) = { { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */ { 0, } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 4e2b925a94cc..37e1b5df5ab8 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -795,7 +795,7 @@ fail: .driver_data = SND_BT87X_BOARD_ ## id } /* driver_data is the card id for that device */ -static struct pci_device_id snd_bt87x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_ids) = { /* Hauppauge WinTV series */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC), /* Hauppauge WinTV series */ @@ -964,7 +964,7 @@ static void __devexit snd_bt87x_remove(struct pci_dev *pci) /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ -static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), { } diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 15e4138bce17..0a3d3d6e77b4 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1875,7 +1875,7 @@ static int snd_ca0106_resume(struct pci_dev *pci) #endif // PCI IDs -static struct pci_device_id snd_ca0106_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ { 0, } }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index a312bae08f52..1ded64e05643 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2796,7 +2796,7 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} #endif -static struct pci_device_id snd_cmipci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cmipci_ids) = { {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, @@ -3018,7 +3018,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc int integrated_midi = 0; char modelstr[16]; int pcm_index, pcm_spdif_index; - static struct pci_device_id intel_82437vx[] = { + static DEFINE_PCI_DEVICE_TABLE(intel_82437vx) = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, { }, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index e2e0359bb056..9edc65059e3e 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -494,7 +494,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_cs4281_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs4281_ids) = { { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 033aec430117..767fa7f06cd0 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -64,7 +64,7 @@ MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); -static struct pci_device_id snd_cs46xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs46xx_ids) = { { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index dc464321d0f3..207479a641cf 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -58,7 +58,7 @@ struct snd_cs5530 { unsigned long pci_base; }; -static struct pci_device_id snd_cs5530_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5530_ids) = { {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0}, {0,} diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 91e7faf69bbb..afb803708416 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); -static struct pci_device_id snd_cs5535audio_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5535audio_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index ed44ed788b60..f42e7e1a1074 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -43,7 +43,7 @@ MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); module_param_array(subsystem, int, NULL, 0444); MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); -static struct pci_device_id ct_pci_dev_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(ct_pci_dev_ids) = { /* only X-Fi is supported, so... */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1), .driver_data = ATC20K1, diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index 8c6db3aa3c1a..a65bafe0800f 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -63,7 +63,7 @@ static const struct firmware card_fw[] = { {0, "darla20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 04cbf3eaf05a..0a6c50bcd758 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "darla24_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index 4022e43a0053..f5142796989b 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -81,7 +81,7 @@ static const struct firmware card_fw[] = { {0, "3g_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */ {0,} }; diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index c0e64b8f52a4..2364f8a1bc21 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "gina20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index c36a78dd0b5e..616b55825a19 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -85,7 +85,7 @@ static const struct firmware card_fw[] = { {0, "gina24_361_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */ {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */ diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index 0a58a7c1fd7c..776175c0bdad 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */ {0,} }; diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index 2db24d29332b..8816b0bd2ba6 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dj_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ {0,} }; diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index 2e44316530a2..b1e3652f2f48 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_djx_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index a60c0a0a89b7..1035125336d6 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_io_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index eb3819f9654a..60b7cb2753cf 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_iox_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 506194688995..8c3f5c5b5301 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -76,7 +76,7 @@ static const struct firmware card_fw[] = { {0, "layla20_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index e09e3ea7781e..ed1cc0abc2b8 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -87,7 +87,7 @@ static const struct firmware card_fw[] = { {0, "layla24_2S_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index f05c8c097aa8..cc2bbfc65327 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -77,7 +77,7 @@ static const struct firmware card_fw[] = { {0, "mia_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */ {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */ {0,} diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index b05bad944901..3e7e01824b40 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -92,7 +92,7 @@ static const struct firmware card_fw[] = { {0, "mona_2_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */ {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */ diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 168af67d938e..4203782d7cb7 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -76,7 +76,7 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ -static struct pci_device_id snd_emu10k1_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1_ids) = { { PCI_VDEVICE(CREATIVE, 0x0002), 0 }, /* EMU10K1 */ { PCI_VDEVICE(CREATIVE, 0x0004), 1 }, /* Audigy */ { PCI_VDEVICE(CREATIVE, 0x0008), 1 }, /* Audigy 2 Value SB0400 */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 1d369ff73805..df47f738098d 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1605,7 +1605,7 @@ static void __devexit snd_emu10k1x_remove(struct pci_dev *pci) } // PCI IDs -static struct pci_device_id snd_emu10k1x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { { PCI_VDEVICE(CREATIVE, 0x0006), 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 2b82c5c723e1..c7fba5379813 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -443,7 +443,7 @@ struct ensoniq { static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_audiopci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_audiopci_ids) = { #ifdef CHIP1370 { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fb83e1ffa5cb..553b75217259 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -243,7 +243,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1938_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1938_ids) = { { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a11f453a6b6d..ecaea9fb48ec 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -551,7 +551,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1968_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1968_ids) = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 83508b3964fb..e1baad74ea4b 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -205,7 +205,7 @@ struct fm801 { #endif }; -static struct pci_device_id snd_fm801_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_fm801_ids) = { { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */ { 0, } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f516e668d88..ac05bef7c2ec 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2664,7 +2664,7 @@ static void __devexit azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static struct pci_device_id azx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* ICH 6..10 */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index fb61943fc4dc..4fc6d8bc637e 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -106,7 +106,7 @@ module_param_array(dxr_enable, int, NULL, 0444); MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); -static const struct pci_device_id snd_ice1712_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ice1712_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ae29073eea93..c1498fa5545f 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -94,7 +94,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ -static const struct pci_device_id snd_vt1724_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vt1724_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index b990143636f1..6433e65c9507 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -420,7 +420,7 @@ struct intel8x0 { u32 int_sta_mask; /* interrupt status mask */ }; -static struct pci_device_id snd_intel8x0_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0_ids) = { { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 9e7d12e7673f..13cec1e5ced9 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -219,7 +219,7 @@ struct intel8x0m { unsigned int pcm_pos_shift; }; -static struct pci_device_id snd_intel8x0m_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 7cc38a11e997..6d795700be79 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -418,7 +418,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); MODULE_AUTHOR("Haroldo Gamal "); -static struct pci_device_id snd_korg1212_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_korg1212_ids) = { { .vendor = 0x10b5, .device = 0x906d, diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 11b8c6514b3d..0cca56038cd9 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -55,7 +55,7 @@ static const char card_name[] = "LX6464ES"; #define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056 -static struct pci_device_id snd_lx6464es_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_lx6464es_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), .subvendor = PCI_VENDOR_ID_DIGIGRAM, .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 75283fbb4b3f..b64e78139d63 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -861,7 +861,7 @@ struct snd_m3 { /* * pci ids */ -static struct pci_device_id snd_m3_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID, diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a83d1968a845..7e8e7da592a9 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -60,7 +60,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); /* */ -static struct pci_device_id snd_mixart_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_mixart_ids) = { { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 97a0731331a1..5a60492ac7b3 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -262,7 +262,7 @@ struct nm256 { /* * PCI ids */ -static struct pci_device_id snd_nm256_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_nm256_ids) = { {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index e3c229b63311..5a87d683691f 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id hifier_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(hifier_ids) = { { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index acbedebcffd9..289cb4dacfc7 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -72,7 +72,7 @@ enum { MODEL_CLARO_HALO, /* HT-Omega Claro halo */ }; -static struct pci_device_id oxygen_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF }, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 563b6f50821f..f03a2f2cffee 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -40,7 +40,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id xonar_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 833e9c7b27c7..95cfde27d25c 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -94,7 +94,7 @@ enum { PCI_ID_LAST }; -static struct pci_device_id pcxhr_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b5ca02e2038c..bb08a2855fce 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -506,7 +506,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); /* */ -static struct pci_device_id snd_riptide_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { { PCI_DEVICE(0x127a, 0x4310) }, { PCI_DEVICE(0x127a, 0x4320) }, { PCI_DEVICE(0x127a, 0x4330) }, @@ -515,7 +515,7 @@ static struct pci_device_id snd_riptide_ids[] = { }; #ifdef SUPPORT_JOYSTICK -static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_joystick_ids) = { { PCI_DEVICE(0x127a, 0x4312) }, { PCI_DEVICE(0x127a, 0x4322) }, { PCI_DEVICE(0x127a, 0x4332) }, diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index f977dba7cbd0..d5e1c6eb7b7b 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -226,7 +226,7 @@ struct rme32 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme32_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme32_ids) = { {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 2ba5c0fd55db..9d5252bc870c 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -231,7 +231,7 @@ struct rme96 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme96_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme96_ids) = { { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7bb827c7d806..52c6eb57cc3f 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -585,7 +585,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_hdsp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdsp_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a1b10d1a384d..3d72c1effeef 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -512,7 +512,7 @@ static char channel_map_madi_ss[HDSPM_MAX_CHANNELS] = { }; -static struct pci_device_id snd_hdspm_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI, diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index bc539abb2105..44a3e2d8c556 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -314,7 +314,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_rme9652_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme9652_ids) = { { .vendor = 0x10ee, .device = 0x3fc4, diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 1a5ff0611072..7e3e8fbc90fe 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); -static struct pci_device_id snd_sis7019_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 1f6406c4534d..337b9facadfd 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -242,7 +242,7 @@ struct sonicvibes { #endif }; -static struct pci_device_id snd_sonic_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sonic_ids) = { { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 21cef97d478d..6d0581841d7a 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); -static struct pci_device_id snd_trident_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_trident_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8a332d2f615c..9595b5b535f3 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -401,7 +401,7 @@ struct via82xx { #endif }; -static struct pci_device_id snd_via82xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_ids) = { /* 0x1106, 0x3058 */ { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 47eb61561dfc..f7e8bbbe3953 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -260,7 +260,7 @@ struct via82xx_modem { struct snd_info_entry *proc_entry; }; -static struct pci_device_id snd_via82xx_modem_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_modem_ids) = { { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index fc9136c3e0d7..99a9a814be0b 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -60,7 +60,7 @@ enum { VX_PCI_VX222_NEW }; -static struct pci_device_id snd_vx222_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vx222_ids) = { { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */ { 0, } diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e6b18b90d451..80c682113381 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address"); module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); -static struct pci_device_id snd_ymfpci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ymfpci_ids) = { { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ -- cgit v1.2.2 From 3ad2f3fbb961429d2aa627465ae4829758bc7e07 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 3 Feb 2010 08:01:28 +0800 Subject: tree-wide: Assorted spelling fixes In particular, several occurances of funny versions of 'success', 'unknown', 'therefore', 'acknowledge', 'argument', 'achieve', 'address', 'beginning', 'desirable', 'separate' and 'necessary' are fixed. Signed-off-by: Daniel Mack Cc: Joe Perches Cc: Junio C Hamano Signed-off-by: Jiri Kosina --- sound/pci/rme9652/hdspm.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a1b10d1a384d..db0ed1cbd982 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2479,7 +2479,7 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol, on MADICARD - playback mixer matrix: [channelout+64] [output] [value] - input(thru) mixer matrix: [channelin] [output] [value] - (better do 2 kontrols for seperation ?) + (better do 2 kontrols for separation ?) */ #define HDSPM_MIXER(xname, xindex) \ diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 341481e0e830..427614a2762b 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -990,7 +990,7 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, reg = snd_soc_read(codec, WM8990_CLOCKING_2); snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); - /* set up N , fractional mode and pre-divisor if neccessary */ + /* set up N , fractional mode and pre-divisor if necessary */ snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | (pll_div.div2?WM8990_PRESCALE:0)); snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); -- cgit v1.2.2 From 71a157e8edca55198e808f8561dd49017a54ee34 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Mon, 1 Feb 2010 21:34:14 -0700 Subject: of: add 'of_' prefix to machine_is_compatible() machine is compatible is an OF-specific call. It should have the of_ prefix to protect the global namespace. Signed-off-by: Grant Likely Acked-by: Michal Simek --- sound/ppc/awacs.c | 24 ++++++++++++------------ sound/ppc/burgundy.c | 4 ++-- sound/ppc/pmac.c | 18 +++++++++--------- sound/soc/fsl/efika-audio-fabric.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- 5 files changed, 25 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 2e156467b814..b36679384b27 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -751,8 +751,8 @@ static void snd_pmac_awacs_suspend(struct snd_pmac *chip) static void snd_pmac_awacs_resume(struct snd_pmac *chip) { - if (machine_is_compatible("PowerBook3,1") - || machine_is_compatible("PowerBook3,2")) { + if (of_machine_is_compatible("PowerBook3,1") + || of_machine_is_compatible("PowerBook3,2")) { msleep(100); snd_pmac_awacs_write_reg(chip, 1, chip->awacs_reg[1] & ~MASK_PAROUT); @@ -780,16 +780,16 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) } #endif /* CONFIG_PM */ -#define IS_PM7500 (machine_is_compatible("AAPL,7500") \ - || machine_is_compatible("AAPL,8500") \ - || machine_is_compatible("AAPL,9500")) -#define IS_PM5500 (machine_is_compatible("AAPL,e411")) -#define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) -#define IS_IMAC1 (machine_is_compatible("PowerMac2,1")) -#define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \ - || machine_is_compatible("PowerMac4,1")) -#define IS_G4AGP (machine_is_compatible("PowerMac3,1")) -#define IS_LOMBARD (machine_is_compatible("PowerBook1,1")) +#define IS_PM7500 (of_machine_is_compatible("AAPL,7500") \ + || of_machine_is_compatible("AAPL,8500") \ + || of_machine_is_compatible("AAPL,9500")) +#define IS_PM5500 (of_machine_is_compatible("AAPL,e411")) +#define IS_BEIGE (of_machine_is_compatible("AAPL,Gossamer")) +#define IS_IMAC1 (of_machine_is_compatible("PowerMac2,1")) +#define IS_IMAC2 (of_machine_is_compatible("PowerMac2,2") \ + || of_machine_is_compatible("PowerMac4,1")) +#define IS_G4AGP (of_machine_is_compatible("PowerMac3,1")) +#define IS_LOMBARD (of_machine_is_compatible("PowerBook1,1")) static int imac1, imac2; diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 0accfe49735b..1f72e1c786bf 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -582,7 +582,7 @@ static int snd_pmac_burgundy_detect_headphone(struct snd_pmac *chip) static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_notify) { if (chip->auto_mute) { - int imac = machine_is_compatible("iMac"); + int imac = of_machine_is_compatible("iMac"); int reg, oreg; reg = oreg = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); @@ -620,7 +620,7 @@ static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_noti */ int __devinit snd_pmac_burgundy_init(struct snd_pmac *chip) { - int imac = machine_is_compatible("iMac"); + int imac = of_machine_is_compatible("iMac"); int i, err; /* Checks to see the chip is alive and kicking */ diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 7bc492ee77ec..85081172403f 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -922,11 +922,11 @@ static void __devinit detect_byte_swap(struct snd_pmac *chip) } /* it seems the Pismo & iBook can't byte-swap in hardware. */ - if (machine_is_compatible("PowerBook3,1") || - machine_is_compatible("PowerBook2,1")) + if (of_machine_is_compatible("PowerBook3,1") || + of_machine_is_compatible("PowerBook2,1")) chip->can_byte_swap = 0 ; - if (machine_is_compatible("PowerBook2,1")) + if (of_machine_is_compatible("PowerBook2,1")) chip->can_duplex = 0; } @@ -959,11 +959,11 @@ static int __devinit snd_pmac_detect(struct snd_pmac *chip) chip->control_mask = MASK_IEPC | MASK_IEE | 0x11; /* default */ /* check machine type */ - if (machine_is_compatible("AAPL,3400/2400") - || machine_is_compatible("AAPL,3500")) + if (of_machine_is_compatible("AAPL,3400/2400") + || of_machine_is_compatible("AAPL,3500")) chip->is_pbook_3400 = 1; - else if (machine_is_compatible("PowerBook1,1") - || machine_is_compatible("AAPL,PowerBook1998")) + else if (of_machine_is_compatible("PowerBook1,1") + || of_machine_is_compatible("AAPL,PowerBook1998")) chip->is_pbook_G3 = 1; chip->node = of_find_node_by_name(NULL, "awacs"); sound = of_node_get(chip->node); @@ -1033,8 +1033,8 @@ static int __devinit snd_pmac_detect(struct snd_pmac *chip) } if (of_device_is_compatible(sound, "tumbler")) { chip->model = PMAC_TUMBLER; - chip->can_capture = machine_is_compatible("PowerMac4,2") - || machine_is_compatible("PowerBook4,1"); + chip->can_capture = of_machine_is_compatible("PowerMac4,2") + || of_machine_is_compatible("PowerBook4,1"); chip->can_duplex = 0; // chip->can_byte_swap = 0; /* FIXME: check this */ chip->num_freqs = ARRAY_SIZE(tumbler_freqs); diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 3326e2a1e863..1a5b8e0d6a34 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -55,7 +55,7 @@ static __init int efika_fabric_init(void) struct platform_device *pdev; int rc; - if (!machine_is_compatible("bplan,efika")) + if (!of_machine_is_compatible("bplan,efika")) return -ENODEV; card.platform = &mpc5200_audio_dma_platform; diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index b928ef7d28eb..6644cba7cbf2 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -55,7 +55,7 @@ static __init int pcm030_fabric_init(void) struct platform_device *pdev; int rc; - if (!machine_is_compatible("phytec,pcm030")) + if (!of_machine_is_compatible("phytec,pcm030")) return -ENODEV; card.platform = &mpc5200_audio_dma_platform; -- cgit v1.2.2 From fed08d036f2aabd8d0c684439de37f8ebec2bbc2 Mon Sep 17 00:00:00 2001 From: Jody Bruchon Date: Sat, 6 Feb 2010 10:46:26 -0500 Subject: ALSA: hda-intel: Avoid divide by zero crash On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by zero for as-yet unknown reasons. A simple check for zero prevents it, though the problem that causes it remains. Since the workaround is harmless and won't affect anyone except victims of this bug, it should be safe; moreover, because this crash can be triggered by a user-mode application, there are denial of service implications on the systems affected by the bug without the patch. Signed-off-by: Jody Bruchon Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b8faa6dc5abe..e767c3f395ab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1893,6 +1893,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (!bdl_pos_adj[chip->dev_index]) return 1; /* no delayed ack */ + if (azx_dev->period_bytes == 0) { + printk(KERN_WARNING + "hda-intel: Divide by zero was avoided " + "in azx_dev->period_bytes.\n"); + return 0; + } if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) return 0; /* NG - it's below the period boundary */ return 1; /* OK, it's fine */ -- cgit v1.2.2 From 22313eafe92aeec1db9839f5afb71675bf2a5c33 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Feb 2010 10:42:33 +0000 Subject: ASoC: add phycore-ac97 sound support This patch adds sound support for Phytec PhyCORE / PhyCARD modules in AC97 mode. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Makefile | 2 + sound/soc/imx/phycore-ac97.c | 90 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) create mode 100644 sound/soc/imx/phycore-ac97.c (limited to 'sound') diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index d05cc95c5cc4..9f8bb92ddfcc 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -8,3 +8,5 @@ endif obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support +snd-soc-phycore-ac97-objs := phycore-ac97.o +obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c new file mode 100644 index 000000000000..a8307d55c70e --- /dev/null +++ b/sound/soc/imx/phycore-ac97.c @@ -0,0 +1,90 @@ +/* + * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode + * + * Copyright 2009 Sascha Hauer, Pengutronix + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/wm9712.h" +#include "imx-ssi.h" + +static struct snd_soc_card imx_phycore; + +static struct snd_soc_ops imx_phycore_hifi_ops = { +}; + +static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { + { + .name = "HiFi", + .stream_name = "HiFi", + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .ops = &imx_phycore_hifi_ops, + }, +}; + +static struct snd_soc_card imx_phycore = { + .name = "PhyCORE-audio", + .platform = &imx_soc_platform, + .dai_link = imx_phycore_dai_ac97, + .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), +}; + +static struct snd_soc_device imx_phycore_snd_devdata = { + .card = &imx_phycore, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *imx_phycore_snd_device; + +static int __init imx_phycore_init(void) +{ + int ret; + + if (!machine_is_pcm043() && !machine_is_pca100()) + /* return happy. We might run on a totally different machine */ + return 0; + + imx_phycore_snd_device = platform_device_alloc("soc-audio", -1); + if (!imx_phycore_snd_device) + return -ENOMEM; + + imx_phycore_dai_ac97[0].cpu_dai = &imx_ssi_pcm_dai[0]; + + platform_set_drvdata(imx_phycore_snd_device, &imx_phycore_snd_devdata); + imx_phycore_snd_devdata.dev = &imx_phycore_snd_device->dev; + ret = platform_device_add(imx_phycore_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(imx_phycore_snd_device); + } + + return ret; +} + +static void __exit imx_phycore_exit(void) +{ + platform_device_unregister(imx_phycore_snd_device); +} + +late_initcall(imx_phycore_init); +module_exit(imx_phycore_exit); + +MODULE_AUTHOR("Sascha Hauer "); +MODULE_DESCRIPTION("PhyCORE ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.2 From c0ff4bcd2e8505b09e0bedc74d08ad2da1e326f8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 9 Feb 2010 02:32:59 +0800 Subject: ASoC: cs4270: enable regulators at probe time Enable the bulk regulators at probe time so we can safely disable them again when going to suspend without confusing the reference counter. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 593bfc7a6986..dfbeb2db61b3 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -629,8 +629,17 @@ static int cs4270_probe(struct platform_device *pdev) if (ret < 0) goto error_free_pcms; + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + goto error_free_regulators; + return 0; +error_free_regulators: + regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + error_free_pcms: snd_soc_free_pcms(socdev); @@ -650,6 +659,7 @@ static int cs4270_remove(struct platform_device *pdev) struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; -- cgit v1.2.2 From c42a59ea277a8898b8f7c83fc89b00be225ea6aa Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Feb 2010 15:24:04 +0200 Subject: ASoC: TWL4030: Add supply for audio serial interface control The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b32aeb38e3a6..277862e480e2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -55,7 +55,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x0c, /* REG_ATXR1PGA (0xB) */ 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ - 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_AUDIO_IF (0xE) */ 0x00, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ @@ -1203,6 +1203,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1337,6 +1339,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1455,6 +1462,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "APLL Enable"}, {"ADC Virtual Right2", NULL, "APLL Enable"}, + {"ADC Virtual Left1", NULL, "AIF Enable"}, + {"ADC Virtual Right1", NULL, "AIF Enable"}, + {"ADC Virtual Left2", NULL, "AIF Enable"}, + {"ADC Virtual Right2", NULL, "AIF Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, -- cgit v1.2.2 From c6848bf566c7217a6090693ff5cc47091fa772f5 Mon Sep 17 00:00:00 2001 From: Paul Menzel Date: Tue, 9 Feb 2010 11:42:27 +0100 Subject: ASoC: Typo. s/Freecale/Freescale/ Signed-off-by: Paul Menzel Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 5f006f0d03dc..c7d0fd9b7de8 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,5 +1,5 @@ config SND_IMX_SOC - tristate "SoC Audio for Freecale i.MX CPUs" + tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC && BROKEN select SND_PCM select FIQ -- cgit v1.2.2 From c3a3e040f01457d2ea4f199f75ca205401001a3b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 11 Feb 2010 17:50:44 +0100 Subject: ALSA: usbmixer - add possibility to remap dB values USB devices tends to represent dB ranges in different way than ALSA expects. Add possibility to override these values and add guessed values for SoundBlaster MP3+. Also rename 'Capture Input Source' control to 'Capture Source' for SoundBlaster MP3+ and Extigy. Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 125 ++++++++++++++++++++++++++++------------------ sound/usb/usbmixer_maps.c | 23 ++++++--- 2 files changed, 93 insertions(+), 55 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index c998220b99c6..c72ad0c82581 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -123,6 +123,7 @@ struct usb_mixer_elem_info { int channels; int val_type; int min, max, res; + int dBmin, dBmax; int cached; int cache_val[MAX_CHANNELS]; u8 initialized; @@ -194,42 +195,50 @@ enum { */ #include "usbmixer_maps.c" -/* get the mapped name if the unit matches */ -static int check_mapped_name(struct mixer_build *state, int unitid, int control, char *buf, int buflen) +static const struct usbmix_name_map * +find_map(struct mixer_build *state, int unitid, int control) { - const struct usbmix_name_map *p; + const struct usbmix_name_map *p = state->map; - if (! state->map) - return 0; + if (!p) + return NULL; for (p = state->map; p->id; p++) { - if (p->id == unitid && p->name && - (! control || ! p->control || control == p->control)) { - buflen--; - return strlcpy(buf, p->name, buflen); - } + if (p->id == unitid && + (!control || !p->control || control == p->control)) + return p; } - return 0; + return NULL; } -/* check whether the control should be ignored */ -static int check_ignored_ctl(struct mixer_build *state, int unitid, int control) +/* get the mapped name if the unit matches */ +static int +check_mapped_name(const struct usbmix_name_map *p, char *buf, int buflen) { - const struct usbmix_name_map *p; + if (!p || !p->name) + return 0; - if (! state->map) + buflen--; + return strlcpy(buf, p->name, buflen); +} + +/* check whether the control should be ignored */ +static inline int +check_ignored_ctl(const struct usbmix_name_map *p) +{ + if (!p || p->name || p->dB) return 0; - for (p = state->map; p->id; p++) { - if (p->id == unitid && ! p->name && - (! control || ! p->control || control == p->control)) { - /* - printk(KERN_DEBUG "ignored control %d:%d\n", - unitid, control); - */ - return 1; - } + return 1; +} + +/* dB mapping */ +static inline void check_mapped_dB(const struct usbmix_name_map *p, + struct usb_mixer_elem_info *cval) +{ + if (p && p->dB) { + cval->dBmin = p->dB->min; + cval->dBmax = p->dB->max; } - return 0; } /* get the mapped selector source name */ @@ -466,20 +475,8 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, if (size < sizeof(scale)) return -ENOMEM; - /* USB descriptions contain the dB scale in 1/256 dB unit - * while ALSA TLV contains in 1/100 dB unit - */ - scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; - scale[3] = (convert_signed_value(cval, cval->max) * 100) / 256; - if (scale[3] <= scale[2]) { - /* something is wrong; assume it's either from/to 0dB */ - if (scale[2] < 0) - scale[3] = 0; - else if (scale[2] > 0) - scale[2] = 0; - else /* totally crap, return an error */ - return -EINVAL; - } + scale[2] = cval->dBmin; + scale[3] = cval->dBmax; if (copy_to_user(_tlv, scale, sizeof(scale))) return -EFAULT; return 0; @@ -720,6 +717,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) cval->min = default_min; cval->max = cval->min + 1; cval->res = 1; + cval->dBmin = cval->dBmax = 0; if (cval->val_type == USB_MIXER_BOOLEAN || cval->val_type == USB_MIXER_INV_BOOLEAN) { @@ -787,6 +785,24 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) cval->initialized = 1; } + + /* USB descriptions contain the dB scale in 1/256 dB unit + * while ALSA TLV contains in 1/100 dB unit + */ + cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 256; + cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 256; + if (cval->dBmin > cval->dBmax) { + /* something is wrong; assume it's either from/to 0dB */ + if (cval->dBmin < 0) + cval->dBmax = 0; + else if (cval->dBmin > 0) + cval->dBmin = 0; + if (cval->dBmin > cval->dBmax) { + /* totally crap, return an error */ + return -EINVAL; + } + } + return 0; } @@ -912,6 +928,7 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, int nameid = desc[desc[0] - 1]; struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; control++; /* change from zero-based to 1-based value */ @@ -920,7 +937,8 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, return; } - if (check_ignored_ctl(state, unitid, control)) + map = find_map(state, unitid, control); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -954,10 +972,11 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, } kctl->private_free = usb_mixer_elem_free; - len = check_mapped_name(state, unitid, control, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); mapped_name = len != 0; if (! len && nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, sizeof(kctl->id.name)); switch (control) { case USB_FEATURE_MUTE: @@ -995,6 +1014,7 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + check_mapped_dB(map, cval); } break; @@ -1122,8 +1142,10 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, unsigned int num_outs = desc[5 + input_pins]; unsigned int i, len; struct snd_kcontrol *kctl; + const struct usbmix_name_map *map; - if (check_ignored_ctl(state, unitid, 0)) + map = find_map(state, unitid, 0); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1152,7 +1174,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, } kctl->private_free = usb_mixer_elem_free; - len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (! len) len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0); if (! len) @@ -1342,6 +1364,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned int i, err, nameid, type, len; struct procunit_info *info; struct procunit_value_info *valinfo; + const struct usbmix_name_map *map; static struct procunit_value_info default_value_info[] = { { 0x01, "Switch", USB_MIXER_BOOLEAN }, { 0 } @@ -1371,7 +1394,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned /* FIXME: bitmap might be longer than 8bit */ if (! (dsc[12 + num_ins] & (1 << (valinfo->control - 1)))) continue; - if (check_ignored_ctl(state, unitid, valinfo->control)) + map = find_map(state, unitid, valinfo->control); + if (check_ignored_ctl(map)) continue; cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (! cval) { @@ -1402,8 +1426,9 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned } kctl->private_free = usb_mixer_elem_free; - if (check_mapped_name(state, unitid, cval->control, kctl->id.name, sizeof(kctl->id.name))) - ; + if (check_mapped_name(map, kctl->id.name, + sizeof(kctl->id.name))) + /* nothing */ ; else if (info->name) strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); else { @@ -1542,6 +1567,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi int err; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; + const struct usbmix_name_map *map; char **namelist; if (! num_ins || desc[0] < 5 + num_ins) { @@ -1557,7 +1583,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi if (num_ins == 1) /* only one ? nonsense! */ return 0; - if (check_ignored_ctl(state, unitid, 0)) + map = find_map(state, unitid, 0); + if (check_ignored_ctl(map)) return 0; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1612,7 +1639,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi kctl->private_free = usb_mixer_selector_elem_free; nameid = desc[desc[0] - 1]; - len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (len) ; else if (nameid) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 77c35885e21c..79e903a60862 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -19,11 +19,16 @@ * */ +struct usbmix_dB_map { + u32 min; + u32 max; +}; struct usbmix_name_map { int id; const char *name; int control; + struct usbmix_dB_map *dB; }; struct usbmix_selector_map { @@ -72,7 +77,7 @@ static struct usbmix_name_map extigy_map[] = { { 8, "Line Playback" }, /* FU */ /* 9: IT mic */ { 10, "Mic Playback" }, /* FU */ - { 11, "Capture Input Source" }, /* SU */ + { 11, "Capture Source" }, /* SU */ { 12, "Capture" }, /* FU */ /* 13: OT pcm capture */ /* 14: MU (w/o controls) */ @@ -102,6 +107,9 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov */ +static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ + static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ /* 2: IT mic */ @@ -110,16 +118,19 @@ static struct usbmix_name_map mp3plus_map[] = { /* 5: OT digital out */ /* 6: OT speaker */ /* 7: OT pcm capture */ - { 8, "Capture Input Source" }, /* FU, default PCM Capture Source */ + { 8, "Capture Source" }, /* FU, default PCM Capture Source */ /* (Mic, Input 1 = Line input, Input 2 = Optical input) */ { 9, "Master Playback" }, /* FU, default Speaker 1 */ /* { 10, "Mic Capture", 1 }, */ /* FU, Mic Capture */ - /* { 10, "Mic Capture", 2 }, */ /* FU, Mic Capture */ + { 10, /* "Mic Capture", */ NULL, 2, .dB = &mp3plus_dB_2 }, + /* FU, Mic Capture */ { 10, "Mic Boost", 7 }, /* FU, default Auto Gain Input */ - { 11, "Line Capture" }, /* FU, default PCM Capture */ + { 11, "Line Capture", .dB = &mp3plus_dB_2 }, + /* FU, default PCM Capture */ { 12, "Digital In Playback" }, /* FU, default PCM 1 */ - /* { 13, "Mic Playback" }, */ /* FU, default Mic Playback */ - { 14, "Line Playback" }, /* FU, default Speaker */ + { 13, /* "Mic Playback", */ .dB = &mp3plus_dB_1 }, + /* FU, default Mic Playback */ + { 14, "Line Playback", .dB = &mp3plus_dB_1 }, /* FU, default Speaker */ /* 15: MU */ { 0 } /* terminator */ }; -- cgit v1.2.2 From 867af973a3b38f2a564d612326efd2694d931f30 Mon Sep 17 00:00:00 2001 From: Thomas Weber Date: Thu, 11 Feb 2010 16:13:59 +0100 Subject: Add ASoC support for Devkit8000 This patch expands the omap3beagle sound soc for the beagle board clone DevKit8000. Signed-off-by: Thomas Weber Acked-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 8 +++++--- sound/soc/omap/omap3beagle.c | 6 +++--- 2 files changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa6cd5a..18ebdc7d0a51 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -94,12 +94,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA Say Y if you want to add support for SoC audio on the OMAP3 Pandora. config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" + depends on TWL4030_CORE && SND_OMAP_SOC + depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Beagleboard. + Say Y if you want to add support for SoC audio on the Beagleboard or + the clone Devkit8000. config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index d88ad5ca526c..240e0975dd6a 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -117,11 +117,11 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!machine_is_omap3_beagle()) { - pr_debug("Not OMAP3 Beagle!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { + pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); return -ENODEV; } - pr_info("OMAP3 Beagle SoC init\n"); + pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); if (!omap3beagle_snd_device) { -- cgit v1.2.2 From 6db29675b1cb60e878d04a1f69aba265189b2e33 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 11 Feb 2010 18:11:10 +0100 Subject: ASoC: fix compile breakage if CONFIG_SH_DMA_API=y && CONFIG_SND_SIU_MIGOR!=n Audio on Migo-R cannot work if CONFIG_SH_DMA_API=y, but compilation should not break anyway. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index a86696bbe179..106674979b53 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_SH4_FSI config SND_SOC_SH4_SIU tristate depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK + select DMA_ENGINE select DMADEVICES select SH_DMAE -- cgit v1.2.2 From 3a66d3877eaa4ab9818000a15c07326adaa9ca79 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 11 Feb 2010 13:27:19 +0000 Subject: ASoC: Add WM2000 driver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The WM2000 is a low power, high quality handset receiver speaker driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It provides enhanced voice communication quality in a noisy environment if the handset acoustics are designed appropriately. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm2000.c | 888 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm2000.h | 79 +++++ 4 files changed, 973 insertions(+) create mode 100644 sound/soc/codecs/wm2000.c create mode 100644 sound/soc/codecs/wm2000.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5ab59219a8de..1743d565e996 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C + select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -265,3 +266,6 @@ config SND_SOC_MAX9877 config SND_SOC_TPA6130A2 tristate + +config SND_SOC_WM2000 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 209dd6c7c254..dd5ce6df6292 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -58,6 +58,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-wm2000-objs := wm2000.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o @@ -119,3 +120,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c new file mode 100644 index 000000000000..217b02680597 --- /dev/null +++ b/sound/soc/codecs/wm2000.c @@ -0,0 +1,888 @@ +/* + * wm2000.c -- WM2000 ALSA Soc Audio driver + * + * Copyright 2008-2010 Wolfson Microelectronics PLC. + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * The download image for the WM2000 will be requested as + * 'wm2000_anc.bin' by default (overridable via platform data) at + * runtime and is expected to be in flat binary format. This is + * generated by Wolfson configuration tools and includes + * system-specific callibration information. If supplied as a + * sequence of ASCII-encoded hexidecimal bytes this can be converted + * into a flat binary with a command such as this on the command line: + * + * perl -e 'while (<>) { s/[\r\n]+// ; printf("%c", hex($_)); }' + * < file > wm2000_anc.bin + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include + +#include "wm2000.h" + +enum wm2000_anc_mode { + ANC_ACTIVE = 0, + ANC_BYPASS = 1, + ANC_STANDBY = 2, + ANC_OFF = 3, +}; + +struct wm2000_priv { + struct i2c_client *i2c; + + enum wm2000_anc_mode anc_mode; + + unsigned int anc_active:1; + unsigned int anc_eng_ena:1; + unsigned int spk_ena:1; + + unsigned int mclk_div:1; + unsigned int speech_clarity:1; + + int anc_download_size; + char *anc_download; +}; + +static struct i2c_client *wm2000_i2c; + +static int wm2000_write(struct i2c_client *i2c, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + int ret; + + data[0] = (reg >> 8) & 0xff; + data[1] = reg & 0xff; + data[2] = value & 0xff; + + dev_vdbg(&i2c->dev, "write %x = %x\n", reg, value); + + ret = i2c_master_send(i2c, data, 3); + if (ret == 3) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r) +{ + struct i2c_msg xfer[2]; + u8 reg[2]; + u8 data; + int ret; + + /* Write register */ + reg[0] = (r >> 8) & 0xff; + reg[1] = r & 0xff; + xfer[0].addr = i2c->addr; + xfer[0].flags = 0; + xfer[0].len = sizeof(reg); + xfer[0].buf = ®[0]; + + /* Read data */ + xfer[1].addr = i2c->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 1; + xfer[1].buf = &data; + + ret = i2c_transfer(i2c->adapter, xfer, 2); + if (ret != 2) { + dev_err(&i2c->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + dev_vdbg(&i2c->dev, "read %x from %x\n", data, r); + + return data; +} + +static void wm2000_reset(struct wm2000_priv *wm2000) +{ + struct i2c_client *i2c = wm2000->i2c; + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_ENG_CLR); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_CLR); + wm2000_write(i2c, WM2000_REG_ID1, 0); + + wm2000->anc_mode = ANC_OFF; +} + +static int wm2000_poll_bit(struct i2c_client *i2c, + unsigned int reg, u8 mask, int timeout) +{ + int val; + + val = wm2000_read(i2c, reg); + + while (!(val & mask) && --timeout) { + msleep(1); + val = wm2000_read(i2c, reg); + } + + if (timeout == 0) + return 0; + else + return 1; +} + +static int wm2000_power_up(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int ret, timeout; + + BUG_ON(wm2000->anc_mode != ANC_OFF); + + dev_dbg(&i2c->dev, "Beginning power up\n"); + + if (!wm2000->mclk_div) { + dev_dbg(&i2c->dev, "Disabling MCLK divider\n"); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, + WM2000_MCLK_DIV2_ENA_CLR); + } else { + dev_dbg(&i2c->dev, "Enabling MCLK divider\n"); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, + WM2000_MCLK_DIV2_ENA_SET); + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_ENG_CLR); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_ENG_SET); + + /* Wait for ANC engine to become ready */ + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, + WM2000_ANC_ENG_IDLE, 1)) { + dev_err(&i2c->dev, "ANC engine failed to reset\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_BOOT_COMPLETE, 1)) { + dev_err(&i2c->dev, "ANC engine failed to initialise\n"); + return -ETIMEDOUT; + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_SET); + + /* Open code download of the data since it is the only bulk + * write we do. */ + dev_dbg(&i2c->dev, "Downloading %d bytes\n", + wm2000->anc_download_size - 2); + + ret = i2c_master_send(i2c, wm2000->anc_download, + wm2000->anc_download_size); + if (ret < 0) { + dev_err(&i2c->dev, "i2c_transfer() failed: %d\n", ret); + return ret; + } + if (ret != wm2000->anc_download_size) { + dev_err(&i2c->dev, "i2c_transfer() failed, %d != %d\n", + ret, wm2000->anc_download_size); + return -EIO; + } + + dev_dbg(&i2c->dev, "Download complete\n"); + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } else { + timeout = 10; + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } + + ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); + if (wm2000->speech_clarity) + ret &= ~WM2000_SPEECH_CLARITY; + else + ret |= WM2000_SPEECH_CLARITY; + wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); + + wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); + wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02); + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_MOUSE_ACTIVE, timeout)) { + dev_err(&i2c->dev, "Timed out waiting for device after %dms\n", + timeout * 10); + return -ETIMEDOUT; + } + + dev_dbg(&i2c->dev, "ANC active\n"); + if (analogue) + dev_dbg(&i2c->dev, "Analogue active\n"); + wm2000->anc_mode = ANC_ACTIVE; + + return 0; +} + +static int wm2000_power_down(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int timeout; + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_POWER_DOWN); + } else { + timeout = 10; + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_POWER_DOWN); + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_POWER_DOWN_COMPLETE, timeout)) { + dev_err(&i2c->dev, "Timeout waiting for ANC power down\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, + WM2000_ANC_ENG_IDLE, 1)) { + dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); + return -ETIMEDOUT; + } + + dev_dbg(&i2c->dev, "powered off\n"); + wm2000->anc_mode = ANC_OFF; + + return 0; +} + +static int wm2000_enter_bypass(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + BUG_ON(wm2000->anc_mode != ANC_ACTIVE); + + if (analogue) { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_BYPASS_ENTRY); + } else { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_BYPASS_ENTRY); + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_ANC_DISABLED, 10)) { + dev_err(&i2c->dev, "Timeout waiting for ANC disable\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, + WM2000_ANC_ENG_IDLE, 1)) { + dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); + return -ETIMEDOUT; + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, WM2000_SYS_STBY); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_CLR); + + wm2000->anc_mode = ANC_BYPASS; + dev_dbg(&i2c->dev, "bypass enabled\n"); + + return 0; +} + +static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + BUG_ON(wm2000->anc_mode != ANC_BYPASS); + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); + + if (analogue) { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } else { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_SET); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_MOUSE_ACTIVE, 10)) { + dev_err(&i2c->dev, "Timed out waiting for MOUSE\n"); + return -ETIMEDOUT; + } + + wm2000->anc_mode = ANC_ACTIVE; + dev_dbg(&i2c->dev, "MOUSE active\n"); + + return 0; +} + +static int wm2000_enter_standby(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int timeout; + + BUG_ON(wm2000->anc_mode != ANC_ACTIVE); + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_STANDBY_ENTRY); + } else { + timeout = 10; + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_STANDBY_ENTRY); + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_ANC_DISABLED, timeout)) { + dev_err(&i2c->dev, + "Timed out waiting for ANC disable after 1ms\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE, + 1)) { + dev_err(&i2c->dev, + "Timed out waiting for standby after %dms\n", + timeout * 10); + return -ETIMEDOUT; + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, WM2000_SYS_STBY); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_CLR); + + wm2000->anc_mode = ANC_STANDBY; + dev_dbg(&i2c->dev, "standby\n"); + if (analogue) + dev_dbg(&i2c->dev, "Analogue disabled\n"); + + return 0; +} + +static int wm2000_exit_standby(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int timeout; + + BUG_ON(wm2000->anc_mode != ANC_STANDBY); + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_MOUSE_ENABLE); + } else { + timeout = 10; + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_MOUSE_ENABLE); + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_SET); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_MOUSE_ACTIVE, timeout)) { + dev_err(&i2c->dev, "Timed out waiting for MOUSE after %dms\n", + timeout * 10); + return -ETIMEDOUT; + } + + wm2000->anc_mode = ANC_ACTIVE; + dev_dbg(&i2c->dev, "MOUSE active\n"); + if (analogue) + dev_dbg(&i2c->dev, "Analogue enabled\n"); + + return 0; +} + +typedef int (*wm2000_mode_fn)(struct i2c_client *i2c, int analogue); + +static struct { + enum wm2000_anc_mode source; + enum wm2000_anc_mode dest; + int analogue; + wm2000_mode_fn step[2]; +} anc_transitions[] = { + { + .source = ANC_OFF, + .dest = ANC_ACTIVE, + .analogue = 1, + .step = { + wm2000_power_up, + }, + }, + { + .source = ANC_OFF, + .dest = ANC_STANDBY, + .step = { + wm2000_power_up, + wm2000_enter_standby, + }, + }, + { + .source = ANC_OFF, + .dest = ANC_BYPASS, + .analogue = 1, + .step = { + wm2000_power_up, + wm2000_enter_bypass, + }, + }, + { + .source = ANC_ACTIVE, + .dest = ANC_BYPASS, + .analogue = 1, + .step = { + wm2000_enter_bypass, + }, + }, + { + .source = ANC_ACTIVE, + .dest = ANC_STANDBY, + .analogue = 1, + .step = { + wm2000_enter_standby, + }, + }, + { + .source = ANC_ACTIVE, + .dest = ANC_OFF, + .analogue = 1, + .step = { + wm2000_power_down, + }, + }, + { + .source = ANC_BYPASS, + .dest = ANC_ACTIVE, + .analogue = 1, + .step = { + wm2000_exit_bypass, + }, + }, + { + .source = ANC_BYPASS, + .dest = ANC_STANDBY, + .analogue = 1, + .step = { + wm2000_exit_bypass, + wm2000_enter_standby, + }, + }, + { + .source = ANC_BYPASS, + .dest = ANC_OFF, + .step = { + wm2000_exit_bypass, + wm2000_power_down, + }, + }, + { + .source = ANC_STANDBY, + .dest = ANC_ACTIVE, + .analogue = 1, + .step = { + wm2000_exit_standby, + }, + }, + { + .source = ANC_STANDBY, + .dest = ANC_BYPASS, + .analogue = 1, + .step = { + wm2000_exit_standby, + wm2000_enter_bypass, + }, + }, + { + .source = ANC_STANDBY, + .dest = ANC_OFF, + .step = { + wm2000_exit_standby, + wm2000_power_down, + }, + }, +}; + +static int wm2000_anc_transition(struct wm2000_priv *wm2000, + enum wm2000_anc_mode mode) +{ + struct i2c_client *i2c = wm2000->i2c; + int i, j; + int ret; + + if (wm2000->anc_mode == mode) + return 0; + + for (i = 0; i < ARRAY_SIZE(anc_transitions); i++) + if (anc_transitions[i].source == wm2000->anc_mode && + anc_transitions[i].dest == mode) + break; + if (i == ARRAY_SIZE(anc_transitions)) { + dev_err(&i2c->dev, "No transition for %d->%d\n", + wm2000->anc_mode, mode); + return -EINVAL; + } + + for (j = 0; j < ARRAY_SIZE(anc_transitions[j].step); j++) { + if (!anc_transitions[i].step[j]) + break; + ret = anc_transitions[i].step[j](i2c, + anc_transitions[i].analogue); + if (ret != 0) + return ret; + } + + return 0; +} + +static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) +{ + struct i2c_client *i2c = wm2000->i2c; + enum wm2000_anc_mode mode; + + if (wm2000->anc_eng_ena && wm2000->spk_ena) + if (wm2000->anc_active) + mode = ANC_ACTIVE; + else + mode = ANC_BYPASS; + else + mode = ANC_STANDBY; + + dev_dbg(&i2c->dev, "Set mode %d (enabled %d, mute %d, active %d)\n", + mode, wm2000->anc_eng_ena, !wm2000->spk_ena, + wm2000->anc_active); + + return wm2000_anc_transition(wm2000, mode); +} + +static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + + ucontrol->value.enumerated.item[0] = wm2000->anc_active; + + return 0; +} + +static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + int anc_active = ucontrol->value.enumerated.item[0]; + + if (anc_active > 1) + return -EINVAL; + + wm2000->anc_active = anc_active; + + return wm2000_anc_set_mode(wm2000); +} + +static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + + ucontrol->value.enumerated.item[0] = wm2000->spk_ena; + + return 0; +} + +static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + int val = ucontrol->value.enumerated.item[0]; + + if (val > 1) + return -EINVAL; + + wm2000->spk_ena = val; + + return wm2000_anc_set_mode(wm2000); +} + +static const struct snd_kcontrol_new wm2000_controls[] = { + SOC_SINGLE_BOOL_EXT("WM2000 ANC Switch", 0, + wm2000_anc_mode_get, + wm2000_anc_mode_put), + SOC_SINGLE_BOOL_EXT("WM2000 Switch", 0, + wm2000_speaker_get, + wm2000_speaker_put), +}; + +static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + + if (SND_SOC_DAPM_EVENT_ON(event)) + wm2000->anc_eng_ena = 1; + + if (SND_SOC_DAPM_EVENT_OFF(event)) + wm2000->anc_eng_ena = 0; + + return wm2000_anc_set_mode(wm2000); +} + +static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { +/* Externally visible pins */ +SND_SOC_DAPM_OUTPUT("WM2000 SPKN"), +SND_SOC_DAPM_OUTPUT("WM2000 SPKP"), + +SND_SOC_DAPM_INPUT("WM2000 LINN"), +SND_SOC_DAPM_INPUT("WM2000 LINP"), + +SND_SOC_DAPM_PGA_E("ANC Engine", SND_SOC_NOPM, 0, 0, NULL, 0, + wm2000_anc_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +}; + +/* Target, Path, Source */ +static const struct snd_soc_dapm_route audio_map[] = { + { "WM2000 SPKN", NULL, "ANC Engine" }, + { "WM2000 SPKP", NULL, "ANC Engine" }, + { "ANC Engine", NULL, "WM2000 LINN" }, + { "ANC Engine", NULL, "WM2000 LINP" }, +}; + +/* Called from the machine driver */ +int wm2000_add_controls(struct snd_soc_codec *codec) +{ + int ret; + + if (!wm2000_i2c) { + pr_err("WM2000 not yet probed\n"); + return -ENODEV; + } + + ret = snd_soc_dapm_new_controls(codec, wm2000_dapm_widgets, + ARRAY_SIZE(wm2000_dapm_widgets)); + if (ret < 0) + return ret; + + ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + if (ret < 0) + return ret; + + return snd_soc_add_controls(codec, wm2000_controls, + ARRAY_SIZE(wm2000_controls)); +} +EXPORT_SYMBOL_GPL(wm2000_add_controls); + +static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct wm2000_priv *wm2000; + struct wm2000_platform_data *pdata; + const char *filename; + const struct firmware *fw; + int reg, ret; + u16 id; + + if (wm2000_i2c) { + dev_err(&i2c->dev, "Another WM2000 is already registered\n"); + return -EINVAL; + } + + wm2000 = kzalloc(sizeof(struct wm2000_priv), GFP_KERNEL); + if (wm2000 == NULL) { + dev_err(&i2c->dev, "Unable to allocate private data\n"); + return -ENOMEM; + } + + /* Verify that this is a WM2000 */ + reg = wm2000_read(i2c, WM2000_REG_ID1); + id = reg << 8; + reg = wm2000_read(i2c, WM2000_REG_ID2); + id |= reg & 0xff; + + if (id != 0x2000) { + dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id); + ret = -ENODEV; + goto err; + } + + reg = wm2000_read(i2c, WM2000_REG_REVISON); + dev_info(&i2c->dev, "revision %c\n", reg + 'A'); + + filename = "wm2000_anc.bin"; + pdata = dev_get_platdata(&i2c->dev); + if (pdata) { + wm2000->mclk_div = pdata->mclkdiv2; + wm2000->speech_clarity = !pdata->speech_enh_disable; + + if (pdata->download_file) + filename = pdata->download_file; + } + + ret = request_firmware(&fw, filename, &i2c->dev); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret); + goto err; + } + + /* Pre-cook the concatenation of the register address onto the image */ + wm2000->anc_download_size = fw->size + 2; + wm2000->anc_download = kmalloc(wm2000->anc_download_size, GFP_KERNEL); + if (wm2000->anc_download == NULL) { + dev_err(&i2c->dev, "Out of memory\n"); + ret = -ENOMEM; + goto err_fw; + } + + wm2000->anc_download[0] = 0x80; + wm2000->anc_download[1] = 0x00; + memcpy(wm2000->anc_download + 2, fw->data, fw->size); + + release_firmware(fw); + + dev_set_drvdata(&i2c->dev, wm2000); + wm2000->anc_eng_ena = 1; + wm2000->i2c = i2c; + + wm2000_reset(wm2000); + + /* This will trigger a transition to standby mode by default */ + wm2000_anc_set_mode(wm2000); + + wm2000_i2c = i2c; + + return 0; + +err_fw: + release_firmware(fw); +err: + kfree(wm2000); + return ret; +} + +static __devexit int wm2000_i2c_remove(struct i2c_client *i2c) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + wm2000_anc_transition(wm2000, ANC_OFF); + + wm2000_i2c = NULL; + kfree(wm2000->anc_download); + kfree(wm2000); + + return 0; +} + +static void wm2000_i2c_shutdown(struct i2c_client *i2c) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + wm2000_anc_transition(wm2000, ANC_OFF); +} + +#ifdef CONFIG_PM +static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + return wm2000_anc_transition(wm2000, ANC_OFF); +} + +static int wm2000_i2c_resume(struct i2c_client *i2c) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + return wm2000_anc_set_mode(wm2000); +} +#else +#define wm2000_i2c_suspend NULL +#define wm2000_i2c_resume NULL +#endif + +static const struct i2c_device_id wm2000_i2c_id[] = { + { "wm2000", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm2000_i2c_id); + +static struct i2c_driver wm2000_i2c_driver = { + .driver = { + .name = "wm2000", + .owner = THIS_MODULE, + }, + .probe = wm2000_i2c_probe, + .remove = __devexit_p(wm2000_i2c_remove), + .suspend = wm2000_i2c_suspend, + .resume = wm2000_i2c_resume, + .shutdown = wm2000_i2c_shutdown, + .id_table = wm2000_i2c_id, +}; + +static int __init wm2000_init(void) +{ + return i2c_add_driver(&wm2000_i2c_driver); +} +module_init(wm2000_init); + +static void __exit wm2000_exit(void) +{ + i2c_del_driver(&wm2000_i2c_driver); +} +module_exit(wm2000_exit); + +MODULE_DESCRIPTION("ASoC WM2000 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h new file mode 100644 index 000000000000..c18e261c3c7f --- /dev/null +++ b/sound/soc/codecs/wm2000.h @@ -0,0 +1,79 @@ +/* + * wm2000.h -- WM2000 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM2000_H +#define _WM2000_H + +struct wm2000_setup_data { + unsigned short i2c_address; + int mclk_div; /* Set to a non-zero value if MCLK_DIV_2 required */ +}; + +extern int wm2000_add_controls(struct snd_soc_codec *codec); + +extern struct snd_soc_dai wm2000_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm2000; + +#define WM2000_REG_SYS_START 0x8000 +#define WM2000_REG_SPEECH_CLARITY 0x8fef +#define WM2000_REG_SYS_WATCHDOG 0x8ff6 +#define WM2000_REG_ANA_VMID_PD_TIME 0x8ff7 +#define WM2000_REG_ANA_VMID_PU_TIME 0x8ff8 +#define WM2000_REG_CAT_FLTR_INDX 0x8ff9 +#define WM2000_REG_CAT_GAIN_0 0x8ffa +#define WM2000_REG_SYS_STATUS 0x8ffc +#define WM2000_REG_SYS_MODE_CNTRL 0x8ffd +#define WM2000_REG_SYS_START0 0x8ffe +#define WM2000_REG_SYS_START1 0x8fff +#define WM2000_REG_ID1 0xf000 +#define WM2000_REG_ID2 0xf001 +#define WM2000_REG_REVISON 0xf002 +#define WM2000_REG_SYS_CTL1 0xf003 +#define WM2000_REG_SYS_CTL2 0xf004 +#define WM2000_REG_ANC_STAT 0xf005 +#define WM2000_REG_IF_CTL 0xf006 + +/* SPEECH_CLARITY */ +#define WM2000_SPEECH_CLARITY 0x01 + +/* SYS_STATUS */ +#define WM2000_STATUS_MOUSE_ACTIVE 0x40 +#define WM2000_STATUS_CAT_FREQ_COMPLETE 0x20 +#define WM2000_STATUS_CAT_GAIN_COMPLETE 0x10 +#define WM2000_STATUS_THERMAL_SHUTDOWN_COMPLETE 0x08 +#define WM2000_STATUS_ANC_DISABLED 0x04 +#define WM2000_STATUS_POWER_DOWN_COMPLETE 0x02 +#define WM2000_STATUS_BOOT_COMPLETE 0x01 + +/* SYS_MODE_CNTRL */ +#define WM2000_MODE_ANA_SEQ_INCLUDE 0x80 +#define WM2000_MODE_MOUSE_ENABLE 0x40 +#define WM2000_MODE_CAT_FREQ_ENABLE 0x20 +#define WM2000_MODE_CAT_GAIN_ENABLE 0x10 +#define WM2000_MODE_BYPASS_ENTRY 0x08 +#define WM2000_MODE_STANDBY_ENTRY 0x04 +#define WM2000_MODE_THERMAL_ENABLE 0x02 +#define WM2000_MODE_POWER_DOWN 0x01 + +/* SYS_CTL1 */ +#define WM2000_SYS_STBY 0x01 + +/* SYS_CTL2 */ +#define WM2000_MCLK_DIV2_ENA_CLR 0x80 +#define WM2000_MCLK_DIV2_ENA_SET 0x40 +#define WM2000_ANC_ENG_CLR 0x20 +#define WM2000_ANC_ENG_SET 0x10 +#define WM2000_ANC_INT_N_CLR 0x08 +#define WM2000_ANC_INT_N_SET 0x04 +#define WM2000_RAM_CLR 0x02 +#define WM2000_RAM_SET 0x01 + +/* ANC_STAT */ +#define WM2000_ANC_ENG_IDLE 0x01 + +#endif -- cgit v1.2.2 From d6d8bf549393484e906913f02fa3c9518a2819b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Feb 2010 18:17:06 +0100 Subject: ALSA: hda - use WARN_ON_ONCE() for zero-division detection Replace the zero-division warning message with WARN_ON_ONCE() per the advice by Linus. This shouldn't happen, but if it happens, it's possible that the bug happens often due to buggy IRQs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e767c3f395ab..3600e9cc9bc6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1893,12 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (!bdl_pos_adj[chip->dev_index]) return 1; /* no delayed ack */ - if (azx_dev->period_bytes == 0) { - printk(KERN_WARNING - "hda-intel: Divide by zero was avoided " - "in azx_dev->period_bytes.\n"); - return 0; - } + if (WARN_ONCE(!azx_dev->period_bytes, + "hda-intel: zero azx_dev->period_bytes")) + return 0; /* this shouldn't happen! */ if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) return 0; /* NG - it's below the period boundary */ return 1; /* OK, it's fine */ -- cgit v1.2.2 From cfd3d8dcf7b4fc783db0806ac3936a7b44735bf7 Mon Sep 17 00:00:00 2001 From: Greg Alexander Date: Sat, 13 Feb 2010 02:02:25 -0500 Subject: ALSA: hda - Add support for Lenovo IdeaPad U150 Add patch for the Conexant 5066 HDA codec to support the Lenovo IdeaPad U150 Signed-off-by: Greg Alexander Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 130 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 126 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 745e35992144..194a28c54992 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -113,7 +113,8 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; - unsigned int dell_vostro; + unsigned int dell_vostro:1; + unsigned int ideapad:1; unsigned int ext_mic_present; unsigned int recording; @@ -2167,6 +2168,34 @@ static void cxt5066_vostro_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_ideapad_automic(struct hda_codec *codec) +{ + unsigned int present; + + struct hda_verb ext_mic_present[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + + present = snd_hda_jack_detect(codec, 0x1b); + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + /* mute internal speaker if HP is plugged */ static void cxt5066_hp_automute(struct hda_codec *codec) { @@ -2216,6 +2245,20 @@ static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_ideapad_automic(codec); + break; + } +} + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -2227,13 +2270,21 @@ static const struct hda_input_mux cxt5066_analog_mic_boost = { }, }; -static int cxt5066_set_mic_boost(struct hda_codec *codec) +static void cxt5066_set_mic_boost(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - return snd_hda_codec_write_cache(codec, 0x17, 0, + snd_hda_codec_write_cache(codec, 0x17, 0, AC_VERB_SET_AMP_GAIN_MUTE, AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | cxt5066_analog_mic_boost.items[spec->mic_boost].index); + if (spec->ideapad) { + /* adjust the internal mic as well...it is not through 0x17 */ + snd_hda_codec_write_cache(codec, 0x23, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_INPUT | + cxt5066_analog_mic_boost. + items[spec->mic_boost].index); + } } static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, @@ -2664,6 +2715,56 @@ static struct hda_verb cxt5066_init_verbs_vostro[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_ideapad[] = { + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ + + /* Speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* HP, Amp */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ + + /* Audio input selector */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ + + /* SPDIF route: PCM */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* internal microphone */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable int mic */ + + /* EAPD */ + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + static struct hda_verb cxt5066_init_verbs_portd_lo[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { } /* end */ @@ -2680,6 +2781,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_hp_automute(codec); if (spec->dell_vostro) cxt5066_vostro_automic(codec); + else if (spec->ideapad) + cxt5066_ideapad_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -2705,6 +2808,7 @@ enum { CXT5066_DELL_LAPTOP, /* Dell Laptop */ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ + CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_MODELS }; @@ -2712,7 +2816,8 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", - [CXT5066_DELL_VOSTO] = "dell-vostro" + [CXT5066_DELL_VOSTO] = "dell-vostro", + [CXT5066_IDEAPAD] = "ideapad", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -2722,6 +2827,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} }; @@ -2810,6 +2916,22 @@ static int patch_cxt5066(struct hda_codec *codec) /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + break; + case CXT5066_IDEAPAD: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_ideapad_event; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->init_verbs[0] = cxt5066_init_verbs_ideapad; + spec->port_d_mode = 0; + spec->ideapad = 1; + spec->mic_boost = 2; /* default 20dB gain */ + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ spec->input_mux = NULL; break; -- cgit v1.2.2 From 19b50063780953563e3c3a2867c39aad7b9e64cf Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:34 +0100 Subject: ALSA: Echoaudio - Add firmware cache #1 Changes the way the firmware is passed through functions. When CONFIG_PM is enabled the firmware cannot be released because the driver will need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then used to locate the firmware in the firmware cache. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/darla20_dsp.c | 2 +- sound/pci/echoaudio/darla24_dsp.c | 2 +- sound/pci/echoaudio/echo3g_dsp.c | 2 +- sound/pci/echoaudio/echoaudio.c | 8 +++++++- sound/pci/echoaudio/echoaudio.h | 6 +++--- sound/pci/echoaudio/echoaudio_3g.c | 5 ++--- sound/pci/echoaudio/echoaudio_dsp.c | 12 +++++++----- sound/pci/echoaudio/gina20_dsp.c | 2 +- sound/pci/echoaudio/gina24_dsp.c | 18 ++++++++--------- sound/pci/echoaudio/indigo_dsp.c | 2 +- sound/pci/echoaudio/indigodj_dsp.c | 2 +- sound/pci/echoaudio/indigodjx_dsp.c | 2 +- sound/pci/echoaudio/indigoio_dsp.c | 2 +- sound/pci/echoaudio/indigoiox_dsp.c | 2 +- sound/pci/echoaudio/layla20_dsp.c | 7 +++---- sound/pci/echoaudio/layla24_dsp.c | 19 +++++++++--------- sound/pci/echoaudio/mia_dsp.c | 2 +- sound/pci/echoaudio/mona_dsp.c | 39 ++++++++++++++++++------------------- 18 files changed, 69 insertions(+), 65 deletions(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index 29043301ebb8..a44135d6acbb 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP]; + chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index 60228731841f..d681da180829 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP]; + chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index 57967e580571..f0071935c0cb 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -61,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP]; + chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get what type of external box is attached */ diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1305f7ca02c3..78fc2637bfa6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -36,11 +36,15 @@ MODULE_PARM_DESC(enable, "Enable " ECHOCARD_NAME " soundcard."); static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999}; static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1); + + static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip) + struct echoaudio *chip, const short fw_index) { int err; char name[30]; + const struct firmware *frm = &card_fw[fw_index]; + DE_ACT(("firmware requested: %s\n", frm->data)); snprintf(name, sizeof(name), "ea/%s", frm->data); if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) @@ -48,6 +52,8 @@ static int get_firmware(const struct firmware **fw_entry, return err; } + + static void free_firmware(const struct firmware *fw_entry) { release_firmware(fw_entry); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index f9490ae36c2e..be76ef3b829a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -442,8 +442,8 @@ struct echoaudio { u16 device_id, subdevice_id; u16 *dsp_code; /* Current DSP code loaded, * NULL if nothing loaded */ - const struct firmware *dsp_code_to_load;/* DSP code to load */ - const struct firmware *asic_code; /* Current ASIC code */ + short dsp_code_to_load; /* DSP code to load */ + short asic_code; /* Current ASIC code */ u32 comm_page_phys; /* Physical address of the * memory seen by DSP */ volatile u32 __iomem *dsp_registers; /* DSP's register base */ @@ -464,7 +464,7 @@ static int load_firmware(struct echoaudio *chip); static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip); + struct echoaudio *chip, const short fw_index); static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index e32a74897921..658db44ef746 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -227,12 +227,11 @@ static int load_asic(struct echoaudio *chip) /* Give the DSP a few milliseconds to settle down */ mdelay(2); - err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, - &card_fw[FW_3G_ASIC]); + err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, FW_3G_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_3G_ASIC]; + chip->asic_code = FW_3G_ASIC; /* Now give the new ASIC some time to set up */ msleep(1000); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 4df51ef5e095..031ef7e9da91 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -175,15 +175,15 @@ static inline int check_asic_status(struct echoaudio *chip) #ifdef ECHOCARD_HAS_ASIC /* Load ASIC code - done after the DSP is loaded */ -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic) +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) { const struct firmware *fw; int err; u32 i, size; u8 *code; - if ((err = get_firmware(&fw, asic, chip)) < 0) { + err = get_firmware(&fw, chip, asic); + if (err < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return err; } @@ -245,7 +245,8 @@ static int install_resident_loader(struct echoaudio *chip) return 0; } - if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) { + i = get_firmware(&fw, chip, FW_361_LOADER); + if (i < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return i; } @@ -485,7 +486,8 @@ static int load_firmware(struct echoaudio *chip) chip->dsp_code = NULL; } - if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0) + err = get_firmware(&fw, chip, chip->dsp_code_to_load); + if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); free_firmware(fw); diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index 3f1e7475faea..c5de88b6792d 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -49,7 +49,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP]; + chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 2fef37a2a5b9..093dd7ba0e81 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -33,8 +33,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,13 +63,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { - chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP]; + chip->dsp_code_to_load = FW_GINA24_361_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; } else { - chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP]; + chip->dsp_code_to_load = FW_GINA24_301_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | @@ -125,7 +124,7 @@ static int load_asic(struct echoaudio *chip) { u32 control_reg; int err; - const struct firmware *fw; + short asic; if (chip->asic_loaded) return 1; @@ -135,14 +134,15 @@ static int load_asic(struct echoaudio *chip) /* Pick the correct ASIC for '301 or '361 Gina24 */ if (chip->device_id == DEVICE_ID_56361) - fw = &card_fw[FW_GINA24_361_ASIC]; + asic = FW_GINA24_361_ASIC; else - fw = &card_fw[FW_GINA24_301_ASIC]; + asic = FW_GINA24_301_ASIC; - if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0) + err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, asic); + if (err < 0) return err; - chip->asic_code = fw; + chip->asic_code = asic; /* Now give the new ASIC a little time to set up */ mdelay(10); diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 0b2cd9c86277..8799d2e6536a 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 08392916691e..cb1c92ca9fef 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index f591fc2ed960..91dbfeb586a7 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 0604c8a85223..134e783d3486 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index f357521c79e6..766cf501799d 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 83750e9fd7b4..07f32454757e 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -31,8 +31,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data); static int set_professional_spdif(struct echoaudio *chip, char prof); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); static int update_flags(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP]; + chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER; @@ -144,7 +143,7 @@ static int load_asic(struct echoaudio *chip) return 0; err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC, - &card_fw[FW_LAYLA20_ASIC]); + FW_LAYLA20_ASIC); if (err < 0) return err; diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index d61b5cbcccad..12dc00adca9f 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -32,8 +32,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP]; + chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -123,18 +122,18 @@ static int load_asic(struct echoaudio *chip) /* Load the ASIC for the PCI card */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC, - &card_fw[FW_LAYLA24_1_ASIC]); + FW_LAYLA24_1_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC]; + chip->asic_code = FW_LAYLA24_2S_ASIC; /* Now give the new ASIC a little time to set up */ mdelay(10); /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, - &card_fw[FW_LAYLA24_2S_ASIC]); + FW_LAYLA24_2S_ASIC); if (err < 0) return FALSE; @@ -299,7 +298,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Depending on what digital mode you want, Layla24 needs different ASICs loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ -static int switch_asic(struct echoaudio *chip, const struct firmware *asic) +static int switch_asic(struct echoaudio *chip, short asic) { s8 *monitors; @@ -335,7 +334,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) { u32 control_reg; int err, incompatible_clock; - const struct firmware *asic; + short asic; /* Set clock to "internal" if it's not compatible with the new mode */ incompatible_clock = FALSE; @@ -344,12 +343,12 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2S_ASIC]; + asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2A_ASIC]; + asic = FW_LAYLA24_2A_ASIC; break; default: DE_ACT(("Digital mode not supported: %d\n", mode)); diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 551405114cbc..d0302f2f00db 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -53,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_MIA_DSP]; + chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index eaa619bd2a03..b28b8e4703cf 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -33,8 +33,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,9 +63,9 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) /* Mona comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) - chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP]; + chip->dsp_code_to_load = FW_MONA_361_DSP; else - chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP]; + chip->dsp_code_to_load = FW_MONA_301_DSP; chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; chip->professional_spdif = FALSE; @@ -120,7 +119,7 @@ static int load_asic(struct echoaudio *chip) { u32 control_reg; int err; - const struct firmware *asic; + short asic; if (chip->asic_loaded) return 0; @@ -128,9 +127,9 @@ static int load_asic(struct echoaudio *chip) mdelay(10); if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic); if (err < 0) @@ -141,7 +140,7 @@ static int load_asic(struct echoaudio *chip) /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC, - &card_fw[FW_MONA_2_ASIC]); + FW_MONA_2_ASIC); if (err < 0) return err; @@ -165,22 +164,22 @@ loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ static int switch_asic(struct echoaudio *chip, char double_speed) { - const struct firmware *asic; int err; + short asic; /* Check the clock detect bits to see if this is a single-speed clock or a double-speed clock; load a new ASIC if necessary. */ if (chip->device_id == DEVICE_ID_56361) { if (double_speed) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; } else { if (double_speed) - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } if (asic != chip->asic_code) { @@ -200,7 +199,7 @@ static int switch_asic(struct echoaudio *chip, char double_speed) static int set_sample_rate(struct echoaudio *chip, u32 rate) { u32 control_reg, clock; - const struct firmware *asic; + short asic; char force_write; /* Only set the clock for internal mode. */ @@ -218,14 +217,14 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EINVAL; if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; } else { if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } force_write = 0; @@ -410,8 +409,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) case DIGITAL_MODE_ADAT: /* If the current ASIC is the 96KHz ASIC, switch the ASIC and set to 48 KHz */ - if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] || - chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) { + if (chip->asic_code == FW_MONA_361_1_ASIC96 || + chip->asic_code == FW_MONA_301_1_ASIC96) { set_sample_rate(chip, 48000); } control_reg |= GML_ADAT_MODE; -- cgit v1.2.2 From 4f8ada444cc7a7ea70cdc81f098b34c5f1f2df41 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:51 +0100 Subject: ALSA: Echoaudio - Add firmware cache #2 This patch implements a simple cache for the firmware files when CONFIG_PM is defined. This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before, that is free_firmware() releases the firmware immediately and free_firmware_cache() does nothing. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 42 +++++++++++++++++++++++++++++++++++++---- sound/pci/echoaudio/echoaudio.h | 3 +++ 2 files changed, 41 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 78fc2637bfa6..79dde9592847 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -43,12 +43,24 @@ static int get_firmware(const struct firmware **fw_entry, { int err; char name[30]; - const struct firmware *frm = &card_fw[fw_index]; - DE_ACT(("firmware requested: %s\n", frm->data)); - snprintf(name, sizeof(name), "ea/%s", frm->data); - if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) +#ifdef CONFIG_PM + if (chip->fw_cache[fw_index]) { + DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); + *fw_entry = chip->fw_cache[fw_index]; + return 0; + } +#endif + + DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data)); + snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); + err = request_firmware(fw_entry, name, pci_device(chip)); + if (err < 0) snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err); +#ifdef CONFIG_PM + else + chip->fw_cache[fw_index] = *fw_entry; +#endif return err; } @@ -56,8 +68,29 @@ static int get_firmware(const struct firmware **fw_entry, static void free_firmware(const struct firmware *fw_entry) { +#ifdef CONFIG_PM + DE_ACT(("firmware not released (kept in cache)\n")); +#else release_firmware(fw_entry); DE_ACT(("firmware released\n")); +#endif +} + + + +static void free_firmware_cache(struct echoaudio *chip) +{ +#ifdef CONFIG_PM + int i; + + for (i = 0; i < 8 ; i++) + if (chip->fw_cache[i]) { + release_firmware(chip->fw_cache[i]); + DE_ACT(("release_firmware(%d)\n", i)); + } + + DE_ACT(("firmware_cache released\n")); +#endif } @@ -1880,6 +1913,7 @@ static int snd_echo_free(struct echoaudio *chip) pci_disable_device(chip->pci); /* release chip data */ + free_firmware_cache(chip); kfree(chip); DE_INIT(("Chip freed.\n")); return 0; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index be76ef3b829a..a84c0d15cc50 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -449,6 +449,9 @@ struct echoaudio { volatile u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ +#ifdef CONFIG_PM + const struct firmware *fw_cache[8]; /* Cached firmwares */ +#endif #ifdef ECHOCARD_HAS_MIDI u16 mtc_state; /* State for MIDI input parsing state machine */ -- cgit v1.2.2 From ad3499f4668f684ef6e5d0222ae14d5e4ade1fdd Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:59 +0100 Subject: ALSA: Echoaudio - Add suspend support #1 Move the controls init code outside the init_hw() function because is must not be called during resume. This patch moves the code that initializes the card's controls with default valued from the init_hw() function into a separated set_mixer_defaults() function (one for each of the 16 supported cards). This change is necessary because during resume we must resurrect the hardware without losing the previous settings. set_mixer_defaults() must be called only once when the module is loaded. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/darla20_dsp.c | 10 +++++++--- sound/pci/echoaudio/darla24_dsp.c | 10 +++++++--- sound/pci/echoaudio/echo3g_dsp.c | 26 ++++++++++++-------------- sound/pci/echoaudio/gina20_dsp.c | 13 ++++++++----- sound/pci/echoaudio/gina24_dsp.c | 20 ++++++++++---------- sound/pci/echoaudio/indigo_dsp.c | 10 +++++++--- sound/pci/echoaudio/indigo_express_dsp.c | 1 + sound/pci/echoaudio/indigodj_dsp.c | 10 +++++++--- sound/pci/echoaudio/indigodjx_dsp.c | 11 +++++++---- sound/pci/echoaudio/indigoio_dsp.c | 10 +++++++--- sound/pci/echoaudio/indigoiox_dsp.c | 11 +++++++---- sound/pci/echoaudio/layla20_dsp.c | 13 ++++++++----- sound/pci/echoaudio/layla24_dsp.c | 18 ++++++++++-------- sound/pci/echoaudio/mia_dsp.c | 10 +++++++--- sound/pci/echoaudio/mona_dsp.c | 22 ++++++++++------------ 15 files changed, 115 insertions(+), 80 deletions(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index a44135d6acbb..20c7cbc89bb3 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -57,15 +57,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + /* The Darla20 has no external clock sources */ static u32 detect_input_clocks(const struct echoaudio *chip) { diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index d681da180829..6da6663e9176 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -56,15 +56,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index f0071935c0cb..3cdc2ee2d1dd 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -97,20 +97,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_phantom_power(chip, 0); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); DE_INIT(("init_hw done\n")); return err; @@ -118,6 +104,18 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->non_audio_spdif = FALSE; + chip->bad_board = FALSE; + chip->phantom_power = FALSE; + return init_line_levels(chip); +} + + + static int set_phantom_power(struct echoaudio *chip, char on) { u32 control_reg = le32_to_cpu(chip->comm_page->control_register); diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index c5de88b6792d..d1615a0579d1 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -62,17 +62,20 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 093dd7ba0e81..98f7cfa81b5f 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -57,9 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | ECHO_CLOCK_BIT_ADAT; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { @@ -81,19 +78,22 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 8799d2e6536a..5e85f14fe5a8 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c index 9ab625e15652..2e4ab3e34a74 100644 --- a/sound/pci/echoaudio/indigo_express_dsp.c +++ b/sound/pci/echoaudio/indigo_express_dsp.c @@ -61,6 +61,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) control_reg |= clock; if (control_reg != old_control_reg) { + DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); chip->comm_page->control_register = cpu_to_le32(control_reg); chip->sample_rate = rate; clear_handshake(chip); diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index cb1c92ca9fef..68f3c8ccc1bf 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 91dbfeb586a7..bb9632c752a9 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 134e783d3486..beb9a5b69892 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 766cf501799d..394c6e76bcbc 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 07f32454757e..53ce94605044 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -64,17 +64,20 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index 12dc00adca9f..8c041647f285 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -61,9 +61,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; if ((err = load_firmware(chip)) < 0) return err; @@ -72,17 +69,22 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index d0302f2f00db..6ebfa6e7ab9e 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -66,15 +66,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip))) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index b28b8e4703cf..6e6a7eb555b8 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -67,28 +67,26 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) else chip->dsp_code_to_load = FW_MONA_301_DSP; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - if ((err = load_firmware(chip)) < 0) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; -- cgit v1.2.2 From 47b5d028fdce8f809bf22852ac900338fb90e8aa Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:16:10 +0100 Subject: ALSA: Echoaudio - Add suspend support #2 This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 153 ++++++++++++++++++++++++++++++++---- sound/pci/echoaudio/echoaudio.h | 2 + sound/pci/echoaudio/echoaudio_dsp.c | 145 +++++++++++++++++++--------------- 3 files changed, 222 insertions(+), 78 deletions(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 79dde9592847..2783ce6c236e 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -753,6 +753,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + DE_ACT(("pcm_trigger resume\n")); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: DE_ACT(("pcm_trigger start\n")); @@ -776,6 +778,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = start_transport(chip, channelmask, chip->pipe_cyclic_mask); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + DE_ACT(("pcm_trigger suspend\n")); case SNDRV_PCM_TRIGGER_STOP: DE_ACT(("pcm_trigger stop\n")); for (i = 0; i < DSP_MAXPIPES; i++) { @@ -1951,18 +1955,27 @@ static __devinit int snd_echo_create(struct snd_card *card, return err; pci_set_master(pci); - /* allocate a chip-specific data */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (!chip) { - pci_disable_device(pci); - return -ENOMEM; + /* Allocate chip if needed */ + if (!*rchip) { + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + pci_disable_device(pci); + return -ENOMEM; + } + DE_INIT(("chip=%p\n", chip)); + spin_lock_init(&chip->lock); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + atomic_set(&chip->opencount, 0); + mutex_init(&chip->mode_mutex); + chip->can_set_rate = 1; + } else { + /* If this was called from the resume function, chip is + * already allocated and it contains current card settings. + */ + chip = *rchip; } - DE_INIT(("chip=%p\n", chip)); - - spin_lock_init(&chip->lock); - chip->card = card; - chip->pci = pci; - chip->irq = -1; /* PCI resource allocation */ chip->dsp_registers_phys = pci_resource_start(pci, 0); @@ -2002,7 +2015,9 @@ static __devinit int snd_echo_create(struct snd_card *card, chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); - if (err) { + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { DE_INIT(("init_hw err=%d\n", err)); snd_echo_free(chip); return err; @@ -2013,9 +2028,6 @@ static __devinit int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return err; } - atomic_set(&chip->opencount, 0); - mutex_init(&chip->mode_mutex); - chip->can_set_rate = 1; *rchip = chip; /* Init done ! */ return 0; @@ -2048,6 +2060,7 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, snd_card_set_dev(card, &pci->dev); + chip = NULL; /* Tells snd_echo_create to allocate chip */ if ((err = snd_echo_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; @@ -2187,6 +2200,112 @@ ctl_error: +#if defined(CONFIG_PM) + +static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + + DE_INIT(("suspend start\n")); + snd_pcm_suspend_all(chip->analog_pcm); + snd_pcm_suspend_all(chip->digital_pcm); + +#ifdef ECHOCARD_HAS_MIDI + /* This call can sleep */ + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 0); +#endif + spin_lock_irq(&chip->lock); + if (wait_handshake(chip)) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + clear_handshake(chip); + if (send_vector(chip, DSP_VC_GO_COMATOSE) < 0) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + spin_unlock_irq(&chip->lock); + + chip->dsp_code = NULL; + free_irq(chip->irq, chip); + chip->irq = -1; + pci_save_state(pci); + pci_disable_device(pci); + + DE_INIT(("suspend done\n")); + return 0; +} + + + +static int snd_echo_resume(struct pci_dev *pci) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); + commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + + err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); + snd_echo_free(chip); + return err; + } + DE_INIT(("resume init OK\n")); + + /* Temporarily set chip->pipe_alloc_mask=0 otherwise + * restore_dsp_settings() fails. + */ + pipe_alloc_mask = chip->pipe_alloc_mask; + chip->pipe_alloc_mask = 0; + err = restore_dsp_rettings(chip); + chip->pipe_alloc_mask = pipe_alloc_mask; + if (err < 0) { + kfree(commpage_bak); + return err; + } + DE_INIT(("resume restore OK\n")); + + memcpy(&commpage->audio_format, &commpage_bak->audio_format, + sizeof(commpage->audio_format)); + memcpy(&commpage->sglist_addr, &commpage_bak->sglist_addr, + sizeof(commpage->sglist_addr)); + memcpy(&commpage->midi_output, &commpage_bak->midi_output, + sizeof(commpage->midi_output)); + kfree(commpage_bak); + + if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, + ECHOCARD_NAME, chip)) { + snd_echo_free(chip); + snd_printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + DE_INIT(("resume irq=%d\n", chip->irq)); + +#ifdef ECHOCARD_HAS_MIDI + if (chip->midi_input_enabled) + enable_midi_input(chip, TRUE); + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 1); +#endif + + DE_INIT(("resume done\n")); + return 0; +} + +#endif /* CONFIG_PM */ + + + static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2209,6 +2328,10 @@ static struct pci_driver driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), +#ifdef CONFIG_PM + .suspend = snd_echo_suspend, + .resume = snd_echo_resume, +#endif /* CONFIG_PM */ }; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index a84c0d15cc50..1df974dcb5f4 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -472,6 +472,8 @@ static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); +static void snd_echo_midi_output_trigger( + struct snd_rawmidi_substream *substream, int up); static int midi_service_irq(struct echoaudio *chip); static int __devinit snd_echo_midi_create(struct snd_card *card, struct echoaudio *chip); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 031ef7e9da91..64417a733220 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -497,9 +497,6 @@ static int load_firmware(struct echoaudio *chip) if ((box_type = load_asic(chip)) < 0) return box_type; /* error */ - if ((err = restore_dsp_rettings(chip)) < 0) - return err; - return box_type; } @@ -659,51 +656,106 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { - int err; + int i, o, err; DE_INIT(("restore_dsp_settings\n")); if ((err = check_asic_status(chip)) < 0) return err; - /* @ Gina20/Darla20 only. Should be harmless for other cards. */ + /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; chip->comm_page->handshake = 0xffffffff; - if ((err = set_sample_rate(chip, chip->sample_rate)) < 0) + /* Restore output busses */ + for (i = 0; i < num_busses_out(chip); i++) { + err = set_output_gain(chip, i, chip->output_gain[i]); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_VMIXER + for (i = 0; i < num_pipes_out(chip); i++) + for (o = 0; o < num_busses_out(chip); o++) { + err = set_vmixer_gain(chip, o, i, + chip->vmixer_gain[o][i]); + if (err < 0) + return err; + } + if (update_vmixer_level(chip) < 0) + return -EIO; +#endif /* ECHOCARD_HAS_VMIXER */ + +#ifdef ECHOCARD_HAS_MONITOR + for (o = 0; o < num_busses_out(chip); o++) + for (i = 0; i < num_busses_in(chip); i++) { + err = set_monitor_gain(chip, o, i, + chip->monitor_gain[o][i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_MONITOR */ + +#ifdef ECHOCARD_HAS_INPUT_GAIN + for (i = 0; i < num_busses_in(chip); i++) { + err = set_input_gain(chip, i, chip->input_gain[i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_INPUT_GAIN */ + + err = update_output_line_level(chip); + if (err < 0) return err; - if (chip->meters_enabled) - if (send_vector(chip, DSP_VC_METERS_ON) < 0) - return -EIO; + err = update_input_line_level(chip); + if (err < 0) + return err; -#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK - if (set_input_clock(chip, chip->input_clock) < 0) + err = set_sample_rate(chip, chip->sample_rate); + if (err < 0) + return err; + + if (chip->meters_enabled) { + err = send_vector(chip, DSP_VC_METERS_ON); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH + if (set_digital_mode(chip, chip->digital_mode) < 0) return -EIO; #endif -#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH - if (set_output_clock(chip, chip->output_clock) < 0) +#ifdef ECHOCARD_HAS_DIGITAL_IO + if (set_professional_spdif(chip, chip->professional_spdif) < 0) return -EIO; #endif - if (update_output_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_PHANTOM_POWER + if (set_phantom_power(chip, chip->phantom_power) < 0) return -EIO; +#endif - if (update_input_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK + /* set_input_clock() also restores automute setting */ + if (set_input_clock(chip, chip->input_clock) < 0) return -EIO; +#endif -#ifdef ECHOCARD_HAS_VMIXER - if (update_vmixer_level(chip) < 0) +#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH + if (set_output_clock(chip, chip->output_clock) < 0) return -EIO; #endif if (wait_handshake(chip) < 0) return -EIO; clear_handshake(chip); + if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) + return -EIO; DE_INIT(("restore_dsp_rettings done\n")); - return send_vector(chip, DSP_VC_UPDATE_FLAGS); + return 0; } @@ -920,9 +972,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) chip->card_name = ECHOCARD_NAME; chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->digital_mode = DIGITAL_MODE_NONE; - chip->input_clock = ECHO_CLOCK_INTERNAL; - chip->output_clock = ECHO_CLOCK_WORD; chip->asic_loaded = FALSE; memset(chip->comm_page, 0, sizeof(struct comm_page)); @@ -933,7 +982,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); - chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE); @@ -944,50 +992,21 @@ static int init_dsp_comm_page(struct echoaudio *chip) -/* This function initializes the several volume controls for busses and pipes. -This MUST be called after the DSP is up and running ! */ +/* This function initializes the chip structure with default values, ie. all + * muted and internal clock source. Then it copies the settings to the DSP. + * This MUST be called after the DSP is up and running ! + */ static int init_line_levels(struct echoaudio *chip) { - int st, i, o; - DE_INIT(("init_line_levels\n")); - - /* Mute output busses */ - for (i = 0; i < num_busses_out(chip); i++) - if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; - -#ifdef ECHOCARD_HAS_VMIXER - /* Mute the Vmixer */ - for (i = 0; i < num_pipes_out(chip); i++) - for (o = 0; o < num_busses_out(chip); o++) - if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_vmixer_level(chip))) - return st; -#endif /* ECHOCARD_HAS_VMIXER */ - -#ifdef ECHOCARD_HAS_MONITOR - /* Mute the monitor mixer */ - for (o = 0; o < num_busses_out(chip); o++) - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_MONITOR */ - -#ifdef ECHOCARD_HAS_INPUT_GAIN - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_input_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_INPUT_GAIN */ - - return 0; + memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); + memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); + memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); + memset(chip->vmixer_gain, ECHOGAIN_MUTED, sizeof(chip->vmixer_gain)); + chip->input_clock = ECHO_CLOCK_INTERNAL; + chip->output_clock = ECHO_CLOCK_WORD; + chip->sample_rate = 44100; + return restore_dsp_rettings(chip); } -- cgit v1.2.2 From 0a27fcfaaf61108d94f0377f91bed81b2dd35f52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Feb 2010 17:05:28 +0100 Subject: ALSA: hda - Correct ASUA blacklist for MSI brokenness The MSI blacklist entry for ASUS mobo added in the commit 8ce28d6abff34886d3797b25324c940471b99164 was based on the alsa-info output wrongly posted. Fix the id to the right one now. Reported-by: Sid Boyce Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3600e9cc9bc6..ff6da6f386d1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2350,7 +2350,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) */ static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ - SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */ + SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ {} }; -- cgit v1.2.2 From 088ef950dc0dd58d2f339e1616c9092fea923f06 Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Fri, 12 Feb 2010 12:26:47 -0800 Subject: omap2: Convert ARCH_OMAP24XX to ARCH_OMAP2 Convert ARCH_OMAP24XX to ARCH_OMAP2 Signed-off-by: Tony Lindgren --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6bbbd2ab0ee7..26e728dc1337 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -82,7 +82,7 @@ static const int omap1_dma_reqs[][2] = {}; static const unsigned long omap1_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP34XX) static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, -- cgit v1.2.2 From a8eb7ca0cbb41c9cd379b8d2a2a5efb503aa65e9 Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Fri, 12 Feb 2010 12:26:48 -0800 Subject: omap3: Replace ARCH_OMAP34XX with ARCH_OMAP3 Replace ARCH_OMAP34XX with ARCH_OMAP3 Signed-off-by: Tony Lindgren --- sound/soc/omap/omap-mcbsp.c | 6 +++--- sound/soc/omap/omap-mcbsp.h | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 26e728dc1337..c0039b35fb25 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -82,11 +82,11 @@ static const int omap1_dma_reqs[][2] = {}; static const unsigned long omap1_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3) static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, @@ -124,7 +124,7 @@ static const unsigned long omap2430_mcbsp_port[][2] = { static const unsigned long omap2430_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP3) static const unsigned long omap34xx_mcbsp_port[][2] = { { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 647d2f981ab0..1968d03bc532 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -50,7 +50,7 @@ enum omap_mcbsp_div { #undef NUM_LINKS #define NUM_LINKS 3 #endif -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) #undef NUM_LINKS #define NUM_LINKS 5 #endif -- cgit v1.2.2 From f167e1d073278fe231bbdd5d6c24fb9d091aa544 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 15 Feb 2010 08:55:28 +0100 Subject: ALSA: usb-audio: reduce MIDI packet size to work around broken firmware Extend the list of devices whose firmware does not expect more than one USB MIDI packet in one USB packet. bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752 Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Jaroslav Kysela --- sound/usb/usbmidi.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6e89b8368d9a..aae50df06232 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1162,10 +1162,22 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, pipe = usb_sndintpipe(umidi->dev, ep_info->out_ep); else pipe = usb_sndbulkpipe(umidi->dev, ep_info->out_ep); - if (umidi->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ - ep->max_transfer = 4; - else + switch (umidi->usb_id) { + default: ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); + break; + /* + * Various chips declare a packet size larger than 4 bytes, but + * do not actually work with larger packets: + */ + case USB_ID(0x0a92, 0x1020): /* ESI M4U */ + case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */ + case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ + case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */ + case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ + ep->max_transfer = 4; + break; + } for (i = 0; i < OUTPUT_URBS; ++i) { buffer = usb_buffer_alloc(umidi->dev, ep->max_transfer, GFP_KERNEL, -- cgit v1.2.2 From d39e82db73eb876c60d00f00219d767b3be30307 Mon Sep 17 00:00:00 2001 From: Sebastien Alaiwan Date: Tue, 16 Feb 2010 08:55:08 +0100 Subject: ALSA: USB MIDI support for Access Music VirusTI Here's a patch that adds MIDI support through USB for one of the Access Music synths, the VirusTI. The synth uses standard USBMIDI protocol on its USB interface 3, although it does signal "vendor specific" class. A magic string has to be sent on interface 3 to enable the sending of MIDI from the synth (this string was found by sniffing usb communication of the Windows driver). This is all my patch does, and it works on my computer. Please note that the synth can also do standard usb audio I/O on its interfaces 2&3, which already works with the current snd-usb-audio driver, except for the audio input from the synth. I'm going to work on it when I have some time. Signed-off-by: Sebastien Alaiwan Signed-off-by: Clemens Ladisch (cosmetics, list terminator) Signed-off-by: Jaroslav Kysela --- sound/usb/usbaudio.c | 32 ++++++++++++++++++++++++++++++++ sound/usb/usbmidi.c | 6 ++++++ sound/usb/usbquirks.h | 27 +++++++++++++++++++++++++++ 3 files changed, 65 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4963defee18a..d01ec188b602 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3326,6 +3326,32 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } +/* + * This call will put the synth in "USB send" mode, i.e it will send MIDI + * messages through USB (this is disabled at startup). The synth will + * acknowledge by sending a sysex on endpoint 0x85 and by displaying a USB + * sign on its LCD. Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) +{ + int err, actual_length; + + /* "midi send" enable */ + static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + + void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (!buf) + return -ENOMEM; + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, + ARRAY_SIZE(seq), &actual_length, 1000); + kfree(buf); + if (err < 0) + return err; + + return 0; +} + /* * Setup quirks */ @@ -3624,6 +3650,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* Access Music VirusTI Desktop */ + if (id == USB_ID(0x133e, 0x0815)) { + if (snd_usb_accessmusic_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6e89b8368d9a..8f5bc1e8dabc 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1407,6 +1407,12 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Access Music Virus TI */ + EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), + PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER), }; static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index a892bda03df9..406b74b65ffb 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2073,6 +2073,33 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Access Music devices */ +{ + /* VirusTI Desktop */ + USB_DEVICE_VENDOR_SPEC(0x133e, 0x0815), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + /* */ { /* aka. Serato Scratch Live DJ Box */ -- cgit v1.2.2 From ebfdeea3df2b8c265975b6acc47996a0b7c507e8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:17:09 +0100 Subject: ALSA: usbmixer - introduce /proc/asound/card#/usbmixer file The usbmixer proc file contains mapping between ALSA control API and USB mixer control units. The purpose of this file is for debugging and a problem diagnostics. Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 75 ++++++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 61 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index dd0c1d7bf3ed..170bfd4fc9f6 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -69,13 +69,16 @@ static const struct rc_config { { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; +#define MAX_ID_ELEMS 256 + struct usb_mixer_interface { struct snd_usb_audio *chip; unsigned int ctrlif; struct list_head list; unsigned int ignore_ctl_error; struct urb *urb; - struct usb_mixer_elem_info **id_elems; /* array[256], indexed by unit id */ + /* array[MAX_ID_ELEMS], indexed by unit id */ + struct usb_mixer_elem_info **id_elems; /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; @@ -1825,6 +1828,45 @@ static void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, info->elem_id); } +static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, + int unitid, + struct usb_mixer_elem_info *cval) +{ + static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", + "S8", "U8", "S16", "U16"}; + snd_iprintf(buffer, " Unit: %i\n", unitid); + if (cval->elem_id) + snd_iprintf(buffer, " Control: name=\"%s\", index=%i\n", + cval->elem_id->name, cval->elem_id->index); + snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " + "channels=%i, type=\"%s\"\n", cval->id, + cval->control, cval->cmask, cval->channels, + val_types[cval->val_type]); + snd_iprintf(buffer, " Volume: min=%i, max=%i, dBmin=%i, dBmax=%i\n", + cval->min, cval->max, cval->dBmin, cval->dBmax); +} + +static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_usb_audio *chip = entry->private_data; + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + snd_iprintf(buffer, + "USB Mixer: ctrlif=%i, ctlerr=%i\n", + mixer->ctrlif, mixer->ignore_ctl_error); + snd_iprintf(buffer, "Card: %s\n", chip->card->longname); + for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { + for (cval = mixer->id_elems[unitid]; cval; + cval = cval->next_id_elem) + snd_usb_mixer_dump_cval(buffer, unitid, cval); + } + } +} + static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, int unitid) { @@ -2187,20 +2229,21 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) } void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, - unsigned char samplerate_id) + unsigned char samplerate_id) { - struct usb_mixer_interface *mixer; - struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid = 12; /* SamleRate ExtensionUnit ID */ - list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer->id_elems[unitid]; - if (cval) { - set_cur_ctl_value(cval, cval->control << 8, samplerate_id); + list_for_each_entry(mixer, &chip->mixer_list, list) { + cval = mixer->id_elems[unitid]; + if (cval) { + set_cur_ctl_value(cval, cval->control << 8, + samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); - } - break; - } + } + break; + } } int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, @@ -2210,6 +2253,7 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, .dev_free = snd_usb_mixer_dev_free }; struct usb_mixer_interface *mixer; + struct snd_info_entry *entry; int err; strcpy(chip->card->mixername, "USB Mixer"); @@ -2236,8 +2280,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { - struct snd_info_entry *entry; - if ((err = snd_audigy2nx_controls_create(mixer)) < 0) goto _error; if (!snd_card_proc_new(chip->card, "audigy2nx", &entry)) @@ -2255,6 +2297,11 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops); if (err < 0) goto _error; + + if (list_empty(&chip->mixer_list) && + !snd_card_proc_new(chip->card, "usbmixer", &entry)) + snd_info_set_text_ops(entry, chip, snd_usb_mixer_proc_read); + list_add(&mixer->list, &chip->mixer_list); return 0; -- cgit v1.2.2 From 3be522a9514f58e0596db34898a514df206cadc5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:55:43 +0100 Subject: ALSA: pcm core - fix fifo_size channels interval check Signed-off-by: Jaroslav Kysela Cc: --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 25b0641e6b8c..f7e1c9f0d3ed 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -315,10 +315,10 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!params->info) params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; if (!params->fifo_size) { - if (snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) == - snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) && - snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS]) == - snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS])) { + m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + if (snd_mask_min(m) == snd_mask_max(m) && + snd_interval_min(i) == snd_interval_max(i)) { changed = substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_FIFO_SIZE, params); if (changed < 0) -- cgit v1.2.2 From 7affdc17d49b5d9e9c350d5d99ee34ab8655c7b4 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:52:27 +0100 Subject: ALSA: usbmixer - add usb_id value to usbmixer proc file Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 170bfd4fc9f6..03f125dca5ff 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1856,8 +1856,9 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, list_for_each_entry(mixer, &chip->mixer_list, list) { snd_iprintf(buffer, - "USB Mixer: ctrlif=%i, ctlerr=%i\n", - mixer->ctrlif, mixer->ignore_ctl_error); + "USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n", + chip->usb_id, mixer->ctrlif, + mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { for (cval = mixer->id_elems[unitid]; cval; -- cgit v1.2.2 From 291186e049d7f8178ad31d43c38a53889f25d79e Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:55:18 +0100 Subject: ALSA: usbmixer - use MAX_ID_ELEMS where possible Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 03f125dca5ff..35b4830fb0c4 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -108,7 +108,7 @@ struct mixer_build { struct usb_mixer_interface *mixer; unsigned char *buffer; unsigned int buflen; - DECLARE_BITMAP(unitbitmap, 256); + DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -2265,7 +2265,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, mixer->chip = chip; mixer->ctrlif = ctrlif; mixer->ignore_ctl_error = ignore_error; - mixer->id_elems = kcalloc(256, sizeof(*mixer->id_elems), GFP_KERNEL); + mixer->id_elems = kcalloc(MAX_ID_ELEMS, sizeof(*mixer->id_elems), + GFP_KERNEL); if (!mixer->id_elems) { kfree(mixer); return -ENOMEM; -- cgit v1.2.2 From 96dd362284ddcb546d2783035ae7eeda73692eda Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Feb 2010 11:05:44 +0000 Subject: ASoC: Make pmdown_time a per-card setting Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ca89c782132d..94b9cde26139 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -542,7 +542,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; schedule_delayed_work(&card->delayed_work, - msecs_to_jiffies(pmdown_time)); + msecs_to_jiffies(card->pmdown_time)); } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(codec, @@ -1039,6 +1039,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) dev_dbg(card->dev, "All components present, instantiating\n"); /* Found everything, bring it up */ + card->pmdown_time = pmdown_time; + if (card->probe) { ret = card->probe(pdev); if (ret < 0) -- cgit v1.2.2 From dbe21408b15f04da4f80fb89a27b7cb067d6103e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Feb 2010 11:37:24 +0000 Subject: ASoC: Make pmdown_time runtime configurable Provide a sysfs file allowing userspace to inspect and change the pmdown_time setting at runtime. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 94b9cde26139..c2008bc9c64a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -130,6 +130,29 @@ static ssize_t codec_reg_show(struct device *dev, static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +static ssize_t pmdown_time_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *socdev = dev_get_drvdata(dev); + struct snd_soc_card *card = socdev->card; + + return sprintf(buf, "%d\n", card->pmdown_time); +} + +static ssize_t pmdown_time_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_soc_device *socdev = dev_get_drvdata(dev); + struct snd_soc_card *card = socdev->card; + + strict_strtol(buf, 10, &card->pmdown_time); + + return count; +} + +static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); + #ifdef CONFIG_DEBUG_FS static int codec_reg_open_file(struct inode *inode, struct file *file) { @@ -1124,6 +1147,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); + ret = device_create_file(card->socdev->dev, &dev_attr_pmdown_time); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); if (ret < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); -- cgit v1.2.2 From e5e878c1c393de917391477bc7627d729f7568fb Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 16 Feb 2010 13:23:15 +0200 Subject: ASoC: tlv320dac33: Clearing FIFOFLUSH flag before playback In repeated playback the FIFOFLUSH bit remained set, and never has been cleared. Clear it during the setup phase. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 1b35d0cf3364..dab7fd5be867 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -734,7 +734,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK); + /* Read FIFO control A, and clear FIFO flush bit */ fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + fifoctrl_a &= ~DAC33_FIFOFLUSH; + fifoctrl_a &= ~DAC33_WIDTH; switch (substream->runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: -- cgit v1.2.2 From 7833ae0edf50b0eb303e95b1bec5fbd63a1e2672 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 16 Feb 2010 13:23:16 +0200 Subject: ASoC: tlv320dac33: Correct the OSCSET calculation OSCSET calculation was not correct in case of 44.1KHz sampling rate. With small adjustment both 48 and 44.1 KHz calculation now gives the correct value. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index dab7fd5be867..f9f367d29a90 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -700,7 +700,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, } #define CALC_OSCSET(rate, refclk) ( \ - ((((rate * 10000) / refclk) * 4096) + 5000) / 10000) + ((((rate * 10000) / refclk) * 4096) + 7000) / 10000) #define CALC_RATIOSET(rate, refclk) ( \ ((((refclk * 100000) / rate) * 16384) + 50000) / 100000) -- cgit v1.2.2 From b721e68bdc5b39c51bf6a1469f8d3663fbe03243 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Wed, 17 Feb 2010 00:57:44 +0100 Subject: ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50 This patch fixes a division by zero error in the irq handler. There is a small window between the hw_params() callback and when runtime->frame_bits is set by ALSA middle layer. When another substream is already running, if an interrupt is delivered during that window the irq handler calls pcm_pointer() which does a division by zero. The patch below makes the irq handler skip substreams that are initialized but not started yet. Cc to Clemens Ladisch because he proposed an alternate fix. For more information, please read the original thread in the linux-kernel mailing list: http://lkml.org/lkml/2010/2/2/187 Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1305f7ca02c3..641d7f07392c 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1821,7 +1821,9 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) /* The hardware doesn't tell us which substream caused the irq, thus we have to check all running substreams. */ for (ss = 0; ss < DSP_MAXPIPES; ss++) { - if ((substream = chip->substream[ss])) { + substream = chip->substream[ss]; + if (substream && ((struct audiopipe *)substream->runtime-> + private_data)->state == PIPE_STATE_STARTED) { period = pcm_pointer(substream) / substream->runtime->period_size; if (period != chip->last_period[ss]) { -- cgit v1.2.2 From e47c796d58a21fc58b00dffb7265bb66de987773 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 17 Feb 2010 09:49:54 +0200 Subject: ASoC: TWL4030: Use codec defaults for Headset initial configuration Disable the amplifiers for the headset outputs, and do not select routings by default to the headset outputs. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 277862e480e2..6f5d4af20052 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -75,8 +75,8 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ 0x00, /* REG_EAR_CTL (0x21) */ - 0x24, /* REG_HS_SEL (0x22) */ - 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x00, /* REG_HS_SEL (0x22) */ + 0x00, /* REG_HS_GAIN_SET (0x23) */ 0x00, /* REG_HS_POPN_SET (0x24) */ 0x00, /* REG_PREDL_CTL (0x25) */ 0x00, /* REG_PREDR_CTL (0x26) */ -- cgit v1.2.2 From 6c5f1fed49f96a0600aa9a97ac3faf972c33a341 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 17 Feb 2010 14:30:44 +0000 Subject: ASoC: Make pmdown_time a long Fixes a warning. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c2008bc9c64a..e1c0336868e1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -136,7 +136,7 @@ static ssize_t pmdown_time_show(struct device *dev, struct snd_soc_device *socdev = dev_get_drvdata(dev); struct snd_soc_card *card = socdev->card; - return sprintf(buf, "%d\n", card->pmdown_time); + return sprintf(buf, "%ld\n", card->pmdown_time); } static ssize_t pmdown_time_set(struct device *dev, -- cgit v1.2.2 From 7fb2d723e65cc793213515fa1da092b7c92a5b48 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Thu, 18 Feb 2010 07:01:20 +0100 Subject: ALSA: cs46xx - Do test writes to register AC97_REC_GAIN in snd_cs46xx_codec_reset() bypassing the register cache, so as to not clobber the cached register value during resume. Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 56fcf00c0e27..9fea5bb448cd 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2266,7 +2266,7 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) return; /* test if we can write to the record gain volume register */ - snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x8a05); + snd_ac97_write(ac97, AC97_REC_GAIN, 0x8a05); if ((err = snd_ac97_read(ac97, AC97_REC_GAIN)) == 0x8a05) return; -- cgit v1.2.2 From 04510a74bfbcbfd53dd48b3094aad89d5eca1d28 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Thu, 18 Feb 2010 07:03:55 +0100 Subject: ALSA: cs46xx - fix some typos Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 9fea5bb448cd..3f99a5e8528c 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2238,11 +2238,11 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) /* set the desired CODEC mode */ if (ac97->num == CS46XX_PRIMARY_CODEC_INDEX) { - snd_printdd("cs46xx: CODOEC1 mode %04x\n",0x0); - snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x0); + snd_printdd("cs46xx: CODEC1 mode %04x\n", 0x0); + snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x0); } else if (ac97->num == CS46XX_SECONDARY_CODEC_INDEX) { - snd_printdd("cs46xx: CODOEC2 mode %04x\n",0x3); - snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x3); + snd_printdd("cs46xx: CODEC2 mode %04x\n", 0x3); + snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x3); } else { snd_BUG(); /* should never happen ... */ } -- cgit v1.2.2 From ba579eb7b30791751f556ee01905636cda50c864 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sat, 20 Feb 2010 11:16:30 -0500 Subject: ALSA: hda: Use 3stack quirk for Toshiba Satellite L40-10Q BugLink: https://bugs.launchpad.net/bugs/524948 The OR has verified that the existing model=laptop-eapd quirk does not function correctly but instead needs model=3stack. Make this change so that manual corrections to module-init-tools file(s) are not required. Reported-by: Lasse Havelund CC: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 21011b5199de..7832f363496f 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1098,7 +1098,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), - SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50), -- cgit v1.2.2 From e458b1fadf9239d1fdb165ff4c4ea0d807041bec Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Fri, 12 Feb 2010 16:28:29 +1100 Subject: ALSA: hda - Add Macmini 3,1 support BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989 Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The pinout is almost identical to the mb5 quirk, except for no microphone and the line-in mixer controls being on a different index. Everything works in 2ch mode, but as I am not sure what needs to be changed for 6ch mode, or whether the Mac Mini's chip supports 6ch mode, I have simply duplicated the code from the mb5 quirk for the mac mini chmode management. The new model parameter for this quirk is "macmini3". Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 136 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 136 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c224977c8cf..b5a6ba025930 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -211,6 +211,7 @@ enum { ALC885_MACPRO, ALC885_MBP3, ALC885_MB5, + ALC885_MACMINI3, ALC885_IMAC24, ALC885_IMAC91, ALC883_3ST_2ch_DIG, @@ -6751,6 +6752,14 @@ static struct hda_input_mux mb5_capture_source = { }, }; +static struct hda_input_mux macmini3_capture_source = { + .num_items = 2, + .items = { + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_3stack_6ch_intel = { .num_items = 4, .items = { @@ -6999,6 +7008,35 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static struct hda_verb alc885_macmini3_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static struct hda_verb alc885_macmini3_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; + /* * 2ch mode @@ -7243,6 +7281,21 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_macmini3_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc885_imac91_mixer[] = { HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7617,6 +7670,53 @@ static struct hda_verb alc885_mb5_init_verbs[] = { { } }; +/* Macmini 3,1 */ +static struct hda_verb alc885_macmini3_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7800,6 +7900,18 @@ static void alc885_mb5_automute(struct hda_codec *codec) } +static void alc885_macmini3_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + static void alc885_mb5_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -7808,6 +7920,14 @@ static void alc885_mb5_unsol_event(struct hda_codec *codec, alc885_mb5_automute(codec); } +static void alc885_macmini3_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -8974,6 +9094,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", + [ALC885_MACMINI3] = "macmini3", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9157,6 +9278,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), {} /* terminator */ }; @@ -9238,6 +9360,20 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc885_mb5_unsol_event, .init_hook = alc885_mb5_automute, }, + [ALC885_MACMINI3] = { + .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_macmini3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_macmini3_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), + .input_mux = &macmini3_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_macmini3_unsol_event, + .init_hook = alc885_macmini3_automute, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, -- cgit v1.2.2 From 9d54f08bc77bf6dfe835b945d03b6e127c9fc5a3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Feb 2010 08:34:40 +0100 Subject: ALSA: hda - Clean up Intel Mac unsol codes Use the standard unsol_event callback with each setup callback for IntelMac models with Realtek ALC885 codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 81 +++++++++---------------------------------- 1 file changed, 17 insertions(+), 64 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b5a6ba025930..f8fb260a2dd7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7879,6 +7879,9 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x1a; } +#define alc885_mb5_setup alc885_imac24_setup +#define alc885_macmini3_setup alc885_imac24_setup + static void alc885_mbp3_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -7887,66 +7890,13 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } -static void alc885_mb5_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - -} - -static void alc885_macmini3_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -static void alc885_mb5_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_mb5_automute(codec); -} - -static void alc885_macmini3_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_mb5_automute(codec); -} - -static void alc885_imac91_automute(struct hda_codec *codec) +static void alc885_imac91_setup(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - -} + struct alc_spec *spec = codec->spec; -static void alc885_imac91_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_imac91_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[1] = 0x1a; } static struct hda_verb alc882_targa_verbs[] = { @@ -9357,8 +9307,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_mb5_unsol_event, - .init_hook = alc885_mb5_automute, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_mb5_setup, + .init_hook = alc_automute_amp, }, [ALC885_MACMINI3] = { .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, @@ -9371,8 +9322,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &macmini3_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_macmini3_unsol_event, - .init_hook = alc885_macmini3_automute, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_macmini3_setup, + .init_hook = alc_automute_amp, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, @@ -9411,8 +9363,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_imac91_unsol_event, - .init_hook = alc885_imac91_automute, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_imac91_setup, + .init_hook = alc_automute_amp, }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, -- cgit v1.2.2 From 2448158ed2ae64ef3219b51e0176a4e1151ba9ec Mon Sep 17 00:00:00 2001 From: Paul Menzel Date: Mon, 8 Feb 2010 20:37:26 +0100 Subject: ALSA: Typo. s/distrubs/disturbs/ Signed-off-by: Paul Menzel Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 06f230f518b7..051cf5145330 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1411,7 +1411,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing * codec often screws up the controller chip, - * and distrubs the further communications. + * and disturbs the further communications. * Thus if an error occurs during probing, * better to reset the controller chip to * get back to the sanity state. -- cgit v1.2.2 From 0708cc582f0fe2578eaab722841caf2b4f8cfe37 Mon Sep 17 00:00:00 2001 From: Paul Menzel Date: Mon, 8 Feb 2010 20:42:46 +0100 Subject: ALSA: hda-intel: Add position_fix quirk for ASUS M2V-MX SE. With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1]. Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE. The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker. $ lspci -vvnn | grep -A10 Audio 20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:8290] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- SERR- Kernel driver in use: HDA Intel [1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user Signed-off-by: Paul Menzel Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 051cf5145330..22dcdc201ede 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2264,6 +2264,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), {} -- cgit v1.2.2 From 40717382e0c1f572553e4fdefb489db4b95a5e7e Mon Sep 17 00:00:00 2001 From: Chris J Arges Date: Wed, 17 Feb 2010 12:12:52 -0600 Subject: ALSA: usbaudio Mbox support, output only Signed-off-by: Chris J Arges Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index e691eba6a83e..fc1d2cd6ccc3 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2215,6 +2215,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Digidesign Mbox */ +{ + /* Thanks to Clemens Ladisch */ + USB_DEVICE(0x0dba, 0x1000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "MBox", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .format = SNDRV_PCM_FORMAT_S24_3BE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x02, + .ep_attr = 0x01, + .maxpacksize = 0x130, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = -1 + } + } + + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v1.2.2 From bf30a4309d4294d3eca248ea8a20c1c3570f5e74 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 22 Feb 2010 10:33:13 +0100 Subject: ALSA: via82xx: add quirk for D1289 motherboard Add a headphones-only quirk for the Fujitsu Siemens D1289. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Marc Haber Cc: Signed-off-by: Jaroslav Kysela --- sound/pci/via82xx.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8a332d2f615c..03d6aea19749 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1790,6 +1790,12 @@ static struct ac97_quirk ac97_quirks[] = { .name = "ASRock K7VT2", .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x110a, + .subdevice = 0x0079, + .name = "Fujitsu Siemens D1289", + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x1019, .subdevice = 0x0a81, -- cgit v1.2.2 From b9dd94a87e5b4d0e864636698931aeeeb3c9d770 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 22 Feb 2010 13:27:13 +0200 Subject: ASoC: core: On resume also check the soc device state Check the card->codec on soc_resume to detect if the soc device is properly initialized. If the card->codec is NULL, than do not continue the resume operation, since the device is not initialized properly. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1c0336868e1..a03bac943aaf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -963,6 +963,12 @@ static int soc_resume(struct device *dev) struct snd_soc_card *card = socdev->card; struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (!card->codec) + return 0; + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume -- cgit v1.2.2 From d01aecdf900574cf6be7c1c6114e708801126baf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Feb 2010 08:07:15 +0100 Subject: ALSA: hda - Remove identical definitions for macmini3 model The channel mode definitions for macmini3 model are identical with mb5. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 +----------------------------- 1 file changed, 1 insertion(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f8fb260a2dd7..c74ca39a0b8e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7008,35 +7008,7 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; -/* - * 2ch - * Speakers/Woofer/HP = Front - * LineIn = Input - */ -static struct hda_verb alc885_macmini3_ch2_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - { } /* end */ -}; - -/* - * 6ch mode - * Speakers/HP = Front - * Woofer = LFE - * LineIn = Surround - */ -static struct hda_verb alc885_macmini3_ch6_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - { } /* end */ -}; - -static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { - { 2, alc885_mb5_ch2_init }, - { 6, alc885_mb5_ch6_init }, -}; - +#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes /* * 2ch mode -- cgit v1.2.2 From 32679f95cac3b1bdf27dce8b5273e06af186fd91 Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Mon, 22 Feb 2010 17:31:09 -0800 Subject: ALSA: hda - enable snoop for Intel Cougar Point This patch enables snoop, eliminating static during playback. This patch supersedes the previous Cougar Point audio patch. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 22dcdc201ede..1adac8cc9592 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -448,6 +448,7 @@ struct azx { /* driver types */ enum { AZX_DRIVER_ICH, + AZX_DRIVER_PCH, AZX_DRIVER_SCH, AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, @@ -462,6 +463,7 @@ enum { static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", + [AZX_DRIVER_PCH] = "HDA Intel PCH", [AZX_DRIVER_SCH] = "HDA Intel MID", [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", @@ -1064,6 +1066,7 @@ static void azx_init_pci(struct azx *chip) 0x01, NVIDIA_HDA_ENABLE_COHBIT); break; case AZX_DRIVER_SCH: + case AZX_DRIVER_PCH: pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) { pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, @@ -2421,6 +2424,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, if (bdl_pos_adj[dev] < 0) { switch (chip->driver_type) { case AZX_DRIVER_ICH: + case AZX_DRIVER_PCH: bdl_pos_adj[dev] = 1; break; default: @@ -2700,7 +2704,7 @@ static struct pci_device_id azx_ids[] = { /* PCH */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, /* CPT */ - { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ -- cgit v1.2.2 From 28e1b773083d349d5223f586a39fa30f5d0f1c36 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:09 +0100 Subject: ALSA: usbaudio: parse USB descriptors with structs In preparation of support for v2.0 audio class, use the structs from linux/usb/audio.h and add some new ones to describe the fields that are actually parsed by the descriptor decoders. Also, factor out code from usb_create_streams(). This makes it easier to adopt the new iteration logic needed for v2.0. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 198 +++++++++++++++++++++++++++++++-------------------- sound/usb/usbmixer.c | 37 +++++----- 2 files changed, 140 insertions(+), 95 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c6b9c8cac59e..f833dea60180 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -2421,15 +2423,17 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * * @fmt: the format type descriptor */ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; int sample_width, sample_bytes; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; /* FIXME: correct endianess and sign? */ pcm_format = -1; - sample_width = fmt[6]; - sample_bytes = fmt[5]; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + switch (format) { case 0: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", @@ -2442,7 +2446,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor sample_width, sample_bytes); } /* check the format byte size */ - switch (fmt[5]) { + switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; break; @@ -2463,8 +2467,8 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor break; default: snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", - chip->dev->devnum, fp->iface, - fp->altsetting, sample_width, sample_bytes); + chip->dev->devnum, fp->iface, fp->altsetting, + sample_width, sample_bytes); break; } break; @@ -2564,11 +2568,12 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - if (fmt[3] == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == USB_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2590,23 +2595,27 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * if (pcm_format < 0) return -1; } + fp->format = pcm_format; - fp->channels = fmt[4]; + fp->channels = fmt->bNrChannels; + if (fp->channels < 1) { snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); return -1; } - return parse_audio_format_rates(chip, fp, fmt, 7); + return parse_audio_format_rates(chip, fp, fmt_raw, 7); } /* - * prase the format type II descriptor + * parse the format type II descriptor */ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int brate, framesize; + struct uac_format_type_ii_discrete_descriptor *fmt = fmt_raw; + switch (format) { case USB_AUDIO_FORMAT_AC3: /* FIXME: there is no AC3 format defined yet */ @@ -2622,20 +2631,25 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; } + fp->channels = 1; - brate = combine_word(&fmt[4]); /* fmt[4,5] : wMaxBitRate (in kbps) */ - framesize = combine_word(&fmt[6]); /* fmt[6,7]: wSamplesPerFrame */ + + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); fp->frame_size = framesize; - return parse_audio_format_rates(chip, fp, fmt, 8); /* fmt[8..] sample rates */ + return parse_audio_format_rates(chip, fp, fmt_raw, 8); /* fmt[8..] sample rates */ } static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream) + int format, void *fmt_raw, int stream) { int err; + /* we only parse the common header of all format types here, + * so it is safe to take a type_i struct */ + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - switch (fmt[3]) { + switch (fmt->bFormatType) { case USB_FORMAT_TYPE_I: case USB_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt); @@ -2645,10 +2659,10 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + chip->dev->devnum, fp->iface, fp->altsetting, fmt->bFormatType); return -1; } - fp->fmt_type = fmt[3]; + fp->fmt_type = fmt->bFormatType; if (err < 0) return err; #if 1 @@ -2659,7 +2673,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt->bFormatType == USB_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2708,6 +2722,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num = 4; for (i = 0; i < num; i++) { + struct uac_as_header_descriptor_v1 *as; + alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); /* skip invalid one */ @@ -2726,7 +2742,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2003) && @@ -2734,20 +2750,21 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; /* get audio formats */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); - if (!fmt) { + as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } - if (fmt[0] < 7) { + if (as->bLength < sizeof(*as)) { snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } - format = (fmt[6] << 8) | fmt[5]; /* remember the format value */ + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ /* get format type */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); @@ -2875,6 +2892,65 @@ static void snd_usb_stream_disconnect(struct list_head *head) } } +static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int interface) +{ + struct usb_device *dev = chip->dev; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_interface *iface = usb_ifnum_to_if(dev, interface); + + if (!iface) { + snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + if (usb_interface_claimed(iface)) { + snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { + snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + + return 0; + } + + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", + dev->devnum, ctrlif, interface, altsd->bInterfaceClass); + /* skip non-supported classes */ + return -EINVAL; + } + + if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { + snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + return -EINVAL; + } + + if (! parse_audio_endpoints(chip, interface)) { + usb_set_interface(dev, interface, 0); /* reset the current interface */ + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return -EINVAL; + } + + return 0; +} + /* * parse audio control descriptor and create pcm/midi streams */ @@ -2882,69 +2958,36 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) { struct usb_device *dev = chip->dev; struct usb_host_interface *host_iface; - struct usb_interface *iface; - unsigned char *p1; - int i, j; + struct uac_ac_header_descriptor_v1 *h1; + void *control_header; + int i; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; - if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) { + control_header = snd_usb_find_csint_desc(host_iface->extra, + host_iface->extralen, + NULL, HEADER); + + if (!control_header) { snd_printk(KERN_ERR "cannot find HEADER\n"); return -EINVAL; } - if (! p1[7] || p1[0] < 8 + p1[7]) { - snd_printk(KERN_ERR "invalid HEADER\n"); + + h1 = control_header; + + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); return -EINVAL; } - /* - * parse all USB audio streaming interfaces - */ - for (i = 0; i < p1[7]; i++) { - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - j = p1[8 + i]; - iface = usb_ifnum_to_if(dev, j); - if (!iface) { - snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", - dev->devnum, ctrlif, j); - continue; - } - if (usb_interface_claimed(iface)) { - snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, j); - continue; - } - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); - if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || - altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { - int err = snd_usbmidi_create(chip->card, iface, - &chip->midi_list, NULL); - if (err < 0) { - snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j); - continue; - } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - continue; - } - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { - snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, j, altsd->bInterfaceClass); - /* skip non-supported classes */ - continue; - } - if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); - continue; - } - if (! parse_audio_endpoints(chip, j)) { - usb_set_interface(dev, j, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - } + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + return -EINVAL; } + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + return 0; } @@ -3607,7 +3650,6 @@ static void *snd_usb_audio_probe(struct usb_device *dev, ifnum = get_iface_desc(alts)->bInterfaceNumber; id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); - if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) goto __err_val; diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 35b4830fb0c4..11636a6112d5 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -32,6 +32,8 @@ #include #include #include +#include + #include #include #include @@ -1086,29 +1088,30 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, * * most of controlls are defined here. */ -static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsigned char *ftr) +static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr) { int channels, i, j; struct usb_audio_term iterm; unsigned int master_bits, first_ch_bits; int err, csize; + struct uac_feature_unit_descriptor *ftr = _ftr; - if (ftr[0] < 7 || ! (csize = ftr[5]) || ftr[0] < 7 + csize) { + if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } /* parse the source unit */ - if ((err = parse_audio_unit(state, ftr[4])) < 0) + if ((err = parse_audio_unit(state, ftr->bSourceID)) < 0) return err; /* determine the input source type and name */ - if (check_input_term(state, ftr[4], &iterm) < 0) + if (check_input_term(state, ftr->bSourceID, &iterm) < 0) return -EINVAL; - channels = (ftr[0] - 7) / csize - 1; + channels = (ftr->bLength - 7) / csize - 1; - master_bits = snd_usb_combine_bytes(ftr + 6, csize); + master_bits = snd_usb_combine_bytes(ftr->controls, csize); /* master configuration quirks */ switch (state->chip->usb_id) { case USB_ID(0x08bb, 0x2702): @@ -1119,21 +1122,21 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig break; } if (channels > 0) - first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); + first_ch_bits = snd_usb_combine_bytes(ftr->controls + csize, csize); else first_ch_bits = 0; /* check all control types */ for (i = 0; i < 10; i++) { unsigned int ch_bits = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(ftr + 6 + csize * (j+1), csize); + unsigned int mask = snd_usb_combine_bytes(ftr->controls + csize * (j+1), csize); if (mask & (1 << i)) ch_bits |= (1 << j); } if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, ftr, ch_bits, i, &iterm, unitid); + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid); if (master_bits & (1 << i)) - build_feature_ctl(state, ftr, 0, i, &iterm, unitid); + build_feature_ctl(state, _ftr, 0, i, &iterm, unitid); } return 0; @@ -1780,7 +1783,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { - unsigned char *desc; + struct uac_output_terminal_descriptor_v1 *desc; struct mixer_build state; int err; const struct usbmix_ctl_map *map; @@ -1805,13 +1808,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) desc = NULL; while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { - if (desc[0] < 9) + if (desc->bLength < 9) continue; /* invalid descriptor? */ - set_bit(desc[3], state.unitbitmap); /* mark terminal ID as visited */ - state.oterm.id = desc[3]; - state.oterm.type = combine_word(&desc[4]); - state.oterm.name = desc[8]; - err = parse_audio_unit(&state, desc[7]); + set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + state.oterm.id = desc->bTerminalID; + state.oterm.type = le16_to_cpu(desc->wTerminalType); + state.oterm.name = desc->iTerminal; + err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) return err; } -- cgit v1.2.2 From 8fee4aff8c89c229593b76a6ab172a9cad24b412 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:10 +0100 Subject: ALSA: usbaudio: introduce new types for audio class v2 This patch adds some definitions for audio class v2. Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have different numerical representations in both standards, so there is need for a _V1 add-on now. usbmixer.c is changed accordingly. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.h | 19 ++++++++++++++++--- sound/usb/usbmixer.c | 14 +++++++------- 2 files changed, 23 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9d8cea48fc5f..4f482939e8e8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,8 +36,17 @@ #define MIXER_UNIT 0x04 #define SELECTOR_UNIT 0x05 #define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT 0x07 -#define EXTENSION_UNIT 0x08 +#define PROCESSING_UNIT_V1 0x07 +#define EXTENSION_UNIT_V1 0x08 + +/* audio class v2 */ +#define EFFECT_UNIT 0x07 +#define PROCESSING_UNIT_V2 0x08 +#define EXTENSION_UNIT_V2 0x09 +#define CLOCK_SOURCE 0x0a +#define CLOCK_SELECTOR 0x0b +#define CLOCK_MULTIPLIER 0x0c +#define SAMPLE_RATE_CONVERTER 0x0d #define AS_GENERAL 0x01 #define FORMAT_TYPE 0x02 @@ -60,7 +69,7 @@ #define EP_CS_ATTR_PITCH_CONTROL 0x02 #define EP_CS_ATTR_FILL_MAX 0x80 -/* Audio Class specific Request Codes */ +/* Audio Class specific Request Codes (v1) */ #define SET_CUR 0x01 #define GET_CUR 0x81 @@ -74,6 +83,10 @@ #define GET_MEM 0x85 #define GET_STAT 0xff +/* Audio Class specific Request Codes (v2) */ +#define CS_CUR 0x01 +#define CS_RANGE 0x02 + /* Terminal Control Selectors */ #define COPY_PROTECT_CONTROL 0x01 diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 11636a6112d5..ca7949598191 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT && p[3] == unit) + if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -607,9 +607,9 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type >> 16) { case SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; case MIXER_UNIT: strcpy(name, "Mixer"); return 5; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT: - case EXTENSION_UNIT: + case PROCESSING_UNIT_V1: + case EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -1747,9 +1747,9 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_selector_unit(state, unitid, p1); case FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); -- cgit v1.2.2 From 53ee98fe8ac77d00bacc1c814d450d83cbd193d4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:11 +0100 Subject: ALSA: usbaudio: implement basic set of class v2.0 parser This adds a number of parsers for audio class v2.0. In particular, the following internals are different and now handled by the code: * the number of streaming interfaces is now reported by an interface association descriptor. The old approach using a proprietary descriptor is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the format_type descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than a fixed value. For now, this is worked around by choosing just one of them. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed im the class request command. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 352 ++++++++++++++++++++++++++++++++++++++++++--------- sound/usb/usbaudio.h | 3 + 2 files changed, 292 insertions(+), 63 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f833dea60180..411a6cf43c21 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2422,17 +2422,53 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * * @format: the format tag (wFormatTag) * @fmt: the format type descriptor */ -static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_i_type(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + int protocol) { - int pcm_format; + int pcm_format, i; int sample_width, sample_bytes; - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; + + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + break; + } + + case UAC_VERSION_2: { + struct uac_format_type_i_ext_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubslotSize; + + /* + * FIXME + * USB audio class v2 devices specify a bitmap of possible + * audio formats rather than one fix value. For now, we just + * pick one of them and report that as the only possible + * value for this setting. + * The bit allocation map is in fact compatible to the + * wFormatTag of the v1 AS streaming descriptors, which is why + * we can simply map the matrix. + */ + + for (i = 0; i < 5; i++) + if (format & (1UL << i)) { + format = i + 1; + break; + } + + break; + } + + default: + return -EINVAL; + } /* FIXME: correct endianess and sign? */ pcm_format = -1; - sample_width = fmt->bBitResolution; - sample_bytes = fmt->bSubframeSize; switch (format) { case 0: /* some devices don't define this correctly... */ @@ -2446,6 +2482,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor sample_width, sample_bytes); } /* check the format byte size */ + printk(" XXXXX SAMPLE BYTES %d\n", sample_bytes); switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; @@ -2500,7 +2537,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor /* * parse the format descriptor and stores the possible sample rates - * on the audioformat table. + * on the audioformat table (audio class v1). * * @dev: usb device * @fp: audioformat record @@ -2508,8 +2545,8 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor * @offset: the start offset of descriptor pointing the rate type * (7 for type I and II, 8 for type II) */ -static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioformat *fp, - unsigned char *fmt, int offset) +static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audioformat *fp, + unsigned char *fmt, int offset) { int nr_rates = fmt[offset]; @@ -2564,14 +2601,86 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform return 0; } +/* + * parse the format descriptor and stores the possible sample rates + * on the audioformat table (audio class v2). + */ +static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, + struct audioformat *fp, + struct usb_host_interface *iface) +{ + struct usb_device *dev = chip->dev; + unsigned char tmp[2], *data; + int i, nr_rates, data_size, ret = 0; + + /* get the number of sample rates first by only fetching 2 bytes */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); + goto err; + } + + nr_rates = (tmp[1] << 8) | tmp[0]; + data_size = 2 + 12 * nr_rates; + data = kzalloc(data_size, GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto err; + } + + /* now get the full information */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + 0x0100, chip->clock_id << 8, data, data_size, 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); + ret = -EINVAL; + goto err_free; + } + + fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + if (!fp->rate_table) { + ret = -ENOMEM; + goto err_free; + } + + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; + + for (i = 0; i < nr_rates; i++) { + int rate = combine_quad(&data[2 + 12 * i]); + + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) + fp->rate_min = rate; + if (!fp->rate_max || rate > fp->rate_max) + fp->rate_max = rate; + fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; + } + +err_free: + kfree(data); +err: + return ret; +} + /* * parse the format type I and III descriptors */ -static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_i(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) { - int pcm_format; - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + int protocol = altsd->bInterfaceProtocol; + int pcm_format, ret; if (fmt->bFormatType == USB_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -2591,30 +2700,49 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * pcm_format = SNDRV_PCM_FORMAT_S16_LE; } } else { - pcm_format = parse_audio_format_i_type(chip, fp, format, fmt); + pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol); if (pcm_format < 0) return -1; } fp->format = pcm_format; - fp->channels = fmt->bNrChannels; + + /* gather possible sample rates */ + /* audio class v1 reports possible sample rates as part of the + * proprietary class specific descriptor. + * audio class v2 uses class specific EP0 range requests for that. + */ + switch (protocol) { + case UAC_VERSION_1: + fp->channels = fmt->bNrChannels; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); + break; + case UAC_VERSION_2: + /* fp->channels is already set in this case */ + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } if (fp->channels < 1) { snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); return -1; } - return parse_audio_format_rates(chip, fp, fmt_raw, 7); + + return ret; } /* * parse the format type II descriptor */ -static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_ii(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) { - int brate, framesize; - struct uac_format_type_ii_discrete_descriptor *fmt = fmt_raw; + int brate, framesize, ret; + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + int protocol = altsd->bInterfaceProtocol; switch (format) { case USB_AUDIO_FORMAT_AC3: @@ -2634,35 +2762,50 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->channels = 1; - brate = le16_to_cpu(fmt->wMaxBitRate); - framesize = le16_to_cpu(fmt->wSamplesPerFrame); - snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); - fp->frame_size = framesize; - return parse_audio_format_rates(chip, fp, fmt_raw, 8); /* fmt[8..] sample rates */ + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 8); /* fmt[8..] sample rates */ + break; + } + case UAC_VERSION_2: { + struct uac_format_type_ii_ext_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } + } + + return ret; } static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw, int stream) + int format, unsigned char *fmt, int stream, + struct usb_host_interface *iface) { int err; - /* we only parse the common header of all format types here, - * so it is safe to take a type_i struct */ - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - switch (fmt->bFormatType) { + switch (fmt[3]) { case USB_FORMAT_TYPE_I: case USB_FORMAT_TYPE_III: - err = parse_audio_format_i(chip, fp, format, fmt); + err = parse_audio_format_i(chip, fp, format, fmt, iface); break; case USB_FORMAT_TYPE_II: - err = parse_audio_format_ii(chip, fp, format, fmt); + err = parse_audio_format_ii(chip, fp, format, fmt, iface); break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt->bFormatType); + chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); return -1; } - fp->fmt_type = fmt->bFormatType; + fp->fmt_type = fmt[3]; if (err < 0) return err; #if 1 @@ -2673,7 +2816,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt->bFormatType == USB_FORMAT_TYPE_I && + if (fmt[3] == USB_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2702,10 +2845,10 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; int i, altno, err, stream; - int format; + int format = 0, num_channels = 0; struct audioformat *fp = NULL; unsigned char *fmt, *csep; - int num; + int num, protocol; dev = chip->dev; @@ -2722,10 +2865,9 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num = 4; for (i = 0; i < num; i++) { - struct uac_as_header_descriptor_v1 *as; - alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); + protocol = altsd->bInterfaceProtocol; /* skip invalid one */ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || @@ -2742,7 +2884,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2003) && @@ -2750,21 +2892,54 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; /* get audio formats */ - as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + switch (protocol) { + case UAC_VERSION_1: { + struct uac_as_header_descriptor_v1 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + break; } - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; + case UAC_VERSION_2: { + struct uac_as_header_descriptor_v2 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + num_channels = as->bNrChannels; + format = le32_to_cpu(as->bmFormats); + + break; } - format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + default: + snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", + dev->devnum, iface_no, altno, protocol); + continue; + } /* get format type */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); @@ -2773,7 +2948,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno); continue; } - if (fmt[0] < 8) { + if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || + ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; @@ -2787,6 +2963,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && fp->format == SNDRV_PCM_FORMAT_S16_LE && + protocol == UAC_VERSION_1 && le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == fp->maxpacksize * 2) continue; @@ -2815,6 +2992,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + /* num_channels is only set for v2 interfaces */ + fp->channels = num_channels; if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); @@ -2850,7 +3029,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } /* ok, let's parse further... */ - if (parse_audio_format(chip, fp, format, fmt, stream) < 0) { + if (parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { kfree(fp->rate_table); kfree(fp); continue; @@ -2958,35 +3137,82 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) { struct usb_device *dev = chip->dev; struct usb_host_interface *host_iface; - struct uac_ac_header_descriptor_v1 *h1; + struct usb_interface_descriptor *altsd; void *control_header; - int i; + int i, protocol; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; control_header = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER); + altsd = get_iface_desc(host_iface); + protocol = altsd->bInterfaceProtocol; if (!control_header) { snd_printk(KERN_ERR "cannot find HEADER\n"); return -EINVAL; } - h1 = control_header; + switch (protocol) { + case UAC_VERSION_1: { + struct uac_ac_header_descriptor_v1 *h1 = control_header; - if (!h1->bInCollection) { - snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); - return -EINVAL; + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); + return -EINVAL; + } + + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + return -EINVAL; + } + + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + + break; } - if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid HEADER (v1)\n"); - return -EINVAL; + case UAC_VERSION_2: { + struct uac_clock_source_descriptor *cs; + struct usb_interface_assoc_descriptor *assoc = + usb_ifnum_to_if(dev, ctrlif)->intf_assoc; + + if (!assoc) { + snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); + return -EINVAL; + } + + /* FIXME: for now, we expect there is at least one clock source + * descriptor and we always take the first one. + * We should properly support devices with multiple clock sources, + * clock selectors and sample rate conversion units. */ + + cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, + NULL, CLOCK_SOURCE); + + if (!cs) { + snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); + return -EINVAL; + } + + chip->clock_id = cs->bClockID; + + for (i = 0; i < assoc->bInterfaceCount; i++) { + int intf = assoc->bFirstInterface + i; + + if (intf != ctrlif) + snd_usb_create_stream(chip, ctrlif, intf); + } + + break; } - for (i = 0; i < h1->bInCollection; i++) - snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + default: + snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); + return -EINVAL; + } return 0; } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4f482939e8e8..26daf68631eb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -142,6 +142,9 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + /* for audio class v2 */ + int clock_id; + struct list_head pcm_list; /* list of pcm streams */ int pcm_devs; -- cgit v1.2.2 From 7b8a043f2686af9f41e313a78ed5e98233e5fded Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:12 +0100 Subject: ALSA: usbmixer: bail out early when parsing audio class v2 descriptors This is just a quick hack that needs to be removed once the new units defined by the audio class v2.0 standard are supported. However, it allows using these devices for now, without mixer support. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ca7949598191..42bb95c739a8 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -2258,7 +2258,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, }; struct usb_mixer_interface *mixer; struct snd_info_entry *entry; - int err; + struct usb_host_interface *host_iface; + int err, protocol; strcpy(chip->card->mixername, "USB Mixer"); @@ -2275,6 +2276,16 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, return -ENOMEM; } + host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; + protocol = host_iface->desc.bInterfaceProtocol; + + /* FIXME! */ + if (protocol != UAC_VERSION_1) { + snd_printk(KERN_WARNING "mixer interface protocol 0x%02x not yet supported\n", + protocol); + return 0; + } + if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) goto _error; -- cgit v1.2.2 From de48c7bc6f93c6c8e0be8612c9d72a2dc92eaa01 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:13 +0100 Subject: ALSA: usbaudio: consolidate header files Use the definitions from linux/usb/audio.h all over the ALSA USB audio driver and add some missing definitions there as well. Use the endpoint attribute macros from linux/usb/ch9 and remove the own things from sound/usb/usbaudio.h. Now things are also nicely prefixed which makes understanding the code easier. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 125 ++++++++++++++++++++++++----------------------- sound/usb/usbaudio.h | 100 ------------------------------------- sound/usb/usbmidi.c | 10 ++-- sound/usb/usbmixer.c | 62 +++++++++++------------ sound/usb/usbquirks.h | 34 ++++++------- sound/usb/usx2y/us122l.c | 6 ++- 6 files changed, 121 insertions(+), 216 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 411a6cf43c21..c539f7fe292f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -47,6 +47,7 @@ #include #include #include +#include #include #include @@ -598,7 +599,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; - if (subs->fmt_type == USB_FORMAT_TYPE_II) { + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ @@ -1106,7 +1107,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri u->packets = (i + 1) * total_packs / subs->nurbs - i * total_packs / subs->nurbs; u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == USB_FORMAT_TYPE_II) + if (subs->fmt_type == UAC_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); if (!u->urb) @@ -1182,7 +1183,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned if (i >= fp->nr_rates) continue; } - attr = fp->ep_attr & EP_ATTR_MASK; + attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; if (! found) { found = fp; cur_attr = attr; @@ -1194,14 +1195,14 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned * M-audio audiophile USB. */ if (attr != cur_attr) { - if ((attr == EP_ATTR_ASYNC && + if ((attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (attr == EP_ATTR_ADAPTIVE && + (attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) continue; - if ((cur_attr == EP_ATTR_ASYNC && + if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (cur_attr == EP_ATTR_ADAPTIVE && + (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) { found = fp; cur_attr = attr; @@ -1231,11 +1232,11 @@ static int init_usb_pitch(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has pitch control, enable it */ - if (fmt->attributes & EP_CS_ATTR_PITCH_CONTROL) { + if (fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL) { data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { + UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", dev->devnum, iface, ep); return err; @@ -1254,21 +1255,21 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has sampling rate control, set it */ - if (fmt->attributes & EP_CS_ATTR_SAMPLE_RATE) { + if (fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE) { int crate; data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ @@ -1386,9 +1387,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) * descriptors which fool us. if it has only one EP, * assume it as adaptive-out or sync-in. */ - attr = fmt->ep_attr & EP_ATTR_MASK; - if (((is_playback && attr == EP_ATTR_ASYNC) || - (! is_playback && attr == EP_ATTR_ADAPTIVE)) && + attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || + (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress @@ -1428,7 +1429,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } /* always fill max packet size */ - if (fmt->attributes & EP_CS_ATTR_FILL_MAX) + if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) subs->fill_max = 1; if ((err = init_usb_pitch(dev, subs->interface, alts, fmt)) < 0) @@ -1886,7 +1887,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.channels_min = fp->channels; if (runtime->hw.channels_max < fp->channels) runtime->hw.channels_max = fp->channels; - if (fp->fmt_type == USB_FORMAT_TYPE_II && fp->frame_size > 0) { + if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) { /* FIXME: there might be more than one audio formats... */ runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = fp->frame_size; @@ -2120,7 +2121,7 @@ static struct usb_device_id usb_audio_ids [] = { #include "usbquirks.h" { .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS), .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { } /* Terminating entry */ }; @@ -2159,7 +2160,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, fp->endpoint & USB_DIR_IN ? "IN" : "OUT", - sync_types[(fp->ep_attr & EP_ATTR_MASK) >> 2]); + sync_types[(fp->ep_attr & USB_ENDPOINT_SYNCTYPE) >> 2]); if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) { snd_iprintf(buffer, " Rates: %d - %d (continuous)\n", fp->rate_min, fp->rate_max); @@ -2471,11 +2472,11 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, pcm_format = -1; switch (format) { - case 0: /* some devices don't define this correctly... */ + case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); /* fall-through */ - case USB_AUDIO_FORMAT_PCM: + case UAC_FORMAT_TYPE_I_PCM: if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, @@ -2509,7 +2510,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, break; } break; - case USB_AUDIO_FORMAT_PCM8: + case UAC_FORMAT_TYPE_I_PCM8: pcm_format = SNDRV_PCM_FORMAT_U8; /* Dallas DS4201 workaround: it advertises U8 format, but really @@ -2517,13 +2518,13 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x04fa, 0x4201)) pcm_format = SNDRV_PCM_FORMAT_S8; break; - case USB_AUDIO_FORMAT_IEEE_FLOAT: + case UAC_FORMAT_TYPE_I_IEEE_FLOAT: pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; break; - case USB_AUDIO_FORMAT_ALAW: + case UAC_FORMAT_TYPE_I_ALAW: pcm_format = SNDRV_PCM_FORMAT_A_LAW; break; - case USB_AUDIO_FORMAT_MU_LAW: + case UAC_FORMAT_TYPE_I_MULAW: pcm_format = SNDRV_PCM_FORMAT_MU_LAW; break; default: @@ -2551,7 +2552,7 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof int nr_rates = fmt[offset]; if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", chip->dev->devnum, fp->iface, fp->altsetting); return -1; } @@ -2614,7 +2615,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, int i, nr_rates, data_size, ret = 0; /* get the number of sample rates first by only fetching 2 bytes */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); @@ -2632,7 +2633,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, } /* now get the full information */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, data, data_size, 1000); @@ -2682,7 +2683,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; int pcm_format, ret; - if (fmt->bFormatType == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2745,12 +2746,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; switch (format) { - case USB_AUDIO_FORMAT_AC3: + case UAC_FORMAT_TYPE_II_AC3: /* FIXME: there is no AC3 format defined yet */ // fp->format = SNDRV_PCM_FORMAT_AC3; fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */ break; - case USB_AUDIO_FORMAT_MPEG: + case UAC_FORMAT_TYPE_II_MPEG: fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: @@ -2793,11 +2794,11 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp int err; switch (fmt[3]) { - case USB_FORMAT_TYPE_I: - case USB_FORMAT_TYPE_III: + case UAC_FORMAT_TYPE_I: + case UAC_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt, iface); break; - case USB_FORMAT_TYPE_II: + case UAC_FORMAT_TYPE_II: err = parse_audio_format_ii(chip, fp, format, fmt, iface); break; default: @@ -2816,7 +2817,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt[3] == UAC_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2871,7 +2872,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* skip invalid one */ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING && + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || altsd->bNumEndpoints < 1 || le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) @@ -2895,16 +2896,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) switch (protocol) { case UAC_VERSION_1: { struct uac_as_header_descriptor_v1 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2915,16 +2916,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) case UAC_VERSION_2: { struct uac_as_header_descriptor_v2 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2942,15 +2943,15 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); if (!fmt) { - snd_printk(KERN_ERR "%d:%u:%d : no FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } @@ -2972,7 +2973,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Creamware Noah has this descriptor after the 2nd endpoint */ if (!csep && altsd->bNumEndpoints >= 2) csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); - if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) { + if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) { snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" " class specific endpoint descriptor\n", dev->devnum, iface_no, altno); @@ -3006,12 +3007,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Optoplay sets the sample rate attribute although * it seems not supporting it in fact. */ - fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE; + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= EP_CS_ATTR_SAMPLE_RATE; + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is @@ -3020,11 +3021,11 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) * plantronics headset and Griffin iMic have set adaptive-in * although it's really not... */ - fp->ep_attr &= ~EP_ATTR_MASK; + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= EP_ATTR_ADAPTIVE; + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; else - fp->ep_attr |= EP_ATTR_SYNC; + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; } @@ -3094,7 +3095,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int altsd = get_iface_desc(alts); if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { int err = snd_usbmidi_create(chip->card, iface, &chip->midi_list, NULL); if (err < 0) { @@ -3109,7 +3110,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING) { snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, interface, altsd->bInterfaceClass); /* skip non-supported classes */ @@ -3145,12 +3146,12 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; control_header = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, HEADER); + NULL, UAC_HEADER); altsd = get_iface_desc(host_iface); protocol = altsd->bInterfaceProtocol; if (!control_header) { - snd_printk(KERN_ERR "cannot find HEADER\n"); + snd_printk(KERN_ERR "cannot find UAC_HEADER\n"); return -EINVAL; } @@ -3164,7 +3165,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + snd_printk(KERN_ERR "invalid UAC_HEADER (v1)\n"); return -EINVAL; } @@ -3190,7 +3191,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) * clock selectors and sample rate conversion units. */ cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, CLOCK_SOURCE); + NULL, UAC_CLOCK_SOURCE); if (!cs) { snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); @@ -3302,7 +3303,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -3394,7 +3395,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, { static const struct audioformat ua1000_format = { .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .attributes = 0, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 26daf68631eb..6b016d4aac6b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -21,106 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ - -/* - */ - -#define USB_SUBCLASS_AUDIO_CONTROL 0x01 -#define USB_SUBCLASS_AUDIO_STREAMING 0x02 -#define USB_SUBCLASS_MIDI_STREAMING 0x03 -#define USB_SUBCLASS_VENDOR_SPEC 0xff - -#define HEADER 0x01 -#define INPUT_TERMINAL 0x02 -#define OUTPUT_TERMINAL 0x03 -#define MIXER_UNIT 0x04 -#define SELECTOR_UNIT 0x05 -#define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT_V1 0x07 -#define EXTENSION_UNIT_V1 0x08 - -/* audio class v2 */ -#define EFFECT_UNIT 0x07 -#define PROCESSING_UNIT_V2 0x08 -#define EXTENSION_UNIT_V2 0x09 -#define CLOCK_SOURCE 0x0a -#define CLOCK_SELECTOR 0x0b -#define CLOCK_MULTIPLIER 0x0c -#define SAMPLE_RATE_CONVERTER 0x0d - -#define AS_GENERAL 0x01 -#define FORMAT_TYPE 0x02 -#define FORMAT_SPECIFIC 0x03 - -#define EP_GENERAL 0x01 - -#define MS_GENERAL 0x01 -#define MIDI_IN_JACK 0x02 -#define MIDI_OUT_JACK 0x03 - -/* endpoint attributes */ -#define EP_ATTR_MASK 0x0c -#define EP_ATTR_ASYNC 0x04 -#define EP_ATTR_ADAPTIVE 0x08 -#define EP_ATTR_SYNC 0x0c - -/* cs endpoint attributes */ -#define EP_CS_ATTR_SAMPLE_RATE 0x01 -#define EP_CS_ATTR_PITCH_CONTROL 0x02 -#define EP_CS_ATTR_FILL_MAX 0x80 - -/* Audio Class specific Request Codes (v1) */ - -#define SET_CUR 0x01 -#define GET_CUR 0x81 -#define SET_MIN 0x02 -#define GET_MIN 0x82 -#define SET_MAX 0x03 -#define GET_MAX 0x83 -#define SET_RES 0x04 -#define GET_RES 0x84 -#define SET_MEM 0x05 -#define GET_MEM 0x85 -#define GET_STAT 0xff - -/* Audio Class specific Request Codes (v2) */ -#define CS_CUR 0x01 -#define CS_RANGE 0x02 - -/* Terminal Control Selectors */ - -#define COPY_PROTECT_CONTROL 0x01 - -/* Endpoint Control Selectors */ - -#define SAMPLING_FREQ_CONTROL 0x01 -#define PITCH_CONTROL 0x02 - -/* Format Types */ -#define USB_FORMAT_TYPE_I 0x01 -#define USB_FORMAT_TYPE_II 0x02 -#define USB_FORMAT_TYPE_III 0x03 - -/* type I */ -#define USB_AUDIO_FORMAT_PCM 0x01 -#define USB_AUDIO_FORMAT_PCM8 0x02 -#define USB_AUDIO_FORMAT_IEEE_FLOAT 0x03 -#define USB_AUDIO_FORMAT_ALAW 0x04 -#define USB_AUDIO_FORMAT_MU_LAW 0x05 - -/* type II */ -#define USB_AUDIO_FORMAT_MPEG 0x1001 -#define USB_AUDIO_FORMAT_AC3 0x1002 - -/* type III */ -#define USB_AUDIO_FORMAT_IEC1937_AC3 0x2001 -#define USB_AUDIO_FORMAT_IEC1937_MPEG1_LAYER1 0x2002 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_NOEXT 0x2003 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_EXT 0x2004 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER1_LS 0x2005 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER23_LS 0x2006 - - /* maximum number of endpoints per interface */ #define MIDI_MAX_ENDPOINTS 2 diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index b2da478a0fae..2c59afd99611 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -1540,7 +1542,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostif->extralen >= 7 && ms_header->bLength >= 7 && ms_header->bDescriptorType == USB_DT_CS_INTERFACE && - ms_header->bDescriptorSubtype == HEADER) + ms_header->bDescriptorSubtype == UAC_HEADER) snd_printdd(KERN_INFO "MIDIStreaming version %02x.%02x\n", ms_header->bcdMSC[1], ms_header->bcdMSC[0]); else @@ -1556,7 +1558,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostep->extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != MS_GENERAL) + ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { @@ -1768,9 +1770,9 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2; cs_desc += cs_desc[0]) { if (cs_desc[1] == USB_DT_CS_INTERFACE) { - if (cs_desc[2] == MIDI_IN_JACK) + if (cs_desc[2] == UAC_MIDI_IN_JACK) endpoint->in_cables = (endpoint->in_cables << 1) | 1; - else if (cs_desc[2] == MIDI_OUT_JACK) + else if (cs_desc[2] == UAC_MIDI_OUT_JACK) endpoint->out_cables = (endpoint->out_cables << 1) | 1; } } diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 42bb95c739a8..8e8f871b74ca 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) + if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC_EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -407,14 +407,14 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value) { - return get_ctl_value(cval, GET_CUR, validx, value); + return get_ctl_value(cval, UAC_GET_CUR, validx, value); } /* channel = 0: master, 1 = first channel */ static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, int channel, int *value) { - return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value); + return get_ctl_value(cval, UAC_GET_CUR, (cval->control << 8) | channel, value); } static int get_cur_mix_value(struct usb_mixer_elem_info *cval, @@ -468,14 +468,14 @@ static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) { - return set_ctl_value(cval, SET_CUR, validx, value); + return set_ctl_value(cval, UAC_SET_CUR, validx, value); } static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; - err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, + err = set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, value); if (err < 0) return err; @@ -605,13 +605,13 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm if (term_only) return 0; switch (iterm->type >> 16) { - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; - case MIXER_UNIT: + case UAC_MIXER_UNIT: strcpy(name, "Mixer"); return 5; default: return sprintf(name, "Unit %d", iterm->id); @@ -650,22 +650,22 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ while ((p1 = find_audio_control_unit(state, id)) != NULL) { term->id = id; switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: term->type = combine_word(p1 + 4); term->channels = p1[7]; term->chconfig = combine_word(p1 + 8); term->name = p1[11]; return 0; - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: id = p1[4]; break; /* continue to parse */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: term->type = p1[2] << 16; /* virtual type */ term->channels = p1[5 + p1[4]]; term->chconfig = combine_word(p1 + 6 + p1[4]); term->name = p1[p1[0] - 1]; return 0; - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: /* call recursively to retrieve the channel info */ if (check_input_term(state, p1[5], term) < 0) return -ENODEV; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT_V1: - case EXTENSION_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -752,23 +752,23 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) break; } } - if (get_ctl_value(cval, GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || - get_ctl_value(cval, GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { + if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || + get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n", cval->id, cval->mixer->ctrlif, cval->control, cval->id); return -EINVAL; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { cval->res = 1; } else { int last_valid_res = cval->res; while (cval->res > 1) { - if (set_ctl_value(cval, SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) + if (set_ctl_value(cval, UAC_SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) break; cval->res /= 2; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) cval->res = last_valid_res; } if (cval->res == 0) @@ -1097,7 +1097,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void struct uac_feature_unit_descriptor *ftr = _ftr; if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { - snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); + snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } @@ -1739,17 +1739,17 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) } switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: return 0; /* NOP */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: return parse_audio_selector_unit(state, unitid, p1); - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); @@ -1779,7 +1779,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) /* * create mixer controls * - * walk through all OUTPUT_TERMINAL descriptors to search for mixers + * walk through all UAC_OUTPUT_TERMINAL descriptors to search for mixers */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { @@ -1807,7 +1807,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } desc = NULL; - while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { + while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, UAC_OUTPUT_TERMINAL)) != NULL) { if (desc->bLength < 9) continue; /* invalid descriptor? */ set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ @@ -2047,7 +2047,7 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) } mixer->rc_setup_packet->bRequestType = USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE; - mixer->rc_setup_packet->bRequest = GET_MEM; + mixer->rc_setup_packet->bRequest = UAC_GET_MEM; mixer->rc_setup_packet->wValue = cpu_to_le16(0); mixer->rc_setup_packet->wIndex = cpu_to_le16(0); mixer->rc_setup_packet->wLength = cpu_to_le16(len); @@ -2170,7 +2170,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s: ", jacks[i].name); err = snd_usb_ctl_msg(mixer->chip->dev, usb_rcvctrlpipe(mixer->chip->dev, 0), - GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | + UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, jacks[i].unitid << 8, buf, 3, 100); if (err == 3 && (buf[0] == 3 || buf[0] == 6)) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index fc1d2cd6ccc3..f06faf7917b9 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -91,7 +91,7 @@ .idVendor = 0x046d, .idProduct = 0x0850, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -100,7 +100,7 @@ .idVendor = 0x046d, .idProduct = 0x08ae, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -109,7 +109,7 @@ .idVendor = 0x046d, .idProduct = 0x08c6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -118,7 +118,7 @@ .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -127,7 +127,7 @@ .idVendor = 0x046d, .idProduct = 0x08f5, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -136,7 +136,7 @@ .idVendor = 0x046d, .idProduct = 0x08f6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { USB_DEVICE(0x046d, 0x0990), @@ -301,7 +301,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_FILL_MAX, + .attributes = UAC_EP_CS_ATTR_FILL_MAX, .endpoint = 0x81, .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -2108,7 +2108,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2122,7 +2122,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2136,7 +2136,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2150,7 +2150,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2164,7 +2164,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2178,7 +2178,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2192,7 +2192,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2206,7 +2206,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-850", @@ -2238,7 +2238,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_SAMPLE_RATE, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x02, .ep_attr = 0x01, .maxpacksize = 0x130, @@ -2268,7 +2268,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .match_flags = USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_MIDI_STREAMING, + .bInterfaceSubClass = USB_SUBCLASS_MIDISTREAMING, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_MIDI_STANDARD_INTERFACE diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb29666d26..44deb21b1777 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -16,6 +16,8 @@ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#include +#include #include #include #include @@ -315,9 +317,9 @@ static int us122l_set_sample_rate(struct usb_device *dev, int rate) data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000); if (err < 0) snd_printk(KERN_ERR "%d: cannot set freq %d to ep 0x%x\n", dev->devnum, rate, ep); -- cgit v1.2.2 From 76e6f5a9efc919f9179163c66403451a789d47ab Mon Sep 17 00:00:00 2001 From: Reimundo Heluani Date: Tue, 23 Feb 2010 01:19:51 -0800 Subject: ALSA: add support for Macbook Air 2,1 internal speaker Add support for Macbook Air 2,1 (late 2008) internal speaker and headphones. Create a "mba21" model for snd-hda-intel. Signed-off-by: Reimundo Heluani Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 64 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 64 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c74ca39a0b8e..5382872eba1f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -209,6 +209,7 @@ enum { ALC882_ASUS_A7J, ALC882_ASUS_A7M, ALC885_MACPRO, + ALC885_MBA21, ALC885_MBP3, ALC885_MB5, ALC885_MACMINI3, @@ -6948,6 +6949,13 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = { { 8, alc882_sixstack_ch8_init }, }; + +/* Macbook Air 2,1 */ + +static struct hda_channel_mode alc885_mba21_ch_modes[1] = { + { 2, NULL }, +}; + /* * macbook pro ALC885 can switch LineIn to LineOut without losing Mic */ @@ -7220,6 +7228,15 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { { } /* end */ }; +/* Macbook Air 2,1 same control for HP and internal Speaker */ + +static struct snd_kcontrol_new alc885_mba21_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), + { } +}; + + static struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7689,6 +7706,29 @@ static struct hda_verb alc885_macmini3_init_verbs[] = { { } }; + +static struct hda_verb alc885_mba21_init_verbs[] = { + /*Internal and HP Speaker Mixer*/ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /*Internal Speaker Pin (0x0c)*/ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: output 0 (0x0e) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)}, + /* Line in (is hp when jack connected)*/ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + { } + }; + + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7854,6 +7894,17 @@ static void alc885_imac24_setup(struct hda_codec *codec) #define alc885_mb5_setup alc885_imac24_setup #define alc885_macmini3_setup alc885_imac24_setup +/* Macbook Air 2,1 */ +static void alc885_mba21_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; +} + + + static void alc885_mbp3_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9017,6 +9068,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", [ALC885_MACMINI3] = "macmini3", + [ALC885_MBA21] = "mba21", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9252,6 +9304,18 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, }, + [ALC885_MBA21] = { + .mixers = { alc885_mba21_mixer }, + .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, + .num_dacs = 2, + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc882_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_mba21_setup, + .init_hook = alc_automute_amp, + }, [ALC885_MBP3] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs, -- cgit v1.2.2 From e17dd32f342d0e876f729b348614320b297cf6f3 Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Mon, 22 Feb 2010 15:09:19 -0600 Subject: ASoC: OMAP: data_type and sync_mode configurable in audio dma Allow client drivers to set the data_type (16, 32) and the sync_mode (element, packet, etc) of the audio dma transferences. McBSP dai driver configures it for a data type of 16 bits and element sync mode. Signed-off-by: Misael Lopez Cruz Signed-off-by: Jorge Eduardo Candelaria Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 2 ++ sound/soc/omap/omap-pcm.c | 15 ++++++++------- sound/soc/omap/omap-pcm.h | 4 +++- 3 files changed, 13 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6bbbd2ab0ee7..d29725664185 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -287,6 +287,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; if (mcbsp_data->configured) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9db2770e9640..825db385f01f 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, @@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_runtime_data *prtd = runtime->private_data; struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + int bytes; /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) return 0; memset(&dma_params, 0, sizeof(dma_params)); - /* - * Note: Regardless of interface data formats supported by OMAP McBSP - * or EAC blocks, internal representation is always fixed 16-bit/sample - */ - dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; + dma_params.data_type = dma_data->data_type; dma_params.trigger = dma_data->dma_req; dma_params.sync_mode = dma_data->sync_mode; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; dma_params.dst_port = OMAP_DMA_PORT_MPUI; + dma_params.dst_fi = dma_data->packet_size; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; @@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; dma_params.src_port = OMAP_DMA_PORT_MPUI; + dma_params.src_fi = dma_data->packet_size; } /* * Set DMA transfer frame size equal to ALSA period size and frame @@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) * we can transfer the whole ALSA buffer with single DMA transfer but * still can get an interrupt at each period bounary */ - dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2; + bytes = snd_pcm_lib_period_bytes(substream); + dma_params.elem_count = bytes >> dma_data->data_type; dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index 38a821dd4118..b19975d26907 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -29,8 +29,10 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ - int sync_mode; /* DMA sync mode */ void (*set_threshold)(struct snd_pcm_substream *substream); + int data_type; /* data type 8,16,32 */ + int sync_mode; /* DMA sync mode */ + int packet_size; /* packet size only in PACKET mode */ }; extern struct snd_soc_platform omap_soc_platform; -- cgit v1.2.2 From b3b0b4580bcb771d1d53b3d5acf689cba9907392 Mon Sep 17 00:00:00 2001 From: "Candelaria Villareal, Jorge" Date: Mon, 22 Feb 2010 17:17:21 -0600 Subject: ASoC: OMAP4: Add support for McPDM McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria Signed-off-by: Margarita Olaya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/mcpdm.c | 484 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 +++++++++++++++ 2 files changed, 635 insertions(+) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h (limited to 'sound') diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c new file mode 100644 index 000000000000..ad8df6cfae88 --- /dev/null +++ b/sound/soc/omap/mcpdm.c @@ -0,0 +1,484 @@ +/* + * mcpdm.c -- McPDM interface driver + * + * Author: Jorge Eduardo Candelaria + * Copyright (C) 2009 - Texas Instruments, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "mcpdm.h" + +static struct omap_mcpdm *mcpdm; + +static inline void omap_mcpdm_write(u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static inline int omap_mcpdm_read(u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +static void omap_mcpdm_reg_dump(void) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); +} + +/* + * Takes the McPDM module in and out of reset state. + * Uplink and downlink can be reset individually. + */ +static void omap_mcpdm_reset_capture(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +static void omap_mcpdm_reset_playback(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +void omap_mcpdm_start(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +void omap_mcpdm_stop(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Configures McPDM uplink for audio recording. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + int ctrl; + + if (!uplink) + return -EINVAL; + + mcpdm->uplink = uplink; + + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; + + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Configures McPDM downlink for audio playback. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + int ctrl; + + if (!downlink) + return -EINVAL; + + mcpdm->downlink = downlink; + + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; + + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Cleans McPDM uplink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + + if (!uplink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + + /* Clear Downlink channels */ + mcpdm->up_channels = 0; + + mcpdm->uplink = NULL; + + return 0; +} + +/* + * Cleans McPDM downlink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + + if (!downlink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + + /* clear Downlink channels */ + mcpdm->dn_channels = 0; + + mcpdm->downlink = NULL; + + return 0; +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; +} + +int omap_mcpdm_request(void) +{ + int ret; + + clk_enable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + ret = -EBUSY; + goto err; + } + mcpdm->free = 0; + + spin_unlock(&mcpdm->lock); + + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + goto err; + } + + return 0; + +err: + clk_disable(mcpdm->clk); + return ret; +} + +void omap_mcpdm_free(void) +{ + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + clk_disable(mcpdm->clk); + + free_irq(mcpdm->irq, (void *)mcpdm); +} + +/* Enable/disable DC offset cancelation for the analog + * headset path (PDM channels 1 and 2). + */ +int omap_mcpdm_set_offset(int offset1, int offset2) +{ + int offset; + + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; + + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + + /* offset cancellation for channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; + + /* offset cancellation for channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; + + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + + return 0; +} + +static int __devinit omap_mcpdm_probe(struct platform_device *pdev) +{ + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_resource; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_resource; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + + mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); + goto err_clk; + } + + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; + +err_clk: + iounmap(mcpdm->io_base); +err_resource: + kfree(mcpdm); +exit: + return ret; +} + +static int __devexit omap_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + + clk_put(mcpdm_ptr->clk); + + iounmap(mcpdm_ptr->io_base); + + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; + + kfree(mcpdm_ptr); + + return 0; +} + +static struct platform_driver omap_mcpdm_driver = { + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, +}; + +static struct platform_device *omap_mcpdm_device; + +static int __init omap_mcpdm_init(void) +{ + return platform_driver_register(&omap_mcpdm_driver); +} +arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h new file mode 100644 index 000000000000..7bb326ef0886 --- /dev/null +++ b/sound/soc/omap/mcpdm.h @@ -0,0 +1,151 @@ +/* + * mcpdm.h -- Defines for McPDM driver + * + * Author: Jorge Eduardo Candelaria + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +/* McPDM registers */ + +#define MCPDM_REVISION 0x00 +#define MCPDM_SYSCONFIG 0x10 +#define MCPDM_IRQSTATUS_RAW 0x24 +#define MCPDM_IRQSTATUS 0x28 +#define MCPDM_IRQENABLE_SET 0x2C +#define MCPDM_IRQENABLE_CLR 0x30 +#define MCPDM_IRQWAKE_EN 0x34 +#define MCPDM_DMAENABLE_SET 0x38 +#define MCPDM_DMAENABLE_CLR 0x3C +#define MCPDM_DMAWAKEEN 0x40 +#define MCPDM_CTRL 0x44 +#define MCPDM_DN_DATA 0x48 +#define MCPDM_UP_DATA 0x4C +#define MCPDM_FIFO_CTRL_DN 0x50 +#define MCPDM_FIFO_CTRL_UP 0x54 +#define MCPDM_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define DMA_DN_ENABLE 0x1 +#define DMA_UP_ENABLE 0x2 + +/* + * MCPDM_CTRL bit fields + */ + +#define PDM_UP1_EN 0x0001 +#define PDM_UP2_EN 0x0002 +#define PDM_UP3_EN 0x0004 +#define PDM_DN1_EN 0x0008 +#define PDM_DN2_EN 0x0010 +#define PDM_DN3_EN 0x0020 +#define PDM_DN4_EN 0x0040 +#define PDM_DN5_EN 0x0080 +#define PDMOUTFORMAT 0x0100 +#define CMD_INT 0x0200 +#define STATUS_INT 0x0400 +#define SW_UP_RST 0x0800 +#define SW_DN_RST 0x1000 +#define PDM_UP_MASK 0x007 +#define PDM_DN_MASK 0x0F8 +#define PDM_CMD_MASK 0x200 +#define PDM_STATUS_MASK 0x400 + + +#define PDMOUTFORMAT_LJUST (0 << 8) +#define PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define UP_THRES_MAX 0xF +#define DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define DN_OFST_RX1_EN 0x0001 +#define DN_OFST_RX2_EN 0x0100 + +#define DN_OFST_RX1 1 +#define DN_OFST_RX2 9 +#define DN_OFST_MAX 0x1F + +#define MCPDM_UPLINK 1 +#define MCPDM_DOWNLINK 2 + +struct omap_mcpdm_link { + int irq_mask; + int threshold; + int format; + int channels; +}; + +struct omap_mcpdm_platform_data { + unsigned long phys_base; + u16 irq; +}; + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + u8 free; + int irq; + + spinlock_t lock; + struct omap_mcpdm_platform_data *pdata; + struct clk *clk; + struct omap_mcpdm_link *downlink; + struct omap_mcpdm_link *uplink; + struct completion irq_completion; + + int dn_channels; + int up_channels; +}; + +extern void omap_mcpdm_start(int stream); +extern void omap_mcpdm_stop(int stream); +extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_request(void); +extern void omap_mcpdm_free(void); +extern int omap_mcpdm_set_offset(int offset1, int offset2); -- cgit v1.2.2 From db72c2f89790f919d65d0adbee390958005c40fc Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Mon, 22 Feb 2010 15:09:22 -0600 Subject: ASoC: OMAP4: Add McPDM platform driver McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz Signed-off-by: Margarita Olaya Signed-off-by: Jorge Eduardo Candelaria Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-mcpdm.c | 251 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 4 files changed, 285 insertions(+) create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 18ebdc7d0a51..f11963c21873 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP +config SND_OMAP_SOC_MCPDM + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 19283e5edfbf..0bc00ca14b37 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,9 +1,11 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 000000000000..25f19e4728bf --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,251 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz + * Contact: Jorge Eduardo Candelaria + * Margarita Olaya + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio playback", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio capture", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_playback_open(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_capture_open(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int err; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + err = omap_mcpdm_playback_close(&mcpdm_links[stream]); + else + err = omap_mcpdm_capture_close(&mcpdm_links[stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 000000000000..73b80d559345 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- cgit v1.2.2 From 47fc9a0a808f23b7b305f6c018e4882118b88d92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 22 Feb 2010 16:41:57 +0900 Subject: ASoC: fsi: Modify over/under run error settlement In current FSI driver, playback function cares only overrun, and capture function cares only underrun. But playback function should had cared about underrun, and capture function should had cared about overrun too. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 46 +++++++++++++++++++++++++--------------------- 1 file changed, 25 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c36d24a6c20..993abb730dfa 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -388,7 +388,7 @@ static void fsi_soft_all_reset(struct fsi_master *master) } /* playback interrupt */ -static int fsi_data_push(struct fsi_priv *fsi) +static int fsi_data_push(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -397,7 +397,7 @@ static int fsi_data_push(struct fsi_priv *fsi) int fifo_free; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -453,24 +453,26 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi->byte_offset += send * width; - ret = 0; status = fsi_reg_read(fsi, DOFF_ST); - if (status & ERR_OVER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "over run error\n"); - fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DOFF_ST, 0); fsi_irq_enable(fsi, 1); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } -static int fsi_data_pop(struct fsi_priv *fsi) +static int fsi_data_pop(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -479,7 +481,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) int fifo_fill; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -534,21 +536,23 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi->byte_offset += fifo_fill * width; - ret = 0; status = fsi_reg_read(fsi, DIFF_ST); - if (status & ERR_UNDER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "under run error\n"); - fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DIFF_ST, 0); fsi_irq_enable(fsi, 0); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } static irqreturn_t fsi_interrupt(int irq, void *data) @@ -562,13 +566,13 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_master_write(master, SOFT_RST, status | 0x00000010); if (int_st & INT_A_OUT) - fsi_data_push(&master->fsia); + fsi_data_push(&master->fsia, 0); if (int_st & INT_B_OUT) - fsi_data_push(&master->fsib); + fsi_data_push(&master->fsib, 0); if (int_st & INT_A_IN) - fsi_data_pop(&master->fsia); + fsi_data_pop(&master->fsia, 0); if (int_st & INT_B_IN) - fsi_data_pop(&master->fsib); + fsi_data_pop(&master->fsib, 0); fsi_master_write(master, INT_ST, 0x0000000); @@ -726,7 +730,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); + ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); -- cgit v1.2.2 From 83905c134571642d7e8a1e51ae9f26bd3a3ad82a Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Mon, 22 Feb 2010 12:21:12 +0000 Subject: ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone Add ASoC interface for OMAP McBSP2 and McBSP3 sidetones. Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Acked-by: Mark Brown Tested-by: Jarkko Nikula Signed-off-by: Tony Lindgren --- sound/soc/omap/omap-mcbsp.c | 138 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 2 files changed, 140 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index c0039b35fb25..8da14f537f49 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,6 +39,14 @@ #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = omap_mcbsp_st_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long) &(struct soc_mixer_control) \ + {.min = xmin, .max = xmax} } + struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; @@ -637,6 +645,136 @@ struct snd_soc_dai omap_mcbsp_dai[] = { EXPORT_SYMBOL_GPL(omap_mcbsp_dai); +int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = min; + uinfo->value.integer.max = max; + return 0; +} + +#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(id, channel) \ +static int \ +omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct soc_mixer_control *mc = \ + (struct soc_mixer_control *)kc->private_value; \ + int max = mc->max; \ + int min = mc->min; \ + int val = uc->value.integer.value[0]; \ + \ + if (val < min || val > max) \ + return -EINVAL; \ + \ + /* OMAP McBSP implementation uses index values 0..4 */ \ + return omap_st_set_chgain((id)-1, channel, val); \ +} + +#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(id, channel) \ +static int \ +omap_mcbsp##id##_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + s16 chgain; \ + \ + if (omap_st_get_chgain((id)-1, channel, &chgain)) \ + return -EAGAIN; \ + \ + uc->value.integer.value[0] = chgain; \ + return 0; \ +} + +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 0) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 1) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 0) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 1) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 0) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 1) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 0) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 1) + +static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 value = ucontrol->value.integer.value[0]; + + if (value == omap_st_is_enabled(mc->reg)) + return 0; + + if (value) + omap_st_enable(mc->reg); + else + omap_st_disable(mc->reg); + + return 1; +} + +static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + ucontrol->value.integer.value[0] = omap_st_is_enabled(mc->reg); + return 0; +} + +static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = { + SOC_SINGLE_EXT("McBSP2 Sidetone Switch", 1, 0, 1, 0, + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume", + -32768, 32767, + omap_mcbsp2_get_st_ch0_volume, + omap_mcbsp2_set_st_ch0_volume), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume", + -32768, 32767, + omap_mcbsp2_get_st_ch1_volume, + omap_mcbsp2_set_st_ch1_volume), +}; + +static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { + SOC_SINGLE_EXT("McBSP3 Sidetone Switch", 2, 0, 1, 0, + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume", + -32768, 32767, + omap_mcbsp3_get_st_ch0_volume, + omap_mcbsp3_set_st_ch0_volume), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume", + -32768, 32767, + omap_mcbsp3_get_st_ch1_volume, + omap_mcbsp3_set_st_ch1_volume), +}; + +int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id) +{ + if (!cpu_is_omap34xx()) + return -ENODEV; + + switch (mcbsp_id) { + case 1: /* McBSP 2 */ + return snd_soc_add_controls(codec, omap_mcbsp2_st_controls, + ARRAY_SIZE(omap_mcbsp2_st_controls)); + case 2: /* McBSP 3 */ + return snd_soc_add_controls(codec, omap_mcbsp3_st_controls, + ARRAY_SIZE(omap_mcbsp3_st_controls)); + default: + break; + } + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); + static int __init snd_omap_mcbsp_init(void) { return snd_soc_register_dais(omap_mcbsp_dai, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 1968d03bc532..6c363e5f4387 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -57,4 +57,6 @@ enum omap_mcbsp_div { extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; +int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id); + #endif -- cgit v1.2.2 From dd2b4a7abf82d88261f8f98e1361388a7db2ffe4 Mon Sep 17 00:00:00 2001 From: "Zhang, Rui" Date: Wed, 24 Feb 2010 09:38:49 +0800 Subject: ALSA: hda - remove unnecessary msleep on power state transitions This will save ~15ms boot time. The first 10ms sleep was introduced in commit d2595d86e5 for (buggy) Cxt codecs, so better to limit the sleep to the problem hardware. For the second 10ms sleep, the HDA spec says: Power State[1:0]: 00: Node Power state (D0) is fully on. 01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog playback) which must remain fully on. 10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state. 11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software control. Note that any low power state set by software must retain sufficient operational capability to properly respond to subsequent software Power State command. So 10ms is actually the max wait time. It should be safe to remove/reduce it and rely on the loop of 1ms-sleeps. CC: Marc Boucher CC: Arjan van de Ven Signed-off-by: Zhang Rui Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 98767df4f03a..76d3c4c049db 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2767,7 +2767,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); /* partial workaround for "azx_get_response timeout" */ - if (power_state == AC_PWRST_D0) + if (power_state == AC_PWRST_D0 && + (codec->vendor_id & 0xffff0000) == 0x14f10000) msleep(10); nid = codec->start_nid; @@ -2801,7 +2802,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (power_state == AC_PWRST_D0) { unsigned long end_time; int state; - msleep(10); /* wait until the codec reachs to D0 */ end_time = jiffies + msecs_to_jiffies(500); do { -- cgit v1.2.2 From 6227cdced0328b0c4322c3170a727af5249393ce Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 25 Feb 2010 08:36:52 +0100 Subject: ALSA: hda - Add ALC670 codec support - Fixed alc_subsystem_id( ) typo and add new function. - !(ass & 0x100000)) ==> Delete this check. It is unnecessary check. - Add porti - ALC670 support Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 40 ++++++++++++++++++++++++---------------- 1 file changed, 24 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5382872eba1f..220a49ff2179 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1254,7 +1254,7 @@ static void alc_init_auto_mic(struct hda_codec *codec) */ static int alc_subsystem_id(struct hda_codec *codec, hda_nid_t porta, hda_nid_t porte, - hda_nid_t portd) + hda_nid_t portd, hda_nid_t porti) { unsigned int ass, tmp, i; unsigned nid; @@ -1280,7 +1280,7 @@ static int alc_subsystem_id(struct hda_codec *codec, snd_printd("realtek: No valid SSID, " "checking pincfg 0x%08x for NID 0x%x\n", ass, nid); - if (!(ass & 1) && !(ass & 0x100000)) + if (!(ass & 1)) return 0; if ((ass >> 30) != 1) /* no physical connection */ return 0; @@ -1340,6 +1340,8 @@ do_sku: nid = porte; else if (tmp == 2) nid = portd; + else if (tmp == 3) + nid = porti; else return 1; for (i = 0; i < spec->autocfg.line_outs; i++) @@ -1354,9 +1356,10 @@ do_sku: } static void alc_ssid_check(struct hda_codec *codec, - hda_nid_t porta, hda_nid_t porte, hda_nid_t portd) + hda_nid_t porta, hda_nid_t porte, + hda_nid_t portd, hda_nid_t porti) { - if (!alc_subsystem_id(codec, porta, porte, portd)) { + if (!alc_subsystem_id(codec, porta, porte, portd, porti)) { struct alc_spec *spec = codec->spec; snd_printd("realtek: " "Enable default setup for auto mode as fallback\n"); @@ -4859,7 +4862,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -6393,7 +6396,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - alc_ssid_check(codec, 0x10, 0x15, 0x0f); + alc_ssid_check(codec, 0x10, 0x15, 0x0f, 0); return 1; } @@ -10224,7 +10227,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -11782,7 +11785,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x14, 0x1b); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -12733,7 +12736,6 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: - case 0x21: dac = 0x03; break; default: @@ -12954,7 +12956,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -13845,11 +13847,11 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { add_verb(spec, alc269vb_init_verbs); real_capsrc_nids = alc269vb_capsrc_nids[0]; - alc_ssid_check(codec, 0x21, 0x1b, 0x14); + alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21); } else { add_verb(spec, alc269_init_verbs); real_capsrc_nids = alc269_capsrc_nids[0]; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); } spec->num_mux_defs = 1; @@ -15013,7 +15015,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(codec); - alc_ssid_check(codec, 0x0e, 0x0f, 0x0b); + alc_ssid_check(codec, 0x0e, 0x0f, 0x0b, 0); return 1; } @@ -15904,7 +15906,7 @@ static struct alc_config_preset alc861vd_presets[] = { static int alc861vd_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0); + return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0); } @@ -16140,7 +16142,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -17627,6 +17629,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), @@ -18257,7 +18260,11 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0x21); + else + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -18407,6 +18414,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, + { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, -- cgit v1.2.2 From 61c2d2b5e7241d4410ab8227ef4f76c1aba8210b Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 25 Feb 2010 08:49:06 +0100 Subject: ALSA: hda - Add/fix ALC269 FSC and Quanta models Specify proper quirk models for FSC and Quanta machines with ALC269 codec. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 220a49ff2179..e8cbe216e912 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13946,8 +13946,14 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), + SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), + SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), + SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), + SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC), + SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC), + SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC), {} }; -- cgit v1.2.2 From 9e4a10d27e89f780539e08abd2b051cb83635dfa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 25 Feb 2010 12:52:09 +0000 Subject: ASoC: Remove a unused variables from i.MX FIQ runtime data Signed-off-by: Mark Brown Acked-by: Sascha Hauer Acked-by: Liam Girdwood --- sound/soc/imx/imx-pcm-fiq.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 5532579ece4d..a1c4ce6ad408 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -35,12 +35,8 @@ struct imx_pcm_runtime_data { int period; int periods; - unsigned long dma_addr; - int dma; unsigned long offset; unsigned long size; - unsigned long period_cnt; - void *buf; struct timer_list timer; int period_time; }; -- cgit v1.2.2 From b4e82b5b785670b68136765059d1afc65c0ae023 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 25 Feb 2010 12:52:10 +0000 Subject: ASoC: Check progress when reporting periods from i.MX FIQ handler Currently the i.MX FIQ handler is reporting periods as elapsed based purely on a timer running in the CPU. This means that any clock mismatch between the CPU and the audio subsystem can result in the status reported to applications drifting away from the actual status of the hardware. This is particularly likely at present since the SSI driver is only capable of operating in slave mode so it's very likely that the interface will be clocked from a different source. Instead check the offset reported by the FIQ and only notify when we have transferred at least one period, re-firing the timer if we didn't do so. Also factor out the calculation of the timer expiry time for make it a bit easier to experiment with. Note that this only improves the situation, problems can still be triggered. Signed-off-by: Mark Brown Acked-by: Sascha Hauer Acked-by: Liam Girdwood --- sound/soc/imx/imx-pcm-fiq.c | 36 ++++++++++++++++++++++++++++++------ 1 file changed, 30 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index a1c4ce6ad408..d9cb9849b033 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -36,17 +36,24 @@ struct imx_pcm_runtime_data { int period; int periods; unsigned long offset; + unsigned long last_offset; unsigned long size; struct timer_list timer; - int period_time; + int poll_time; }; +static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd) +{ + iprtd->timer.expires = jiffies + iprtd->poll_time; +} + static void imx_ssi_timer_callback(unsigned long data) { struct snd_pcm_substream *substream = (void *)data; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; struct pt_regs regs; + unsigned long delta; get_fiq_regs(®s); @@ -55,9 +62,25 @@ static void imx_ssi_timer_callback(unsigned long data) else iprtd->offset = regs.ARM_r9 & 0xffff; - iprtd->timer.expires = jiffies + iprtd->period_time; + /* How much data have we transferred since the last period report? */ + if (iprtd->offset >= iprtd->last_offset) + delta = iprtd->offset - iprtd->last_offset; + else + delta = runtime->buffer_size + iprtd->offset + - iprtd->last_offset; + + /* If we've transferred at least a period then report it and + * reset our poll time */ + if (delta >= runtime->period_size) { + snd_pcm_period_elapsed(substream); + iprtd->last_offset = iprtd->offset; + + imx_ssi_set_next_poll(iprtd); + } + + /* Restart the timer; if we didn't report we'll run on the next tick */ add_timer(&iprtd->timer); - snd_pcm_period_elapsed(substream); + } static struct fiq_handler fh = { @@ -72,9 +95,10 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params); + iprtd->period = params_period_bytes(params) ; iprtd->offset = 0; - iprtd->period_time = HZ / (params_rate(params) / params_period_size(params)); + iprtd->last_offset = 0; + iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params)); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); @@ -110,7 +134,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - iprtd->timer.expires = jiffies + iprtd->period_time; + imx_ssi_set_next_poll(iprtd); add_timer(&iprtd->timer); if (++fiq_enable == 1) enable_fiq(imx_pcm_fiq); -- cgit v1.2.2 From ea071cc705e8bfba0c8bf84be8d4f9f4e9da6962 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Tue, 13 Oct 2009 20:22:34 +0200 Subject: MIPS: Alchemy: remove dbdma compat macros Remove dbdma compat macros, move remaining users over to default queueing functions and -flags. (Queueing function signature has changed in order to give a build failure instead of silent functional changes due to the no longer implicitly specified DDMA_FLAGS_IE flag) Signed-off-by: Manuel Lauss Signed-off-by: Ralf Baechle --- sound/oss/au1550_ac97.c | 12 +++++++----- sound/soc/au1x/dbdma2.c | 8 ++++---- 2 files changed, 11 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index 4191acccbcdb..b9ff0b798032 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -614,7 +614,8 @@ start_adc(struct au1550_state *s) /* Put two buffers on the ring to get things started. */ for (i=0; i<2; i++) { - au1xxx_dbdma_put_dest(db->dmanr, db->nextIn, db->dma_fragsize); + au1xxx_dbdma_put_dest(db->dmanr, db->nextIn, + db->dma_fragsize, DDMA_FLAGS_IE); db->nextIn += db->dma_fragsize; if (db->nextIn >= db->rawbuf + db->dmasize) @@ -733,7 +734,7 @@ static void dac_dma_interrupt(int irq, void *dev_id) if (db->count >= db->fragsize) { if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut, - db->fragsize) == 0) { + db->fragsize, DDMA_FLAGS_IE) == 0) { err("qcount < 2 and no ring room!"); } db->nextOut += db->fragsize; @@ -777,7 +778,8 @@ static void adc_dma_interrupt(int irq, void *dev_id) /* Put a new empty buffer on the destination DMA. */ - au1xxx_dbdma_put_dest(dp->dmanr, dp->nextIn, dp->dma_fragsize); + au1xxx_dbdma_put_dest(dp->dmanr, dp->nextIn, + dp->dma_fragsize, DDMA_FLAGS_IE); dp->nextIn += dp->dma_fragsize; if (dp->nextIn >= dp->rawbuf + dp->dmasize) @@ -1177,8 +1179,8 @@ au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos) * we know the dma has stopped. */ while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) { - if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut, - db->fragsize) == 0) { + if (au1xxx_dbdma_put_source(db->dmanr, + db->nextOut, db->fragsize, DDMA_FLAGS_IE) == 0) { err("qcount < 2 and no ring room!"); } db->nextOut += db->fragsize; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 19e4d37eba1c..2929f1c42264 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -94,7 +94,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_source_flags(cd->ddma_chan, + au1xxx_dbdma_put_source(cd->ddma_chan, (void *)phys_to_virt(cd->dma_area), cd->period_bytes, DDMA_FLAGS_IE); @@ -109,9 +109,9 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_dest_flags(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), - cd->period_bytes, DDMA_FLAGS_IE); + au1xxx_dbdma_put_dest(cd->ddma_chan, + (void *)phys_to_virt(cd->dma_area), + cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ ++cd->q_period; -- cgit v1.2.2 From 963accbc82a0912b39de39d59e2fd6741db3aa4b Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Tue, 13 Oct 2009 20:22:35 +0200 Subject: MIPS: Alchemy: change dbdma to accept physical memory addresses DMA can only be done from physical addresses; move the "virt_to_phys" source/destination buffer address translation from the dbdma queueing functions (since the hardware can only DMA to/from physical addresses) to their respective users. Signed-off-by: Manuel Lauss Signed-off-by: Ralf Baechle --- sound/oss/au1550_ac97.c | 12 +++++++----- sound/soc/au1x/dbdma2.c | 12 +++++------- 2 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index b9ff0b798032..c1070e33b32f 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -614,7 +614,7 @@ start_adc(struct au1550_state *s) /* Put two buffers on the ring to get things started. */ for (i=0; i<2; i++) { - au1xxx_dbdma_put_dest(db->dmanr, db->nextIn, + au1xxx_dbdma_put_dest(db->dmanr, virt_to_phys(db->nextIn), db->dma_fragsize, DDMA_FLAGS_IE); db->nextIn += db->dma_fragsize; @@ -733,8 +733,9 @@ static void dac_dma_interrupt(int irq, void *dev_id) db->dma_qcount--; if (db->count >= db->fragsize) { - if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut, - db->fragsize, DDMA_FLAGS_IE) == 0) { + if (au1xxx_dbdma_put_source(db->dmanr, + virt_to_phys(db->nextOut), db->fragsize, + DDMA_FLAGS_IE) == 0) { err("qcount < 2 and no ring room!"); } db->nextOut += db->fragsize; @@ -778,7 +779,7 @@ static void adc_dma_interrupt(int irq, void *dev_id) /* Put a new empty buffer on the destination DMA. */ - au1xxx_dbdma_put_dest(dp->dmanr, dp->nextIn, + au1xxx_dbdma_put_dest(dp->dmanr, virt_to_phys(dp->nextIn), dp->dma_fragsize, DDMA_FLAGS_IE); dp->nextIn += dp->dma_fragsize; @@ -1180,7 +1181,8 @@ au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos) */ while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) { if (au1xxx_dbdma_put_source(db->dmanr, - db->nextOut, db->fragsize, DDMA_FLAGS_IE) == 0) { + virt_to_phys(db->nextOut), db->fragsize, + DDMA_FLAGS_IE) == 0) { err("qcount < 2 and no ring room!"); } db->nextOut += db->fragsize; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 2929f1c42264..6d9f4c624949 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -51,8 +51,8 @@ struct au1xpsc_audio_dmadata { struct snd_pcm_substream *substream; unsigned long curr_period; /* current segment DDMA is working on */ unsigned long q_period; /* queue period(s) */ - unsigned long dma_area; /* address of queued DMA area */ - unsigned long dma_area_s; /* start address of DMA area */ + dma_addr_t dma_area; /* address of queued DMA area */ + dma_addr_t dma_area_s; /* start address of DMA area */ unsigned long pos; /* current byte position being played */ unsigned long periods; /* number of SG segments in total */ unsigned long period_bytes; /* size in bytes of one SG segment */ @@ -94,8 +94,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_source(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), + au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area, cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ @@ -109,8 +108,7 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_dest(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), + au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area, cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ @@ -233,7 +231,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, pcd->substream = substream; pcd->period_bytes = params_period_bytes(params); pcd->periods = params_periods(params); - pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr; + pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; pcd->q_period = 0; pcd->curr_period = 0; pcd->pos = 0; -- cgit v1.2.2 From 05ae3231801df8fdb4e1c0aa4aa6b8d7278eddde Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 2 Nov 2009 21:21:44 +0100 Subject: MIPS/SOUND: Alchemy: DB1200 AC97+I2S audio support. Machine driver for DB1200 AC97 and I2S audio systems, intended as a proper reference asoc machine for Alchemy-based systems. AC97/I2S can be selected at boot time by setting switch S6.7. Signed-off-by: Manuel Lauss Cc: Linux-MIPS Cc: alsa-devel@alsa-project.org Cc: Mark Brown Acked-by: Mark Brown Signed-off-by: Ralf Baechle --- sound/soc/au1x/Kconfig | 10 +-- sound/soc/au1x/Makefile | 4 +- sound/soc/au1x/db1200.c | 141 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/au1x/sample-ac97.c | 144 ------------------------------------------- 4 files changed, 149 insertions(+), 150 deletions(-) create mode 100644 sound/soc/au1x/db1200.c delete mode 100644 sound/soc/au1x/sample-ac97.c (limited to 'sound') diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 410a893aa66b..4b67140fdec3 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97 ## ## Boards ## -config SND_SOC_SAMPLE_PSC_AC97 - tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" +config SND_SOC_DB1200 + tristate "DB1200 AC97+I2S audio support" depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC + select SND_SOC_AU1XPSC_I2S + select SND_SOC_WM8731 help - This is a sample AC97 sound machine for use in Au12x0/Au1550 - based systems which have audio on PSC1 (e.g. Db1200 demoboard). + Select this option to enable audio (AC97 or I2S) on the + Alchemy/AMD/RMI DB1200 demoboard. diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 6c6950b8003a..16873076e8c4 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o # Boards -snd-soc-sample-ac97-objs := sample-ac97.o +snd-soc-db1200-objs := db1200.o -obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o +obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c new file mode 100644 index 000000000000..cdf7be1b9b91 --- /dev/null +++ b/sound/soc/au1x/db1200.c @@ -0,0 +1,141 @@ +/* + * DB1200 ASoC audio fabric support code. + * + * (c) 2008-9 Manuel Lauss + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/ac97.h" +#include "../codecs/wm8731.h" +#include "psc.h" + +/*------------------------- AC97 PART ---------------------------*/ + +static struct snd_soc_dai_link db1200_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card db1200_ac97_machine = { + .name = "DB1200_AC97", + .dai_link = &db1200_ac97_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_ac97_devdata = { + .card = &db1200_ac97_machine, + .codec_dev = &soc_codec_dev_ac97, +}; + +/*------------------------- I2S PART ---------------------------*/ + +static int db1200_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* WM8731 has its own 12MHz crystal */ + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + + /* codec is bitclock and lrclk master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = 0; +out: + return ret; +} + +static struct snd_soc_ops db1200_i2s_wm8731_ops = { + .startup = db1200_i2s_startup, +}; + +static struct snd_soc_dai_link db1200_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &au1xpsc_i2s_dai, + .codec_dai = &wm8731_dai, + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1200_i2s_machine = { + .name = "DB1200_I2S", + .dai_link = &db1200_i2s_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_i2s_devdata = { + .card = &db1200_i2s_machine, + .codec_dev = &soc_codec_dev_wm8731, +}; + +/*------------------------- COMMON PART ---------------------------*/ + +static struct platform_device *db1200_asoc_dev; + +static int __init db1200_audio_load(void) +{ + int ret; + + ret = -ENOMEM; + db1200_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!db1200_asoc_dev) + goto out; + + /* DB1200 board setup set PSC1MUX to preferred audio device */ + if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX) + platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata); + else + platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata); + + db1200_ac97_devdata.dev = &db1200_asoc_dev->dev; + db1200_i2s_devdata.dev = &db1200_asoc_dev->dev; + ret = platform_device_add(db1200_asoc_dev); + + if (ret) { + platform_device_put(db1200_asoc_dev); + db1200_asoc_dev = NULL; + } +out: + return ret; +} + +static void __exit db1200_audio_unload(void) +{ + platform_device_unregister(db1200_asoc_dev); +} + +module_init(db1200_audio_load); +module_exit(db1200_audio_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1200 ASoC audio support"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c deleted file mode 100644 index 27683eb7905e..000000000000 --- a/sound/soc/au1x/sample-ac97.c +++ /dev/null @@ -1,144 +0,0 @@ -/* - * Sample Au12x0/Au1550 PSC AC97 sound machine. - * - * Copyright (c) 2007-2008 Manuel Lauss - * - * This program is free software; you can redistribute it and/or modify - * it under the terms outlined in the file COPYING at the root of this - * source archive. - * - * This is a very generic AC97 sound machine driver for boards which - * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "../codecs/ac97.h" -#include "psc.h" - -static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_sync(codec); - return 0; -} - -static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ - .codec_dai = &ac97_dai, /* see codecs/ac97.c */ - .init = au1xpsc_sample_ac97_init, - .ops = NULL, -}; - -static struct snd_soc_card au1xpsc_sample_ac97_machine = { - .name = "Au1xxx PSC AC97 Audio", - .dai_link = &au1xpsc_sample_ac97_dai, - .num_links = 1, -}; - -static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct resource au1xpsc_psc1_res[] = { - [0] = { - .start = CPHYSADDR(PSC1_BASE_ADDR), - .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, - .flags = IORESOURCE_MEM, - }, - [1] = { -#ifdef CONFIG_SOC_AU1200 - .start = AU1200_PSC1_INT, - .end = AU1200_PSC1_INT, -#elif defined(CONFIG_SOC_AU1550) - .start = AU1550_PSC1_INT, - .end = AU1550_PSC1_INT, -#endif - .flags = IORESOURCE_IRQ, - }, - [2] = { - .start = DSCR_CMD0_PSC1_TX, - .end = DSCR_CMD0_PSC1_TX, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = DSCR_CMD0_PSC1_RX, - .end = DSCR_CMD0_PSC1_RX, - .flags = IORESOURCE_DMA, - }, -}; - -static struct platform_device *au1xpsc_sample_ac97_dev; - -static int __init au1xpsc_sample_ac97_load(void) -{ - int ret; - -#ifdef CONFIG_SOC_AU1200 - unsigned long io; - - /* modify sys_pinfunc for AC97 on PSC1 */ - io = au_readl(SYS_PINFUNC); - io |= SYS_PINFUNC_P1C; - io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); - au_writel(io, SYS_PINFUNC); - au_sync(); -#endif - - ret = -ENOMEM; - - /* setup PSC clock source for AC97 part: external clock provided - * by codec. The psc-ac97.c driver depends on this setting! - */ - au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); - au_sync(); - - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; - - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; - - platform_set_drvdata(au1xpsc_sample_ac97_dev, - &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); - - if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; - } - -out: - return ret; -} - -static void __exit au1xpsc_sample_ac97_exit(void) -{ - platform_device_unregister(au1xpsc_sample_ac97_dev); -} - -module_init(au1xpsc_sample_ac97_load); -module_exit(au1xpsc_sample_ac97_exit); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss "); -- cgit v1.2.2 From e584bc3cf6865e005bbb4dbabae0bf4b3df59500 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Mar 2010 16:20:37 +0100 Subject: ALSA: ua101: add Edirol UA-1000 support Add support for the Edirol UA-1000 to the UA-101 driver. Both devices behave the same, so we just have to shuffle around some interface numbers and name strings. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/usb/Kconfig | 6 +++--- sound/usb/ua101.c | 45 +++++++++++++++++++++++++++++++------------ sound/usb/usbaudio.c | 53 --------------------------------------------------- sound/usb/usbaudio.h | 3 +-- sound/usb/usbquirks.h | 30 ----------------------------- 5 files changed, 37 insertions(+), 100 deletions(-) (limited to 'sound') diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 8c2925814ce4..c570ae3e6d55 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -22,13 +22,13 @@ config SND_USB_AUDIO will be called snd-usb-audio. config SND_USB_UA101 - tristate "Edirol UA-101 driver (EXPERIMENTAL)" + tristate "Edirol UA-101/UA-1000 driver (EXPERIMENTAL)" depends on EXPERIMENTAL select SND_PCM select SND_RAWMIDI help - Say Y here to include support for the Edirol UA-101 audio/MIDI - interface. + Say Y here to include support for the Edirol UA-101 and UA-1000 + audio/MIDI interfaces. To compile this driver as a module, choose M here: the module will be called snd-ua101. diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 4f4ccdf70dd0..047dc1ca84d0 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -1,5 +1,5 @@ /* - * Edirol UA-101 driver + * Edirol UA-101/UA-1000 driver * Copyright (c) Clemens Ladisch * * This driver is free software: you can redistribute it and/or modify @@ -25,10 +25,10 @@ #include #include "usbaudio.h" -MODULE_DESCRIPTION("Edirol UA-101 driver"); +MODULE_DESCRIPTION("Edirol UA-101/1000 driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101}}"); +MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101},{Edirol,UA-1000}}"); /* I use my UA-1A for testing because I don't have a UA-101 ... */ #define UA1A_HACK @@ -1200,13 +1200,30 @@ static int ua101_probe(struct usb_interface *interface, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &midi_ep }; + static const int intf_numbers[2][3] = { + { /* UA-101 */ + [INTF_PLAYBACK] = 0, + [INTF_CAPTURE] = 1, + [INTF_MIDI] = 2, + }, + { /* UA-1000 */ + [INTF_CAPTURE] = 1, + [INTF_PLAYBACK] = 2, + [INTF_MIDI] = 3, + }, + }; struct snd_card *card; struct ua101 *ua; unsigned int card_index, i; + int is_ua1000; + const char *name; char usb_path[32]; int err; - if (interface->altsetting->desc.bInterfaceNumber != 0) + is_ua1000 = usb_id->idProduct == 0x0044; + + if (interface->altsetting->desc.bInterfaceNumber != + intf_numbers[is_ua1000][0]) return -ENODEV; mutex_lock(&devices_mutex); @@ -1250,9 +1267,11 @@ static int ua101_probe(struct usb_interface *interface, #endif ua->intf[0] = interface; for (i = 1; i < ARRAY_SIZE(ua->intf); ++i) { - ua->intf[i] = usb_ifnum_to_if(ua->dev, i); + ua->intf[i] = usb_ifnum_to_if(ua->dev, + intf_numbers[is_ua1000][i]); if (!ua->intf[i]) { - dev_err(&ua->dev->dev, "interface %u not found\n", i); + dev_err(&ua->dev->dev, "interface %u not found\n", + intf_numbers[is_ua1000][i]); err = -ENXIO; goto probe_error; } @@ -1292,11 +1311,12 @@ static int ua101_probe(struct usb_interface *interface, } #endif + name = usb_id->idProduct == 0x0044 ? "UA-1000" : "UA-101"; strcpy(card->driver, "UA-101"); - strcpy(card->shortname, "UA-101"); + strcpy(card->shortname, name); usb_make_path(ua->dev, usb_path, sizeof(usb_path)); snprintf(ua->card->longname, sizeof(ua->card->longname), - "EDIROL UA-101 (serial %s), %u Hz at %s, %s speed", + "EDIROL %s (serial %s), %u Hz at %s, %s speed", name, ua->dev->serial ? ua->dev->serial : "?", ua->rate, usb_path, ua->dev->speed == USB_SPEED_HIGH ? "high" : "full"); @@ -1314,11 +1334,11 @@ static int ua101_probe(struct usb_interface *interface, if (err < 0) goto probe_error; - err = snd_pcm_new(card, "UA-101", 0, 1, 1, &ua->pcm); + err = snd_pcm_new(card, name, 0, 1, 1, &ua->pcm); if (err < 0) goto probe_error; ua->pcm->private_data = ua; - strcpy(ua->pcm->name, "UA-101"); + strcpy(ua->pcm->name, name); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); @@ -1389,8 +1409,9 @@ static struct usb_device_id ua101_ids[] = { #ifdef UA1A_HACK { USB_DEVICE(0x0582, 0x0018) }, #endif - { USB_DEVICE(0x0582, 0x007d) }, - { USB_DEVICE(0x0582, 0x008d) }, + { USB_DEVICE(0x0582, 0x0044) }, /* UA-1000 high speed */ + { USB_DEVICE(0x0582, 0x007d) }, /* UA-101 high speed */ + { USB_DEVICE(0x0582, 0x008d) }, /* UA-101 full speed */ { } }; MODULE_DEVICE_TABLE(usb, ua101_ids); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8a8f62515b80..7ad8089b233e 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3116,58 +3116,6 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, return 0; } -/* - * Create a stream for an Edirol UA-1000 interface. - */ -static int create_ua1000_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - static const struct audioformat ua1000_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - }; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct audioformat *fp; - int stream, err; - - if (iface->num_altsetting != 2) - return -ENXIO; - alts = &iface->altsetting[1]; - altsd = get_iface_desc(alts); - if (alts->extralen != 11 || alts->extra[1] != USB_DT_CS_INTERFACE || - altsd->bNumEndpoints != 1) - return -ENXIO; - - fp = kmemdup(&ua1000_format, sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; - - fp->channels = alts->extra[4]; - fp->iface = altsd->bInterfaceNumber; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - fp->rate_max = fp->rate_min = combine_triple(&alts->extra[8]); - - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp); - return err; - } - /* FIXME: playback must be synchronized to capture */ - usb_set_interface(chip->dev, fp->iface, 0); - return 0; -} - static int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, const struct snd_usb_audio_quirk *quirk); @@ -3416,7 +3364,6 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk }; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9d8cea48fc5f..96c558a76ba6 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,7 +159,6 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UAXX, QUIRK_AUDIO_ALIGN_TRANSFER, @@ -196,7 +195,7 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_AUDIO/MIDI_STANDARD_INTERFACE, data is NULL */ -/* for QUIRK_AUDIO_EDIROL_UA700_UA25/UA1000, data is NULL */ +/* for QUIRK_AUDIO_EDIROL_UAXX, data is NULL */ /* for QUIRK_IGNORE_INTERFACE, data is NULL */ diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index e691eba6a83e..977d980fb117 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1015,36 +1015,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -{ - USB_DEVICE(0x0582, 0x0044), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "UA-1000", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA1000 - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA1000 - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } - }, - { - .ifnum = -1 - } - } - } -}, { /* has ID 0x0049 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0047), -- cgit v1.2.2 From e1aed7ca555af7412ca1336241b918d78485232f Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Mon, 4 Jan 2010 16:26:32 +0800 Subject: [ARM] pxa: remove the unnecessary restoring of MFP registers MFP registers are saved and restored by the mfp sys_device before all other platform devices, and it is unnecessary here. Cc: Dmitry Eremin-Solenikov Cc: Mark Brown Signed-off-by: Eric Miao --- sound/arm/pxa2xx-ac97-lib.c | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fdca97186e7..7587a748ea06 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -345,16 +345,6 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - if (cpu_is_pxa25x() || cpu_is_pxa27x()) { - pxa_gpio_mode(GPIO31_SYNC_AC97_MD); - pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); - pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); - pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); - } - if (cpu_is_pxa27x()) { - /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); - } clk_enable(ac97_clk); return 0; } -- cgit v1.2.2 From fb1bf8cd13bfa7ed0364ab0d82f717fc020d35f6 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Mon, 4 Jan 2010 16:30:58 +0800 Subject: [ARM] pxa: introduce processor specific pxa27x_assert_ac97reset() This is really pxa27x specific and should be kept in pxa27x.c. With this newly introduced function, the original set_resetgpio_mode() is deprecated. Cc: Dmitry Eremin-Solenikov Cc: Mark Brown Signed-off-by: Eric Miao --- sound/arm/pxa2xx-ac97-lib.c | 50 ++++++--------------------------------------- 1 file changed, 6 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 7587a748ea06..ee687283b6a1 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -32,6 +32,8 @@ static struct clk *ac97_clk; static struct clk *ac97conf_clk; static int reset_gpio; +extern void pxa27x_assert_ac97reset(int reset_gpio, int on); + /* * Beware PXA27x bugs: * @@ -42,45 +44,6 @@ static int reset_gpio; * 1 jiffy timeout if interrupt never comes). */ -enum { - RESETGPIO_FORCE_HIGH, - RESETGPIO_FORCE_LOW, - RESETGPIO_NORMAL_ALTFUNC -}; - -/** - * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA - * @mode: chosen action - * - * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line - * must be done to insure proper work of AC97 reset line. This function - * computes the correct gpio_mode for further use by reset functions, and - * applied the change through pxa_gpio_mode. - */ -static void set_resetgpio_mode(int resetgpio_action) -{ - int mode = 0; - - if (reset_gpio) - switch (resetgpio_action) { - case RESETGPIO_NORMAL_ALTFUNC: - if (reset_gpio == 113) - mode = 113 | GPIO_ALT_FN_2_OUT; - if (reset_gpio == 95) - mode = 95 | GPIO_ALT_FN_1_OUT; - break; - case RESETGPIO_FORCE_LOW: - mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; - break; - case RESETGPIO_FORCE_HIGH: - mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; - break; - }; - - if (mode) - pxa_gpio_mode(mode); -} - unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -174,12 +137,11 @@ static inline void pxa_ac97_warm_pxa27x(void) { gsr_bits = 0; - /* warm reset broken on Bulverde, - so manually keep AC97 reset high */ - set_resetgpio_mode(RESETGPIO_FORCE_HIGH); + /* warm reset broken on Bulverde, so manually keep AC97 reset high */ + pxa27x_assert_ac97reset(reset_gpio, 1); udelay(10); GCR |= GCR_WARM_RST; - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_assert_ac97reset(reset_gpio, 0); udelay(500); } @@ -385,7 +347,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_assert_ac97reset(reset_gpio, 0); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- cgit v1.2.2 From 846c864cac520eaa10e845f585f05af643aa848a Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Mon, 4 Jan 2010 17:14:21 +0800 Subject: [ARM] pxa: remove now unnecessary pxa_gpio_mode() calls in ac97 Now most (if not all) PXA platforms have been switched to the new MFP API, it's rather safe to remove these unnecessary pxa_gpio_mode() calls in pxa2xx-ac97-lib.c now. Cc: Dmitry Eremin-Solenikov Cc: Mark Brown Signed-off-by: Eric Miao --- sound/arm/pxa2xx-ac97-lib.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index ee687283b6a1..88eec3847df2 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -22,7 +22,6 @@ #include #include -#include #include static DEFINE_MUTEX(car_mutex); @@ -338,13 +337,6 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) reset_gpio = 113; } - if (cpu_is_pxa25x() || cpu_is_pxa27x()) { - pxa_gpio_mode(GPIO31_SYNC_AC97_MD); - pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); - pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); - pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); - } - if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ pxa27x_assert_ac97reset(reset_gpio, 0); -- cgit v1.2.2 From a056bef45529810183f56944dcea8b4e297c2dc3 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Tue, 9 Feb 2010 11:10:10 +0800 Subject: [ARM] pxa: introduce PXA_SSP_LEGACY for legacy SSP API Signed-off-by: Eric Miao --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 376e14a9c273..89de27578416 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -23,6 +23,7 @@ config SND_PXA2XX_SOC_I2S config SND_PXA_SOC_SSP tristate + select PXA_SSP_LEGACY config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" -- cgit v1.2.2 From f9efc9df94fd126f7d585339e64edec0c03e904b Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Tue, 9 Feb 2010 19:46:01 +0800 Subject: ASoC: Remove legacy SSP API usage from pxa-ssp.c Acked-by: Mark Brown Signed-off-by: Eric Miao --- sound/soc/pxa/Kconfig | 1 - sound/soc/pxa/pxa-ssp.c | 90 +++++++++++++++++++++++++++++++++---------------- 2 files changed, 61 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 89de27578416..376e14a9c273 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -23,7 +23,6 @@ config SND_PXA2XX_SOC_I2S config SND_PXA_SOC_SSP tristate - select PXA_SSP_LEGACY config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3bd7712f029b..cf00df9c40f4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -42,11 +42,14 @@ * SSP audio private data */ struct ssp_priv { - struct ssp_dev dev; + struct ssp_device *ssp; unsigned int sysclk; int dai_fmt; #ifdef CONFIG_PM - struct ssp_state state; + uint32_t cr0; + uint32_t cr1; + uint32_t to; + uint32_t psp; #endif }; @@ -61,6 +64,22 @@ static void dump_registers(struct ssp_device *ssp) ssp_read_reg(ssp, SSACD)); } +static void ssp_enable(struct ssp_device *ssp) +{ + uint32_t sscr0; + + sscr0 = __raw_readl(ssp->mmio_base + SSCR0) | SSCR0_SSE; + __raw_writel(sscr0, ssp->mmio_base + SSCR0); +} + +static void ssp_disable(struct ssp_device *ssp) +{ + uint32_t sscr0; + + sscr0 = __raw_readl(ssp->mmio_base + SSCR0) & ~SSCR0_SSE; + __raw_writel(sscr0, ssp->mmio_base + SSCR0); +} + struct pxa2xx_pcm_dma_data { struct pxa2xx_pcm_dma_params params; char name[20]; @@ -94,13 +113,12 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; int ret = 0; if (!cpu_dai->active) { - priv->dev.port = cpu_dai->id + 1; - priv->dev.irq = NO_IRQ; - clk_enable(priv->dev.ssp->clk); - ssp_disable(&priv->dev); + clk_enable(ssp->clk); + ssp_disable(ssp); } if (cpu_dai->dma_data) { @@ -116,10 +134,11 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) { - ssp_disable(&priv->dev); - clk_disable(priv->dev.ssp->clk); + ssp_disable(ssp); + clk_disable(ssp->clk); } if (cpu_dai->dma_data) { @@ -133,26 +152,39 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) { struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) return 0; - ssp_save_state(&priv->dev, &priv->state); - clk_disable(priv->dev.ssp->clk); + priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); + priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); + priv->to = __raw_readl(ssp->mmio_base + SSTO); + priv->psp = __raw_readl(ssp->mmio_base + SSPSP); + + ssp_disable(ssp); + clk_disable(ssp->clk); return 0; } static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) { struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; if (!cpu_dai->active) return 0; - clk_enable(priv->dev.ssp->clk); - ssp_restore_state(&priv->dev, &priv->state); - ssp_enable(&priv->dev); + clk_enable(ssp->clk); + + __raw_writel(sssr, ssp->mmio_base + SSSR); + __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); + __raw_writel(priv->cr1, ssp->mmio_base + SSCR1); + __raw_writel(priv->to, ssp->mmio_base + SSTO); + __raw_writel(priv->psp, ssp->mmio_base + SSPSP); + __raw_writel(priv->cr0 | SSCR0_SSE, ssp->mmio_base + SSCR0); return 0; } @@ -201,7 +233,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; u32 sscr0 = ssp_read_reg(ssp, SSCR0) & @@ -242,11 +274,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (!cpu_is_pxa3xx()) - clk_disable(priv->dev.ssp->clk); + clk_disable(ssp->clk); val = ssp_read_reg(ssp, SSCR0) | sscr0; ssp_write_reg(ssp, SSCR0, val); if (!cpu_is_pxa3xx()) - clk_enable(priv->dev.ssp->clk); + clk_enable(ssp->clk); return 0; } @@ -258,7 +290,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; switch (div_id) { @@ -309,7 +341,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; #if defined(CONFIG_PXA3xx) @@ -378,7 +410,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr0; sscr0 = ssp_read_reg(ssp, SSCR0); @@ -413,7 +445,7 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, int tristate) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr1; sscr1 = ssp_read_reg(ssp, SSCR1); @@ -435,7 +467,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr0; u32 sscr1; u32 sspsp; @@ -530,7 +562,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int chn = params_channels(params); u32 sscr0; u32 sspsp; @@ -640,12 +672,12 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ssp_enable(&priv->dev); + ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ssp_read_reg(ssp, SSCR1); @@ -664,7 +696,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, else val |= SSCR1_RSRE; ssp_write_reg(ssp, SSCR1, val); - ssp_enable(&priv->dev); + ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_STOP: val = ssp_read_reg(ssp, SSCR1); @@ -675,7 +707,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, ssp_write_reg(ssp, SSCR1, val); break; case SNDRV_PCM_TRIGGER_SUSPEND: - ssp_disable(&priv->dev); + ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: val = ssp_read_reg(ssp, SSCR1); @@ -705,8 +737,8 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); - if (priv->dev.ssp == NULL) { + priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + if (priv->ssp == NULL) { ret = -ENODEV; goto err_priv; } @@ -725,7 +757,7 @@ static void pxa_ssp_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct ssp_priv *priv = dai->private_data; - ssp_free(priv->dev.ssp); + ssp_free(priv->ssp); } #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- cgit v1.2.2 From 8b1935e6a36b0967efc593d67ed3aebbfbc1f5b1 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 11 Feb 2010 16:50:14 +0000 Subject: dmaengine: shdma: separate DMA headers. Separate SH DMA headers into ones, commonly used by both drivers, and ones, specific to each of them. This will make the future development of the dmaengine driver easier. Signed-off-by: Guennadi Liakhovetski Acked-by: Mark Brown Signed-off-by: Paul Mundt --- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h index 9cc04ab2bce7..c0bfab8fed3d 100644 --- a/sound/soc/sh/siu.h +++ b/sound/soc/sh/siu.h @@ -72,7 +72,7 @@ struct siu_firmware { #include #include -#include +#include #include #include diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index c5efc30f0136..ba7f8d05d977 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -32,7 +32,7 @@ #include #include -#include +#include #include #include "siu.h" -- cgit v1.2.2 From 20645d70bdcdcc29b1b92011780d233008a8adcf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Mar 2010 11:14:01 +0100 Subject: ALSA: hda - Add missing hp_pins definitions for ALC269 quirks In 2.6.33 ACL269 unsol event handler was changed to look up the pre-defined pins, but the headphone pins aren't defined properly in each quirk. This patch adds the missing definitions, and fixes the speaker auto-mute regression on some ASUS (and possibly other) laptops. Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e8cbe216e912..b9f4689ccd9a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13561,6 +13561,8 @@ static void alc269_lifebook_unsol_event(struct hda_codec *codec, static void alc269_quanta_fl1_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x19; @@ -13656,6 +13658,8 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec, static void alc269_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; @@ -13666,6 +13670,8 @@ static void alc269_laptop_dmic_setup(struct hda_codec *codec) static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; @@ -13676,6 +13682,8 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) static void alc269_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x19; -- cgit v1.2.2 From 28aedaf7bf6e4b629aea333978e8bb440bd1eb4f Mon Sep 17 00:00:00 2001 From: Norberto Lopes Date: Sun, 28 Feb 2010 20:16:53 +0100 Subject: ALSA: sound/pci/hda/hda_codec.c: various coding style fixes Signed-off-by: Norberto Lopes Acked-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++++--------------------- 1 file changed, 38 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 76d3c4c049db..5bd7cf45f3a5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -978,8 +978,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * * Returns 0 if successful, or a negative error code. */ -int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp) +int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, + unsigned int codec_addr, + struct hda_codec **codecp) { struct hda_codec *codec; char component[31]; @@ -1186,7 +1187,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); */ /* FIXME: more better hash key? */ -#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_KEY(nid, dir, idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) #define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) @@ -1356,7 +1357,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) if (!codec->no_trigger_sense) { pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); } return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); @@ -1372,8 +1374,8 @@ EXPORT_SYMBOL_HDA(snd_hda_pin_sense); */ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return !!(sense & AC_PINSENSE_PRESENCE); + u32 sense = snd_hda_pin_sense(codec, nid); + return !!(sense & AC_PINSENSE_PRESENCE); } EXPORT_SYMBOL_HDA(snd_hda_jack_detect); @@ -1952,7 +1954,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - + for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; int i = 0; @@ -2439,27 +2441,27 @@ static struct snd_kcontrol_new dig_mixes[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_cmask_get, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_pmask_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_default_get, .put = snd_hda_spdif_default_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), .info = snd_hda_spdif_out_switch_info, .get = snd_hda_spdif_out_switch_get, .put = snd_hda_spdif_out_switch_put, @@ -2610,7 +2612,7 @@ static int snd_hda_spdif_in_status_get(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new dig_in_ctls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, SWITCH), .info = snd_hda_spdif_in_switch_info, .get = snd_hda_spdif_in_switch_get, .put = snd_hda_spdif_in_switch_put, @@ -2618,7 +2620,7 @@ static struct snd_kcontrol_new dig_in_ctls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_in_status_get, }, @@ -2883,7 +2885,7 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) int err = snd_hda_codec_build_controls(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build controls" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -2979,8 +2981,12 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, val |= channels - 1; switch (snd_pcm_format_width(format)) { - case 8: val |= 0x00; break; - case 16: val |= 0x10; break; + case 8: + val |= 0x00; + break; + case 16: + val |= 0x10; + break; case 20: case 24: case 32: @@ -3298,7 +3304,8 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; - snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); + snd_printk(KERN_WARNING "Too many %s devices\n", + snd_hda_pcm_type_name[type]); return -EAGAIN; } @@ -3336,7 +3343,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) err = codec->patch_ops.build_pcms(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build PCMs" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -3466,8 +3473,8 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_config); /** * snd_hda_check_board_codec_sid_config - compare the current codec - subsystem ID with the - config table + subsystem ID with the + config table This is important for Gateway notebooks with SB450 HDA Audio where the vendor ID of the PCI device is: @@ -3607,7 +3614,7 @@ void snd_hda_update_power_acct(struct hda_codec *codec) * * Increment the power-up counter and power up the hardware really when * not turned on yet. - */ + */ void snd_hda_power_up(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; @@ -3636,7 +3643,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); * * Decrement the power-up counter and schedules the power-off work if * the counter rearches to zero. - */ + */ void snd_hda_power_down(struct hda_codec *codec) { --codec->power_count; @@ -3662,7 +3669,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_down); * * This function is supposed to be set or called from the check_power_status * patch ops. - */ + */ int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, hda_nid_t nid) @@ -3830,7 +3837,7 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, { /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) - set_dig_out_convert(codec, nid, + set_dig_out_convert(codec, nid, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff, -1); snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); @@ -4089,13 +4096,13 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) /* * Sort an associated group of pins according to their sequence numbers. */ -static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, +static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, int num_pins) { int i, j; short seq; hda_nid_t nid; - + for (i = 0; i < num_pins; i++) { for (j = i + 1; j < num_pins; j++) { if (sequences[i] > sequences[j]) { @@ -4123,7 +4130,7 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, * is detected, one of speaker of HP pins is assigned as the primary * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. - * + * * The analog input pins are assigned to input_pins array. * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. @@ -4186,9 +4193,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (! assoc) + if (!assoc) continue; - if (! assoc_speaker) + if (!assoc_speaker) assoc_speaker = assoc; else if (assoc_speaker != assoc) continue; @@ -4286,7 +4293,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->speaker_outs); sort_pins_by_sequence(cfg->hp_pins, sequences_hp, cfg->hp_outs); - + /* if we have only one mic, make it AUTO_PIN_MIC */ if (!cfg->input_pins[AUTO_PIN_MIC] && cfg->input_pins[AUTO_PIN_FRONT_MIC]) { @@ -4436,7 +4443,7 @@ EXPORT_SYMBOL_HDA(snd_hda_resume); /** * snd_array_new - get a new element from the given array * @array: the array object - * + * * Get a new element from the given array. If it exceeds the * pre-allocated array size, re-allocate the array. * -- cgit v1.2.2 From 76b53774c51c4eaec646578a2e1b3716befedf1c Mon Sep 17 00:00:00 2001 From: Andrea Gelmini Date: Sat, 27 Feb 2010 17:51:03 +0100 Subject: sound/oss/v_midi.h: Checkpatch cleanup sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible sound/oss/v_midi.h:7: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini Signed-off-by: Takashi Iwai --- sound/oss/v_midi.h | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/v_midi.h b/sound/oss/v_midi.h index 1b86cb45c607..08e2185ee816 100644 --- a/sound/oss/v_midi.h +++ b/sound/oss/v_midi.h @@ -2,9 +2,9 @@ typedef struct vmidi_devc { int dev; /* State variables */ - int opened; + int opened; spinlock_t lock; - + /* MIDI fields */ int my_mididev; int pair_mididev; @@ -12,4 +12,3 @@ typedef struct vmidi_devc { int intr_active; void (*midi_input_intr) (int dev, unsigned char data); } vmidi_devc; - -- cgit v1.2.2 From 3ea49652f679c2b571ca214c605ec80cb056ec10 Mon Sep 17 00:00:00 2001 From: Andrea Gelmini Date: Sat, 27 Feb 2010 17:51:19 +0100 Subject: sound/oss/coproc.h: Checkpatch cleanup sound/oss/coproc.h:7: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini Signed-off-by: Takashi Iwai --- sound/oss/coproc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/coproc.h b/sound/oss/coproc.h index 7306346e9ac4..7bec21bbdd88 100644 --- a/sound/oss/coproc.h +++ b/sound/oss/coproc.h @@ -4,7 +4,7 @@ */ /* - * Coprocessor access types + * Coprocessor access types */ #define COPR_CUSTOM 0x0001 /* Custom applications */ #define COPR_MIDI 0x0002 /* MIDI (MPU-401) emulation */ -- cgit v1.2.2 From 7f9320d415fab5c05097c77eea7a77f2f6341f24 Mon Sep 17 00:00:00 2001 From: Andrea Gelmini Date: Sat, 27 Feb 2010 17:51:29 +0100 Subject: ALSA: sound/usb/caiaq/midi.h: Checkpatch cleanup sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar" Signed-off-by: Andrea Gelmini Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/midi.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/caiaq/midi.h b/sound/usb/caiaq/midi.h index 9d16db027fc3..380f984babc9 100644 --- a/sound/usb/caiaq/midi.h +++ b/sound/usb/caiaq/midi.h @@ -3,6 +3,6 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *dev); void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len); -void snd_usb_caiaq_midi_output_done(struct urb* urb); +void snd_usb_caiaq_midi_output_done(struct urb *urb); #endif /* CAIAQ_MIDI_H */ -- cgit v1.2.2 From 0a566ec25627bdd360f7294aa2e52f9d121233b4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 Mar 2010 08:47:20 +0100 Subject: ALSA: ua101: removing debugging code Remove some code that is no longer needed now that the relevant parts of the driver have been tested. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/usb/ua101.c | 55 ------------------------------------------------------- 1 file changed, 55 deletions(-) (limited to 'sound') diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 047dc1ca84d0..3d458d3b9962 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -30,9 +30,6 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101},{Edirol,UA-1000}}"); -/* I use my UA-1A for testing because I don't have a UA-101 ... */ -#define UA1A_HACK - /* * Should not be lower than the minimum scheduling delay of the host * controller. Some Intel controllers need more than one frame; as long as @@ -132,9 +129,6 @@ struct ua101 { dma_addr_t dma; } buffers[MAX_MEMORY_BUFFERS]; } capture, playback; - - unsigned int fps[10]; - unsigned int frame_counter; }; static DEFINE_MUTEX(devices_mutex); @@ -424,16 +418,6 @@ static void capture_urb_complete(struct urb *urb) if (do_period_elapsed) snd_pcm_period_elapsed(stream->substream); - /* for debugging: measure the sample rate relative to the USB clock */ - ua->fps[ua->frame_counter++ / ua->packets_per_second] += frames; - if (ua->frame_counter >= ARRAY_SIZE(ua->fps) * ua->packets_per_second) { - printk(KERN_DEBUG "capture rate:"); - for (frames = 0; frames < ARRAY_SIZE(ua->fps); ++frames) - printk(KERN_CONT " %u", ua->fps[frames]); - printk(KERN_CONT "\n"); - memset(ua->fps, 0, sizeof(ua->fps)); - ua->frame_counter = 0; - } return; stream_stopped: @@ -1256,15 +1240,6 @@ static int ua101_probe(struct usb_interface *interface, init_waitqueue_head(&ua->rate_feedback_wait); init_waitqueue_head(&ua->alsa_playback_wait); -#ifdef UA1A_HACK - if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { - ua->intf[2] = interface; - ua->intf[0] = usb_ifnum_to_if(ua->dev, 1); - ua->intf[1] = usb_ifnum_to_if(ua->dev, 2); - usb_driver_claim_interface(&ua101_driver, ua->intf[0], ua); - usb_driver_claim_interface(&ua101_driver, ua->intf[1], ua); - } else { -#endif ua->intf[0] = interface; for (i = 1; i < ARRAY_SIZE(ua->intf); ++i) { ua->intf[i] = usb_ifnum_to_if(ua->dev, @@ -1283,33 +1258,12 @@ static int ua101_probe(struct usb_interface *interface, goto probe_error; } } -#ifdef UA1A_HACK - } -#endif snd_card_set_dev(card, &interface->dev); -#ifdef UA1A_HACK - if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { - ua->format_bit = SNDRV_PCM_FMTBIT_S16_LE; - ua->rate = 44100; - ua->packets_per_second = 1000; - ua->capture.channels = 2; - ua->playback.channels = 2; - ua->capture.frame_bytes = 4; - ua->playback.frame_bytes = 4; - ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, 2); - ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, 1); - ua->capture.max_packet_bytes = 192; - ua->playback.max_packet_bytes = 192; - } else { -#endif err = detect_usb_format(ua); if (err < 0) goto probe_error; -#ifdef UA1A_HACK - } -#endif name = usb_id->idProduct == 0x0044 ? "UA-1000" : "UA-101"; strcpy(card->driver, "UA-101"); @@ -1342,16 +1296,10 @@ static int ua101_probe(struct usb_interface *interface, snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); -#ifdef UA1A_HACK - if (ua->dev->descriptor.idProduct != cpu_to_le16(0x0018)) { -#endif err = snd_usbmidi_create(card, ua->intf[INTF_MIDI], &ua->midi_list, &midi_quirk); if (err < 0) goto probe_error; -#ifdef UA1A_HACK - } -#endif err = snd_card_register(card); if (err < 0) @@ -1406,9 +1354,6 @@ static void ua101_disconnect(struct usb_interface *interface) } static struct usb_device_id ua101_ids[] = { -#ifdef UA1A_HACK - { USB_DEVICE(0x0582, 0x0018) }, -#endif { USB_DEVICE(0x0582, 0x0044) }, /* UA-1000 high speed */ { USB_DEVICE(0x0582, 0x007d) }, /* UA-101 high speed */ { USB_DEVICE(0x0582, 0x008d) }, /* UA-101 full speed */ -- cgit v1.2.2 From 864c11080cf365720103042444534a1e94d42bac Mon Sep 17 00:00:00 2001 From: Arseniy Lartsev Date: Tue, 2 Mar 2010 14:52:28 +0300 Subject: ALSA: usbaudio: Fix wrong bitrate for Creative Creative VF0470 Live Cam This patch works around misbehaviour of Creative Creative VF0470 Live Cam which reports 16 kHz sample rate for audio capture while actually producing 8 kHz stream. Signed-off-by: Arseniy Lartsev Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 20b656e9f90d..ea3eaa53d637 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2581,6 +2581,9 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; + /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ + if (rate == 16000 && chip->usb_id == USB_ID(0x041e, 0x4068)) + rate = 8000; fp->rate_table[fp->nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) fp->rate_min = rate; -- cgit v1.2.2 From bb1c04784d39b95a4382bd283f3048c4eb859b58 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Thu, 25 Feb 2010 11:24:53 +0900 Subject: ASoC: soc_pcm_open: Add missing bailout tag The codec_dai needs to be shutdown should the machine startup fails. This patch adds another bailout tag for that case and rename the tag for configuration failures. Signed-off-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a03bac943aaf..c8b0556ef431 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -427,24 +427,24 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (!runtime->hw.rates) { printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", codec_dai->name, cpu_dai->name); - goto machine_err; + goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", codec_dai->name, cpu_dai->name); - goto machine_err; + goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", codec_dai->name, cpu_dai->name); - goto machine_err; + goto config_err; } /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active || codec_dai->active) { ret = soc_pcm_apply_symmetry(substream); if (ret != 0) - goto machine_err; + goto config_err; } pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); @@ -464,10 +464,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&pcm_mutex); return 0; -machine_err: +config_err: if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); +machine_err: + if (codec_dai->ops->shutdown) + codec_dai->ops->shutdown(substream, codec_dai); + codec_dai_err: if (platform->pcm_ops->close) platform->pcm_ops->close(substream); -- cgit v1.2.2 From e555317c083fda01f516d2153589e82514e20e70 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 26 Feb 2010 14:36:54 +0800 Subject: ASoC: fix ak4104 register array access Don't touch the variable 'reg' to construct the value for the actual SPI transport. This variable is again used to access the driver's register cache, and so random memory is overwritten. Compute the value in-place instead. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Cc: stable@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b9ef7e45891d..b68d99fb6af0 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -90,12 +90,10 @@ static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, if (reg >= codec->reg_cache_size) return -EINVAL; - reg &= AK4104_REG_MASK; - reg |= AK4104_WRITE; - /* only write to the hardware if value has changed */ if (cache[reg] != value) { - u8 tmp[2] = { reg, value }; + u8 tmp[2] = { (reg & AK4104_REG_MASK) | AK4104_WRITE, value }; + if (spi_write(spi, tmp, sizeof(tmp))) { dev_err(&spi->dev, "SPI write failed\n"); return -EIO; -- cgit v1.2.2 From fd8d47351d2e241f3168eeb697ce55cc28c75b78 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 3 Mar 2010 19:41:44 +0100 Subject: ALSA: opti92x: use PnP data to select Master Control port The Master Control port (MC) is available as the last PnP resource (OPT005). Use this value instead fo guessing. Also, add some comments to the code. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 120 ++++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 43 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b865e45a8f9b..5913717c1be6 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1558,7 +1558,7 @@ static int __devinit snd_card_miro_pnp(struct snd_miro *chip, err = pnp_activate_dev(devmc); if (err < 0) { - snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); return err; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index a4af53b5c1cf..becd90d7536d 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -144,12 +144,8 @@ struct snd_opti9xx { spinlock_t lock; + long wss_base; int irq; - -#ifdef CONFIG_PNP - struct pnp_dev *dev; - struct pnp_dev *devmpu; -#endif /* CONFIG_PNP */ }; static int snd_opti9xx_pnp_is_probed; @@ -159,12 +155,17 @@ static int snd_opti9xx_pnp_is_probed; static struct pnp_card_device_id snd_opti9xx_pnpids[] = { #ifndef OPTi93X /* OPTi 82C924 */ - { .id = "OPT0924", .devs = { { "OPT0000" }, { "OPT0002" } }, .driver_data = 0x0924 }, + { .id = "OPT0924", + .devs = { { "OPT0000" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0924 }, /* OPTi 82C925 */ - { .id = "OPT0925", .devs = { { "OPT9250" }, { "OPT0002" } }, .driver_data = 0x0925 }, + { .id = "OPT0925", + .devs = { { "OPT9250" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0925 }, #else /* OPTi 82C931/3 */ - { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, .driver_data = 0x0931 }, + { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, + .driver_data = 0x0931 }, #endif /* OPTi93X */ { .id = "" } }; @@ -207,24 +208,34 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, chip->hardware = hardware; strcpy(chip->name, snd_opti9xx_names[hardware]); - chip->mc_base_size = opti9xx_mc_size[hardware]; - spin_lock_init(&chip->lock); chip->irq = -1; +#ifndef OPTi93X +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; +#endif /* CONFIG_PNP */ + else { + chip->mc_base = 0xf8c; + chip->mc_base_size = opti9xx_mc_size[hardware]; + } +#else + chip->mc_base_size = opti9xx_mc_size[hardware]; +#endif + switch (hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = (hardware == OPTi9XX_HW_82C928) ? 0xe2 : 0xe3; chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: case OPTi9XX_HW_82C925: - chip->mc_base = 0xf8c; chip->password = 0xe5; chip->pwd_reg = 3; break; @@ -292,7 +303,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, spin_unlock_irqrestore(&chip->lock, flags); return retval; } - + static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, unsigned char value) { @@ -341,7 +352,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, - long wss_base, + long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) { @@ -354,16 +365,23 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, switch (chip->hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C924: + /* opti 929 mode (?), OPL3 clock output, audio enable */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); + /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); case OPTi9XX_HW_82C925: + /* enable WSS mode */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + /* OPL3 FM synthesis */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x00, 0x20); + /* disable Sound Blaster IRQ and DMA */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); #ifdef CS4231 + /* cs4231/4248 fix enabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); #else + /* cs4231/4248 fix disabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x00, 0x02); #endif /* CS4231 */ break; @@ -411,21 +429,26 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, return -EINVAL; } - switch (wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (port & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", port); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -487,7 +510,7 @@ __skip_base: #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, wss_base); + outb(irq_bits << 3 | dma_bits, chip->wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ @@ -729,15 +752,15 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, { struct pnp_dev *pdev; int err; + struct pnp_dev *devmpu; +#ifndef OPTi93X + struct pnp_dev *devmc; +#endif - chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (chip->dev == NULL) + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - - pdev = chip->dev; - err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); @@ -750,9 +773,24 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else - if (pid->driver_data != 0x0924) - port = pnp_port_start(pdev, 1); + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); + return err; + } + + port = pnp_port_start(pdev, 1); fm_port = pnp_port_start(pdev, 2) + 8; + /* + * The MC(0) is never accessed and card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; #endif /* OPTi93X */ irq = pnp_irq(pdev, 0); dma1 = pnp_dma(pdev, 0); @@ -760,16 +798,16 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, dma2 = pnp_dma(pdev, 1); #endif /* CS4231 || OPTi93X */ - pdev = chip->devmpu; - if (pdev && mpu_port > 0) { - err = pnp_activate_dev(pdev); + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + + if (devmpu && mpu_port > 0) { + err = pnp_activate_dev(devmpu); if (err < 0) { - snd_printk(KERN_ERR "AUDIO pnp configure failure\n"); + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); mpu_port = -1; - chip->devmpu = NULL; } else { - mpu_port = pnp_port_start(pdev, 0); - mpu_irq = pnp_irq(pdev, 0); + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); } } return pid->driver_data; @@ -824,7 +862,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error) return error; - error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, + error = snd_wss_create(card, chip->wss_base + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -865,10 +903,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, port + 4, irq, dma1, xdma2); + card->shortname, pcm->name, + chip->wss_base + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, port + 4, irq, dma1); + card->shortname, pcm->name, chip->wss_base + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) @@ -1062,9 +1101,6 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, snd_card_free(card); return error; } - if (hw <= OPTi9XX_HW_82C930) - chip->mc_base -= 0x80; - error = snd_opti9xx_read_check(chip); if (error) { snd_printk(KERN_ERR "OPTI chip not found\n"); -- cgit v1.2.2 From faf4eb23d5fcb9a4728766a1e7bce9c6f2b43bd8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 3 Mar 2010 09:16:18 +0100 Subject: ALSA: oxygen: change || to && In the original code the condition was always true (hopefully) because WM8776_HPLVOL is zero. Signed-off-by: Dan Carpenter Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 7754db166d9e..dbc4b89d74e4 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -68,7 +68,7 @@ static void wm8776_write(struct oxygen *chip, OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); if (reg < ARRAY_SIZE(data->wm8776_regs)) { - if (reg >= WM8776_HPLVOL || reg <= WM8776_DACMASTER) + if (reg >= WM8776_HPLVOL && reg <= WM8776_DACMASTER) value &= ~WM8776_UPDATE; data->wm8776_regs[reg] = value; } -- cgit v1.2.2 From b30477d5e2961bfd90ad4146c517871ca8a6bebc Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 3 Mar 2010 11:05:55 +0100 Subject: ALSA: timer - pass real event in snd_timer_notify1() to instance callback Do not use hardcoded SNDRV_TIMER_EVENT_START value. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 8f8b17ac074d..73943651caed 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -393,7 +393,7 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) event == SNDRV_TIMER_EVENT_CONTINUE) resolution = snd_timer_resolution(ti); if (ti->ccallback) - ti->ccallback(ti, SNDRV_TIMER_EVENT_START, &tstamp, resolution); + ti->ccallback(ti, event, &tstamp, resolution); if (ti->flags & SNDRV_TIMER_IFLG_SLAVE) return; timer = ti->timer; -- cgit v1.2.2 From e61e642c2a0dc283c52cec76a223ac0699773633 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 3 Mar 2010 11:11:57 +0100 Subject: ALSA: usbaudio - remove debug "SAMPLE BYTES" printk line Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index ea3eaa53d637..11b0826b8fe6 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2483,7 +2483,6 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, sample_width, sample_bytes); } /* check the format byte size */ - printk(" XXXXX SAMPLE BYTES %d\n", sample_bytes); switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; -- cgit v1.2.2 From 282572b5ab99cf27073210ca60b80dd085e1a469 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 3 Mar 2010 10:13:49 +0300 Subject: ALSA: riptide: clean up while loop If getpaths() returned an odd number this would be a buffer under-run and an endless loop. It turns out that getpaths() can only return even numbers, but let's make it easy for people auditing code. With the new code you don't need to look at getpaths(). This silences a smatch warning. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 960a227eb653..ad4462677615 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1974,9 +1974,9 @@ snd_riptide_proc_read(struct snd_info_entry *entry, } snd_iprintf(buffer, "Paths:\n"); i = getpaths(cif, p); - while (i--) { - snd_iprintf(buffer, "%x->%x ", p[i - 1], p[i]); - i--; + while (i >= 2) { + i -= 2; + snd_iprintf(buffer, "%x->%x ", p[i], p[i + 1]); } snd_iprintf(buffer, "\n"); } -- cgit v1.2.2 From 7445dfc159f90b4bc82fd7d898b53d74520e2f83 Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Wed, 3 Mar 2010 15:05:53 +0800 Subject: ALSA: hda - Support max codecs to 8 for nvidia hda controller Support max codecs to 8 for nvidia hda controller. Change AZX_MAX_CODECS to 8, and add "#define AZX_DEFAULT_CODECS 4" for default driver. Set azx_max_codecs to 8 for nvidia controller. Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1adac8cc9592..b1047570e78d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -267,7 +267,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define RIRB_INT_MASK 0x05 /* STATESTS int mask: S3,SD2,SD1,SD0 */ -#define AZX_MAX_CODECS 4 +#define AZX_MAX_CODECS 8 +#define AZX_DEFAULT_CODECS 4 #define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) /* SD_CTL bits */ @@ -1367,6 +1368,7 @@ static void azx_bus_reset(struct hda_bus *bus) /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { + [AZX_DRIVER_NVIDIA] = 8, [AZX_DRIVER_TERA] = 1, }; @@ -1399,7 +1401,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) codecs = 0; max_slots = azx_max_codecs[chip->driver_type]; if (!max_slots) - max_slots = AZX_MAX_CODECS; + max_slots = AZX_DEFAULT_CODECS; /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { -- cgit v1.2.2 From 25045705d4053925a617ed71c5e4b6888e468765 Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Wed, 3 Mar 2010 15:11:40 +0800 Subject: ALSA: hda - Support NVIDIA MCP89 and GT21x hdmi audio Support nvidia MCP89 and GT21x 8ch hdmi audio. Add some eld support. Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 2 +- sound/pci/hda/Makefile | 2 +- sound/pci/hda/patch_nvhdmi.c | 1038 ++++++++++++++++++++++++++++++++++++++++-- 3 files changed, 990 insertions(+), 52 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 556cff937be7..567348b05b5a 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -157,7 +157,7 @@ config SND_HDA_CODEC_INTELHDMI config SND_HDA_ELD def_bool y - depends on SND_HDA_CODEC_INTELHDMI + depends on SND_HDA_CODEC_INTELHDMI || SND_HDA_CODEC_NVHDMI config SND_HDA_CODEC_CIRRUS bool "Build Cirrus Logic codec support" diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 315a1c4f8998..199f4405b3ad 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -17,7 +17,7 @@ snd-hda-codec-cirrus-objs := patch_cirrus.o snd-hda-codec-ca0110-objs := patch_ca0110.o snd-hda-codec-conexant-objs := patch_conexant.o snd-hda-codec-via-objs := patch_via.o -snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o +snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o hda_eld.o snd-hda-codec-intelhdmi-objs := patch_intelhdmi.o hda_eld.o # common driver diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 6afdab09bab7..1c774f942407 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -32,10 +32,11 @@ /* define below to restrict the supported rates and formats */ /* #define LIMITED_RATE_FMT_SUPPORT */ -struct nvhdmi_spec { - struct hda_multi_out multiout; - - struct hda_pcm pcm_rec; +enum HDACodec { + HDA_CODEC_NVIDIA_MCP7X, + HDA_CODEC_NVIDIA_MCP89, + HDA_CODEC_NVIDIA_GT21X, + HDA_CODEC_INVALID }; #define Nv_VERB_SET_Channel_Allocation 0xF79 @@ -43,15 +44,18 @@ struct nvhdmi_spec { #define Nv_VERB_SET_Audio_Protection_On 0xF98 #define Nv_VERB_SET_Audio_Protection_Off 0xF99 -#define Nv_Master_Convert_nid 0x04 -#define Nv_Master_Pin_nid 0x05 +#define nvhdmi_master_con_nid_7x 0x04 +#define nvhdmi_master_pin_nid_7x 0x05 -static hda_nid_t nvhdmi_convert_nids[4] = { +#define nvhdmi_master_con_nid_89 0x04 +#define nvhdmi_master_pin_nid_89 0x05 + +static hda_nid_t nvhdmi_con_nids_7x[4] = { /*front, rear, clfe, rear_surr */ 0x6, 0x8, 0xa, 0xc, }; -static struct hda_verb nvhdmi_basic_init[] = { +static struct hda_verb nvhdmi_basic_init_7x[] = { /* set audio protect on */ { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, /* enable digital output on pin widget */ @@ -79,6 +83,796 @@ static struct hda_verb nvhdmi_basic_init[] = { (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) #endif +#define NVIDIA_89_HDMI_CVTS 1 +#define NVIDIA_89_HDMI_PINS 1 + +static char *nvhdmi_pcm_names[NVIDIA_89_HDMI_CVTS] = { + "NVIDIA HDMI", +}; + +struct nvhdmi_spec { + int num_cvts; + int num_pins; + hda_nid_t cvt[NVIDIA_89_HDMI_CVTS+1]; /* audio sources */ + hda_nid_t pin[NVIDIA_89_HDMI_PINS+1]; /* audio sinks */ + hda_nid_t pin_cvt[NVIDIA_89_HDMI_PINS+1]; + struct hda_pcm pcm_rec[NVIDIA_89_HDMI_CVTS]; + struct hdmi_eld sink_eld[NVIDIA_89_HDMI_PINS]; + struct hda_multi_out multiout; + unsigned int codec_type; +}; + +struct hdmi_audio_infoframe { + u8 type; /* 0x84 */ + u8 ver; /* 0x01 */ + u8 len; /* 0x0a */ + + u8 checksum; /* PB0 */ + u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ + u8 SS01_SF24; + u8 CXT04; + u8 CA; + u8 LFEPBL01_LSV36_DM_INH7; +}; + +/* + * CEA speaker placement: + * + * FLH FCH FRH + * FLW FL FLC FC FRC FR FRW + * + * LFE + * TC + * + * RL RLC RC RRC RR + * + * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to + * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. + */ +enum cea_speaker_placement { + FL = (1 << 0), /* Front Left */ + FC = (1 << 1), /* Front Center */ + FR = (1 << 2), /* Front Right */ + FLC = (1 << 3), /* Front Left Center */ + FRC = (1 << 4), /* Front Right Center */ + RL = (1 << 5), /* Rear Left */ + RC = (1 << 6), /* Rear Center */ + RR = (1 << 7), /* Rear Right */ + RLC = (1 << 8), /* Rear Left Center */ + RRC = (1 << 9), /* Rear Right Center */ + LFE = (1 << 10), /* Low Frequency Effect */ + FLW = (1 << 11), /* Front Left Wide */ + FRW = (1 << 12), /* Front Right Wide */ + FLH = (1 << 13), /* Front Left High */ + FCH = (1 << 14), /* Front Center High */ + FRH = (1 << 15), /* Front Right High */ + TC = (1 << 16), /* Top Center */ +}; + +/* + * ELD SA bits in the CEA Speaker Allocation data block + */ +static int eld_speaker_allocation_bits[] = { + [0] = FL | FR, + [1] = LFE, + [2] = FC, + [3] = RL | RR, + [4] = RC, + [5] = FLC | FRC, + [6] = RLC | RRC, + /* the following are not defined in ELD yet */ + [7] = FLW | FRW, + [8] = FLH | FRH, + [9] = TC, + [10] = FCH, +}; + +struct cea_channel_speaker_allocation { + int ca_index; + int speakers[8]; + + /* derived values, just for convenience */ + int channels; + int spk_mask; +}; + +/* + * ALSA sequence is: + * + * surround40 surround41 surround50 surround51 surround71 + * ch0 front left = = = = + * ch1 front right = = = = + * ch2 rear left = = = = + * ch3 rear right = = = = + * ch4 LFE center center center + * ch5 LFE LFE + * ch6 side left + * ch7 side right + * + * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} + */ +static int hdmi_channel_mapping[0x32][8] = { + /* stereo */ + [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* 2.1 */ + [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* Dolby Surround */ + [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* surround40 */ + [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, + /* 4ch */ + [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, + /* surround41 */ + [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround50 */ + [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround51 */ + [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, + /* 7.1 */ + [0x13] = { 0x00, 0x11, 0x32, 0x23, 0x64, 0x75, 0x46, 0x57 }, +}; + +/* + * This is an ordered list! + * + * The preceding ones have better chances to be selected by + * hdmi_setup_channel_allocation(). + */ +static struct cea_channel_speaker_allocation channel_allocations[] = { +/* channel: 7 6 5 4 3 2 1 0 */ +{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, + /* 2.1 */ +{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, + /* Dolby Surround */ +{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, + /* surround40 */ +{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, + /* surround41 */ +{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, + /* surround50 */ +{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, + /* surround51 */ +{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, + /* 6.1 */ +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, + +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, +{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, +{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, +}; + +/* + * HDA/HDMI auto parsing + */ + +static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) +{ + int i; + + for (i = 0; nids[i]; i++) + if (nids[i] == nid) + return i; + + snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); + return -EINVAL; +} + +static int nvhdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct nvhdmi_spec *spec = codec->spec; + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int conn_len, curr; + int index; + + if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { + snd_printk(KERN_WARNING + "HDMI: pin %d wcaps %#x " + "does not support connection list\n", + pin_nid, get_wcaps(codec, pin_nid)); + return -EINVAL; + } + + conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, + HDA_MAX_CONNECTIONS); + if (conn_len > 1) + curr = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + else + curr = 0; + + index = hda_node_index(spec->pin, pin_nid); + if (index < 0) + return -EINVAL; + + spec->pin_cvt[index] = conn_list[curr]; + + return 0; +} + +static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) + snd_hdmi_show_eld(eld); +} + +static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + int present = snd_hda_pin_sense(codec, pin_nid); + + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + eld->eld_valid = !!(present & AC_PINSENSE_ELDV); + + if (present & AC_PINSENSE_ELDV) + hdmi_get_show_eld(codec, pin_nid, eld); +} + +static int nvhdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct nvhdmi_spec *spec = codec->spec; + + if (spec->num_pins >= NVIDIA_89_HDMI_PINS) { + snd_printk(KERN_WARNING + "HDMI: no space for pin %d \n", pin_nid); + return -EINVAL; + } + + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); + + spec->pin[spec->num_pins] = pin_nid; + spec->num_pins++; + + /* + * It is assumed that converter nodes come first in the node list and + * hence have been registered and usable now. + */ + return nvhdmi_read_pin_conn(codec, pin_nid); +} + +static int nvhdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) +{ + struct nvhdmi_spec *spec = codec->spec; + + if (spec->num_cvts >= NVIDIA_89_HDMI_CVTS) { + snd_printk(KERN_WARNING + "HDMI: no space for converter %d \n", nid); + return -EINVAL; + } + + spec->cvt[spec->num_cvts] = nid; + spec->num_cvts++; + + return 0; +} + + +static int nvhdmi_parse_codec(struct hda_codec *codec) +{ + hda_nid_t nid; + int i, nodes; + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + if (!nid || nodes < 0) { + snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); + return -EINVAL; + } + + for (i = 0; i < nodes; i++, nid++) { + unsigned int caps; + unsigned int type; + + caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + type = get_wcaps_type(caps); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + switch (type) { + case AC_WID_AUD_OUT: + if (nvhdmi_add_cvt(codec, nid) < 0) + return -EINVAL; + break; + case AC_WID_PIN: + caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) + continue; + if (nvhdmi_add_pin(codec, nid) < 0) + return -EINVAL; + break; + } + } + + /* + * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event + * can be lost and presence sense verb will become inaccurate if the + * HDA link is powered off at hot plug or hw initialization time. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & + AC_PWRST_EPSS)) + codec->bus->power_keep_link_on = 1; +#endif + + return 0; +} + +/* + * HDMI routines + */ + +#ifdef BE_PARANOID +static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, + int *packet_index, int *byte_index) +{ + int val; + + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_INDEX, 0); + + *packet_index = val >> 5; + *byte_index = val & 0x1f; +} +#endif + +static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, + int packet_index, int byte_index) +{ + int val; + + val = (packet_index << 5) | (byte_index & 0x1f); + + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); +} + +static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, + unsigned char val) +{ + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); +} + +static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) +{ + /* Unmute */ + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + /* Enable pin out */ + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); +} + +/* + * Enable Audio InfoFrame Transmission + */ +static void hdmi_start_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) +{ + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_BEST); +} + +/* + * Disable Audio InfoFrame Transmission + */ +static void hdmi_stop_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) +{ + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_DISABLE); +} + +static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) +{ + return 1 + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CVT_CHAN_COUNT, 0); +} + +static void hdmi_set_channel_count(struct hda_codec *codec, + hda_nid_t nid, int chs) +{ + if (chs != hdmi_get_channel_count(codec, nid)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); +} + +static void hdmi_debug_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int slot; + + for (i = 0; i < 8; i++) { + slot = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_CHAN_SLOT, i); + printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", + slot >> 4, slot & 0xf); + } +#endif +} + + +/* + * Audio InfoFrame routines + */ + +static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int size; + + size = snd_hdmi_get_eld_size(codec, pin_nid); + printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); + + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); + } +#endif +} + +static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) +{ +#ifdef BE_PARANOID + int i, j; + int size; + int pi, bi; + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + if (size == 0) + continue; + + hdmi_set_dip_index(codec, pin_nid, i, 0x0); + for (j = 1; j < 1000; j++) { + hdmi_write_dip_byte(codec, pin_nid, 0x0); + hdmi_get_dip_index(codec, pin_nid, &pi, &bi); + if (pi != i) + snd_printd(KERN_INFO "dip index %d: %d != %d\n", + bi, pi, i); + if (bi == 0) /* byte index wrapped around */ + break; + } + snd_printd(KERN_INFO + "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", + i, size, j); + } +#endif +} + +static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) +{ + ai->checksum = 0; +} + +static void hdmi_fill_audio_infoframe(struct hda_codec *codec, + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + int i; + + hdmi_debug_dip_size(codec, pin_nid); + hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ + + hdmi_checksum_audio_infoframe(ai); + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) + hdmi_write_dip_byte(codec, pin_nid, bytes[i]); +} + +/* + * Compute derived values in channel_allocations[]. + */ +static void init_channel_allocations(void) +{ + int i, j; + struct cea_channel_speaker_allocation *p; + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + p = channel_allocations + i; + p->channels = 0; + p->spk_mask = 0; + for (j = 0; j < ARRAY_SIZE(p->speakers); j++) + if (p->speakers[j]) { + p->channels++; + p->spk_mask |= p->speakers[j]; + } + } +} + +/* + * The transformation takes two steps: + * + * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask + * spk_mask => (channel_allocations[]) => ai->CA + * + * TODO: it could select the wrong CA from multiple candidates. +*/ +static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, + struct hdmi_audio_infoframe *ai) +{ + struct nvhdmi_spec *spec = codec->spec; + struct hdmi_eld *eld; + int i; + int spk_mask = 0; + int channels = 1 + (ai->CC02_CT47 & 0x7); + char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; + + /* + * CA defaults to 0 for basic stereo audio + */ + if (channels <= 2) + return 0; + + i = hda_node_index(spec->pin_cvt, nid); + if (i < 0) + return 0; + eld = &spec->sink_eld[i]; + + /* + * HDMI sink's ELD info cannot always be retrieved for now, e.g. + * in console or for audio devices. Assume the highest speakers + * configuration, to _not_ prohibit multi-channel audio playback. + */ + if (!eld->spk_alloc) + eld->spk_alloc = 0xffff; + + /* + * expand ELD's speaker allocation mask + * + * ELD tells the speaker mask in a compact(paired) form, + * expand ELD's notions to match the ones used by Audio InfoFrame. + */ + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (eld->spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* search for the first working match in the CA table */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels && + (spk_mask & channel_allocations[i].spk_mask) == + channel_allocations[i].spk_mask) { + ai->CA = channel_allocations[i].ca_index; + break; + } + } + + snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); + snd_printdd(KERN_INFO + "HDMI: select CA 0x%x for %d-channel allocation: %s\n", + ai->CA, channels, buf); + + return ai->CA; +} + +static void hdmi_setup_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + int i; + int ca = ai->CA; + int err; + + if (hdmi_channel_mapping[ca][1] == 0) { + for (i = 0; i < channel_allocations[ca].channels; i++) + hdmi_channel_mapping[ca][i] = i | (i << 4); + for (; i < 8; i++) + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + } + + for (i = 0; i < 8; i++) { + err = snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + hdmi_channel_mapping[ca][i]); + if (err) { + snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); + break; + } + } + + hdmi_debug_channel_mapping(codec, pin_nid); +} + +static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 val; + int i; + + if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) + != AC_DIPXMIT_BEST) + return false; + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) { + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_DATA, 0); + if (val != bytes[i]) + return false; + } + + return true; +} + +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, + struct snd_pcm_substream *substream) +{ + struct nvhdmi_spec *spec = codec->spec; + hda_nid_t pin_nid; + int i; + struct hdmi_audio_infoframe ai = { + .type = 0x84, + .ver = 0x01, + .len = 0x0a, + .CC02_CT47 = substream->runtime->channels - 1, + }; + + hdmi_setup_channel_allocation(codec, nid, &ai); + + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != nid) + continue; + if (!spec->sink_eld[i].monitor_present) + continue; + + pin_nid = spec->pin[i]; + if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + hdmi_setup_channel_mapping(codec, pin_nid, &ai); + hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_start_infoframe_trans(codec, pin_nid); + } + } +} + +/* + * Unsolicited events + */ + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + struct nvhdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int pind = !!(res & AC_UNSOL_RES_PD); + int eldv = !!(res & AC_UNSOL_RES_ELDV); + int index; + + printk(KERN_INFO + "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", + tag, pind, eldv); + + index = hda_node_index(spec->pin, tag); + if (index < 0) + return; + + spec->sink_eld[index].monitor_present = pind; + spec->sink_eld[index].eld_valid = eldv; + + if (eldv) { + spec->sink_eld[index].monitor_present = 1; + hdmi_get_show_eld(codec, spec->pin[index], + &spec->sink_eld[index]); + /* TODO: do real things about ELD */ + } +} + +static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); + int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); + + printk(KERN_INFO + "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + tag, + subtag, + cp_state, + cp_ready); + + /* TODO */ + if (cp_state) + ; + if (cp_ready) + ; +} + +static void nvhdmi_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct nvhdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + + if (hda_node_index(spec->pin, tag) < 0) { + snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); + return; + } + + if (subtag == 0) + hdmi_intrinsic_event(codec, res); + else + hdmi_non_intrinsic_event(codec, res); +} + +/* + * Callbacks + */ + +static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, + u32 stream_tag, int format) +{ + int tag; + int fmt; + + tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; + fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); + + snd_printdd("hdmi_setup_stream: " + "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", + nid, + tag == stream_tag ? "" : "new-", + stream_tag, + fmt == format ? "" : "new-", + format); + + if (tag != stream_tag) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + stream_tag << 4); + if (fmt != format) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, format); +} + /* * Controls */ @@ -86,20 +880,58 @@ static int nvhdmi_build_controls(struct hda_codec *codec) { struct nvhdmi_spec *spec = codec->spec; int err; + int i; - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); - if (err < 0) - return err; + if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) + || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { + for (i = 0; i < codec->num_pcms; i++) { + err = snd_hda_create_spdif_out_ctls(codec, + spec->cvt[i]); + if (err < 0) + return err; + } + } else { + err = snd_hda_create_spdif_out_ctls(codec, + spec->multiout.dig_out_nid); + if (err < 0) + return err; + } return 0; } static int nvhdmi_init(struct hda_codec *codec) { - snd_hda_sequence_write(codec, nvhdmi_basic_init); + struct nvhdmi_spec *spec = codec->spec; + int i; + if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) + || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { + for (i = 0; spec->pin[i]; i++) { + hdmi_enable_output(codec, spec->pin[i]); + snd_hda_codec_write(codec, spec->pin[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | spec->pin[i]); + } + } else { + snd_hda_sequence_write(codec, nvhdmi_basic_init_7x); + } return 0; } +static void nvhdmi_free(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec = codec->spec; + int i; + + if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) + || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); + } + + kfree(spec); +} + /* * Digital out */ @@ -111,21 +943,21 @@ static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -static int nvhdmi_dig_playback_pcm_close_8ch(struct hda_pcm_stream *hinfo, +static int nvhdmi_dig_playback_pcm_close_8ch_7x(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct nvhdmi_spec *spec = codec->spec; int i; - snd_hda_codec_write(codec, Nv_Master_Convert_nid, + snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); for (i = 0; i < 4; i++) { /* set the stream id */ - snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0, + snd_hda_codec_write(codec, nvhdmi_con_nids_7x[i], 0, AC_VERB_SET_CHANNEL_STREAMID, 0); /* set the stream format */ - snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0, + snd_hda_codec_write(codec, nvhdmi_con_nids_7x[i], 0, AC_VERB_SET_STREAM_FORMAT, 0); } @@ -140,6 +972,21 @@ static int nvhdmi_dig_playback_pcm_close_2ch(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_dig_close(codec, &spec->multiout); } +static int nvhdmi_dig_playback_pcm_prepare_8ch_89(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + hdmi_set_channel_count(codec, hinfo->nid, + substream->runtime->channels); + + hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); + + hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); + return 0; +} + static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -181,29 +1028,29 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) snd_hda_codec_write(codec, - Nv_Master_Convert_nid, + nvhdmi_master_con_nid_7x, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); /* set the stream id */ - snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0, + snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0, AC_VERB_SET_CHANNEL_STREAMID, (stream_tag << 4) | 0x0); /* set the stream format */ - snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0, + snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0, AC_VERB_SET_STREAM_FORMAT, format); /* turn on again (if needed) */ /* enable and set the channel status audio/data flag */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, - Nv_Master_Convert_nid, + nvhdmi_master_con_nid_7x, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); snd_hda_codec_write(codec, - Nv_Master_Convert_nid, + nvhdmi_master_con_nid_7x, 0, AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); } @@ -220,19 +1067,19 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) snd_hda_codec_write(codec, - nvhdmi_convert_nids[i], + nvhdmi_con_nids_7x[i], 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); /* set the stream id */ snd_hda_codec_write(codec, - nvhdmi_convert_nids[i], + nvhdmi_con_nids_7x[i], 0, AC_VERB_SET_CHANNEL_STREAMID, (stream_tag << 4) | channel_id); /* set the stream format */ snd_hda_codec_write(codec, - nvhdmi_convert_nids[i], + nvhdmi_con_nids_7x[i], 0, AC_VERB_SET_STREAM_FORMAT, format); @@ -241,12 +1088,12 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, - nvhdmi_convert_nids[i], + nvhdmi_con_nids_7x[i], 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); snd_hda_codec_write(codec, - nvhdmi_convert_nids[i], + nvhdmi_con_nids_7x[i], 0, AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); } @@ -261,6 +1108,13 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, return 0; } +static int nvhdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + return 0; +} + static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -272,17 +1126,29 @@ static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, format, substream); } -static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = { +static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = { + .substreams = 1, + .channels_min = 2, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, + .ops = { + .prepare = nvhdmi_dig_playback_pcm_prepare_8ch_89, + .cleanup = nvhdmi_playback_pcm_cleanup, + }, +}; + +static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_7x = { .substreams = 1, .channels_min = 2, .channels_max = 8, - .nid = Nv_Master_Convert_nid, + .nid = nvhdmi_master_con_nid_7x, .rates = SUPPORTED_RATES, .maxbps = SUPPORTED_MAXBPS, .formats = SUPPORTED_FORMATS, .ops = { .open = nvhdmi_dig_playback_pcm_open, - .close = nvhdmi_dig_playback_pcm_close_8ch, + .close = nvhdmi_dig_playback_pcm_close_8ch_7x, .prepare = nvhdmi_dig_playback_pcm_prepare_8ch }, }; @@ -291,7 +1157,7 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .nid = Nv_Master_Convert_nid, + .nid = nvhdmi_master_con_nid_7x, .rates = SUPPORTED_RATES, .maxbps = SUPPORTED_MAXBPS, .formats = SUPPORTED_FORMATS, @@ -302,10 +1168,36 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { }, }; -static int nvhdmi_build_pcms_8ch(struct hda_codec *codec) +static int nvhdmi_build_pcms_8ch_89(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + int i; + + codec->num_pcms = spec->num_cvts; + codec->pcm_info = info; + + for (i = 0; i < codec->num_pcms; i++, info++) { + unsigned int chans; + + chans = get_wcaps(codec, spec->cvt[i]); + chans = get_wcaps_channels(chans); + + info->name = nvhdmi_pcm_names[i]; + info->pcm_type = HDA_PCM_TYPE_HDMI; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] + = nvhdmi_pcm_digital_playback_8ch_89; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->cvt[i]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; + } + + return 0; +} + +static int nvhdmi_build_pcms_8ch_7x(struct hda_codec *codec) { struct nvhdmi_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm_rec; + struct hda_pcm *info = spec->pcm_rec; codec->num_pcms = 1; codec->pcm_info = info; @@ -313,7 +1205,7 @@ static int nvhdmi_build_pcms_8ch(struct hda_codec *codec) info->name = "NVIDIA HDMI"; info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] - = nvhdmi_pcm_digital_playback_8ch; + = nvhdmi_pcm_digital_playback_8ch_7x; return 0; } @@ -321,7 +1213,7 @@ static int nvhdmi_build_pcms_8ch(struct hda_codec *codec) static int nvhdmi_build_pcms_2ch(struct hda_codec *codec) { struct nvhdmi_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm_rec; + struct hda_pcm *info = spec->pcm_rec; codec->num_pcms = 1; codec->pcm_info = info; @@ -334,14 +1226,17 @@ static int nvhdmi_build_pcms_2ch(struct hda_codec *codec) return 0; } -static void nvhdmi_free(struct hda_codec *codec) -{ - kfree(codec->spec); -} +static struct hda_codec_ops nvhdmi_patch_ops_8ch_89 = { + .build_controls = nvhdmi_build_controls, + .build_pcms = nvhdmi_build_pcms_8ch_89, + .init = nvhdmi_init, + .free = nvhdmi_free, + .unsol_event = nvhdmi_unsol_event, +}; -static struct hda_codec_ops nvhdmi_patch_ops_8ch = { +static struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { .build_controls = nvhdmi_build_controls, - .build_pcms = nvhdmi_build_pcms_8ch, + .build_pcms = nvhdmi_build_pcms_8ch_7x, .init = nvhdmi_init, .free = nvhdmi_free, }; @@ -353,7 +1248,34 @@ static struct hda_codec_ops nvhdmi_patch_ops_2ch = { .free = nvhdmi_free, }; -static int patch_nvhdmi_8ch(struct hda_codec *codec) +static int patch_nvhdmi_8ch_89(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec; + int i; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->codec_type = HDA_CODEC_NVIDIA_MCP89; + + if (nvhdmi_parse_codec(codec) < 0) { + codec->spec = NULL; + kfree(spec); + return -EINVAL; + } + codec->patch_ops = nvhdmi_patch_ops_8ch_89; + + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i); + + init_channel_allocations(); + + return 0; +} + +static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) { struct nvhdmi_spec *spec; @@ -365,9 +1287,10 @@ static int patch_nvhdmi_8ch(struct hda_codec *codec) spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 8; - spec->multiout.dig_out_nid = Nv_Master_Convert_nid; + spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; + spec->codec_type = HDA_CODEC_NVIDIA_MCP7X; - codec->patch_ops = nvhdmi_patch_ops_8ch; + codec->patch_ops = nvhdmi_patch_ops_8ch_7x; return 0; } @@ -384,7 +1307,8 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = Nv_Master_Convert_nid; + spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; + spec->codec_type = HDA_CODEC_NVIDIA_MCP7X; codec->patch_ops = nvhdmi_patch_ops_2ch; @@ -395,13 +1319,24 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, - { .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, - { .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, - { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, - { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de0002, .name = "MCP77/78 HDMI", + .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0003, .name = "MCP77/78 HDMI", + .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0005, .name = "MCP77/78 HDMI", + .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0006, .name = "MCP77/78 HDMI", + .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0007, .name = "MCP79/7A HDMI", + .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de000c, .name = "MCP89 HDMI", + .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000b, .name = "GT21x HDMI", + .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000d, .name = "GT240 HDMI", + .patch = patch_nvhdmi_8ch_89 }, {} /* terminator */ }; @@ -412,9 +1347,12 @@ MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); +MODULE_ALIAS("snd-hda-codec-id:10de000c"); +MODULE_ALIAS("snd-hda-codec-id:10de000b"); +MODULE_ALIAS("snd-hda-codec-id:10de000d"); MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Nvidia HDMI HD-audio codec"); +MODULE_DESCRIPTION("NVIDIA HDMI HD-audio codec"); static struct hda_codec_preset_list nvhdmi_list = { .preset = snd_hda_preset_nvhdmi, -- cgit v1.2.2 From dd74b4653597d1d321efa13935cb029b4d819343 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Mar 2010 16:05:24 +0100 Subject: ALSA: hda - Build hda_eld into snd-hda-codec module Now two modules require hda_eld.o, so we need to put it to the common place instead of building into two individual modules. Signed-off-by: Takashi Iwai --- sound/pci/hda/Makefile | 6 +++--- sound/pci/hda/hda_eld.c | 6 ++++++ 2 files changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 199f4405b3ad..24bc195b02da 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -3,7 +3,7 @@ snd-hda-intel-objs := hda_intel.o snd-hda-codec-y := hda_codec.o snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o -# snd-hda-codec-$(CONFIG_SND_HDA_ELD) += hda_eld.o +snd-hda-codec-$(CONFIG_SND_HDA_ELD) += hda_eld.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o @@ -17,8 +17,8 @@ snd-hda-codec-cirrus-objs := patch_cirrus.o snd-hda-codec-ca0110-objs := patch_ca0110.o snd-hda-codec-conexant-objs := patch_conexant.o snd-hda-codec-via-objs := patch_via.o -snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o hda_eld.o -snd-hda-codec-intelhdmi-objs := patch_intelhdmi.o hda_eld.o +snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o +snd-hda-codec-intelhdmi-objs := patch_intelhdmi.o # common driver obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 4228f2fe5956..dcd22446cfc7 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -331,6 +331,7 @@ int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid) return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, AC_DIPSIZE_ELD_BUF); } +EXPORT_SYMBOL_HDA(snd_hdmi_get_eld_size); int snd_hdmi_get_eld(struct hdmi_eld *eld, struct hda_codec *codec, hda_nid_t nid) @@ -366,6 +367,7 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, kfree(buf); return ret; } +EXPORT_SYMBOL_HDA(snd_hdmi_get_eld); static void hdmi_show_short_audio_desc(struct cea_sad *a) { @@ -404,6 +406,7 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) } buf[j] = '\0'; /* necessary when j == 0 */ } +EXPORT_SYMBOL_HDA(snd_print_channel_allocation); void snd_hdmi_show_eld(struct hdmi_eld *e) { @@ -422,6 +425,7 @@ void snd_hdmi_show_eld(struct hdmi_eld *e) for (i = 0; i < e->sad_count; i++) hdmi_show_short_audio_desc(e->sad + i); } +EXPORT_SYMBOL_HDA(snd_hdmi_show_eld); #ifdef CONFIG_PROC_FS @@ -580,6 +584,7 @@ int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, return 0; } +EXPORT_SYMBOL_HDA(snd_hda_eld_proc_new); void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) { @@ -588,5 +593,6 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) eld->proc_entry = NULL; } } +EXPORT_SYMBOL_HDA(snd_hda_eld_proc_free); #endif /* CONFIG_PROC_FS */ -- cgit v1.2.2 From 9919c7619c52d01e89103bca405cc3d4a2b1ac31 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 3 Mar 2010 18:24:26 -0500 Subject: ALSA: hda: Use LPIB for Dell Latitude 131L BugLink: https://launchpad.net/bugs/530346 The OR has verified that position_fix=1 is necessary to work around errors on his machine. Reported-by: Tom Louwrier Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b1047570e78d..531a0b6a66c1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2268,6 +2268,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), -- cgit v1.2.2 From facf92695dcf40836973ce09b7f62d3cc3a89152 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Mar 2010 19:57:59 +0000 Subject: ASoC: Fix S3C64xx IIS driver for Samsung header reorg The reorgs of the Samsung headers have moved the GPIO bank definitions from plat/ to mach/ - the IIS driver needs to be updated to take care of this. Signed-off-by: Mark Brown Signed-off-by: Ben Dooks --- sound/soc/s3c24xx/s3c64xx-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index cc7edb5f792d..22fdb799c883 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -28,8 +28,8 @@ #include #include -#include -#include +#include +#include #include #include -- cgit v1.2.2 From 50152dfaa7d09da85588b66fee7e8c7f541f631d Mon Sep 17 00:00:00 2001 From: Meelis Roos Date: Thu, 4 Mar 2010 20:33:07 +0200 Subject: ALSA: fix jazz16 compile (udelay) While trying to compile jazz16 isa sound driver on alpha (2.6.33+git), I found a compile failure in jazz16.c (udelay is unknown). Fix it by including delay.h. Signed-foo-by: Meelis Roos Signed-off-by: Takashi Iwai --- sound/isa/sb/jazz16.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 8d21a3feda3a..8ccbcddf08e1 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include -- cgit v1.2.2 From 0321b69569eadbc13242922925a4316754c5f744 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 5 Mar 2010 09:04:49 -0500 Subject: ALSA: hda: Use LPIB for a Biostar Microtech board BugLink: https://launchpad.net/bugs/523953 The OR has verified that position_fix=1 is necessary to work around errors on his machine. Reported-by: MMarking Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 531a0b6a66c1..c24bffa08c84 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2273,6 +2273,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), {} }; -- cgit v1.2.2 From d2db09b87eb7b547136d5d25ff1df06820e070bf Mon Sep 17 00:00:00 2001 From: Frederik Deweerdt Date: Fri, 5 Mar 2010 16:34:31 +0100 Subject: ALSA: hda: uninitialized variable fix Commit eaa9b3a748539651f50e3a234c8854e1b42a839a introduced the following uninitialized warning: sound/pci/hda/patch_realtek.c: In function 'set_capture_mixer': sound/pci/hda/patch_realtek.c:4928: warning: 'pin' is used uninitialized in this function sound/pci/hda/patch_realtek.c:4918: note: 'pin' was declared here It appears indeed that 'pin' needs to be initialized to 0. Signed-off-by: Frederik Deweerdt Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b9f4689ccd9a..5d2fbb87b871 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4915,7 +4915,7 @@ static void fixup_automic_adc(struct hda_codec *codec) static void fixup_single_adc(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; + hda_nid_t pin = 0; int i; /* search for the input pin; there must be only one */ -- cgit v1.2.2 From 984b3f5746ed2cde3d184651dabf26980f2b66e5 Mon Sep 17 00:00:00 2001 From: Akinobu Mita Date: Fri, 5 Mar 2010 13:41:37 -0800 Subject: bitops: rename for_each_bit() to for_each_set_bit() Rename for_each_bit to for_each_set_bit in the kernel source tree. To permit for_each_clear_bit(), should that ever be added. The patch includes a macro to map the old for_each_bit() onto the new for_each_set_bit(). This is a (very) temporary thing to ease the migration. [akpm@linux-foundation.org: add temporary for_each_bit()] Suggested-by: Alexey Dobriyan Suggested-by: Andrew Morton Signed-off-by: Akinobu Mita Cc: "David S. Miller" Cc: Russell King Cc: David Woodhouse Cc: Artem Bityutskiy Cc: Stephen Rothwell Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/soc/codecs/uda1380.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a2763c2e7348..9cd0a66b7663 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -137,7 +137,7 @@ static void uda1380_flush_work(struct work_struct *work) { int bit, reg; - for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) { + for_each_set_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) { reg = 0x10 + bit; pr_debug("uda1380: flush reg %x val %x:\n", reg, uda1380_read_reg_cache(uda1380_codec, reg)); -- cgit v1.2.2 From 4193d13b2c2b694aa59e629e6daf6269d7922f13 Mon Sep 17 00:00:00 2001 From: Michele Ballabio Date: Sat, 6 Mar 2010 21:06:46 +0100 Subject: ALSA: hda - Add ASRock mobo to MSI blacklist This avoids a lockup at boot. Signed-off-by: Michele Ballabio Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 94b444e6fed3..e37bffec749a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2358,6 +2358,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ + SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ {} }; -- cgit v1.2.2 From f99344fc69c3df46786a39ea4283a4175ea40b3f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Jan 2010 13:59:07 +0000 Subject: mfd: Add a data argument to the WM8350 IRQ free function To better match genirq. Signed-off-by: Mark Brown Signed-off-by: Samuel Ortiz --- sound/soc/codecs/wm8350.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 718ef912e758..079bf745bf05 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1521,8 +1521,8 @@ static int wm8350_remove(struct platform_device *pdev) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); - wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); - wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); priv->hpl.jack = NULL; priv->hpr.jack = NULL; -- cgit v1.2.2 From 59f25070df0325067d7916b467ad15725657fedc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Jan 2010 19:24:25 +0000 Subject: mfd: Update WM8350 drivers for changed interrupt numbers The headphone detect and charger are using the IRQ numbers so need to take account of irq_base with the genirq conversion. I obviously picked the wrong system for initial testing. Signed-off-by: Mark Brown Signed-off-by: Samuel Ortiz --- sound/soc/codecs/wm8350.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 079bf745bf05..df2c6d9617fb 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1349,7 +1349,7 @@ static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) int mask; struct wm8350_jack_data *jack = NULL; - switch (irq) { + switch (irq - wm8350->irq_base) { case WM8350_IRQ_CODEC_JCK_DET_L: jack = &priv->hpl; mask = WM8350_JACK_L_LVL; @@ -1424,7 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); /* Sync status */ - wm8350_hp_jack_handler(irq, priv); + wm8350_hp_jack_handler(irq + wm8350->irq_base, priv); return 0; } -- cgit v1.2.2 From 079d88ccc374d2c1a850b8a83595ba4c907fb3df Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Mon, 8 Mar 2010 10:44:23 +0800 Subject: ALSA: hdmi - merge common code for intelhdmi and nvhdmi Create patch_hdmi.c to hold common code from intelhdmi and nvhdmi. For now the patch_hdmi.c file is simply included by patch_intelhdmi.c and patch_nvhdmi.c, and does not represent a real codec. There are no behavior changes to intelhdmi. However nvhdmi made several changes when copying code out of intelhdmi, which are all reverted in this patch. Wei Ni confirmed that the reverted code actually works fine. Tested-by: Wei Ni Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 845 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/patch_intelhdmi.c | 821 +------------------------------------- sound/pci/hda/patch_nvhdmi.c | 829 ++------------------------------------- 3 files changed, 882 insertions(+), 1613 deletions(-) create mode 100644 sound/pci/hda/patch_hdmi.c (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c new file mode 100644 index 000000000000..b2ab39670dda --- /dev/null +++ b/sound/pci/hda/patch_hdmi.c @@ -0,0 +1,845 @@ +/* + * + * patch_hdmi.c - routines for HDMI/DisplayPort codecs + * + * Copyright(c) 2008-2010 Intel Corporation. All rights reserved. + * + * Authors: + * Wu Fengguang + * + * Maintained by: + * Wu Fengguang + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY + * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License + * for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software Foundation, + * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + + +struct hdmi_spec { + int num_cvts; + int num_pins; + hda_nid_t cvt[MAX_HDMI_CVTS+1]; /* audio sources */ + hda_nid_t pin[MAX_HDMI_PINS+1]; /* audio sinks */ + + /* + * source connection for each pin + */ + hda_nid_t pin_cvt[MAX_HDMI_PINS+1]; + + /* + * HDMI sink attached to each pin + */ + struct hdmi_eld sink_eld[MAX_HDMI_PINS]; + + /* + * export one pcm per pipe + */ + struct hda_pcm pcm_rec[MAX_HDMI_CVTS]; + + /* + * nvhdmi specific + */ + struct hda_multi_out multiout; + unsigned int codec_type; +}; + + +struct hdmi_audio_infoframe { + u8 type; /* 0x84 */ + u8 ver; /* 0x01 */ + u8 len; /* 0x0a */ + + u8 checksum; /* PB0 */ + u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ + u8 SS01_SF24; + u8 CXT04; + u8 CA; + u8 LFEPBL01_LSV36_DM_INH7; + u8 reserved[5]; /* PB6 - PB10 */ +}; + +/* + * CEA speaker placement: + * + * FLH FCH FRH + * FLW FL FLC FC FRC FR FRW + * + * LFE + * TC + * + * RL RLC RC RRC RR + * + * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to + * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. + */ +enum cea_speaker_placement { + FL = (1 << 0), /* Front Left */ + FC = (1 << 1), /* Front Center */ + FR = (1 << 2), /* Front Right */ + FLC = (1 << 3), /* Front Left Center */ + FRC = (1 << 4), /* Front Right Center */ + RL = (1 << 5), /* Rear Left */ + RC = (1 << 6), /* Rear Center */ + RR = (1 << 7), /* Rear Right */ + RLC = (1 << 8), /* Rear Left Center */ + RRC = (1 << 9), /* Rear Right Center */ + LFE = (1 << 10), /* Low Frequency Effect */ + FLW = (1 << 11), /* Front Left Wide */ + FRW = (1 << 12), /* Front Right Wide */ + FLH = (1 << 13), /* Front Left High */ + FCH = (1 << 14), /* Front Center High */ + FRH = (1 << 15), /* Front Right High */ + TC = (1 << 16), /* Top Center */ +}; + +/* + * ELD SA bits in the CEA Speaker Allocation data block + */ +static int eld_speaker_allocation_bits[] = { + [0] = FL | FR, + [1] = LFE, + [2] = FC, + [3] = RL | RR, + [4] = RC, + [5] = FLC | FRC, + [6] = RLC | RRC, + /* the following are not defined in ELD yet */ + [7] = FLW | FRW, + [8] = FLH | FRH, + [9] = TC, + [10] = FCH, +}; + +struct cea_channel_speaker_allocation { + int ca_index; + int speakers[8]; + + /* derived values, just for convenience */ + int channels; + int spk_mask; +}; + +/* + * ALSA sequence is: + * + * surround40 surround41 surround50 surround51 surround71 + * ch0 front left = = = = + * ch1 front right = = = = + * ch2 rear left = = = = + * ch3 rear right = = = = + * ch4 LFE center center center + * ch5 LFE LFE + * ch6 side left + * ch7 side right + * + * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} + */ +static int hdmi_channel_mapping[0x32][8] = { + /* stereo */ + [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* 2.1 */ + [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* Dolby Surround */ + [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* surround40 */ + [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, + /* 4ch */ + [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, + /* surround41 */ + [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround50 */ + [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround51 */ + [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, + /* 7.1 */ + [0x13] = { 0x00, 0x11, 0x26, 0x37, 0x43, 0x52, 0x64, 0x75 }, +}; + +/* + * This is an ordered list! + * + * The preceding ones have better chances to be selected by + * hdmi_setup_channel_allocation(). + */ +static struct cea_channel_speaker_allocation channel_allocations[] = { +/* channel: 7 6 5 4 3 2 1 0 */ +{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, + /* 2.1 */ +{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, + /* Dolby Surround */ +{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, + /* surround40 */ +{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, + /* surround41 */ +{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, + /* surround50 */ +{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, + /* surround51 */ +{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, + /* 6.1 */ +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, + +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, +{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, +{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, +}; + + +/* + * HDMI routines + */ + +static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) +{ + int i; + + for (i = 0; nids[i]; i++) + if (nids[i] == nid) + return i; + + snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); + return -EINVAL; +} + +static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) + snd_hdmi_show_eld(eld); +} + +#ifdef BE_PARANOID +static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, + int *packet_index, int *byte_index) +{ + int val; + + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_INDEX, 0); + + *packet_index = val >> 5; + *byte_index = val & 0x1f; +} +#endif + +static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, + int packet_index, int byte_index) +{ + int val; + + val = (packet_index << 5) | (byte_index & 0x1f); + + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); +} + +static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, + unsigned char val) +{ + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); +} + +static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) +{ + /* Unmute */ + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + /* Enable pin out */ + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); +} + +static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) +{ + return 1 + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CVT_CHAN_COUNT, 0); +} + +static void hdmi_set_channel_count(struct hda_codec *codec, + hda_nid_t nid, int chs) +{ + if (chs != hdmi_get_channel_count(codec, nid)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); +} + + +/* + * Channel mapping routines + */ + +/* + * Compute derived values in channel_allocations[]. + */ +static void init_channel_allocations(void) +{ + int i, j; + struct cea_channel_speaker_allocation *p; + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + p = channel_allocations + i; + p->channels = 0; + p->spk_mask = 0; + for (j = 0; j < ARRAY_SIZE(p->speakers); j++) + if (p->speakers[j]) { + p->channels++; + p->spk_mask |= p->speakers[j]; + } + } +} + +/* + * The transformation takes two steps: + * + * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask + * spk_mask => (channel_allocations[]) => ai->CA + * + * TODO: it could select the wrong CA from multiple candidates. +*/ +static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, + struct hdmi_audio_infoframe *ai) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_eld *eld; + int i; + int spk_mask = 0; + int channels = 1 + (ai->CC02_CT47 & 0x7); + char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; + + /* + * CA defaults to 0 for basic stereo audio + */ + if (channels <= 2) + return 0; + + i = hda_node_index(spec->pin_cvt, nid); + if (i < 0) + return 0; + eld = &spec->sink_eld[i]; + + /* + * HDMI sink's ELD info cannot always be retrieved for now, e.g. + * in console or for audio devices. Assume the highest speakers + * configuration, to _not_ prohibit multi-channel audio playback. + */ + if (!eld->spk_alloc) + eld->spk_alloc = 0xffff; + + /* + * expand ELD's speaker allocation mask + * + * ELD tells the speaker mask in a compact(paired) form, + * expand ELD's notions to match the ones used by Audio InfoFrame. + */ + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (eld->spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* search for the first working match in the CA table */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels && + (spk_mask & channel_allocations[i].spk_mask) == + channel_allocations[i].spk_mask) { + ai->CA = channel_allocations[i].ca_index; + break; + } + } + + snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); + snd_printdd(KERN_INFO + "HDMI: select CA 0x%x for %d-channel allocation: %s\n", + ai->CA, channels, buf); + + return ai->CA; +} + +static void hdmi_debug_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int slot; + + for (i = 0; i < 8; i++) { + slot = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_CHAN_SLOT, i); + printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", + slot >> 4, slot & 0xf); + } +#endif +} + + +static void hdmi_setup_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + int i; + int ca = ai->CA; + int err; + + if (hdmi_channel_mapping[ca][1] == 0) { + for (i = 0; i < channel_allocations[ca].channels; i++) + hdmi_channel_mapping[ca][i] = i | (i << 4); + for (; i < 8; i++) + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + } + + for (i = 0; i < 8; i++) { + err = snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + hdmi_channel_mapping[ca][i]); + if (err) { + snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); + break; + } + } + + hdmi_debug_channel_mapping(codec, pin_nid); +} + + +/* + * Audio InfoFrame routines + */ + +/* + * Enable Audio InfoFrame Transmission + */ +static void hdmi_start_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) +{ + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_BEST); +} + +/* + * Disable Audio InfoFrame Transmission + */ +static void hdmi_stop_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) +{ + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_DISABLE); +} + +static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int size; + + size = snd_hdmi_get_eld_size(codec, pin_nid); + printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); + + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); + } +#endif +} + +static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) +{ +#ifdef BE_PARANOID + int i, j; + int size; + int pi, bi; + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + if (size == 0) + continue; + + hdmi_set_dip_index(codec, pin_nid, i, 0x0); + for (j = 1; j < 1000; j++) { + hdmi_write_dip_byte(codec, pin_nid, 0x0); + hdmi_get_dip_index(codec, pin_nid, &pi, &bi); + if (pi != i) + snd_printd(KERN_INFO "dip index %d: %d != %d\n", + bi, pi, i); + if (bi == 0) /* byte index wrapped around */ + break; + } + snd_printd(KERN_INFO + "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", + i, size, j); + } +#endif +} + +static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 sum = 0; + int i; + + ai->checksum = 0; + + for (i = 0; i < sizeof(*ai); i++) + sum += bytes[i]; + + ai->checksum = -sum; +} + +static void hdmi_fill_audio_infoframe(struct hda_codec *codec, + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + int i; + + hdmi_debug_dip_size(codec, pin_nid); + hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ + + hdmi_checksum_audio_infoframe(ai); + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) + hdmi_write_dip_byte(codec, pin_nid, bytes[i]); +} + +static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 val; + int i; + + if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) + != AC_DIPXMIT_BEST) + return false; + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) { + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_DATA, 0); + if (val != bytes[i]) + return false; + } + + return true; +} + +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t pin_nid; + int i; + struct hdmi_audio_infoframe ai = { + .type = 0x84, + .ver = 0x01, + .len = 0x0a, + .CC02_CT47 = substream->runtime->channels - 1, + }; + + hdmi_setup_channel_allocation(codec, nid, &ai); + + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != nid) + continue; + if (!spec->sink_eld[i].monitor_present) + continue; + + pin_nid = spec->pin[i]; + if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + hdmi_setup_channel_mapping(codec, pin_nid, &ai); + hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_start_infoframe_trans(codec, pin_nid); + } + } +} + + +/* + * Unsolicited events + */ + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + struct hdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int pind = !!(res & AC_UNSOL_RES_PD); + int eldv = !!(res & AC_UNSOL_RES_ELDV); + int index; + + printk(KERN_INFO + "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", + tag, pind, eldv); + + index = hda_node_index(spec->pin, tag); + if (index < 0) + return; + + spec->sink_eld[index].monitor_present = pind; + spec->sink_eld[index].eld_valid = eldv; + + if (pind && eldv) { + hdmi_get_show_eld(codec, spec->pin[index], + &spec->sink_eld[index]); + /* TODO: do real things about ELD */ + } +} + +static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); + int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); + + printk(KERN_INFO + "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + tag, + subtag, + cp_state, + cp_ready); + + /* TODO */ + if (cp_state) + ; + if (cp_ready) + ; +} + + +static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct hdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + + if (hda_node_index(spec->pin, tag) < 0) { + snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); + return; + } + + if (subtag == 0) + hdmi_intrinsic_event(codec, res); + else + hdmi_non_intrinsic_event(codec, res); +} + +/* + * Callbacks + */ + +static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, + u32 stream_tag, int format) +{ + int tag; + int fmt; + + tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; + fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); + + snd_printdd("hdmi_setup_stream: " + "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", + nid, + tag == stream_tag ? "" : "new-", + stream_tag, + fmt == format ? "" : "new-", + format); + + if (tag != stream_tag) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + stream_tag << 4); + if (fmt != format) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, format); +} + +/* + * HDA/HDMI auto parsing + */ + +static int hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int conn_len, curr; + int index; + + if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { + snd_printk(KERN_WARNING + "HDMI: pin %d wcaps %#x " + "does not support connection list\n", + pin_nid, get_wcaps(codec, pin_nid)); + return -EINVAL; + } + + conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, + HDA_MAX_CONNECTIONS); + if (conn_len > 1) + curr = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + else + curr = 0; + + index = hda_node_index(spec->pin, pin_nid); + if (index < 0) + return -EINVAL; + + spec->pin_cvt[index] = conn_list[curr]; + + return 0; +} + +static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + int present = snd_hda_pin_sense(codec, pin_nid); + + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + eld->eld_valid = !!(present & AC_PINSENSE_ELDV); + + if (present & AC_PINSENSE_ELDV) + hdmi_get_show_eld(codec, pin_nid, eld); +} + +static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec->num_pins >= MAX_HDMI_PINS) { + snd_printk(KERN_WARNING + "HDMI: no space for pin %d\n", pin_nid); + return -EINVAL; + } + + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); + + spec->pin[spec->num_pins] = pin_nid; + spec->num_pins++; + + /* + * It is assumed that converter nodes come first in the node list and + * hence have been registered and usable now. + */ + return hdmi_read_pin_conn(codec, pin_nid); +} + +static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec->num_cvts >= MAX_HDMI_CVTS) { + snd_printk(KERN_WARNING + "HDMI: no space for converter %d\n", nid); + return -EINVAL; + } + + spec->cvt[spec->num_cvts] = nid; + spec->num_cvts++; + + return 0; +} + +static int hdmi_parse_codec(struct hda_codec *codec) +{ + hda_nid_t nid; + int i, nodes; + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + if (!nid || nodes < 0) { + snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); + return -EINVAL; + } + + for (i = 0; i < nodes; i++, nid++) { + unsigned int caps; + unsigned int type; + + caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + type = get_wcaps_type(caps); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + switch (type) { + case AC_WID_AUD_OUT: + if (hdmi_add_cvt(codec, nid) < 0) + return -EINVAL; + break; + case AC_WID_PIN: + caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) + continue; + if (hdmi_add_pin(codec, nid) < 0) + return -EINVAL; + break; + } + } + + /* + * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event + * can be lost and presence sense verb will become inaccurate if the + * HDA link is powered off at hot plug or hw initialization time. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & + AC_PWRST_EPSS)) + codec->bus->power_keep_link_on = 1; +#endif + + return 0; +} + diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 918f40378d52..88d035104cc5 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -40,815 +40,20 @@ * * The HDA correspondence of pipes/ports are converter/pin nodes. */ -#define INTEL_HDMI_CVTS 2 -#define INTEL_HDMI_PINS 3 +#define MAX_HDMI_CVTS 2 +#define MAX_HDMI_PINS 3 -static char *intel_hdmi_pcm_names[INTEL_HDMI_CVTS] = { +#include "patch_hdmi.c" + +static char *intel_hdmi_pcm_names[MAX_HDMI_CVTS] = { "INTEL HDMI 0", "INTEL HDMI 1", }; -struct intel_hdmi_spec { - int num_cvts; - int num_pins; - hda_nid_t cvt[INTEL_HDMI_CVTS+1]; /* audio sources */ - hda_nid_t pin[INTEL_HDMI_PINS+1]; /* audio sinks */ - - /* - * source connection for each pin - */ - hda_nid_t pin_cvt[INTEL_HDMI_PINS+1]; - - /* - * HDMI sink attached to each pin - */ - struct hdmi_eld sink_eld[INTEL_HDMI_PINS]; - - /* - * export one pcm per pipe - */ - struct hda_pcm pcm_rec[INTEL_HDMI_CVTS]; -}; - -struct hdmi_audio_infoframe { - u8 type; /* 0x84 */ - u8 ver; /* 0x01 */ - u8 len; /* 0x0a */ - - u8 checksum; /* PB0 */ - u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ - u8 SS01_SF24; - u8 CXT04; - u8 CA; - u8 LFEPBL01_LSV36_DM_INH7; - u8 reserved[5]; /* PB6 - PB10 */ -}; - -/* - * CEA speaker placement: - * - * FLH FCH FRH - * FLW FL FLC FC FRC FR FRW - * - * LFE - * TC - * - * RL RLC RC RRC RR - * - * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to - * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. - */ -enum cea_speaker_placement { - FL = (1 << 0), /* Front Left */ - FC = (1 << 1), /* Front Center */ - FR = (1 << 2), /* Front Right */ - FLC = (1 << 3), /* Front Left Center */ - FRC = (1 << 4), /* Front Right Center */ - RL = (1 << 5), /* Rear Left */ - RC = (1 << 6), /* Rear Center */ - RR = (1 << 7), /* Rear Right */ - RLC = (1 << 8), /* Rear Left Center */ - RRC = (1 << 9), /* Rear Right Center */ - LFE = (1 << 10), /* Low Frequency Effect */ - FLW = (1 << 11), /* Front Left Wide */ - FRW = (1 << 12), /* Front Right Wide */ - FLH = (1 << 13), /* Front Left High */ - FCH = (1 << 14), /* Front Center High */ - FRH = (1 << 15), /* Front Right High */ - TC = (1 << 16), /* Top Center */ -}; - -/* - * ELD SA bits in the CEA Speaker Allocation data block - */ -static int eld_speaker_allocation_bits[] = { - [0] = FL | FR, - [1] = LFE, - [2] = FC, - [3] = RL | RR, - [4] = RC, - [5] = FLC | FRC, - [6] = RLC | RRC, - /* the following are not defined in ELD yet */ - [7] = FLW | FRW, - [8] = FLH | FRH, - [9] = TC, - [10] = FCH, -}; - -struct cea_channel_speaker_allocation { - int ca_index; - int speakers[8]; - - /* derived values, just for convenience */ - int channels; - int spk_mask; -}; - -/* - * ALSA sequence is: - * - * surround40 surround41 surround50 surround51 surround71 - * ch0 front left = = = = - * ch1 front right = = = = - * ch2 rear left = = = = - * ch3 rear right = = = = - * ch4 LFE center center center - * ch5 LFE LFE - * ch6 side left - * ch7 side right - * - * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} - */ -static int hdmi_channel_mapping[0x32][8] = { - /* stereo */ - [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* 2.1 */ - [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* Dolby Surround */ - [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* surround40 */ - [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, - /* 4ch */ - [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, - /* surround41 */ - [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround50 */ - [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround51 */ - [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, - /* 7.1 */ - [0x13] = { 0x00, 0x11, 0x26, 0x37, 0x43, 0x52, 0x64, 0x75 }, -}; - -/* - * This is an ordered list! - * - * The preceding ones have better chances to be selected by - * hdmi_setup_channel_allocation(). - */ -static struct cea_channel_speaker_allocation channel_allocations[] = { -/* channel: 7 6 5 4 3 2 1 0 */ -{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, - /* 2.1 */ -{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, - /* Dolby Surround */ -{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, - /* surround40 */ -{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, - /* surround41 */ -{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, - /* surround50 */ -{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, - /* surround51 */ -{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, -{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, -{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, -{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, -}; - -/* - * HDA/HDMI auto parsing - */ - -static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) -{ - int i; - - for (i = 0; nids[i]; i++) - if (nids[i] == nid) - return i; - - snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); - return -EINVAL; -} - -static int intel_hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct intel_hdmi_spec *spec = codec->spec; - hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; - int conn_len, curr; - int index; - - if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { - snd_printk(KERN_WARNING - "HDMI: pin %d wcaps %#x " - "does not support connection list\n", - pin_nid, get_wcaps(codec, pin_nid)); - return -EINVAL; - } - - conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, - HDA_MAX_CONNECTIONS); - if (conn_len > 1) - curr = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONNECT_SEL, 0); - else - curr = 0; - - index = hda_node_index(spec->pin, pin_nid); - if (index < 0) - return -EINVAL; - - spec->pin_cvt[index] = conn_list[curr]; - - return 0; -} - -static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - if (!snd_hdmi_get_eld(eld, codec, pin_nid)) - snd_hdmi_show_eld(eld); -} - -static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - int present = snd_hda_pin_sense(codec, pin_nid); - - eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); - eld->eld_valid = !!(present & AC_PINSENSE_ELDV); - - if (present & AC_PINSENSE_ELDV) - hdmi_get_show_eld(codec, pin_nid, eld); -} - -static int intel_hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct intel_hdmi_spec *spec = codec->spec; - - if (spec->num_pins >= INTEL_HDMI_PINS) { - snd_printk(KERN_WARNING - "HDMI: no space for pin %d \n", pin_nid); - return -EINVAL; - } - - hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); - - spec->pin[spec->num_pins] = pin_nid; - spec->num_pins++; - - /* - * It is assumed that converter nodes come first in the node list and - * hence have been registered and usable now. - */ - return intel_hdmi_read_pin_conn(codec, pin_nid); -} - -static int intel_hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) -{ - struct intel_hdmi_spec *spec = codec->spec; - - if (spec->num_cvts >= INTEL_HDMI_CVTS) { - snd_printk(KERN_WARNING - "HDMI: no space for converter %d \n", nid); - return -EINVAL; - } - - spec->cvt[spec->num_cvts] = nid; - spec->num_cvts++; - - return 0; -} - -static int intel_hdmi_parse_codec(struct hda_codec *codec) -{ - hda_nid_t nid; - int i, nodes; - - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); - if (!nid || nodes < 0) { - snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); - return -EINVAL; - } - - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; - - caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); - type = get_wcaps_type(caps); - - if (!(caps & AC_WCAP_DIGITAL)) - continue; - - switch (type) { - case AC_WID_AUD_OUT: - if (intel_hdmi_add_cvt(codec, nid) < 0) - return -EINVAL; - break; - case AC_WID_PIN: - caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) - continue; - if (intel_hdmi_add_pin(codec, nid) < 0) - return -EINVAL; - break; - } - } - - /* - * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event - * can be lost and presence sense verb will become inaccurate if the - * HDA link is powered off at hot plug or hw initialization time. - */ -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & - AC_PWRST_EPSS)) - codec->bus->power_keep_link_on = 1; -#endif - - return 0; -} - -/* - * HDMI routines - */ - -#ifdef BE_PARANOID -static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int *packet_index, int *byte_index) -{ - int val; - - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_INDEX, 0); - - *packet_index = val >> 5; - *byte_index = val & 0x1f; -} -#endif - -static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int packet_index, int byte_index) -{ - int val; - - val = (packet_index << 5) | (byte_index & 0x1f); - - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); -} - -static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, - unsigned char val) -{ - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); -} - -static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) -{ - /* Unmute */ - if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable pin out */ - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); -} - -/* - * Enable Audio InfoFrame Transmission - */ -static void hdmi_start_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_BEST); -} - -/* - * Disable Audio InfoFrame Transmission - */ -static void hdmi_stop_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_DISABLE); -} - -static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) -{ - return 1 + snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CVT_CHAN_COUNT, 0); -} - -static void hdmi_set_channel_count(struct hda_codec *codec, - hda_nid_t nid, int chs) -{ - if (chs != hdmi_get_channel_count(codec, nid)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); -} - -static void hdmi_debug_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int slot; - - for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_CHAN_SLOT, i); - printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", - slot >> 4, slot & 0xf); - } -#endif -} - - -/* - * Audio InfoFrame routines - */ - -static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int size; - - size = snd_hdmi_get_eld_size(codec, pin_nid); - printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); - - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); - } -#endif -} - -static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef BE_PARANOID - int i, j; - int size; - int pi, bi; - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - if (size == 0) - continue; - - hdmi_set_dip_index(codec, pin_nid, i, 0x0); - for (j = 1; j < 1000; j++) { - hdmi_write_dip_byte(codec, pin_nid, 0x0); - hdmi_get_dip_index(codec, pin_nid, &pi, &bi); - if (pi != i) - snd_printd(KERN_INFO "dip index %d: %d != %d\n", - bi, pi, i); - if (bi == 0) /* byte index wrapped around */ - break; - } - snd_printd(KERN_INFO - "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", - i, size, j); - } -#endif -} - -static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - u8 sum = 0; - int i; - - ai->checksum = 0; - - for (i = 0; i < sizeof(*ai); i++) - sum += bytes[i]; - - ai->checksum = - sum; -} - -static void hdmi_fill_audio_infoframe(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - int i; - - hdmi_debug_dip_size(codec, pin_nid); - hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ - - hdmi_checksum_audio_infoframe(ai); - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) - hdmi_write_dip_byte(codec, pin_nid, bytes[i]); -} - -/* - * Compute derived values in channel_allocations[]. - */ -static void init_channel_allocations(void) -{ - int i, j; - struct cea_channel_speaker_allocation *p; - - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - p = channel_allocations + i; - p->channels = 0; - p->spk_mask = 0; - for (j = 0; j < ARRAY_SIZE(p->speakers); j++) - if (p->speakers[j]) { - p->channels++; - p->spk_mask |= p->speakers[j]; - } - } -} - -/* - * The transformation takes two steps: - * - * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask - * spk_mask => (channel_allocations[]) => ai->CA - * - * TODO: it could select the wrong CA from multiple candidates. -*/ -static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, - struct hdmi_audio_infoframe *ai) -{ - struct intel_hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld; - int i; - int spk_mask = 0; - int channels = 1 + (ai->CC02_CT47 & 0x7); - char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; - - /* - * CA defaults to 0 for basic stereo audio - */ - if (channels <= 2) - return 0; - - i = hda_node_index(spec->pin_cvt, nid); - if (i < 0) - return 0; - eld = &spec->sink_eld[i]; - - /* - * HDMI sink's ELD info cannot always be retrieved for now, e.g. - * in console or for audio devices. Assume the highest speakers - * configuration, to _not_ prohibit multi-channel audio playback. - */ - if (!eld->spk_alloc) - eld->spk_alloc = 0xffff; - - /* - * expand ELD's speaker allocation mask - * - * ELD tells the speaker mask in a compact(paired) form, - * expand ELD's notions to match the ones used by Audio InfoFrame. - */ - for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { - if (eld->spk_alloc & (1 << i)) - spk_mask |= eld_speaker_allocation_bits[i]; - } - - /* search for the first working match in the CA table */ - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - if (channels == channel_allocations[i].channels && - (spk_mask & channel_allocations[i].spk_mask) == - channel_allocations[i].spk_mask) { - ai->CA = channel_allocations[i].ca_index; - break; - } - } - - snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); - snd_printdd(KERN_INFO - "HDMI: select CA 0x%x for %d-channel allocation: %s\n", - ai->CA, channels, buf); - - return ai->CA; -} - -static void hdmi_setup_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - int i; - int ca = ai->CA; - int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); - for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); - } - - for (i = 0; i < 8; i++) { - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); - if (err) { - snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); - break; - } - } - - hdmi_debug_channel_mapping(codec, pin_nid); -} - -static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - u8 val; - int i; - - if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) - != AC_DIPXMIT_BEST) - return false; - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) { - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_DATA, 0); - if (val != bytes[i]) - return false; - } - - return true; -} - -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, - struct snd_pcm_substream *substream) -{ - struct intel_hdmi_spec *spec = codec->spec; - hda_nid_t pin_nid; - int i; - struct hdmi_audio_infoframe ai = { - .type = 0x84, - .ver = 0x01, - .len = 0x0a, - .CC02_CT47 = substream->runtime->channels - 1, - }; - - hdmi_setup_channel_allocation(codec, nid, &ai); - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_cvt[i] != nid) - continue; - if (!spec->sink_eld[i].monitor_present) - continue; - - pin_nid = spec->pin[i]; - if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { - hdmi_setup_channel_mapping(codec, pin_nid, &ai); - hdmi_stop_infoframe_trans(codec, pin_nid); - hdmi_fill_audio_infoframe(codec, pin_nid, &ai); - hdmi_start_infoframe_trans(codec, pin_nid); - } - } -} - - /* - * Unsolicited events + * HDMI callbacks */ -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - struct intel_hdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pind = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); - int index; - - printk(KERN_INFO - "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - tag, pind, eldv); - - index = hda_node_index(spec->pin, tag); - if (index < 0) - return; - - spec->sink_eld[index].monitor_present = pind; - spec->sink_eld[index].eld_valid = eldv; - - if (pind && eldv) { - hdmi_get_show_eld(codec, spec->pin[index], &spec->sink_eld[index]); - /* TODO: do real things about ELD */ - } -} - -static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); - int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); - - printk(KERN_INFO - "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", - tag, - subtag, - cp_state, - cp_ready); - - /* TODO */ - if (cp_state) - ; - if (cp_ready) - ; -} - - -static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct intel_hdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - - if (hda_node_index(spec->pin, tag) < 0) { - snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); - return; - } - - if (subtag == 0) - hdmi_intrinsic_event(codec, res); - else - hdmi_non_intrinsic_event(codec, res); -} - -/* - * Callbacks - */ - -static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, int format) -{ - int tag; - int fmt; - - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); - - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); -} - static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -882,7 +87,7 @@ static struct hda_pcm_stream intel_hdmi_pcm_playback = { static int intel_hdmi_build_pcms(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; int i; @@ -908,7 +113,7 @@ static int intel_hdmi_build_pcms(struct hda_codec *codec) static int intel_hdmi_build_controls(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int err; int i; @@ -923,7 +128,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) static int intel_hdmi_init(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; for (i = 0; spec->pin[i]; i++) { @@ -937,7 +142,7 @@ static int intel_hdmi_init(struct hda_codec *codec) static void intel_hdmi_free(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; for (i = 0; i < spec->num_pins; i++) @@ -951,12 +156,12 @@ static struct hda_codec_ops intel_hdmi_patch_ops = { .free = intel_hdmi_free, .build_pcms = intel_hdmi_build_pcms, .build_controls = intel_hdmi_build_controls, - .unsol_event = intel_hdmi_unsol_event, + .unsol_event = hdmi_unsol_event, }; static int patch_intel_hdmi(struct hda_codec *codec) { - struct intel_hdmi_spec *spec; + struct hdmi_spec *spec; int i; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -964,7 +169,7 @@ static int patch_intel_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - if (intel_hdmi_parse_codec(codec) < 0) { + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 1c774f942407..70669a246902 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -29,6 +29,15 @@ #include "hda_codec.h" #include "hda_local.h" +#define MAX_HDMI_CVTS 1 +#define MAX_HDMI_PINS 1 + +#include "patch_hdmi.c" + +static char *nvhdmi_pcm_names[MAX_HDMI_CVTS] = { + "NVIDIA HDMI", +}; + /* define below to restrict the supported rates and formats */ /* #define LIMITED_RATE_FMT_SUPPORT */ @@ -83,802 +92,12 @@ static struct hda_verb nvhdmi_basic_init_7x[] = { (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) #endif -#define NVIDIA_89_HDMI_CVTS 1 -#define NVIDIA_89_HDMI_PINS 1 - -static char *nvhdmi_pcm_names[NVIDIA_89_HDMI_CVTS] = { - "NVIDIA HDMI", -}; - -struct nvhdmi_spec { - int num_cvts; - int num_pins; - hda_nid_t cvt[NVIDIA_89_HDMI_CVTS+1]; /* audio sources */ - hda_nid_t pin[NVIDIA_89_HDMI_PINS+1]; /* audio sinks */ - hda_nid_t pin_cvt[NVIDIA_89_HDMI_PINS+1]; - struct hda_pcm pcm_rec[NVIDIA_89_HDMI_CVTS]; - struct hdmi_eld sink_eld[NVIDIA_89_HDMI_PINS]; - struct hda_multi_out multiout; - unsigned int codec_type; -}; - -struct hdmi_audio_infoframe { - u8 type; /* 0x84 */ - u8 ver; /* 0x01 */ - u8 len; /* 0x0a */ - - u8 checksum; /* PB0 */ - u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ - u8 SS01_SF24; - u8 CXT04; - u8 CA; - u8 LFEPBL01_LSV36_DM_INH7; -}; - -/* - * CEA speaker placement: - * - * FLH FCH FRH - * FLW FL FLC FC FRC FR FRW - * - * LFE - * TC - * - * RL RLC RC RRC RR - * - * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to - * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. - */ -enum cea_speaker_placement { - FL = (1 << 0), /* Front Left */ - FC = (1 << 1), /* Front Center */ - FR = (1 << 2), /* Front Right */ - FLC = (1 << 3), /* Front Left Center */ - FRC = (1 << 4), /* Front Right Center */ - RL = (1 << 5), /* Rear Left */ - RC = (1 << 6), /* Rear Center */ - RR = (1 << 7), /* Rear Right */ - RLC = (1 << 8), /* Rear Left Center */ - RRC = (1 << 9), /* Rear Right Center */ - LFE = (1 << 10), /* Low Frequency Effect */ - FLW = (1 << 11), /* Front Left Wide */ - FRW = (1 << 12), /* Front Right Wide */ - FLH = (1 << 13), /* Front Left High */ - FCH = (1 << 14), /* Front Center High */ - FRH = (1 << 15), /* Front Right High */ - TC = (1 << 16), /* Top Center */ -}; - -/* - * ELD SA bits in the CEA Speaker Allocation data block - */ -static int eld_speaker_allocation_bits[] = { - [0] = FL | FR, - [1] = LFE, - [2] = FC, - [3] = RL | RR, - [4] = RC, - [5] = FLC | FRC, - [6] = RLC | RRC, - /* the following are not defined in ELD yet */ - [7] = FLW | FRW, - [8] = FLH | FRH, - [9] = TC, - [10] = FCH, -}; - -struct cea_channel_speaker_allocation { - int ca_index; - int speakers[8]; - - /* derived values, just for convenience */ - int channels; - int spk_mask; -}; - -/* - * ALSA sequence is: - * - * surround40 surround41 surround50 surround51 surround71 - * ch0 front left = = = = - * ch1 front right = = = = - * ch2 rear left = = = = - * ch3 rear right = = = = - * ch4 LFE center center center - * ch5 LFE LFE - * ch6 side left - * ch7 side right - * - * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} - */ -static int hdmi_channel_mapping[0x32][8] = { - /* stereo */ - [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* 2.1 */ - [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* Dolby Surround */ - [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* surround40 */ - [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, - /* 4ch */ - [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, - /* surround41 */ - [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround50 */ - [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround51 */ - [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, - /* 7.1 */ - [0x13] = { 0x00, 0x11, 0x32, 0x23, 0x64, 0x75, 0x46, 0x57 }, -}; - -/* - * This is an ordered list! - * - * The preceding ones have better chances to be selected by - * hdmi_setup_channel_allocation(). - */ -static struct cea_channel_speaker_allocation channel_allocations[] = { -/* channel: 7 6 5 4 3 2 1 0 */ -{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, - /* 2.1 */ -{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, - /* Dolby Surround */ -{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, - /* surround40 */ -{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, - /* surround41 */ -{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, - /* surround50 */ -{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, - /* surround51 */ -{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, -{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, -{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, -{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, -}; - -/* - * HDA/HDMI auto parsing - */ - -static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) -{ - int i; - - for (i = 0; nids[i]; i++) - if (nids[i] == nid) - return i; - - snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); - return -EINVAL; -} - -static int nvhdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct nvhdmi_spec *spec = codec->spec; - hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; - int conn_len, curr; - int index; - - if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { - snd_printk(KERN_WARNING - "HDMI: pin %d wcaps %#x " - "does not support connection list\n", - pin_nid, get_wcaps(codec, pin_nid)); - return -EINVAL; - } - - conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, - HDA_MAX_CONNECTIONS); - if (conn_len > 1) - curr = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONNECT_SEL, 0); - else - curr = 0; - - index = hda_node_index(spec->pin, pin_nid); - if (index < 0) - return -EINVAL; - - spec->pin_cvt[index] = conn_list[curr]; - - return 0; -} - -static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - if (!snd_hdmi_get_eld(eld, codec, pin_nid)) - snd_hdmi_show_eld(eld); -} - -static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - int present = snd_hda_pin_sense(codec, pin_nid); - - eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); - eld->eld_valid = !!(present & AC_PINSENSE_ELDV); - - if (present & AC_PINSENSE_ELDV) - hdmi_get_show_eld(codec, pin_nid, eld); -} - -static int nvhdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct nvhdmi_spec *spec = codec->spec; - - if (spec->num_pins >= NVIDIA_89_HDMI_PINS) { - snd_printk(KERN_WARNING - "HDMI: no space for pin %d \n", pin_nid); - return -EINVAL; - } - - hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); - - spec->pin[spec->num_pins] = pin_nid; - spec->num_pins++; - - /* - * It is assumed that converter nodes come first in the node list and - * hence have been registered and usable now. - */ - return nvhdmi_read_pin_conn(codec, pin_nid); -} - -static int nvhdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) -{ - struct nvhdmi_spec *spec = codec->spec; - - if (spec->num_cvts >= NVIDIA_89_HDMI_CVTS) { - snd_printk(KERN_WARNING - "HDMI: no space for converter %d \n", nid); - return -EINVAL; - } - - spec->cvt[spec->num_cvts] = nid; - spec->num_cvts++; - - return 0; -} - - -static int nvhdmi_parse_codec(struct hda_codec *codec) -{ - hda_nid_t nid; - int i, nodes; - - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); - if (!nid || nodes < 0) { - snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); - return -EINVAL; - } - - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; - - caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); - type = get_wcaps_type(caps); - - if (!(caps & AC_WCAP_DIGITAL)) - continue; - - switch (type) { - case AC_WID_AUD_OUT: - if (nvhdmi_add_cvt(codec, nid) < 0) - return -EINVAL; - break; - case AC_WID_PIN: - caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) - continue; - if (nvhdmi_add_pin(codec, nid) < 0) - return -EINVAL; - break; - } - } - - /* - * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event - * can be lost and presence sense verb will become inaccurate if the - * HDA link is powered off at hot plug or hw initialization time. - */ -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & - AC_PWRST_EPSS)) - codec->bus->power_keep_link_on = 1; -#endif - - return 0; -} - -/* - * HDMI routines - */ - -#ifdef BE_PARANOID -static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int *packet_index, int *byte_index) -{ - int val; - - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_INDEX, 0); - - *packet_index = val >> 5; - *byte_index = val & 0x1f; -} -#endif - -static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int packet_index, int byte_index) -{ - int val; - - val = (packet_index << 5) | (byte_index & 0x1f); - - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); -} - -static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, - unsigned char val) -{ - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); -} - -static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) -{ - /* Unmute */ - if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable pin out */ - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); -} - -/* - * Enable Audio InfoFrame Transmission - */ -static void hdmi_start_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_BEST); -} - -/* - * Disable Audio InfoFrame Transmission - */ -static void hdmi_stop_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_DISABLE); -} - -static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) -{ - return 1 + snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CVT_CHAN_COUNT, 0); -} - -static void hdmi_set_channel_count(struct hda_codec *codec, - hda_nid_t nid, int chs) -{ - if (chs != hdmi_get_channel_count(codec, nid)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); -} - -static void hdmi_debug_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int slot; - - for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_CHAN_SLOT, i); - printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", - slot >> 4, slot & 0xf); - } -#endif -} - - -/* - * Audio InfoFrame routines - */ - -static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int size; - - size = snd_hdmi_get_eld_size(codec, pin_nid); - printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); - - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); - } -#endif -} - -static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef BE_PARANOID - int i, j; - int size; - int pi, bi; - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - if (size == 0) - continue; - - hdmi_set_dip_index(codec, pin_nid, i, 0x0); - for (j = 1; j < 1000; j++) { - hdmi_write_dip_byte(codec, pin_nid, 0x0); - hdmi_get_dip_index(codec, pin_nid, &pi, &bi); - if (pi != i) - snd_printd(KERN_INFO "dip index %d: %d != %d\n", - bi, pi, i); - if (bi == 0) /* byte index wrapped around */ - break; - } - snd_printd(KERN_INFO - "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", - i, size, j); - } -#endif -} - -static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) -{ - ai->checksum = 0; -} - -static void hdmi_fill_audio_infoframe(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - int i; - - hdmi_debug_dip_size(codec, pin_nid); - hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ - - hdmi_checksum_audio_infoframe(ai); - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) - hdmi_write_dip_byte(codec, pin_nid, bytes[i]); -} - -/* - * Compute derived values in channel_allocations[]. - */ -static void init_channel_allocations(void) -{ - int i, j; - struct cea_channel_speaker_allocation *p; - - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - p = channel_allocations + i; - p->channels = 0; - p->spk_mask = 0; - for (j = 0; j < ARRAY_SIZE(p->speakers); j++) - if (p->speakers[j]) { - p->channels++; - p->spk_mask |= p->speakers[j]; - } - } -} - -/* - * The transformation takes two steps: - * - * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask - * spk_mask => (channel_allocations[]) => ai->CA - * - * TODO: it could select the wrong CA from multiple candidates. -*/ -static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, - struct hdmi_audio_infoframe *ai) -{ - struct nvhdmi_spec *spec = codec->spec; - struct hdmi_eld *eld; - int i; - int spk_mask = 0; - int channels = 1 + (ai->CC02_CT47 & 0x7); - char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; - - /* - * CA defaults to 0 for basic stereo audio - */ - if (channels <= 2) - return 0; - - i = hda_node_index(spec->pin_cvt, nid); - if (i < 0) - return 0; - eld = &spec->sink_eld[i]; - - /* - * HDMI sink's ELD info cannot always be retrieved for now, e.g. - * in console or for audio devices. Assume the highest speakers - * configuration, to _not_ prohibit multi-channel audio playback. - */ - if (!eld->spk_alloc) - eld->spk_alloc = 0xffff; - - /* - * expand ELD's speaker allocation mask - * - * ELD tells the speaker mask in a compact(paired) form, - * expand ELD's notions to match the ones used by Audio InfoFrame. - */ - for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { - if (eld->spk_alloc & (1 << i)) - spk_mask |= eld_speaker_allocation_bits[i]; - } - - /* search for the first working match in the CA table */ - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - if (channels == channel_allocations[i].channels && - (spk_mask & channel_allocations[i].spk_mask) == - channel_allocations[i].spk_mask) { - ai->CA = channel_allocations[i].ca_index; - break; - } - } - - snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); - snd_printdd(KERN_INFO - "HDMI: select CA 0x%x for %d-channel allocation: %s\n", - ai->CA, channels, buf); - - return ai->CA; -} - -static void hdmi_setup_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - int i; - int ca = ai->CA; - int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); - for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); - } - - for (i = 0; i < 8; i++) { - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); - if (err) { - snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); - break; - } - } - - hdmi_debug_channel_mapping(codec, pin_nid); -} - -static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - u8 val; - int i; - - if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) - != AC_DIPXMIT_BEST) - return false; - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) { - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_DATA, 0); - if (val != bytes[i]) - return false; - } - - return true; -} - -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, - struct snd_pcm_substream *substream) -{ - struct nvhdmi_spec *spec = codec->spec; - hda_nid_t pin_nid; - int i; - struct hdmi_audio_infoframe ai = { - .type = 0x84, - .ver = 0x01, - .len = 0x0a, - .CC02_CT47 = substream->runtime->channels - 1, - }; - - hdmi_setup_channel_allocation(codec, nid, &ai); - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_cvt[i] != nid) - continue; - if (!spec->sink_eld[i].monitor_present) - continue; - - pin_nid = spec->pin[i]; - if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { - hdmi_setup_channel_mapping(codec, pin_nid, &ai); - hdmi_stop_infoframe_trans(codec, pin_nid); - hdmi_fill_audio_infoframe(codec, pin_nid, &ai); - hdmi_start_infoframe_trans(codec, pin_nid); - } - } -} - -/* - * Unsolicited events - */ - -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - struct nvhdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pind = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); - int index; - - printk(KERN_INFO - "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - tag, pind, eldv); - - index = hda_node_index(spec->pin, tag); - if (index < 0) - return; - - spec->sink_eld[index].monitor_present = pind; - spec->sink_eld[index].eld_valid = eldv; - - if (eldv) { - spec->sink_eld[index].monitor_present = 1; - hdmi_get_show_eld(codec, spec->pin[index], - &spec->sink_eld[index]); - /* TODO: do real things about ELD */ - } -} - -static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); - int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); - - printk(KERN_INFO - "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", - tag, - subtag, - cp_state, - cp_ready); - - /* TODO */ - if (cp_state) - ; - if (cp_ready) - ; -} - -static void nvhdmi_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct nvhdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - - if (hda_node_index(spec->pin, tag) < 0) { - snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); - return; - } - - if (subtag == 0) - hdmi_intrinsic_event(codec, res); - else - hdmi_non_intrinsic_event(codec, res); -} - -/* - * Callbacks - */ - -static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, int format) -{ - int tag; - int fmt; - - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); - - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); -} - /* * Controls */ static int nvhdmi_build_controls(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int err; int i; @@ -902,7 +121,7 @@ static int nvhdmi_build_controls(struct hda_codec *codec) static int nvhdmi_init(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { @@ -920,7 +139,7 @@ static int nvhdmi_init(struct hda_codec *codec) static void nvhdmi_free(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) @@ -939,7 +158,7 @@ static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); } @@ -947,7 +166,7 @@ static int nvhdmi_dig_playback_pcm_close_8ch_7x(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, @@ -968,7 +187,7 @@ static int nvhdmi_dig_playback_pcm_close_2ch(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -1121,7 +340,7 @@ static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, unsigned int format, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -1170,7 +389,7 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { static int nvhdmi_build_pcms_8ch_89(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; int i; @@ -1196,7 +415,7 @@ static int nvhdmi_build_pcms_8ch_89(struct hda_codec *codec) static int nvhdmi_build_pcms_8ch_7x(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; codec->num_pcms = 1; @@ -1212,7 +431,7 @@ static int nvhdmi_build_pcms_8ch_7x(struct hda_codec *codec) static int nvhdmi_build_pcms_2ch(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; codec->num_pcms = 1; @@ -1231,7 +450,7 @@ static struct hda_codec_ops nvhdmi_patch_ops_8ch_89 = { .build_pcms = nvhdmi_build_pcms_8ch_89, .init = nvhdmi_init, .free = nvhdmi_free, - .unsol_event = nvhdmi_unsol_event, + .unsol_event = hdmi_unsol_event, }; static struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { @@ -1250,7 +469,7 @@ static struct hda_codec_ops nvhdmi_patch_ops_2ch = { static int patch_nvhdmi_8ch_89(struct hda_codec *codec) { - struct nvhdmi_spec *spec; + struct hdmi_spec *spec; int i; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -1260,7 +479,7 @@ static int patch_nvhdmi_8ch_89(struct hda_codec *codec) codec->spec = spec; spec->codec_type = HDA_CODEC_NVIDIA_MCP89; - if (nvhdmi_parse_codec(codec) < 0) { + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; @@ -1277,7 +496,7 @@ static int patch_nvhdmi_8ch_89(struct hda_codec *codec) static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) { - struct nvhdmi_spec *spec; + struct hdmi_spec *spec; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1297,7 +516,7 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) static int patch_nvhdmi_2ch(struct hda_codec *codec) { - struct nvhdmi_spec *spec; + struct hdmi_spec *spec; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) -- cgit v1.2.2 From 2abbf4391fb56dfa97221ed6796782537d15196f Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Mon, 8 Mar 2010 10:45:38 +0800 Subject: ALSA: hdmi - show debug message on changing audio infoframe Also change printk level for the two others. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b2ab39670dda..2c2bafbf0258 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -398,9 +398,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, } snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); - snd_printdd(KERN_INFO - "HDMI: select CA 0x%x for %d-channel allocation: %s\n", - ai->CA, channels, buf); + snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", + ai->CA, channels, buf); return ai->CA; } @@ -442,7 +441,8 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, AC_VERB_SET_HDMI_CHAN_SLOT, hdmi_channel_mapping[ca][i]); if (err) { - snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); + snd_printdd(KERN_NOTICE + "HDMI: channel mapping failed\n"); break; } } @@ -599,6 +599,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, pin_nid = spec->pin[i]; if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + snd_printdd("hdmi_setup_audio_infoframe: " + "cvt=%d pin=%d channels=%d\n", + nid, pin_nid, + substream->runtime->channels); hdmi_setup_channel_mapping(codec, pin_nid, &ai); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, &ai); -- cgit v1.2.2 From 50ae0aa8f55813b2cc5e5b7f589f328b8fcd45ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Mar 2010 12:09:59 +0100 Subject: ALSA: hda - Fix wrong model range check for ALC268 Fix a wrong value passed to snd_hda_check_board_codec_sid_config() as the upper-limit in parse_alc268(), so that any wrong value can't be passed. So far, no bogus value was set in the quirk entries, so this won't give any behavioral changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5d2fbb87b871..dcd8a2cadd99 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13201,7 +13201,7 @@ static int patch_alc268(struct hda_codec *codec) if (board_config < 0 || board_config >= ALC268_MODEL_LAST) board_config = snd_hda_check_board_codec_sid_config(codec, - ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); + ALC268_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", -- cgit v1.2.2 From 5311114d4867113c00f78829d4ce14be458ec925 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Mar 2010 12:13:07 +0100 Subject: ALSA: hda - Fix input source elements of secondary ADCs on Realtek Since alc_auto_create_input_ctls() doesn't set the elements for the secondary ADCs, "Input Source" elemtns for these also get empty, resulting in buggy outputs of alsactl like: control.14 { comment.access 'read write' comment.type ENUMERATED comment.count 1 iface MIXER name 'Input Source' index 1 value 0 } This patch fixes alc_mux_enum_*() (and others) to fall back to the first entry if the secondary input mux is empty. Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd8a2cadd99..3a8371990d75 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -411,6 +411,8 @@ static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, unsigned int mux_idx = snd_ctl_get_ioffidx(kcontrol, &uinfo->id); if (mux_idx >= spec->num_mux_defs) mux_idx = 0; + if (!spec->input_mux[mux_idx].num_items && mux_idx > 0) + mux_idx = 0; return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo); } @@ -439,6 +441,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; + if (!imux->num_items && mux_idx > 0) + imux = &spec->input_mux[0]; type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_AUD_MIX) { @@ -10105,6 +10109,8 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) continue; mux_idx = c >= spec->num_mux_defs ? 0 : c; imux = &spec->input_mux[mux_idx]; + if (!imux->num_items && mux_idx > 0) + imux = &spec->input_mux[0]; for (idx = 0; idx < conns; idx++) { /* if the current connection is the selected one, * unmute it as default - otherwise mute it -- cgit v1.2.2 From 89c0ac7cab2440a771ba1e2ab953186bc9c29786 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Mon, 8 Mar 2010 09:32:42 -0800 Subject: sound: fix opti92x-ad1848 build Fix 'else' placement in ifdef block so that build succeeds: sound/isa/opti9xx/opti92x-ad1848.c:221: error: 'else' without a previous 'if' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index becd90d7536d..4d2d0405bdc7 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -217,8 +217,9 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, if (isapnp && chip->mc_base) /* PnP resource gives the least 10 bits */ chip->mc_base |= 0xc00; + else #endif /* CONFIG_PNP */ - else { + { chip->mc_base = 0xf8c; chip->mc_base_size = opti9xx_mc_size[hardware]; } -- cgit v1.2.2 From ecd216260f87dd8c14b2580a16f055554644bbea Mon Sep 17 00:00:00 2001 From: Ralf Gerbig Date: Tue, 9 Mar 2010 18:25:47 +0100 Subject: ALSA: hda - Sound MSI fallout on a Asus mobo NVIDIA MCP55 without the following patch audio ssttuutteerrs on ASUS M2N32-SLI PREMIUM ACPI BIOS Revision 1304 the sound device is: 00:0e.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2) worked with 2.6.32 Signed-off-by: Ralf Gerbig Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e37bffec749a..10bbb534d3ca 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2358,6 +2358,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ + SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ {} }; -- cgit v1.2.2 From c602c8ad45d6ee6ad91fc544513cc96f70790983 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Mon, 15 Mar 2010 09:01:26 +0100 Subject: ALSA: hda - New Intel HDA controller Added a PCI controller id on new Dell laptops. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 10bbb534d3ca..926815201885 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2706,6 +2706,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, /* PCH */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x3b57), .driver_data = AZX_DRIVER_ICH }, /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ -- cgit v1.2.2 From 28d1a85e136985982448b2f9b1342bae85ad1c98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Mar 2010 09:05:46 +0100 Subject: ALSA: hda - Add an error message for invalid mapping NID Add an error message to snd_hda_add_nid() for invalid mapping NID to make easier to hunt the buggy code. Also added a missing space to the error message in snd_hda_build_controls() Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5bd7cf45f3a5..0e76ac2b2ace 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1806,6 +1806,8 @@ int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl, item->nid = nid; return 0; } + printk(KERN_ERR "hda-codec: no NID for mapping control %s:%d:%d\n", + kctl->id.name, kctl->id.index, index); return -EINVAL; } EXPORT_SYMBOL_HDA(snd_hda_add_nid); @@ -2884,7 +2886,7 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { int err = snd_hda_codec_build_controls(codec); if (err < 0) { - printk(KERN_ERR "hda_codec: cannot build controls" + printk(KERN_ERR "hda_codec: cannot build controls " "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { -- cgit v1.2.2 From 9c4cc0bdede1c39bde60a0d5d9251aac71fbe719 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Mar 2010 09:07:52 +0100 Subject: ALSA: hda - Fix secondary ADC of ALC260 basic model Fix adc_nids[] for ALC260 basic model to match with num_adc_nids. Otherwise you get an invalid NID in the secondary "Input Source" mixer element. Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a8371990d75..ba45868d5242 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6477,7 +6477,7 @@ static struct alc_config_preset alc260_presets[] = { .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_adc_nids, + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, -- cgit v1.2.2 From b43f6e5e258d67acae5961896d10bbe38c271070 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Wed, 10 Mar 2010 19:17:46 +0100 Subject: ALSA: hda - Add PCI quirks for MSI NetOn AP1900 and Wind Top AE2220 This should make the speakers and jack detection work on MSI all-in-one computers NetOn AP1900 and Wind Top AE2220. Signed-off-by: Anisse Astier Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba45868d5242..07637c4aa46f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9195,6 +9195,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), @@ -9204,6 +9205,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), -- cgit v1.2.2 From 80c43ed724797627d8f86855248c497a6161a214 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Mar 2010 15:51:53 +0100 Subject: ALSA: hda - Disable MSI for Nvidia controller Judging from the member of enable_msi white-list, Nvidia controller seems to cause troubles with MSI enabled, e.g. boot hang up or other serious issue may come up. It's safer to disable MSI as default for Nvidia controllers again for now. Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/hda_intel.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 926815201885..027d3f4c1c59 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2378,6 +2378,13 @@ static void __devinit check_msi(struct azx *chip) "hda_intel: msi for device %04x:%04x set to %d\n", q->subvendor, q->subdevice, q->value); chip->msi = q->value; + return; + } + + /* NVidia chipsets seem to cause troubles with MSI */ + if (chip->driver_type == AZX_DRIVER_NVIDIA) { + printk(KERN_INFO "hda_intel: Disable MSI for Nvidia chipset\n"); + chip->msi = 0; } } -- cgit v1.2.2 From 572c0e3c73341755f3e7dfaaef6b26df12bd709c Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 14 Mar 2010 23:44:03 -0400 Subject: ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212 BugLink: https://bugs.launchpad.net/bugs/538895 The OR has verified that both position_fix=1 and model=6stack-dig are necessary to have capture function properly. (The existing 3stack-6ch model quirk seems to be incorrect.) Reported-by: Reuben Bailey Tested-by: Reuben Bailey Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 027d3f4c1c59..1766ad2926d6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2271,6 +2271,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 07637c4aa46f..4ec57633af88 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9237,7 +9237,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL), SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL), SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL), - SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG), {} }; -- cgit v1.2.2 From fb40b496ad8bbe60a60c25eb2fce20f3cc114679 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 16 Mar 2010 09:46:23 +0300 Subject: sound: sequencer: clean up remove bogus check A few lines earlier bend is limited to 2399. So semitones is always less than 24 here. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sequencer.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index c79874696bec..e85789e53816 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1631,8 +1631,6 @@ unsigned long compute_finetune(unsigned long base_freq, int bend, int range, } semitones = bend / 100; - if (semitones > 99) - semitones = 99; cents = bend % 100; amount = (int) (semitone_tuning[semitones] * multiplier * cent_tuning[cents]) / 10000; -- cgit v1.2.2 From da3b062e306452ffb74cf5e9e5128f9f1e0502ab Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 18 Mar 2010 09:39:59 +0100 Subject: ASoC: SIU driver shall select FW_LOADER The SIU ASoC driver must load firmware to program the DSP, therefore it has to select FW_LOADER in its Kconfig entry. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 106674979b53..f07f6d8b93e1 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_SH4_SIU select DMA_ENGINE select DMADEVICES select SH_DMAE + select FW_LOADER ## ## Boards -- cgit v1.2.2 From 44f497b4e0bba6ce1b73a107cc13636393344252 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 19 Mar 2010 11:10:19 +0200 Subject: ASoC: tlv320dac33: Fix DSP modes To make DSP_A mode working correctly the data delay should be configured to 0. DSP_B mode thus can not be used with DAC33, so remove it. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f9f367d29a90..00d6f36aabc9 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1038,11 +1038,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_DSP_A: aictrl_a |= DAC33_AFMT_DSP; aictrl_b &= ~DAC33_DATA_DELAY_MASK; - aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */ - break; - case SND_SOC_DAIFMT_DSP_B: - aictrl_a |= DAC33_AFMT_DSP; - aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */ + aictrl_b |= DAC33_DATA_DELAY(0); break; case SND_SOC_DAIFMT_RIGHT_J: aictrl_a |= DAC33_AFMT_RIGHT_J; -- cgit v1.2.2 From fdb6b1e195757a66670801702e4b5fcc66ed3d72 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 19 Mar 2010 11:10:20 +0200 Subject: ASoC: tlv320dac33: Internal clocking changes During validation of the internal clocking setup it has been found that the following settings were not configured in an optimal way: ASRC_CTRL_A: SRCLKDIV was incorrect, instad of divide ratio 3, ratio of 2 has to be used (as the comment stated) DAC_CTRL_A: Fs = Fsref is the desired configuration instead of Fs = Fsref / 1.5 Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 00d6f36aabc9..d50f1699ccb2 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -778,7 +778,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) if (dac33->fifo_mode) { /* Generic for all FIFO modes */ /* 50-51 : ASRC Control registers */ - dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCLKDIV(1)); dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ /* Write registers 0x34 and 0x35 (MSB, LSB) */ @@ -1062,7 +1062,7 @@ static void dac33_init_chip(struct snd_soc_codec *codec) { /* 44-46: DAC Control Registers */ /* A : DAC sample rate Fsref/1.5 */ - dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1)); + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0)); /* B : DAC src=normal, not muted */ dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | DAC33_DACSRCL_LEFT); -- cgit v1.2.2 From 6937c947d31186750f72c9f8c942bbcc6fe63585 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Mar 2010 12:25:35 +0000 Subject: ASoC: Bail out of wm_hubs DC servo if calibration fails We're keeping track of the number of times we've iterated but never actually using this to bail out if the chip looks stuck. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 0ad9f5d536c6..486bdd21a98a 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -74,7 +74,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) msleep(1); reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & WM8993_DCS_DATAPATH_BUSY); + } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400); if (reg & WM8993_DCS_DATAPATH_BUSY) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); -- cgit v1.2.2 From 8727b909bb2348d29e62c599cd7a5d610da3760f Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 28 Feb 2010 10:42:38 +0800 Subject: ASoC: pxa-pcm-lib: initialize DMA channel to -1 This fixes a warning ("pxa_free_dma: trying to free channel 0 which is already freed") when a device was opened but the hw_params() call failed. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/arm/pxa2xx-pcm-lib.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 743ac6a29065..fd51fa8b06a1 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -205,6 +205,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (!rtd->dma_desc_array) goto err1; + rtd->dma_ch = -1; runtime->private_data = rtd; return 0; -- cgit v1.2.2 From fc8aa7b16a5fcfe9c6d0be9bb587f1fcedd9145f Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 18 Mar 2010 07:53:11 +0100 Subject: sound/oss/vidc.c: change the field used with DMA_ACTIVE The constant DMA_ACTIVE is defined with the dma_buffparams structure rather than with the audio_operations structure. Takashi Iwai suggested that the dmap_out field of the audio_operations structure should be used instead. This is not tested. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/oss/vidc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index 725fef0f59a3..a4127bab9231 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -363,13 +363,13 @@ static void vidc_audio_trigger(int dev, int enable_bits) struct audio_operations *adev = audio_devs[dev]; if (enable_bits & PCM_ENABLE_OUTPUT) { - if (!(adev->flags & DMA_ACTIVE)) { + if (!(adev->dmap_out->flags & DMA_ACTIVE)) { unsigned long flags; local_irq_save(flags); /* prevent recusion */ - adev->flags |= DMA_ACTIVE; + adev->dmap_out->flags |= DMA_ACTIVE; dma_interrupt = vidc_audio_dma_interrupt; vidc_sound_dma_irq(0, NULL); -- cgit v1.2.2 From e3d2530a6cea80987f77b75d8784a00f3aaf22ff Mon Sep 17 00:00:00 2001 From: Kunal Gangakhedkar Date: Sat, 20 Mar 2010 23:08:01 +0530 Subject: ALSA: hda - Add PCI quirk for HP dv6-1110ax. Adding this PCI quirk fixes the board config detection. This also fixes jack sensing by using "hp_detect=1" via properly detected board config. Signed-off-by: Kunal Gangakhedkar Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c416bb18a57..c4be3fab94e5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1730,6 +1730,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, "HP dv6", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061, + "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- cgit v1.2.2 From 025f206c9e0f96cc41567b01c07fb852d8900da1 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 21 Mar 2010 18:34:43 -0400 Subject: ALSA: hda: Fix 0 dB offset for HP laptops using CX20551 (Waikiki) BugLink: https://launchpad.net/bugs/420578 The OR has verified that his hardware distorts because of the 0 dB offset not corresponding to the highest PCM level. Fix this by capping said PCM level to 0 dB similarly to what we do for CX20549 (Venice). Reported-by: Mike Pontillo Tested-by: Mike Pontillo Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 194a28c54992..61682e1d09da 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1591,6 +1591,21 @@ static int patch_cxt5047(struct hda_codec *codec) #endif } spec->vmaster_nid = 0x13; + + switch (codec->subsystem_id >> 16) { + case 0x103c: + /* HP laptops have really bad sound over 0 dB on NID 0x10. + * Fix max PCM level to 0 dB (originally it has 0x1e steps + * with 0 dB offset 0x17) + */ + snd_hda_override_amp_caps(codec, 0x10, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; + } + return 0; } -- cgit v1.2.2 From e933e9e5238b79870b04718024416a6dcf602a27 Mon Sep 17 00:00:00 2001 From: Derek Kelly Date: Mon, 22 Mar 2010 08:04:19 +0100 Subject: ALSA: hda - Add support of Nvidia GT220 HDMI This patch adds the device id for Nvidia GT220 cards to the nvhdmi driver. I have tested it and confirmed it to be working. Original patch download link: https://gist.github.com/324070/ Signed-off-by: Derek Kelly Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 70669a246902..9e47717c8e2a 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -554,6 +554,8 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000b, .name = "GT21x HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000a, .name = "GT220 HDMI", + .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000d, .name = "GT240 HDMI", .patch = patch_nvhdmi_8ch_89 }, {} /* terminator */ @@ -568,6 +570,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000b"); +MODULE_ALIAS("snd-hda-codec-id:10de000a"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); MODULE_LICENSE("GPL"); -- cgit v1.2.2 From ea823c08912cfb6d4af2fa8b6dd5d8deb2fb486a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Mar 2010 08:07:55 +0100 Subject: ALSA: hda - Sort codec entry list of Nvidia HDMI Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 9e47717c8e2a..3c10c0b149f4 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -538,8 +538,6 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0003, .name = "MCP77/78 HDMI", @@ -550,14 +548,16 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de000c, .name = "MCP89 HDMI", + { .id = 0x10de000a, .name = "GT220 HDMI", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000b, .name = "GT21x HDMI", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000a, .name = "GT220 HDMI", + { .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000d, .name = "GT240 HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ }; @@ -566,12 +566,12 @@ MODULE_ALIAS("snd-hda-codec-id:10de0003"); MODULE_ALIAS("snd-hda-codec-id:10de0005"); MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); -MODULE_ALIAS("snd-hda-codec-id:10de0067"); -MODULE_ALIAS("snd-hda-codec-id:10de8001"); -MODULE_ALIAS("snd-hda-codec-id:10de000c"); -MODULE_ALIAS("snd-hda-codec-id:10de000b"); MODULE_ALIAS("snd-hda-codec-id:10de000a"); +MODULE_ALIAS("snd-hda-codec-id:10de000b"); +MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("NVIDIA HDMI HD-audio codec"); -- cgit v1.2.2 From bae84e70d66fe46c12231082cf1c4848ea22f3ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Mar 2010 08:30:20 +0100 Subject: ALSA: hda - Fix access-after-free in patch_realtek.c alc_free_kctls() has to be called after all jobs done in alc_build_controls(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4ec57633af88..053d53d8c8b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2532,8 +2532,6 @@ static int alc_build_controls(struct hda_codec *codec) return err; } - alc_free_kctls(codec); /* no longer needed */ - /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); if (!kctl) @@ -2602,6 +2600,9 @@ static int alc_build_controls(struct hda_codec *codec) } } } + + alc_free_kctls(codec); /* no longer needed */ + return 0; } -- cgit v1.2.2 From 3cc4e53f86dab635166929bfa47cc68d59b28c26 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Feb 2010 14:39:36 +0000 Subject: ASoC: Remove BROKEN from i.MX audio after dependencies merged Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index c7d0fd9b7de8..7174b4c710de 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,6 +1,6 @@ config SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC && BROKEN + depends on ARCH_MXC select SND_PCM select FIQ select SND_SOC_AC97_BUS -- cgit v1.2.2 From 1c583063a5c769fe2ec604752e383972c69e6d9b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 24 Mar 2010 07:10:54 +0100 Subject: ALSA: cmipci: work around invalid PCM pointer When the CMI8738 FRAME2 register is read, the chip sometimes (probably when wrapping around) returns an invalid value that would be outside the programmed DMA buffer. This leads to an inconsistent PCM pointer that is likely to result in an underrun. To work around this, read the register multiple times until we get a valid value; the error state seems to be very short-lived. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Matija Nalis Cc: Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1ded64e05643..329968edca9b 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -941,13 +941,21 @@ static snd_pcm_uframes_t snd_cmipci_pcm_pointer(struct cmipci *cm, struct cmipci struct snd_pcm_substream *substream) { size_t ptr; - unsigned int reg; + unsigned int reg, rem, tries; + if (!rec->running) return 0; #if 1 // this seems better.. reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2; - ptr = rec->dma_size - (snd_cmipci_read_w(cm, reg) + 1); - ptr >>= rec->shift; + for (tries = 0; tries < 3; tries++) { + rem = snd_cmipci_read_w(cm, reg); + if (rem < rec->dma_size) + goto ok; + } + printk(KERN_ERR "cmipci: invalid PCM pointer: %#x\n", rem); + return SNDRV_PCM_POS_XRUN; +ok: + ptr = (rec->dma_size - (rem + 1)) >> rec->shift; #else reg = rec->ch ? CM_REG_CH1_FRAME1 : CM_REG_CH0_FRAME1; ptr = snd_cmipci_read(cm, reg) - rec->offset; -- cgit v1.2.2 From a8462bde78fdb77c8ede61e1af99617905a78ccf Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 24 Mar 2010 14:58:34 +0300 Subject: ASoC: wm8994: playback => capture Sparse caught that initialize "playback" two times instead of initializing "capture". Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771c33a4..d10d65191fd2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3401,7 +3401,7 @@ struct snd_soc_dai wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, - .playback = { + .capture = { .stream_name = "AIF3 Capture", .channels_min = 2, .channels_max = 2, -- cgit v1.2.2 From 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Mar 2010 15:00:15 +0100 Subject: ALSA: hda - Don't set invalid connection index in Realtek initialiaiton Skip initialization of connections of DAC widgets that aren't used, which resulted in invalid verb parameters. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 053d53d8c8b2..9a23444e9e7a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10043,8 +10043,11 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, alc_set_pin_output(codec, nid, pin_type); if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; - else + else { + if (spec->multiout.num_dacs >= dac_idx) + return; idx = spec->multiout.dac_nids[dac_idx] - 2; + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } -- cgit v1.2.2 From e1f7f02b45cf33a774d56e505ce1718af9392f5e Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Thu, 25 Mar 2010 22:38:15 -0700 Subject: ALSA: ac97: Add IBM ThinkPad R40e to Headphone/Line Jack Sense blacklist BugLink: https://launchpad.net/bugs/303789 This model needs both 'Headphone Jack Sense' and 'Line Jack Sense' muted for audible audio, so just add its SSID to the blacklist and don't enumerate the controls. Signed-off-by: Daniel T Chen Cc: Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1caf5e3c1f6a..1a59b71c5432 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1852,6 +1852,7 @@ static unsigned int ad1981_jacks_blacklist[] = { 0x10140523, /* Thinkpad R40 */ 0x10140534, /* Thinkpad X31 */ 0x10140537, /* Thinkpad T41p */ + 0x1014053e, /* Thinkpad R40e */ 0x10140554, /* Thinkpad T42p/R50p */ 0x10140567, /* Thinkpad T43p 2668-G7U */ 0x10140581, /* Thinkpad X41-2527 */ -- cgit v1.2.2 From 0f17014b340b98465fcf0de4c0d6c84a002ec53b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 26 Mar 2010 16:07:25 +0200 Subject: ALSA: pcm_lib - fix xrun functionality The commit 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 broke the interrupt time xrun functionality (stream stop etc.) if the CONFIG_SND_PCM_XRUN_DEBUG is not set. This is because the xrun() is null defined without it. Fix this by letting the function xrun() to be always defined as it was before. Signed-off-by: Jarkko Nikula Cc: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b546ac2660f9..a2ff86189d2a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -148,6 +148,9 @@ static void pcm_debug_name(struct snd_pcm_substream *substream, #define xrun_debug(substream, mask) \ ((substream)->pstr->xrun_debug & (mask)) +#else +#define xrun_debug(substream, mask) 0 +#endif #define dump_stack_on_xrun(substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ @@ -169,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream) } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG #define hw_ptr_error(substream, fmt, args...) \ do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ @@ -255,8 +259,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ -#define xrun_debug(substream, mask) 0 -#define xrun(substream) do { } while (0) #define hw_ptr_error(substream, fmt, args...) do { } while (0) #define xrun_log(substream, pos) do { } while (0) #define xrun_log_show(substream) do { } while (0) -- cgit v1.2.2 From 5cd165e7057020884e430941c24454d3df9a799d Mon Sep 17 00:00:00 2001 From: Daniel Chen Date: Sun, 28 Mar 2010 13:32:34 -0700 Subject: ALSA: ac97: Add Toshiba P500 to ac97 jack sense blacklist BugLink: https://launchpad.net/bugs/481058 The OR has verified that both 'Headphone Jack Sense' and 'Line Jack Sense' need to be muted for sound to be audible, so just add the machine's SSID to the ac97 jack sense blacklist. Reported-by: Richard Gagne Tested-by: Richard Gagne Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1a59b71c5432..e68c98ef4041 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1859,6 +1859,7 @@ static unsigned int ad1981_jacks_blacklist[] = { 0x10280160, /* Dell Dimension 2400 */ 0x104380b0, /* Asus A7V8X-MX */ 0x11790241, /* Toshiba Satellite A-15 S127 */ + 0x1179ff10, /* Toshiba P500 */ 0x144dc01a, /* Samsung NP-X20C004/SEG */ 0 /* end */ }; -- cgit v1.2.2 From 9ec8ddad59fadd8021adfea4cb716a49b0e232e9 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 28 Mar 2010 02:34:40 -0400 Subject: ALSA: hda: Use LPIB for ga-ma770-ud3 board BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=575669 The OR states that position_fix=1 is necessary to work around glitching during volume adjustments using PulseAudio. Reported-by: Carlos Laviola Tested-by: Carlos Laviola Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8b2915631cc3..4bb90675f70f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2269,6 +2269,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), -- cgit v1.2.2 From 5dbd5ec6e1cf2e49128025d80813a275744a7ac5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2010 09:16:24 +0200 Subject: ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo() The mask and value parameters passed to snd_hda_codec_amp_stereo() should be 8-bit values for mute and volume. Passing AMP_IN_MUTE() is wrong, which is found in many places in patch_realtek.c as a left-over from the conversion to snd_hda_codec_amp_stereo(). Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 52 +++++++++++++++++++++---------------------- 1 file changed, 26 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a23444e9e7a..bc55c1e96df5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12459,11 +12459,11 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc268_acer_lc_unsol_event(struct hda_codec *codec, @@ -13482,11 +13482,11 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13511,11 +13511,11 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec) /* Check port replicator headphone socket */ present |= snd_hda_jack_detect(codec, 0x1a); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13646,11 +13646,11 @@ static void alc269_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, nid); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } /* unsolicited event for HP jack sensing */ @@ -17115,9 +17115,9 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) @@ -17128,13 +17128,13 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) @@ -17145,13 +17145,13 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc662_f5z_speaker_automute(struct hda_codec *codec) @@ -17190,14 +17190,14 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) if (present1 || present2) { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); } } -- cgit v1.2.2 From 6694635d3ae1b038d7a0e38b80637db867c7c8e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2010 17:21:45 +0200 Subject: ALSA: hda - Fix ADC/MUX assignment of ALC269 codec ALC269 codec has a few different variants, and each of them may have different ADC and MUX widgets. For example, one model has ADC 0x08 with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or 0x24. The difference of ADC appears usually as the capability of the digital mic pin (0x12), and the current driver sometimes misses the internal mic pin due to the mismatching ADC. This patch adds a bit more clever way to find the matching ADC instead of the static list. Now the driver checks all active input pins and fills only the ADC/MUX's that contain all of them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 95 ++++++++++++++++++++++++++++++++++++------- 1 file changed, 80 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc55c1e96df5..22aea7b089c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4984,6 +4984,69 @@ static void set_capture_mixer(struct hda_codec *codec) } } +/* fill adc_nids (and capsrc_nids) containing all active input pins */ +static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, + int num_nids) +{ + struct alc_spec *spec = codec->spec; + int n; + hda_nid_t fallback_adc = 0, fallback_cap = 0; + + for (n = 0; n < num_nids; n++) { + hda_nid_t adc, cap; + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int nconns, i, j; + + adc = nids[n]; + if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN) + continue; + cap = adc; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + if (nconns == 1) { + cap = conn[0]; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + } + if (nconns <= 0) + continue; + if (!fallback_adc) { + fallback_adc = adc; + fallback_cap = cap; + } + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if (!nid) + continue; + for (j = 0; j < nconns; j++) { + if (conn[j] == nid) + break; + } + if (j >= nconns) + break; + } + if (i >= AUTO_PIN_LAST) { + int num_adcs = spec->num_adc_nids; + spec->private_adc_nids[num_adcs] = adc; + spec->private_capsrc_nids[num_adcs] = cap; + spec->num_adc_nids++; + spec->adc_nids = spec->private_adc_nids; + if (adc != cap) + spec->capsrc_nids = spec->private_capsrc_nids; + } + } + if (!spec->num_adc_nids) { + printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" + " using fallback 0x%x\n", fallback_adc); + spec->private_adc_nids[0] = fallback_adc; + spec->adc_nids = spec->private_adc_nids; + if (fallback_adc != fallback_cap) { + spec->private_capsrc_nids[0] = fallback_cap; + spec->capsrc_nids = spec->private_adc_nids; + } + } +} + #ifdef CONFIG_SND_HDA_INPUT_BEEP #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) @@ -13333,9 +13396,9 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { 0x22, }; -/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24), - * not a mux! - */ +static hda_nid_t alc269_adc_candidates[] = { + 0x08, 0x09, 0x07, +}; #define alc269_modes alc260_modes #define alc269_capture_source alc880_lg_lw_capture_source @@ -13842,7 +13905,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; - hda_nid_t real_capsrc_nids; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc269_ignore); @@ -13866,18 +13928,19 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { add_verb(spec, alc269vb_init_verbs); - real_capsrc_nids = alc269vb_capsrc_nids[0]; alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21); } else { add_verb(spec, alc269_init_verbs); - real_capsrc_nids = alc269_capsrc_nids[0]; alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); } spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + fillup_priv_adc_nids(codec, alc269_adc_candidates, + sizeof(alc269_adc_candidates)); + /* set default input source */ - snd_hda_codec_write_cache(codec, real_capsrc_nids, + snd_hda_codec_write_cache(codec, spec->capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, spec->input_mux->items[0].index); @@ -14156,14 +14219,16 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; - if (!is_alc269vb) { - spec->adc_nids = alc269_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->capsrc_nids = alc269_capsrc_nids; - } else { - spec->adc_nids = alc269vb_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); - spec->capsrc_nids = alc269vb_capsrc_nids; + if (!spec->adc_nids) { /* wasn't filled automatically? use default */ + if (!is_alc269vb) { + spec->adc_nids = alc269_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->capsrc_nids = alc269_capsrc_nids; + } else { + spec->adc_nids = alc269vb_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); + spec->capsrc_nids = alc269vb_capsrc_nids; + } } if (!spec->cap_mixer) -- cgit v1.2.2 From fb48e3c6a4d8888aff61fbf567aadac7d206e973 Mon Sep 17 00:00:00 2001 From: Graham Gower Date: Thu, 25 Mar 2010 10:52:12 +1030 Subject: ASoC: Fix passing platform_data to ac97 bus users and fix a leak [The issue is an attempt to write the pdata without the AC97 device allocated when using ac97.c - also added a comment in soc-core.c for the special case for ac97. -- broonie] Signed-off-by: Graham Gower Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 15 +++++++++------ sound/soc/soc-core.c | 3 ++- 2 files changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index a1bbe16b7f96..bcfa53271673 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -80,9 +80,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; struct snd_soc_codec *codec; struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; + int i; int ret = 0; printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); @@ -102,12 +104,6 @@ static int ac97_soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); - if (ret < 0) { - printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n"); - goto err; - } - /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) @@ -123,6 +119,13 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].codec_dai->ac97_control) { + snd_ac97_dev_add_pdata(codec->ac97, + card->dai_link[i].cpu_dai->ac97_pdata); + } + } + return 0; bus_err: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556ef431..d0efd5eaaa0b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1548,7 +1548,8 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) mutex_unlock(&codec->mutex); return ret; } - if (card->dai_link[i].codec_dai->ac97_control) { + /* Check for codec->ac97 to handle the ac97.c fun */ + if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) { snd_ac97_dev_add_pdata(codec->ac97, card->dai_link[i].cpu_dai->ac97_pdata); } -- cgit v1.2.2 From 1f85d72d2c9c9a1d6d32cf325936bc224ad5d591 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Mar 2010 07:48:05 +0200 Subject: ALSA: hda - Add missing printk argument in previous patch Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 22aea7b089c6..ca93c4cc144e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5037,7 +5037,8 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, } if (!spec->num_adc_nids) { printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" - " using fallback 0x%x\n", fallback_adc); + " using fallback 0x%x\n", + codec->chip_name, fallback_adc); spec->private_adc_nids[0] = fallback_adc; spec->adc_nids = spec->private_adc_nids; if (fallback_adc != fallback_cap) { -- cgit v1.2.2 From 5a0e3ad6af8660be21ca98a971cd00f331318c05 Mon Sep 17 00:00:00 2001 From: Tejun Heo Date: Wed, 24 Mar 2010 17:04:11 +0900 Subject: include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo Guess-its-ok-by: Christoph Lameter Cc: Ingo Molnar Cc: Lee Schermerhorn --- sound/aoa/codecs/onyx.c | 1 + sound/aoa/codecs/tas.c | 1 + sound/aoa/codecs/toonie.c | 1 + sound/aoa/core/gpio-pmf.c | 1 + sound/aoa/fabrics/layout.c | 1 + sound/aoa/soundbus/i2sbus/control.c | 1 + sound/aoa/soundbus/i2sbus/core.c | 1 + sound/aoa/soundbus/i2sbus/pcm.c | 1 + sound/arm/pxa2xx-pcm-lib.c | 1 + sound/core/control_compat.c | 1 + sound/core/hrtimer.c | 1 + sound/core/info.c | 1 + sound/core/jack.c | 1 + sound/core/misc.c | 1 + sound/core/oss/route.c | 1 - sound/core/pcm_compat.c | 1 + sound/core/pcm_memory.c | 1 + sound/core/seq/oss/seq_oss_init.c | 1 + sound/core/seq/oss/seq_oss_midi.c | 1 + sound/core/seq/oss/seq_oss_readq.c | 1 + sound/core/seq/oss/seq_oss_synth.c | 1 + sound/core/seq/oss/seq_oss_timer.c | 1 + sound/core/seq/oss/seq_oss_writeq.c | 1 + sound/core/seq/seq_compat.c | 1 + sound/core/seq/seq_system.c | 1 + sound/drivers/ml403-ac97cr.c | 1 + sound/drivers/mtpav.c | 1 - sound/drivers/mts64.c | 1 + sound/drivers/opl3/opl3_oss.c | 1 - sound/drivers/opl3/opl3_synth.c | 1 + sound/drivers/opl4/opl4_lib.c | 1 + sound/drivers/pcsp/pcsp_lib.c | 1 + sound/drivers/portman2x4.c | 1 + sound/drivers/vx/vx_hwdep.c | 1 + sound/i2c/other/tea575x-tuner.c | 1 + sound/isa/cmi8330.c | 1 - sound/isa/cs423x/cs4236.c | 1 - sound/isa/es18xx.c | 1 - sound/isa/gus/interwave.c | 1 - sound/isa/msnd/msnd_midi.c | 1 + sound/isa/opl3sa2.c | 1 - sound/isa/opti9xx/miro.c | 1 - sound/isa/opti9xx/opti92x-ad1848.c | 1 - sound/isa/sb/emu8000_pcm.c | 1 + sound/isa/sb/sb16.c | 1 - sound/isa/sb/sb8.c | 1 - sound/isa/wavefront/wavefront.c | 1 - sound/isa/wavefront/wavefront_fx.c | 1 + sound/isa/wavefront/wavefront_synth.c | 1 + sound/mips/hal2.c | 1 + sound/mips/sgio2audio.c | 2 +- sound/oss/ad1848.c | 1 + sound/oss/dmabuf.c | 1 + sound/oss/kahlua.c | 1 + sound/oss/mpu401.c | 1 + sound/oss/msnd.c | 1 - sound/oss/msnd_pinnacle.c | 2 +- sound/oss/opl3.c | 1 + sound/oss/sb_card.c | 1 + sound/oss/sb_common.c | 1 + sound/oss/sb_midi.c | 1 + sound/oss/sb_mixer.c | 2 ++ sound/oss/soundcard.c | 1 - sound/oss/uart401.c | 1 + sound/oss/v_midi.c | 1 + sound/oss/vidc.c | 1 + sound/oss/vwsnd.c | 1 + sound/oss/waveartist.c | 1 + sound/pci/ac97/ac97_proc.c | 1 - sound/pci/als4000.c | 1 - sound/pci/aw2/aw2-saa7146.c | 1 - sound/pci/ca0106/ca0106_mixer.c | 1 - sound/pci/ca0106/ca0106_proc.c | 1 - sound/pci/cs5530.c | 1 + sound/pci/cs5535audio/cs5535audio_pcm.c | 1 - sound/pci/cs5535audio/cs5535audio_pm.c | 1 - sound/pci/ctxfi/ctatc.c | 1 + sound/pci/ctxfi/ctpcm.c | 1 + sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigodjx.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/indigoiox.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/emu10k1/memory.c | 1 + sound/pci/hda/hda_beep.c | 1 + sound/pci/hda/hda_eld.c | 1 + sound/pci/ice1712/ak4xxx.c | 1 + sound/pci/ice1712/amp.c | 1 - sound/pci/ice1712/vt1720_mobo.c | 1 - sound/pci/ice1712/wtm.c | 1 - sound/pci/lx6464es/lx6464es.c | 1 + sound/pci/mixart/mixart.c | 1 + sound/pci/mixart/mixart_hwdep.c | 1 + sound/pci/oxygen/oxygen_lib.c | 1 + sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 1 - sound/pci/rme9652/hdsp.c | 1 - sound/pci/rme9652/rme9652.c | 1 - sound/pci/sis7019.c | 1 + sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 1 + sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 - sound/pcmcia/vx/vxpocket.c | 1 + sound/ppc/burgundy.c | 1 - sound/ppc/keywest.c | 1 - sound/ppc/snd_ps3.c | 2 +- sound/sh/sh_dac_audio.c | 1 + sound/soc/au1x/psc-ac97.c | 1 + sound/soc/au1x/psc-i2s.c | 1 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 1 + sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 2 +- sound/soc/codecs/ac97.c | 1 + sound/soc/codecs/ad1836.c | 1 + sound/soc/codecs/ad1938.c | 1 + sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/ad73311.c | 1 + sound/soc/codecs/ads117x.c | 1 + sound/soc/codecs/ak4104.c | 1 + sound/soc/codecs/ak4535.c | 1 + sound/soc/codecs/ak4642.c | 1 + sound/soc/codecs/ak4671.c | 1 + sound/soc/codecs/cs4270.c | 1 + sound/soc/codecs/cx20442.c | 1 + sound/soc/codecs/da7210.c | 1 + sound/soc/codecs/pcm3008.c | 1 + sound/soc/codecs/ssm2602.c | 1 + sound/soc/codecs/stac9766.c | 1 + sound/soc/codecs/tlv320aic23.c | 1 + sound/soc/codecs/tlv320aic26.c | 1 + sound/soc/codecs/tlv320aic3x.c | 1 + sound/soc/codecs/tlv320dac33.c | 1 + sound/soc/codecs/tpa6130a2.c | 1 + sound/soc/codecs/twl4030.c | 1 + sound/soc/codecs/uda134x.c | 1 + sound/soc/codecs/wm2000.c | 1 + sound/soc/codecs/wm8350.c | 1 + sound/soc/codecs/wm8400.c | 1 + sound/soc/codecs/wm8510.c | 1 + sound/soc/codecs/wm8523.c | 1 + sound/soc/codecs/wm8580.c | 1 + sound/soc/codecs/wm8711.c | 1 + sound/soc/codecs/wm8727.c | 1 + sound/soc/codecs/wm8728.c | 1 + sound/soc/codecs/wm8731.c | 1 + sound/soc/codecs/wm8750.c | 1 + sound/soc/codecs/wm8753.c | 1 + sound/soc/codecs/wm8776.c | 1 + sound/soc/codecs/wm8900.c | 1 + sound/soc/codecs/wm8903.c | 1 + sound/soc/codecs/wm8904.c | 1 + sound/soc/codecs/wm8940.c | 1 + sound/soc/codecs/wm8955.c | 1 + sound/soc/codecs/wm8960.c | 1 + sound/soc/codecs/wm8961.c | 1 + sound/soc/codecs/wm8971.c | 1 + sound/soc/codecs/wm8974.c | 1 + sound/soc/codecs/wm8978.c | 1 + sound/soc/codecs/wm8988.c | 1 + sound/soc/codecs/wm8990.c | 1 + sound/soc/codecs/wm8993.c | 1 + sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm9081.c | 1 + sound/soc/codecs/wm9705.c | 1 + sound/soc/codecs/wm9712.c | 1 + sound/soc/codecs/wm9713.c | 1 + sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/fsl/fsl_dma.c | 1 + sound/soc/fsl/fsl_ssi.c | 1 + sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc8610_hpcd.c | 1 + sound/soc/fsl/soc-of-simple.c | 1 + sound/soc/imx/imx-pcm-dma-mx2.c | 1 + sound/soc/imx/imx-pcm-fiq.c | 1 + sound/soc/imx/imx-ssi.c | 1 + sound/soc/omap/mcpdm.c | 1 + sound/soc/omap/omap-pcm.c | 1 + sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/s6000/s6000-i2s.c | 1 + sound/soc/sh/dma-sh7760.c | 1 + sound/soc/sh/fsi.c | 1 + sound/soc/sh/siu_dai.c | 1 + sound/soc/sh/siu_pcm.c | 1 - sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 1 + sound/soc/txx9/txx9aclc-ac97.c | 1 + sound/soc/txx9/txx9aclc.c | 1 + sound/sound_firmware.c | 1 - sound/sparc/cs4231.c | 1 - sound/sparc/dbri.c | 1 + sound/synth/emux/emux_proc.c | 1 - sound/usb/caiaq/audio.c | 1 + sound/usb/caiaq/device.c | 1 + sound/usb/caiaq/midi.c | 1 + sound/usb/usx2y/us122l.c | 1 + sound/usb/usx2y/usX2Yhwdep.c | 1 + sound/usb/usx2y/usb_stream.c | 1 + sound/usb/usx2y/usbusx2y.c | 1 + sound/usb/usx2y/usbusx2yaudio.c | 1 + sound/usb/usx2y/usx2yhwdeppcm.c | 1 + 210 files changed, 176 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 84bb07d39a7f..91852e49910e 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -33,6 +33,7 @@ */ #include #include +#include MODULE_AUTHOR("Johannes Berg "); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 1dd66ddffcaf..fd2188c3df2b 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -66,6 +66,7 @@ #include #include #include +#include MODULE_AUTHOR("Johannes Berg "); MODULE_LICENSE("GPL"); diff --git a/sound/aoa/codecs/toonie.c b/sound/aoa/codecs/toonie.c index f13827e17562..69d2cb601f2a 100644 --- a/sound/aoa/codecs/toonie.c +++ b/sound/aoa/codecs/toonie.c @@ -11,6 +11,7 @@ */ #include #include +#include MODULE_AUTHOR("Johannes Berg "); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("toonie codec driver for snd-aoa"); diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index 1dd0c28d1fb7..6776d1c12b63 100644 --- a/sound/aoa/core/gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c @@ -6,6 +6,7 @@ * GPL v2, can be found in COPYING. */ +#include #include #include #include "../aoa.h" diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 7a437da05646..1cd9b301df03 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -12,6 +12,7 @@ #include #include #include +#include #include "../aoa.h" #include "../soundbus/soundbus.h" diff --git a/sound/aoa/soundbus/i2sbus/control.c b/sound/aoa/soundbus/i2sbus/control.c index 87beb4ad4d63..47f854c2001f 100644 --- a/sound/aoa/soundbus/i2sbus/control.c +++ b/sound/aoa/soundbus/i2sbus/control.c @@ -8,6 +8,7 @@ #include #include +#include #include #include diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 4e3b819d4993..9d6f3b176ed1 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -7,6 +7,7 @@ */ #include +#include #include #include #include diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 59bacd365733..be838993926d 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -8,6 +8,7 @@ #include #include +#include #include #include #include diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index fd51fa8b06a1..8808b82311b1 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -4,6 +4,7 @@ * published by the Free Software Foundation. */ +#include #include #include diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 368dc9c4aef8..426874429a5e 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -21,6 +21,7 @@ /* this file included from control.c */ #include +#include struct snd_ctl_elem_list32 { u32 offset; diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 7f4d744ae40a..7730575bfadd 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -19,6 +19,7 @@ */ #include +#include #include #include #include diff --git a/sound/core/info.c b/sound/core/info.c index d749a0d394a7..cc4a53d4b7f8 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/core/jack.c b/sound/core/jack.c index f705eec7372a..14b8a4ee690d 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -20,6 +20,7 @@ */ #include +#include #include #include diff --git a/sound/core/misc.c b/sound/core/misc.c index 3da4f92427d8..2c41825c836e 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -21,6 +21,7 @@ #include #include +#include #include #include diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index 0dcc2870d537..bbe25d8c450a 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 08bfed594a83..5fb2e28e796f 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -21,6 +21,7 @@ /* This file included from pcm_native.c */ #include +#include static int snd_pcm_ioctl_delay_compat(struct snd_pcm_substream *substream, s32 __user *src) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index d6d49d6651f9..917e4055ee30 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index d0d721c22eac..685712276ac9 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -29,6 +29,7 @@ #include "seq_oss_event.h" #include #include +#include /* * common variables diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 9dfb2f77be60..677dc84590c7 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -28,6 +28,7 @@ #include #include "../seq_lock.h" #include +#include /* diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index f5de79f29f1e..73661c4ab82a 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -25,6 +25,7 @@ #include #include "../seq_lock.h" #include +#include /* * constants diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 945a27c34a9d..ee44ab9593c0 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -24,6 +24,7 @@ #include "seq_oss_midi.h" #include "../seq_lock.h" #include +#include /* * constants diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index c440fdacec93..ab59cbfbcaf2 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -23,6 +23,7 @@ #include "seq_oss_timer.h" #include "seq_oss_event.h" #include +#include /* */ diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c index 217424858191..d50338bbc21f 100644 --- a/sound/core/seq/oss/seq_oss_writeq.c +++ b/sound/core/seq/oss/seq_oss_writeq.c @@ -27,6 +27,7 @@ #include "../seq_lock.h" #include "../seq_clientmgr.h" #include +#include /* diff --git a/sound/core/seq/seq_compat.c b/sound/core/seq/seq_compat.c index c956fe462569..81f7c109dc46 100644 --- a/sound/core/seq/seq_compat.c +++ b/sound/core/seq/seq_compat.c @@ -21,6 +21,7 @@ /* This file included from seq.c */ #include +#include struct snd_seq_port_info32 { struct snd_seq_addr addr; /* client/port numbers */ diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c index 77884e62b648..c38b90cf3cb0 100644 --- a/sound/core/seq/seq_system.c +++ b/sound/core/seq/seq_system.c @@ -20,6 +20,7 @@ */ #include +#include #include #include "seq_system.h" #include "seq_timer.h" diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 1950ffce2b54..a1282c1c0591 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -39,6 +39,7 @@ #include #include +#include #include #include diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 2f8f295d6b0c..da03597fc893 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -54,7 +54,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 9284829bf927..8539ab0a0893 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index a54b1dc5cc78..ade3ca52422e 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -19,7 +19,6 @@ */ #include "opl3_voice.h" -#include static int snd_opl3_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure); static int snd_opl3_close_seq_oss(struct snd_seq_oss_arg *arg); diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 6d57b6441dec..301acb6b9cf9 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -19,6 +19,7 @@ * */ +#include #include #include diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 01997f24c895..f07e38da59b8 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -20,6 +20,7 @@ #include "opl4_local.h" #include #include +#include #include #include diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index e1145ac6e908..d77ffa9a9387 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -7,6 +7,7 @@ */ #include +#include #include #include #include diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 60158e2e0eaf..f2b0ba22d9ce 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -42,6 +42,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 46df8817c18f..f7a6fbd313e3 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -22,6 +22,7 @@ #include #include +#include #include #include #include diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index c4c6ef73f9bf..ee538f1ae846 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 8246aae32ab4..fe79a169acb5 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -46,7 +46,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index cc15d1d65a22..999dc1e0fdbd 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 9a43baae7250..fb4d6b34bbca 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -80,7 +80,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 534a6eced2b8..c7b80e4730fc 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index 4be562b2cf21..787495674235 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -25,6 +25,7 @@ */ #include +#include #include #include #include diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 0481a55334b9..265abcce9dba 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 5913717c1be6..8c24102d0d93 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 4d2d0405bdc7..c35dc68930dc 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index 91dc3d83e2cf..ccedbfed061a 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -20,6 +20,7 @@ #include "emu8000_local.h" #include +#include #include #include diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 519c36346dec..4d1c5a300ff8 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -21,7 +21,6 @@ #include #include -#include #include #include #include diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 3cd57ee54660..81284a8fa0ce 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index a34ae7b1f7d0..711670e4a425 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -21,7 +21,6 @@ #include #include -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index 2bb1cee09255..657e2d6c01ac 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 5d4ff48c4345..4fb7b19ff393 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 9a88cdfd952a..453d343550a8 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 6aff217379d9..717604c00f0a 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -25,11 +25,11 @@ #include #include #include -#include #include #include #include #include +#include #include #include diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index d12bd98a37ba..24793c5b65ac 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -45,6 +45,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index 1bfcf7e88546..bcc3e8e07122 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -26,6 +26,7 @@ #define SAMPLE_ROUNDUP 0 #include +#include #include "sound_config.h" #define DMAP_FREE_ON_CLOSE 0 diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 24d152ccf80d..52d06a334e8f 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -31,6 +31,7 @@ #include #include #include +#include #include "sound_config.h" diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 0af9d24feb8f..25e4609f8339 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -19,6 +19,7 @@ */ #include +#include #include #include #include diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c index 21eb6dce46df..c0cc951ba97d 100644 --- a/sound/oss/msnd.c +++ b/sound/oss/msnd.c @@ -24,7 +24,6 @@ #include #include -#include #include #include #include diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index bf27e008f465..a1e3f9671bea 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -35,12 +35,12 @@ #include #include -#include #include #include #include #include #include +#include #include #include #include "sound_config.h" diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 7781c13c1476..938c48c43585 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -24,6 +24,7 @@ */ #include +#include #include #include diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c index 7de18b58f2cd..84ef4d06c1c2 100644 --- a/sound/oss/sb_card.c +++ b/sound/oss/sb_card.c @@ -24,6 +24,7 @@ #include #include +#include #include #include "sound_config.h" #include "sb_mixer.h" diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index ce4db49291f7..7d42c5418d1b 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -31,6 +31,7 @@ #include #include #include +#include #include "sound_config.h" #include "sound_firmware.h" diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c index 8b796704e112..f139028e85c0 100644 --- a/sound/oss/sb_midi.c +++ b/sound/oss/sb_midi.c @@ -12,6 +12,7 @@ */ #include +#include #include "sound_config.h" diff --git a/sound/oss/sb_mixer.c b/sound/oss/sb_mixer.c index fad1a4f25ad6..2039d31b7e22 100644 --- a/sound/oss/sb_mixer.c +++ b/sound/oss/sb_mixer.c @@ -16,6 +16,8 @@ * Stanislav Voronyi : Support for AWE 3DSE device (Jun 7 1999) */ +#include + #include "sound_config.h" #define __SB_MIXER_C__ diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index fde7c12fe5da..2d9c51312622 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -36,7 +36,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index a446b826d5fc..8e514a676a0d 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include "sound_config.h" diff --git a/sound/oss/v_midi.c b/sound/oss/v_midi.c index 103940fd5b4f..f0b4151d9b17 100644 --- a/sound/oss/v_midi.c +++ b/sound/oss/v_midi.c @@ -21,6 +21,7 @@ #include #include +#include #include #include "sound_config.h" diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index a4127bab9231..ac39a531df19 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -17,6 +17,7 @@ * We currently support a mixer device, but it is currently non-functional. */ +#include #include #include #include diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 6713110bdc75..20b3b325aa80 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -149,6 +149,7 @@ #include #include #include +#include #include diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 2c63bb9da74a..e688dde6bbde 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -35,6 +35,7 @@ #include #include +#include #include #include #include diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 73b17d526c8b..6320bf084e47 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -22,7 +22,6 @@ * */ -#include #include #include diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index d75cf7b06426..6cf1de8042e8 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -68,7 +68,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c index 296123ab74f7..8afd8b5d1ac7 100644 --- a/sound/pci/aw2/aw2-saa7146.c +++ b/sound/pci/aw2/aw2-saa7146.c @@ -25,7 +25,6 @@ #include #include -#include #include #include #include diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 8f443a9d61ec..85fd315d9999 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -63,7 +63,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 0470461cc03e..ba96428c9f4c 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -63,7 +63,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 207479a641cf..bc07e275d4d4 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 0f48a871f17b..f16bc8aad6ed 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -23,7 +23,6 @@ */ #include -#include #include #include #include diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index 564c33b60953..a3301cc4ab82 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -19,7 +19,6 @@ */ #include -#include #include #include #include diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 480cb1e905b6..1bff80cde0a2 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -24,6 +24,7 @@ #include "ctdaio.h" #include "cttimer.h" #include +#include #include #include #include diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index d0dc227fbdd3..85ab43e89212 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -17,6 +17,7 @@ #include "ctpcm.h" #include "cttimer.h" +#include #include /* Hardware descriptions for playback */ diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index a65bafe0800f..fe7ad64dccd7 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -40,9 +40,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 0a6c50bcd758..d1fd34b1a8e3 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -44,9 +44,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index f5142796989b..1dffdc54416d 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -51,9 +51,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index 2364f8a1bc21..050e54aa693f 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -44,9 +44,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index 616b55825a19..5748fc6d29d6 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -50,9 +50,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index 776175c0bdad..4ae5e35cb5f1 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -42,9 +42,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index 8816b0bd2ba6..3550715bab1c 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -42,9 +42,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index b1e3652f2f48..19b191fd0120 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -42,10 +42,10 @@ #include #include #include -#include #include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index 1035125336d6..a9fcedf317a4 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -43,9 +43,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index 60b7cb2753cf..bcdfac63212c 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -43,10 +43,10 @@ #include #include #include -#include #include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 8c3f5c5b5301..d3a98c5dac86 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -49,9 +49,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index ed1cc0abc2b8..2a1dca6dce17 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -51,9 +51,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index cc2bbfc65327..9cdf14cfdd74 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -50,9 +50,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index 3e7e01824b40..1047be405ebe 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -48,9 +48,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 6a47672f930a..ffb1ddb8dc28 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -22,6 +22,7 @@ */ #include +#include #include #include diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index e4581a42ace5..29714c818b53 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -21,6 +21,7 @@ #include #include +#include #include #include #include "hda_beep.h" diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index dcd22446cfc7..d8da18a9e98b 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -22,6 +22,7 @@ */ #include +#include #include #include #include "hda_codec.h" diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index 03391da8c8c7..90d560c3df13 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 6da21a2bcade..e328cfb7620c 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include "ice1712.h" diff --git a/sound/pci/ice1712/vt1720_mobo.c b/sound/pci/ice1712/vt1720_mobo.c index 7f9674b641c0..4c551e147c08 100644 --- a/sound/pci/ice1712/vt1720_mobo.c +++ b/sound/pci/ice1712/vt1720_mobo.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include "ice1712.h" diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 5af9e84456d1..e618f789026e 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,7 +29,6 @@ #include #include #include -#include #include #include "ice1712.h" diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 0cca56038cd9..ef9af3f4ace2 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7e8e7da592a9..55e9315d4ccd 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8c939c..bf2696aa5d49 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include "mixart.h" diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9c5e6450eebb..fad03d64e3ad 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index d5e1c6eb7b7b..3c04524de37c 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -70,10 +70,10 @@ #include +#include #include #include #include -#include #include #include diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 9d5252bc870c..d19dc052c391 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 52c6eb57cc3f..b92adef8e81e 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 44a3e2d8c556..c492af5b25f3 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 7e3e8fbc90fe..9cc1b5aa0148 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index 5d2afa0b0ce4..9dce0bde5c05 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -19,6 +19,7 @@ */ #include +#include #include #include #include "pdaudiocf.h" diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 0d668f471620..43f995a3f960 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7be3b3357045..cfd1438bcc64 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -21,6 +21,7 @@ #include #include +#include #include #include "vxpocket.h" #include diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 1f72e1c786bf..00e2d5166d0a 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -21,7 +21,6 @@ #include #include -#include #include #include #include "pmac.h" diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index d06f780bd7e8..8f064c7ce745 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include "pmac.h" diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 53c81a547613..2f12da4da561 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -20,10 +20,10 @@ #include #include +#include #include #include #include -#include #include #include diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 76d9ad27d91c..68e0dee4ff05 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 340311d7fed5..a61ccd2d505f 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -17,6 +17,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 0cf2ca61c776..495be6e71931 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -18,6 +18,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 67cbfe7283da..5e7aacf3bb5a 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -29,8 +29,8 @@ #include #include #include -#include #include +#include #include #include diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index e69322978739..523b7fc33f4e 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index c6c6a4a7d948..1d2a1adf2575 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -29,8 +29,8 @@ #include #include #include -#include #include +#include #include #include diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 5e03bb2f3cd7..6bac1ac1a315 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -29,8 +29,8 @@ #include #include #include -#include #include +#include #include #include diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index a1bbe16b7f96..fd101d450d56 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -13,6 +13,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 3c80137d5938..11b62dee842c 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -17,6 +17,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index c233810d463d..240cd155b313 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -27,6 +27,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 39c0f7584e65..042072738cdc 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -12,6 +12,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index d2fcc601722c..475807bea2c2 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -11,6 +11,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index cc96411ca3e6..f8e75edb27b7 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -11,6 +11,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b68d99fb6af0..bdeb10dfd887 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -10,6 +10,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index ff966567e2ba..352d1d08dbd9 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3ef16bbc8c83..729859cf6ca8 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 82fca284d007..926797a014c7 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index dfbeb2db61b3..81a62d198b70 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -23,6 +23,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index e000cdfec1ec..9f169c477108 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -14,6 +14,7 @@ */ #include +#include #include #include diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cf2975a7294a..366daf1d044e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 2afcd0a8669d..5a5f187a2657 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index d2ff1cde6883..29d0906a924a 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 81b8c9dfe7fc..3293629dcb3b 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -15,6 +15,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index da589d8664d0..776b79cde904 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 357b609196e3..b5b7d6a03844 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e4b946a19ea3..4a6d56c3fed9 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d50f1699ccb2..d1e0e81ef30c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -31,6 +31,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 958d49c969ac..569ad8758a84 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6f5d4af20052..520ffd6536c3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 3e99fe5131dd..a8dcd5a5bbcb 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -15,6 +15,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 217b02680597..a34cbcf7904f 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -32,6 +32,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index df2c6d9617fb..2e0772f9c456 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index b432f4d4a324..6acc885cf9b7 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index af8cb6995a1f..9000b1d19afb 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index d3a61d7ea0c5..19cd47293424 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d077df6f5e75..8cc9042965eb 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 24a35603bcf7..8ca3812f2f2f 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 63a254e293ca..1072621e93fd 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -13,6 +13,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 3fb653ba363a..07adc375a706 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5a2619dbf283..e7c6bf163185 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 475c67ac7818..2916ed4d3844 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index c2444e7c8480..613199a0f799 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 44e7d9d82f87..60b1b3e1094b 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index dbc368c08263..b7fd96adac64 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3595bd57c4eb..fa5f99fde68b 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 593e47d0e0eb..c6f0abcc5711 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 31e39ffd1d8e..0c04b476487f 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 615dab2b62ef..c8d7a809af4d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d07bcc1e1c60..f1e63e01b04d 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index d2342c5e0425..50634ab76a5c 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index d9540d55fc89..a65b781af512 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index ee637af4737a..69708c4cc004 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 28bb59ea6ea1..526f56b09066 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 2862e4dced27..bb18c3ecfeb9 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 056b787b6ee0..831f4730bfd5 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index bf022f68b84f..03e8b1a6a56c 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771c33a4..8d1c63754be4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index c468497314ba..3a184fcb702b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ec54c6da9856..8793341849d1 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -10,6 +10,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e237bf615129..2f48a8aae22c 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -11,6 +11,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index ceb86b4ddb25..2fca514fde58 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -16,6 +16,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 6362ca05506e..62af7e025e7f 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ab6518d86f18..6c80cc35ecad 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b1a3a278819f..410c7496a18d 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 93f0f38a32c9..762c1b8e8e4e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 30ed568afb2e..d639e55c5124 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -8,6 +8,7 @@ #include #include +#include #include diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index ef67d1cdffe7..83de1c81c8c4 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -9,6 +9,7 @@ * express or implied. */ +#include #include #include #include diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c index 8bc5cd9e972f..3bc13fd89096 100644 --- a/sound/soc/fsl/soc-of-simple.c +++ b/sound/soc/fsl/soc-of-simple.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 19452e44afdc..86668ab3f4d4 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index d9cb9849b033..f96a373699cf 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 56f46a75d297..6546b06cbd2a 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index ad8df6cfae88..1dab4c14874d 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 825db385f01f..ba8acbb0a7fa 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -23,6 +23,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 9e95e5117c88..d5fc52d0a3c4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -16,6 +16,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index c5cda187ecab..0664fac7612a 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index baddb1242c71..0d8bdf07729c 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -13,6 +13,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 993abb730dfa..8dc966f45c36 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 5452d19607e1..d86ee1bfc03a 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index ba7f8d05d977..8f85719212f9 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556ef431..2320153bd923 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6c3351095786..7c28f401f436 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -38,6 +38,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 0f83bdb9b16f..612e18b4bf4e 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index efed64b8b026..49cc7ea9a518 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/sound_firmware.c b/sound/sound_firmware.c index 96deaefaa897..340a0bc5303e 100644 --- a/sound/sound_firmware.c +++ b/sound/sound_firmware.c @@ -2,7 +2,6 @@ #include #include #include -#include #include #include #include "oss/sound_firmware.h" diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 8d13d933087d..7dcc06512e86 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -10,7 +10,6 @@ #include #include -#include #include #include #include diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 1d2e51b3f918..2eab6ce48852 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -58,6 +58,7 @@ #include #include #include +#include #include #include diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c index 687e6a13689e..58a32a10d115 100644 --- a/sound/synth/emux/emux_proc.c +++ b/sound/synth/emux/emux_proc.c @@ -19,7 +19,6 @@ */ #include -#include #include #include #include diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 86b2c3b92df5..4328cad6c3a2 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -17,6 +17,7 @@ */ #include +#include #include #include #include diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index a3f02dd97440..afc5aeb68005 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 538e8c00d31a..2f218c77fff2 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -17,6 +17,7 @@ */ #include +#include #include #include #include diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 44deb21b1777..9ca9a13a78da 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -16,6 +16,7 @@ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#include #include #include #include diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 1879b72c40f8..04aafb43a13c 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -21,6 +21,7 @@ */ #include +#include #include #include #include diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 12ae0340adc0..c400ade3ff08 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -17,6 +17,7 @@ */ #include +#include #include "usb_stream.h" diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index c42350eed2eb..cbd37f2c76d0 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -133,6 +133,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 74a67a85aa81..5d37d1ccf813 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -32,6 +32,7 @@ #include +#include #include #include #include diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 9ed6c3956ca7..2a528e56afd5 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -51,6 +51,7 @@ */ #include +#include #include "usbusx2yaudio.c" #if defined(USX2Y_NRPACKS_VARIABLE) || (!defined(USX2Y_NRPACKS_VARIABLE) && USX2Y_NRPACKS == 1) -- cgit v1.2.2 From b8e80cf386419453678b01bef830f53445ebb15d Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Tue, 30 Mar 2010 13:29:28 -0400 Subject: ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981 BugLink: https://launchpad.net/bugs/551606 The OR's hardware distorts at PCM 100% because it does not correspond to 0 dB. Fix this in patch_ad1981() for all models using the Thinkpad quirk. Reported-by: Jane Silber Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e6d1bdff1b6e..af34606c30c3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1896,6 +1896,14 @@ static int patch_ad1981(struct hda_codec *codec) case AD1981_THINKPAD: spec->mixers[0] = ad1981_thinkpad_mixers; spec->input_mux = &ad1981_thinkpad_capture_source; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); break; case AD1981_TOSHIBA: spec->mixers[0] = ad1981_hp_mixers; -- cgit v1.2.2 From b5442a75deee293d10c2ab8f4a77013973c4c9e0 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sun, 28 Mar 2010 22:29:29 +0200 Subject: ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code With recent (2.6.34) chnages in PCM handling, capture stopped working on my OMAP1510 based Amstrad Delta videophone. Using 2.6.34-rc2, I was able to correct the problem in 3 different ways: 1. reverting commit 7b3a177b0d4f92b3431b8dca777313a07533a710, 2. enabling additional jiffies check with echo 4 >/proc/asound/card0/pcm0c0/xrun_debug 3. applying the patch below. Since I wasn't able to reproduce the problem on my i686 PC, I guess the problem is probably machine specific. The patch reuses the method for software emulation of missing hardware pointer, already implemented for playback on OMAP1510. It's possible that event if a hardware pointer is available for capture on this machine, its behaviour may be not compatible with what upper layer expects. If you think the problem may be more general and should be solved differently, on a higher level, I can try to work more on it if you give me a hint. If the patch gets accepted, I suggest it goes as a fix in the current release cycle. Created and tested against linux-2.6.34-rc2. Signed-off-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 825db385f01f..bdd1097c7b13 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -60,12 +60,11 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data) struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { + if ((cpu_is_omap1510())) { /* * OMAP1510 doesn't fully support DMA progress counter * and there is no software emulation implemented yet, - * so have to maintain our own playback progress counter + * so have to maintain our own progress counters * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); @@ -189,8 +188,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else @@ -248,14 +246,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (cpu_is_omap1510()) { + offset = prtd->period_index * runtime->period_size; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else if (!(cpu_is_omap1510())) { + } else { ptr = omap_get_dma_src_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else - offset = prtd->period_index * runtime->period_size; + } if (offset >= runtime->buffer_size) offset = 0; -- cgit v1.2.2 From 3815595e78d2baae6feb866e737f92d8ef48b337 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 4 Apr 2010 12:14:03 +0200 Subject: ALSA: hda - Add MSI blacklist for Aopen MZ915-M The device needs MSI disablement. Added to the quirk list. Reported-by: Harald Dunkel Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4bb90675f70f..f8fd586ae024 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2362,6 +2362,7 @@ static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ + SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} }; -- cgit v1.2.2 From f11947c7c5b8abffd328739996dfdffef2b3e03f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 2 Apr 2010 14:29:23 +0300 Subject: ALSA: i2c: cleanup: change parameter to pointer We actually pass an array of 7 chars not 5. This silences a smatch warning. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4113.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index fff62cc8607c..971a84a4fa77 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -70,7 +70,7 @@ static int snd_ak4113_dev_free(struct snd_device *device) } int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, - ak4113_write_t *write, const unsigned char pgm[5], + ak4113_write_t *write, const unsigned char *pgm, void *private_data, struct ak4113 **r_ak4113) { struct ak4113 *chip; -- cgit v1.2.2 From a0fd4345f928d72a56e27b23e4cd28c94bf36be5 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Fri, 2 Apr 2010 14:47:59 +0200 Subject: ALSA: echoaudio - Eliminate use after free Use the call to snd_card_free in the error handling code at the end of the function, as in the other error cases. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ expression E,E2; @@ snd_card_free(E) ... ( E = E2 | * E ) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 8dab82d7d19d..668a5ec04499 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2184,10 +2184,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, goto ctl_error; #endif - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); + err = snd_card_register(card); + if (err < 0) goto ctl_error; - } snd_printk(KERN_INFO "Card registered: %s\n", card->longname); pci_set_drvdata(pci, chip); -- cgit v1.2.2 From 3fa49e3ad9ac20b15edfb0c51bbad36e45a84b17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 15:24:40 +0100 Subject: ASoC: Avoid wraparound in wm_hubs DC servo correction If the correction wraps around then a substantial offset would be introduced. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 486bdd21a98a..3729a12b151f 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -113,13 +113,15 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* HPOUT1L */ reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) & WM8993_DCS_INTEG_CHAN_0_MASK;; - reg += hubs->dcs_codes; + if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) + reg += hubs->dcs_codes; dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & WM8993_DCS_INTEG_CHAN_1_MASK; - reg += hubs->dcs_codes; + if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) + reg += hubs->dcs_codes; dcs_cfg |= reg; /* Do it */ -- cgit v1.2.2 From 8437f7006b9cfa249791e2fd57596683d4561843 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 17:09:45 +0100 Subject: ASoC: Support second DC servo readback method for wm_hubs More recent Wolfson hubs devices add the ability to read back the DC servo calibration information from the register used to write offsets, and later still ones remove the old readback registers. Add support for the new scheme, and use it for WM8994 device revisions that support it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 3 ++- sound/soc/codecs/wm_hubs.c | 41 ++++++++++++++++++++++++++++++----------- sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 33 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d10d65191fd2..c80218f23bb9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3730,11 +3730,12 @@ static int wm8994_codec_probe(struct platform_device *pdev) case 3: wm8994->hubs.dcs_codes = -5; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.dcs_readback_mode = 1; break; default: + wm8994->hubs.dcs_readback_mode = 1; break; } - /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 3729a12b151f..2b5c0924f615 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -86,7 +86,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = codec->private_data; - u16 reg, dcs_cfg; + u16 reg, reg_l, reg_r, dcs_cfg; /* Set for 32 series updates */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, @@ -110,19 +110,38 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Applying %d code DC servo correction\n", hubs->dcs_codes); + /* Different chips in the family support different + * readback methods. + */ + switch (hubs->dcs_readback_mode) { + case 0: + reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) + & WM8993_DCS_INTEG_CHAN_0_MASK;; + reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) + & WM8993_DCS_INTEG_CHAN_1_MASK; + break; + case 1: + reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; + reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + break; + default: + WARN(1, "Unknown DCS readback method"); + break; + } + /* HPOUT1L */ - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) & - WM8993_DCS_INTEG_CHAN_0_MASK;; - if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) - reg += hubs->dcs_codes; - dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + if (reg_l + hubs->dcs_codes > 0 && + reg_l + hubs->dcs_codes < 0xff) + reg_l += hubs->dcs_codes; + dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & - WM8993_DCS_INTEG_CHAN_1_MASK; - if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) - reg += hubs->dcs_codes; - dcs_cfg |= reg; + if (reg_r + hubs->dcs_codes > 0 && + reg_r + hubs->dcs_codes < 0xff) + reg_r += hubs->dcs_codes; + dcs_cfg |= reg_r; /* Do it */ snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 420104fe9c90..e51c16683589 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -21,6 +21,7 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; /* This *must* be the first element of the codec->private_data struct */ struct wm_hubs_data { int dcs_codes; + int dcs_readback_mode; int hp_startup_mode; }; -- cgit v1.2.2 From ae9d8607fe24253efc9f14b696f51cfd683801be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 16:34:42 +0100 Subject: ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction If we need to offset correct the DC servo then don't use runtime recalibration since that is likely to introduce further offsets which will be evident on powerdown. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2b5c0924f615..e81ba6d2d7cd 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -162,10 +162,16 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm_hubs_data *hubs = codec->private_data; int ret; ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + /* If we're applying an offset correction then updating the + * callibration would be likely to introduce further offsets. */ + if (hubs->dcs_codes) + return ret; + /* Only need to do this if the outputs are active */ if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1) & (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA)) -- cgit v1.2.2 From 4dcc93d0ede49fae32dd0ee41c685db1be14c529 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 17:18:41 +0100 Subject: ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices The DCS_DATAPATH_BUSY bit used to monitor the completion of DC servo operations has been deprecated and with some more recente revisions may perform incorrectly, especially when only analogue bypass paths are in use. Switch to using readback from the DC servo command register instead, which is supported for all devices. Without this unacceptably long timeouts may be observed in some circumstances. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 38 +++++++++++++++----------------------- 1 file changed, 15 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e81ba6d2d7cd..e1f225a3ac46 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -62,21 +62,27 @@ static const char *speaker_mode_text[] = { static const struct soc_enum speaker_mode = SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text); -static void wait_for_dc_servo(struct snd_soc_codec *codec) +static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { unsigned int reg; int count = 0; + unsigned int val; + + val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; + + /* Trigger the command */ + snd_soc_write(codec, WM8993_DC_SERVO_0, val); dev_dbg(codec->dev, "Waiting for DC servo...\n"); do { count++; msleep(1); - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400); + } while (reg & op && count < 400); - if (reg & WM8993_DCS_DATAPATH_BUSY) + if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); } @@ -92,18 +98,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_SERIES_NO_01_MASK, 32 << WM8993_DCS_SERIES_NO_01_SHIFT); - - /* Enable the DC servo. Write all bits to avoid triggering startup - * or write calibration. - */ - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - 0xFFFF, - WM8993_DCS_ENA_CHAN_0 | - WM8993_DCS_ENA_CHAN_1 | - WM8993_DCS_TRIG_SERIES_1 | - WM8993_DCS_TRIG_SERIES_0); - - wait_for_dc_servo(codec); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1); /* Apply correction to DC servo result */ if (hubs->dcs_codes) { @@ -145,13 +141,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Do it */ snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - WM8993_DCS_TRIG_DAC_WR_0 | - WM8993_DCS_TRIG_DAC_WR_1, - WM8993_DCS_TRIG_DAC_WR_0 | - WM8993_DCS_TRIG_DAC_WR_1); - - wait_for_dc_servo(codec); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1); } } -- cgit v1.2.2 From d522ffbfb9fccf6eca283cd2e8b03cf3d21fb616 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 30 Mar 2010 14:29:14 +0100 Subject: ASoC: Only do WM8994 bias off transition from standby Otherwise we may try to power down multiple times when the using idle bias off and the driver is removed. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 53 ++++++++++++++++++++++++++--------------------- 1 file changed, 29 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c80218f23bb9..f8355ac76a42 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3007,34 +3007,39 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - /* Switch over to startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (1 << WM8994_VMID_RAMP_SHIFT)); - - /* Disable main biases */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0); + if (codec->bias_level == SND_SOC_BIAS_STANDBY) { + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); - /* Discharge line */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH); + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, 0); - msleep(5); + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); - /* Switch off startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, 0); + msleep(5); + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + } break; } codec->bias_level = level; -- cgit v1.2.2 From d12841827a6de120199609dadb6ff4ec99bd90ea Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 5 Apr 2010 16:30:43 +0100 Subject: ALSA: hda - Enable amplifiers on Acer Inspire 6530G After more tests it appears that EAPD needs to be enabled on both the 0x14 and 0x15 NIDs to enable the main speaker and headphone amplifiers. The maximum volume setting is now equal to what the machine achieves under other operating systems. Disabling Front or LFE playback triggers EAPD and disables the amplifier. As such, these two playback switches have been removed from the mixer. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ca93c4cc144e..547206296d7b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1621,6 +1621,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { */ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Route to built-in subwoofer as well as speakers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Bias voltage on for external mic port */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, /* Front Mic: set to PIN_IN (empty by default) */ @@ -1632,10 +1637,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { /* Enable speaker output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, /* Enable headphone output */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -8462,9 +8469,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), -- cgit v1.2.2 From 5f712b2b73a9fc87fcc52124cfe8adefaa0c92f5 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Mar 2010 10:11:15 +0100 Subject: ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream This fixes a memory corruption when ASoC devices are used in full-duplex mode. Specifically for pxa-ssp code, where this pointer is dynamically allocated for each direction and destroyed upon each stream start. All other platforms are fixed blindly, I couldn't even compile-test them. Sorry for any breakage I may have caused. [Note that this is a backported version for 2.6.34. Upstream commit is fd23b7dee] Signed-off-by: Daniel Mack Reported-by: Sven Neumann Reported-by: Michael Hirsch Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 6 +++--- sound/soc/davinci/davinci-i2s.c | 3 ++- sound/soc/davinci/davinci-mcasp.c | 3 ++- sound/soc/davinci/davinci-pcm.c | 4 +++- sound/soc/imx/imx-pcm-dma-mx2.c | 8 ++++++-- sound/soc/imx/imx-ssi.c | 7 +++++-- sound/soc/omap/omap-mcbsp.c | 4 +++- sound/soc/omap/omap-mcpdm.c | 3 ++- sound/soc/omap/omap-pcm.c | 4 +++- sound/soc/pxa/pxa-ssp.c | 23 +++++++++++----------- sound/soc/pxa/pxa2xx-ac97.c | 17 ++++++++++++----- sound/soc/pxa/pxa2xx-i2s.c | 7 +++++-- sound/soc/pxa/pxa2xx-pcm.c | 4 +++- sound/soc/s3c24xx/s3c-ac97.c | 21 +++++++++++--------- sound/soc/s3c24xx/s3c-dma.c | 4 +++- sound/soc/s3c24xx/s3c-i2s-v2.c | 13 ++++++++----- sound/soc/s3c24xx/s3c-pcm.c | 7 +++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 19 ++++++++++--------- sound/soc/s6000/s6000-i2s.c | 3 ++- sound/soc/s6000/s6000-pcm.c | 40 ++++++++++++++++++++++++++++----------- 21 files changed, 131 insertions(+), 71 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 9ef6b96373f5..3e6628c8e665 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); - prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e588e63f18d2..0b59806905d1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, ssc_p->dma_params[dir] = dma_params; /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() + * The snd_soc_pcm_stream->dma_data field is only used to communicate + * the appropriate DMA parameters to the pcm driver hw_params() * function. It should not be used for other purposes * as it is common to all substreams. */ - rtd->dai->cpu_dai->dma_data = dma_params; + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params); channels = params_channels(params); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 6362ca05506e..4aad7ecc90a2 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -585,7 +585,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; - davinci_i2s_dai.dma_data = dev->dma_params; + davinci_i2s_dai.capture.dma_data = dev->dma_params; + davinci_i2s_dai.playback.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ab6518d86f18..c056bfbe0340 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -917,7 +917,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; - davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 80c7fdf2f521..2dc406f42fe7 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -649,8 +649,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_hardware *ppcm; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *pa; struct davinci_pcm_dma_params *params; + + pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); if (!pa) return -ENODEV; params = &pa[substream->stream]; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 19452e44afdc..c78c000e2afe 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -83,11 +83,13 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err) static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int ret; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); if (iprtd->dma < 0) { pr_err("Failed to claim the audio DMA\n"); @@ -192,10 +194,12 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int err; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->substream = substream; iprtd->buf = (unsigned int *)substream->dma_buffer.area; iprtd->period_cnt = 0; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 56f46a75d297..28e55c7b14b4 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -234,17 +234,20 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct imx_ssi *ssi = cpu_dai->private_data; + struct imx_pcm_dma_params *dma_data; u32 reg, sccr; /* Tx/Rx config */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { reg = SSI_STCCR; - cpu_dai->dma_data = &ssi->dma_params_tx; + dma_data = &ssi->dma_params_tx; } else { reg = SSI_SRCCR; - cpu_dai->dma_data = &ssi->dma_params_rx; + dma_data = &ssi->dma_params_rx; } + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; /* DAI data (word) size */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index e814a9591f78..8ad9dc901007 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -297,7 +297,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; omap_mcbsp_dai_dma_params[id][substream->stream].data_type = OMAP_DMA_DATA_TYPE_S16; - cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; + + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcbsp_dai_dma_params[id][substream->stream]); if (mcbsp_data->configured) { /* McBSP already configured by another stream */ diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 25f19e4728bf..b7f4f7e015f3 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -150,7 +150,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; int channels, err, link_mask = 0; - cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcpdm_dai_dma_params[stream]); channels = params_channels(params); switch (channels) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index bdd1097c7b13..39456447132c 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -99,9 +99,11 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_runtime_data *prtd = runtime->private_data; - struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data; + struct omap_pcm_dma_data *dma_data; int err = 0; + dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma_data) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 9e95e5117c88..6959c5199160 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -121,10 +121,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, ssp_disable(ssp); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); + return ret; } @@ -141,10 +140,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable(ssp->clk); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } #ifdef CONFIG_PM @@ -569,19 +566,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + struct pxa2xx_pcm_dma_params *dma_data; + + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); + kfree(dma_data); /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + dma_data = ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_dma_data(dai, substream, dma_data); + /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index e9ae7b3a7e00..d314115e3dd7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -122,11 +122,14 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; + dma_data = &pxa2xx_ac97_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in; + dma_data = &pxa2xx_ac97_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -137,11 +140,14 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; + dma_data = &pxa2xx_ac97_pcm_aux_mono_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -156,7 +162,8 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &pxa2xx_ac97_pcm_mic_mono_in); return 0; } diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b8f655d1ad8..c1a5275721e4 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -164,6 +164,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); @@ -171,9 +172,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; + dma_data = &pxa2xx_i2s_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in; + dma_data = &pxa2xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d38e39575f51..adc7e6f15f93 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -25,9 +25,11 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct pxa2xx_pcm_dma_params *dma; int ret; + dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma) diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c index ee8ed9d7e703..ecf4fd04ae96 100644 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -224,11 +224,14 @@ static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &s3c_ac97_pcm_out; + dma_data = &s3c_ac97_pcm_out; else - cpu_dai->dma_data = &s3c_ac97_pcm_in; + dma_data = &s3c_ac97_pcm_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -238,8 +241,8 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -265,7 +268,7 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } @@ -280,7 +283,7 @@ static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &s3c_ac97_mic_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); return 0; } @@ -290,8 +293,8 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -311,7 +314,7 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 7725e26d6c91..1b61c23ff300 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -145,10 +145,12 @@ static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); int ret = 0; + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index e994d8374fe6..88515946b6c0 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -339,14 +339,17 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = i2s->dma_playback; + dma_data = i2s->dma_playback; else - dai->cpu_dai->dma_data = i2s->dma_capture; + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(i2s->regs + S3C2412_IISMOD); @@ -394,8 +397,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -431,7 +434,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, * of the auto reload mechanism of S3C24XX. * This call won't bother S3C64XX. */ - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index a98f40c3cd29..326f0a9e7e30 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -178,6 +178,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -187,9 +188,11 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = pcm->dma_playback; + dma_data = pcm->dma_playback; else - dai->cpu_dai->dma_data = pcm->dma_capture; + dma_data = pcm->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Strictly check for sample size */ switch (params_format(params)) { diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0bc5950b9f02..c3ac890a3986 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -242,14 +242,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; + dma_data = &s3c24xx_i2s_pcm_stereo_out; else - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in; + dma_data = &s3c24xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -258,13 +261,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 1; + dma_data->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 2; + dma_data->dma_size = 2; break; default: return -EINVAL; @@ -280,8 +281,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -300,7 +301,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index c5cda187ecab..fa23854c5f3a 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -518,7 +518,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) s6000_i2s_dai.dev = &pdev->dev; s6000_i2s_dai.private_data = dev; - s6000_i2s_dai.dma_data = &dev->dma_params; + s6000_i2s_dai.capture.dma_data = &dev->dma_params; + s6000_i2s_dai.playback.dma_data = &dev->dma_params; dev->sifbase = sifmem->start; dev->scbbase = mmio; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1d61109e09fa..9c7f7f00cebb 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -58,13 +58,15 @@ static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int channel; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; dma_addr_t src, dst; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; @@ -101,7 +103,8 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data) { struct snd_pcm *pcm = data; struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); struct s6000_runtime_data *prtd; unsigned int has_xrun; int i, ret = IRQ_NONE; @@ -172,11 +175,13 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream) { struct s6000_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; unsigned long flags; int srcinc; u32 dma; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -212,10 +217,12 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream) { struct s6000_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; unsigned long flags; u32 channel; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) channel = par->dma_out; else @@ -236,9 +243,11 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream) static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + ret = par->trigger(substream, cmd, 0); if (ret < 0) return ret; @@ -275,13 +284,15 @@ static int s6000_pcm_prepare(struct snd_pcm_substream *substream) static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd = runtime->private_data; unsigned long flags; unsigned int offset; dma_addr_t count; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -305,11 +316,12 @@ static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) static int s6000_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); ret = snd_pcm_hw_constraint_step(runtime, 0, @@ -364,7 +376,7 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int ret; ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -373,6 +385,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, return ret; } + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (par->same_rate) { spin_lock(&par->lock); if (par->rate == -1 || @@ -392,7 +406,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); spin_lock(&par->lock); par->in_use &= ~(1 << substream->stream); @@ -417,7 +432,8 @@ static struct snd_pcm_ops s6000_pcm_ops = { static void s6000_pcm_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); free_irq(params->irq, pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -429,9 +445,11 @@ static int s6000_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params; int res; + params = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) -- cgit v1.2.2 From f9700d5a4575e7fb343df10a1d29d425e4b81082 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Apr 2010 23:25:13 +0200 Subject: ALSA: hda - Fix a wrong array range check in patch_realtek.c The commit 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 introduced a wrong comparision for the array range check, which effectively skips the whole initialization of DAC connections. Fixed now. Reference: bko#15689 https://bugzilla.kernel.org/show_bug.cgi?id=15689 Reported-by: Adrian Ulrich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 547206296d7b..c7730dbb9ddb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10110,13 +10110,12 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, int idx; alc_set_pin_output(codec, nid, pin_type); + if (dac_idx >= spec->multiout.num_dacs) + return; if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; - else { - if (spec->multiout.num_dacs >= dac_idx) - return; + else idx = spec->multiout.dac_nids[dac_idx] - 2; - } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } -- cgit v1.2.2 From b0cc58a25d04160d39a80e436847eaa2fbc5aa09 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 6 Apr 2010 19:31:26 +0300 Subject: ALSA: mixart: range checking proc file The original code doesn't take into consideration that the value of MIXART_BA0_SIZE - pos can be less than zero which would lead to a large unsigned value for "count". Also I moved the check that read size is a multiple of 4 bytes below the code that adjusts "count". Signed-off-by: Dan Carpenter Cc: Acked-by: Linus Torvalds Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7e8e7da592a9..ea4256b08a38 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1161,13 +1161,15 @@ static long snd_mixart_BA0_read(struct snd_info_entry *entry, void *file_private unsigned long count, unsigned long pos) { struct mixart_mgr *mgr = entry->private_data; + unsigned long maxsize; - count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (pos >= MIXART_BA0_SIZE) return 0; - if(pos + count > MIXART_BA0_SIZE) - count = (long)(MIXART_BA0_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count)) + maxsize = MIXART_BA0_SIZE - pos; + if (count > maxsize) + count = maxsize; + count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ + if (copy_to_user_fromio(buf, MIXART_MEM(mgr, pos), count)) return -EFAULT; return count; } @@ -1180,13 +1182,15 @@ static long snd_mixart_BA1_read(struct snd_info_entry *entry, void *file_private unsigned long count, unsigned long pos) { struct mixart_mgr *mgr = entry->private_data; + unsigned long maxsize; - count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (pos > MIXART_BA1_SIZE) return 0; - if(pos + count > MIXART_BA1_SIZE) - count = (long)(MIXART_BA1_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count)) + maxsize = MIXART_BA1_SIZE - pos; + if (count > maxsize) + count = maxsize; + count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ + if (copy_to_user_fromio(buf, MIXART_REG(mgr, pos), count)) return -EFAULT; return count; } -- cgit v1.2.2 From 7ad7b218f4aae4f395b3b4cef261572556bbd20a Mon Sep 17 00:00:00 2001 From: Maurus Cuelenaere Date: Tue, 6 Apr 2010 18:12:52 +0200 Subject: ALSA: hda: Add support for Medion WIM2160 This adds support for the Medion WIM2160 soundcard. There's no PCI quirk added because it has the same PCI id as the Medion MD2. Signed-off-by: Maurus Cuelenaere Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 53 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7730dbb9ddb..2971e48e50ad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -230,6 +230,7 @@ enum { ALC888_ACER_ASPIRE_7730G, ALC883_MEDION, ALC883_MEDION_MD2, + ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, @@ -8455,6 +8456,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc883_medion_wim2160_verbs[] = { + /* Unmute front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Set speaker pin to front mixer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Init headphone pin */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc883_medion_wim2160_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1a; + spec->autocfg.speaker_pins[0] = 0x15; +} + static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9164,6 +9201,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", + [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", @@ -9818,6 +9856,21 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc883_medion_md2_setup, .init_hook = alc_automute_amp, }, + [ALC883_MEDION_WIM2160] = { + .mixers = { alc883_medion_wim2160_mixer }, + .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc883_medion_wim2160_setup, + .init_hook = alc_automute_amp, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, -- cgit v1.2.2 From 78e4fd26ef8b85c8cbb6803e18b6b1f970420e06 Mon Sep 17 00:00:00 2001 From: Huang Weiyi Date: Thu, 8 Apr 2010 19:50:08 +0800 Subject: ASoC: wm2000: remove unused #include Remove unused #include ('s) in sound/soc/codecs/wm2000.c Signed-off-by: Huang Weiyi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 217b02680597..8de866618bf4 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -23,7 +23,6 @@ #include #include -#include #include #include #include -- cgit v1.2.2 From 206b60e189c7cc2b4675687d66f167299a13a4d4 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 8 Apr 2010 11:31:24 +0200 Subject: ASoC: imx-ssi: honor IMX_SSI_DMA flag When checking if we are DMA capable we have to check for the IMX_SSI_DMA flag which is already set from platform_data instead of setting it again when we want to do DMA. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 28e55c7b14b4..1bf9dc88babf 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -655,7 +655,8 @@ static int imx_ssi_probe(struct platform_device *pdev) dai->private_data = ssi; if ((cpu_is_mx27() || cpu_is_mx21()) && - !(ssi->flags & IMX_SSI_USE_AC97)) { + !(ssi->flags & IMX_SSI_USE_AC97) && + (ssi->flags & IMX_SSI_DMA)) { ssi->flags |= IMX_SSI_DMA; platform = imx_ssi_dma_mx2_init(pdev, ssi); } else -- cgit v1.2.2 From 671999cb5d8817611f865f3877f5a5b81372f61e Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 8 Apr 2010 11:31:25 +0200 Subject: ASoC: imx-pcm-dma-mx2: restart DMA after an error Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index c78c000e2afe..93272966b848 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -70,7 +70,12 @@ static void imx_ssi_dma_callback(int channel, void *data) static void snd_imx_dma_err_callback(int channel, void *data, int err) { - pr_err("DMA error callback called\n"); + struct snd_pcm_substream *substream = data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int ret; pr_err("DMA timeout on channel %d -%s%s%s%s\n", channel, @@ -78,6 +83,14 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err) err & IMX_DMA_ERR_REQUEST ? " request" : "", err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); + + imx_dma_disable(iprtd->dma); + ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, + IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_MODE_WRITE : DMA_MODE_READ); + if (!ret) + imx_dma_enable(iprtd->dma); } static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) -- cgit v1.2.2 From 43a3cec01354573517f1348383e0ab6e6067076b Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 8 Apr 2010 11:31:26 +0200 Subject: ASoC: imx-ssi: Use a hrtimer in FIQ mode Using a regular timer results in poll times < 1 jiffie with small buffers, so we loaded the timer with the actual jiffie value. We can be more accurate using a hrtimer. Also, we have to call snd_pcm_period_elapsed after playing period_bytes and not runtime->period_size (which is in samples and not in bytes). Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 45 +++++++++++++++++++++------------------------ 1 file changed, 21 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index d9cb9849b033..64df813b9af8 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -38,20 +38,17 @@ struct imx_pcm_runtime_data { unsigned long offset; unsigned long last_offset; unsigned long size; - struct timer_list timer; - int poll_time; + struct hrtimer hrt; + int poll_time_ns; + struct snd_pcm_substream *substream; }; -static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd) +static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { - iprtd->timer.expires = jiffies + iprtd->poll_time; -} - -static void imx_ssi_timer_callback(unsigned long data) -{ - struct snd_pcm_substream *substream = (void *)data; + struct imx_pcm_runtime_data *iprtd = + container_of(hrt, struct imx_pcm_runtime_data, hrt); + struct snd_pcm_substream *substream = iprtd->substream; struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; struct pt_regs regs; unsigned long delta; @@ -71,16 +68,14 @@ static void imx_ssi_timer_callback(unsigned long data) /* If we've transferred at least a period then report it and * reset our poll time */ - if (delta >= runtime->period_size) { + if (delta >= iprtd->period) { snd_pcm_period_elapsed(substream); iprtd->last_offset = iprtd->offset; - - imx_ssi_set_next_poll(iprtd); } - /* Restart the timer; if we didn't report we'll run on the next tick */ - add_timer(&iprtd->timer); + hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); + return HRTIMER_RESTART; } static struct fiq_handler fh = { @@ -98,8 +93,8 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->period = params_period_bytes(params) ; iprtd->offset = 0; iprtd->last_offset = 0; - iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params)); - + iprtd->poll_time_ns = 1000000000 / params_rate(params) * + params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; @@ -134,8 +129,8 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - imx_ssi_set_next_poll(iprtd); - add_timer(&iprtd->timer); + hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), + HRTIMER_MODE_REL); if (++fiq_enable == 1) enable_fiq(imx_pcm_fiq); @@ -144,7 +139,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - del_timer(&iprtd->timer); + hrtimer_cancel(&iprtd->hrt); if (--fiq_enable == 0) disable_fiq(imx_pcm_fiq); @@ -193,9 +188,10 @@ static int snd_imx_open(struct snd_pcm_substream *substream) iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); runtime->private_data = iprtd; - init_timer(&iprtd->timer); - iprtd->timer.data = (unsigned long)substream; - iprtd->timer.function = imx_ssi_timer_callback; + iprtd->substream = substream; + + hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + iprtd->hrt.function = snd_hrtimer_callback; ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -211,7 +207,8 @@ static int snd_imx_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - del_timer_sync(&iprtd->timer); + hrtimer_cancel(&iprtd->hrt); + kfree(iprtd); return 0; -- cgit v1.2.2 From 531d8791accf1464bc6854ff69d08dd866189d17 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 9 Apr 2010 10:57:33 +0200 Subject: ALSA: hda - Fix auto-parser of ALC269vb for HP pin NID 0x21 ALC269vb has an alternative HP pin 0x21 in addition. Fix the parser to recognize it. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2971e48e50ad..fbbdfbc8a1ca 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12869,6 +12869,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x21: /* ALC269vb has this pin, too */ dac = 0x03; break; default: -- cgit v1.2.2 From 226b1ec8c18bcb6d1aa448a29b2c8aeae1946228 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 9 Apr 2010 11:01:20 +0200 Subject: ALSA: hda - Fix setup for ALC269vb amic and dmic models Corrected HP and mic pins for ALC269vb amic and dmic models. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fbbdfbc8a1ca..9b58f29833e6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13789,19 +13789,19 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec, } } -static void alc269_laptop_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; - spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 5; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; spec->auto_mic = 1; } -static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; @@ -13809,14 +13809,14 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 6; + spec->int_mic.mux_idx = 5; spec->auto_mic = 1; } -static void alc269_laptop_amic_setup(struct hda_codec *codec) +static void alc269vb_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -13825,6 +13825,18 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec) spec->auto_mic = 1; } +static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 6; + spec->auto_mic = 1; +} + static void alc269_laptop_inithook(struct hda_codec *codec) { alc269_speaker_automute(codec); @@ -14162,7 +14174,7 @@ static struct alc_config_preset alc269_presets[] = { .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, .unsol_event = alc269_laptop_unsol_event, - .setup = alc269_laptop_amic_setup, + .setup = alc269vb_laptop_amic_setup, .init_hook = alc269_laptop_inithook, }, [ALC269VB_DMIC] = { -- cgit v1.2.2 From 7f311a46916a3be00a1a8e3f1bdf461d08f1d263 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Apr 2010 17:32:23 +0200 Subject: ALSA: hda - Fix initial capture source connections of ALC880/260 The widget connections of ADC of ALC880 and ALC2260 aren't initialized, thus it might point to invalid pin. This can be a problem when mode=auto and there is only one input pin. Then user can't change the connection at all. This patch adds the code to initialize the input pin connection of these codecs. Reference: Novell bnc#594363 https://bugzilla.novell.com/show_bug.cgi?id=594363 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b58f29833e6..8d60b1f25ce1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4809,6 +4809,25 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) } } +static void alc880_auto_init_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int c; + + for (c = 0; c < spec->num_adc_nids; c++) { + unsigned int mux_idx; + const struct hda_input_mux *imux; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; + if (!imux->num_items && mux_idx > 0) + imux = &spec->input_mux[0]; + if (imux) + snd_hda_codec_write(codec, spec->adc_nids[c], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[0].index); + } +} + /* parse the BIOS configuration and set up the alc_spec */ /* return 1 if successful, 0 if the proper config is not found, * or a negative error code @@ -4887,6 +4906,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); + alc880_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -6398,6 +6418,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) } } +#define alc260_auto_init_input_src alc880_auto_init_input_src + /* * generic initialization of ADC, input mixers and output mixers */ @@ -6484,6 +6506,7 @@ static void alc260_auto_init(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); + alc260_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } -- cgit v1.2.2 From 29aac005ff4dc8a5f50b80f4e5c4f59b21c0fb50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 10 Apr 2010 21:27:23 +0200 Subject: ALSA: usb - Fix Oops after usb-midi disconnection usb-midi causes sometimes Oops at snd_usbmidi_output_drain() after disconnection. This is due to the access to the endpoints which have been already released at disconnection while the files are still alive. This patch fixes the problem by checking disconnection state at snd_usbmidi_output_drain() and by releasing urbs but keeping the endpoint instances until really all freed. Tested-by: Tvrtko Ursulin Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbmidi.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2c59afd99611..9e28b20cb2ce 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) DEFINE_WAIT(wait); long timeout = msecs_to_jiffies(50); + if (ep->umidi->disconnected) + return; /* * The substream buffer is empty, but some data might still be in the * currently active URBs, so we have to wait for those to complete. @@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, * Frees an output endpoint. * May be called when ep hasn't been initialized completely. */ -static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep) { unsigned int i; for (i = 0; i < OUTPUT_URBS; ++i) - if (ep->urbs[i].urb) + if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); + ep->urbs[i].urb = NULL; + } +} + +static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep) +{ + snd_usbmidi_out_endpoint_clear(ep); kfree(ep); } @@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out); + ep->out->active_urbs = 0; + if (ep->out->drain_urbs) { + ep->out->drain_urbs = 0; + wake_up(&ep->out->drain_wait); + } } if (ep->in) for (j = 0; j < INPUT_URBS; ++j) usb_kill_urb(ep->in->urbs[j]); /* free endpoints here; later call can result in Oops */ - if (ep->out) { - snd_usbmidi_out_endpoint_delete(ep->out); - ep->out = NULL; - } + if (ep->out) + snd_usbmidi_out_endpoint_clear(ep->out); if (ep->in) { snd_usbmidi_in_endpoint_delete(ep->in); ep->in = NULL; -- cgit v1.2.2 From 7fa90e873f520dad5ec58f47340996cda083e875 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Apr 2010 08:49:00 +0200 Subject: ALSA: hda - Enhance fix-up table for Realtek codecs A few enhancement / fixes for fix-up table of some Realtek codecs: - Apply fix-ups only for the auto model - Apply additional verbs after normal init verbs - Add a debug print to show the fix-up application This is basically a preliminary work for the next fix for Sony VAIO. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 35 ++++++++++++++++++++++++++++------- 1 file changed, 28 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8d60b1f25ce1..cff57710d1fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1390,22 +1390,31 @@ struct alc_fixup { static void alc_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, - const struct alc_fixup *fix) + const struct alc_fixup *fix, + int pre_init) { const struct alc_pincfg *cfg; quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (!quirk) return; - fix += quirk->value; cfg = fix->pins; - if (cfg) { + if (pre_init && cfg) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n", + codec->chip_name, quirk->name); +#endif for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - if (fix->verbs) + if (!pre_init && fix->verbs) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n", + codec->chip_name, quirk->name); +#endif add_verb(codec->spec, fix->verbs); + } } static int alc_read_coef_idx(struct hda_codec *codec, @@ -10439,7 +10448,8 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1); if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ @@ -10512,6 +10522,9 @@ static int patch_alc882(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0); + spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; @@ -15417,7 +15430,8 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups); + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1); if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ @@ -15454,6 +15468,9 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) { spec->init_hook = alc861_auto_init; @@ -16388,7 +16405,8 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups); + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1); if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ @@ -16436,6 +16454,9 @@ static int patch_alc861vd(struct hda_codec *codec) spec->vmaster_nid = 0x02; + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861VD_AUTO) -- cgit v1.2.2 From ff818c24c2af370153646d302d831b69b023816f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Apr 2010 08:59:25 +0200 Subject: ALSA: hda - Add fix-up for Sony VAIO with ALC269 Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF ground or Hi-Z to make the headphone working. Other than that, model=auto works fine, so let's use model=auto with a specific fix-up table. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cff57710d1fb..4b35176d3454 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14077,6 +14077,27 @@ static void alc269_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC269_FIXUP_SONY_VAIO, +}; + +const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = { + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, + {} +}; + +static const struct alc_fixup alc269_fixups[] = { + [ALC269_FIXUP_SONY_VAIO] = { + .verbs = alc269_sony_vaio_fixup_verbs + }, +}; + +static struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + {} +}; + + /* * configuration and preset */ @@ -14136,7 +14157,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC), + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), @@ -14290,6 +14311,9 @@ static int patch_alc269(struct hda_codec *codec) board_config = ALC269_AUTO; } + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1); + if (board_config == ALC269_AUTO) { /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); @@ -14342,6 +14366,9 @@ static int patch_alc269(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0); + spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; -- cgit v1.2.2 From b68b58fd6a341c2115ff5fb466fe9fc0b581980e Mon Sep 17 00:00:00 2001 From: Philby John Date: Fri, 26 Mar 2010 21:37:51 +0530 Subject: ALSA: aaci - Fix alignment faults on ARM Cortex introduced by commit 29a4f2d3 The commit 29a4f2d3 used writel() at offset 0x26 which is half-word aligned causing unaligned exceptions on a Cortex-A8. The original patch solved the "aaci-pl041 fpga:04: ac97 read back fail" issue on a soft reset. Reading from any arbitrary aaci register seems to solve this issue. Signed-off-by: Philby John Acked-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 656e474dca47..91acc9a243ec 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -863,7 +863,6 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; - writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ @@ -1047,7 +1046,11 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) writel(0x1fff, aaci->base + AACI_INTCLR); writel(aaci->maincr, aaci->base + AACI_MAINCR); - + /* + * Fix: ac97 read back fail errors by reading + * from any arbitrary aaci register. + */ + readl(aaci->base + AACI_CSCH1); ret = aaci_probe_ac97(aaci); if (ret) goto out; -- cgit v1.2.2 From b331439dfd41dc813b3557ca5927a3a644f35792 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Apr 2010 14:33:57 +0200 Subject: ALSA: hda - Fix control element allocations in VIA codec parser The commit 5b0cb1d850c26893b1468b3a519433a1b7a176be ALSA: hda - add more NID->Control mapping breaks the control element allocation by returning a wrong value. Let's fix it. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9ddc37300f6b..be1295438989 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -476,7 +476,7 @@ static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, knew->name = kstrdup(tmpl->name, GFP_KERNEL); if (!knew->name) return NULL; - return 0; + return knew; } static void via_free_kctls(struct hda_codec *codec) -- cgit v1.2.2 From 3d83e577a8206f0f3822a3840e12f76477142ba2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Apr 2010 14:36:23 +0200 Subject: ALSA: hda - Avoid invalid "Independent HP" control for VIA codecs Some VIA codecs have no multiple source selection for headphone pins, thus it's useless (and wrong) to create "Independent HP" control on them. This patch adds the check of connections to skip the control in such a case. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 39 +++++++++++++++++++++++---------------- 1 file changed, 23 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index be1295438989..73453814e098 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1215,14 +1215,13 @@ static struct snd_kcontrol_new via_hp_mixer[2] = { }, }; -static int via_hp_build(struct via_spec *spec) +static int via_hp_build(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; struct snd_kcontrol_new *knew; hda_nid_t nid; - - knew = via_clone_control(spec, &via_hp_mixer[0]); - if (knew == NULL) - return -ENOMEM; + int nums; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; switch (spec->codec_type) { case VT1718S: @@ -1239,6 +1238,14 @@ static int via_hp_build(struct via_spec *spec) break; } + nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + + knew = via_clone_control(spec, &via_hp_mixer[0]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; knew->private_value = nid; @@ -2561,7 +2568,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -3087,7 +3094,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -3654,7 +3661,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -4140,7 +4147,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -4510,7 +4517,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } @@ -4930,7 +4937,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); @@ -5425,7 +5432,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); @@ -5781,7 +5788,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } @@ -6000,12 +6007,12 @@ static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, /* Line-Out: PortE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, - "Master Front Playback Volume", + "Front Playback Volume", HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, - "Master Front Playback Switch", + "Front Playback Switch", HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -6130,7 +6137,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } -- cgit v1.2.2 From 565a79f74af96ae90dfec411da14dc38d2cd56bc Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 14 Apr 2010 09:17:31 +0200 Subject: ASoC: imx-ssi: increase minimum periods to 4 Currently the notification of elapsed periods is not very exact. Increase minimum periods to 4 as suggested by Liam Girdwood. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 64df813b9af8..98ab33109527 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -174,7 +174,7 @@ static struct snd_pcm_hardware snd_imx_hardware = { .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 16 * 1024, - .periods_min = 2, + .periods_min = 4, .periods_max = 255, .fifo_size = 0, }; -- cgit v1.2.2 From d1501ea844eefdf925f6b711875b4b2b928fddf8 Mon Sep 17 00:00:00 2001 From: Joerg Schirottke Date: Thu, 15 Apr 2010 08:37:41 +0200 Subject: ALSA: hda - add a quirk for Clevo M570U laptop Added the matching model for Clevo laptop M570U. Signed-off-by: Joerg Schirottke Tested-by: Maximilian Gerhard Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4b35176d3454..aad1627f56f1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9350,6 +9350,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), -- cgit v1.2.2 From 8815cd030fdd73932a791d1f06194c8db807cde7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Apr 2010 09:02:41 +0200 Subject: ALSA: hda - Add position_fix quirk for Biostar mobo The Biostar mobo seems to give a wrong DMA position, resulting in stuttering or skipping sounds on 2.6.34. Since the commit 7b3a177b0d4f92b3431b8dca777313a07533a710, "ALSA: pcm_lib: fix "something must be really wrong" condition", makes the position check more strictly, the DMA position problem is revealed more clearly now. The fix is to use only LPIB for obtaining the position, i.e. passing position_fix=1. This patch adds a static quirk to achieve it as default. Reported-by: Frank Griffin Cc: Eric Piel Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f8fd586ae024..f669442b7c82 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2272,6 +2272,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; -- cgit v1.2.2 From 8392609969b3b37a4da5cff08161661f7a8c16af Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 14 Apr 2010 09:17:30 +0200 Subject: ASoC: imx-ssi: do not call hrtimer_disable in trigger function Doing so causes a deadlock, so just signal the timer to stop using an atomic variable. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 98ab33109527..ecec332121f2 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -41,6 +41,7 @@ struct imx_pcm_runtime_data { struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; + atomic_t running; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) @@ -52,6 +53,9 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct pt_regs regs; unsigned long delta; + if (!atomic_read(&iprtd->running)) + return HRTIMER_NORESTART; + get_fiq_regs(®s); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -129,6 +133,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + atomic_set(&iprtd->running, 1); hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), HRTIMER_MODE_REL); if (++fiq_enable == 1) @@ -139,11 +144,11 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - hrtimer_cancel(&iprtd->hrt); + atomic_set(&iprtd->running, 0); + if (--fiq_enable == 0) disable_fiq(imx_pcm_fiq); - break; default: return -EINVAL; @@ -190,6 +195,7 @@ static int snd_imx_open(struct snd_pcm_substream *substream) iprtd->substream = substream; + atomic_set(&iprtd->running, 0); hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); iprtd->hrt.function = snd_hrtimer_callback; -- cgit v1.2.2 From b7d2526f5c20385894a5e57b1a4292f5a1741f1b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Apr 2010 18:11:29 +0200 Subject: ALSA: hda - Fix resume from StR of HP 2510p with docking-station When HP laptop with AD1981 codec is suspended and the docking-station is connected before the resume, the outputs get confused, and wrongly routed still to the speaker. This is because of a change in 2.6.34-rc1 ea52bf260ecbb175339af3178c15788df21b7516 ALSA: hda: Add powerdown for Analog Devices HDA codecs The problem was the added resume callback that doesn't consider the modified init hook. The fix is simply remove the resume callback here and make the resume normally. This doesn't change any behavior intended in the commit above (for shutting down the sound at suspend) but only fixes the resume. Reported-and-tested-by: Frans Pop Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index af34606c30c3..e9fdfc4b1c57 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -519,14 +519,6 @@ static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) ad198x_power_eapd(codec); return 0; } - -static int ad198x_resume(struct hda_codec *codec) -{ - ad198x_init(codec); - snd_hda_codec_resume_amp(codec); - snd_hda_codec_resume_cache(codec); - return 0; -} #endif static struct hda_codec_ops ad198x_patch_ops = { @@ -539,7 +531,6 @@ static struct hda_codec_ops ad198x_patch_ops = { #endif #ifdef SND_HDA_NEEDS_RESUME .suspend = ad198x_suspend, - .resume = ad198x_resume, #endif .reboot_notify = ad198x_shutup, }; -- cgit v1.2.2 From aac78daf8f37256283f56820ae858add7139c56c Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 21 Apr 2010 20:41:52 -0400 Subject: ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio XPS 1645 BugLink: https://launchpad.net/bugs/553002 The OR has verified that the dell-m6 model quirk is necessary for audio to be audible by default on the Dell Studio XPS 1645. This change is necessary for 2.6.32.11 and 2.6.33.2 alike. Reported-by: Robert Chambers Tested-by: Robert Chambers Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c4be3fab94e5..81ecd9388a80 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1607,6 +1607,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "Dell Studio 1555", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd, "Dell Studio 1557", STAC_DELL_M6_DMIC), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe, + "Dell Studio XPS 1645", STAC_DELL_M6_BOTH), {} /* terminator */ }; -- cgit v1.2.2 From 3353541fe533350a22a03e2fb7dc085b35912575 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Thu, 22 Apr 2010 07:15:26 -0400 Subject: ALSA: hda: Use ALC880_F1734 quirk for Fujitsu Siemens AMILO Xi 1526 BugLink: https://launchpad.net/bugs/567494 The OR has verified that the existing model quirk, ALC880_UNIWILL, is insufficient for audible playback and capture by default. Instead, the ALC880_F1734 model quirk needs to be used. This change is necessary for both 2.6.32.11 and 2.6.33.2. Reported-by: Arnaud Malpeyre Tested-by: Arnaud Malpeyre Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aad1627f56f1..7404dba16f83 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4143,7 +4143,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), - SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), + SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), -- cgit v1.2.2 From 7efbfd1ae98ef9efe06352e2a1ad83e8c14ceeb1 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 21 Apr 2010 11:04:06 -0400 Subject: ALSA: snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C Without this quirk sound stops working after suspend resume. With this quirk, one still needs to manually unmute the master volume control after a suspend / / resume cycle. That is fixed in another patch in this set. Note that this patch was submitted to the alsa bug tracker a long time ago: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319 Signed-off-by: Hans de Goede CC: Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index b64e78139d63..728de232e091 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -884,6 +884,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { MODULE_DEVICE_TABLE(pci, snd_m3_ids); static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = { + SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c), SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d), SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d), SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03), -- cgit v1.2.2 From 715aa675338ce6e1a3b4f77cf87ea611f93058a8 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 21 Apr 2010 11:04:08 -0400 Subject: ALSA: snd-meastro3: Ignore spurious HV interrupts during suspend / resume Ignore spurious HV interrupts during suspend / resume, this avoids mistaking them for a mute button press. This is not very pretty but it seems the only way to fix the master volume control gets muted after suspend issue I'm seeing. Note that the es1968 driver is doing exactly the same. Signed-off-by: Hans de Goede Cc: Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 728de232e091..b56e33676780 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -849,6 +849,7 @@ struct snd_m3 { struct snd_kcontrol *master_switch; struct snd_kcontrol *master_volume; struct tasklet_struct hwvol_tq; + unsigned int in_suspend; #ifdef CONFIG_PM u16 *suspend_mem; @@ -1614,6 +1615,11 @@ static void snd_m3_update_hw_volume(unsigned long private_data) outb(0x88, chip->iobase + SHADOW_MIX_REG_MASTER); outb(0x88, chip->iobase + HW_VOL_COUNTER_MASTER); + /* Ignore spurious HV interrupts during suspend / resume, this avoids + mistaking them for a mute button press. */ + if (chip->in_suspend) + return; + if (!chip->master_switch || !chip->master_volume) return; @@ -2425,6 +2431,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state) if (chip->suspend_mem == NULL) return 0; + chip->in_suspend = 1; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_pcm_suspend_all(chip->pcm); snd_ac97_suspend(chip->ac97); @@ -2498,6 +2505,7 @@ static int m3_resume(struct pci_dev *pci) snd_m3_hv_init(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D0); + chip->in_suspend = 0; return 0; } #endif /* CONFIG_PM */ -- cgit v1.2.2 From 0e0280dc2b0c7395a880d25544b47f3e3e3f79db Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 21 Apr 2010 19:55:43 -0400 Subject: ALSA: hda: Use LPIB quirk for DG965OT board version AAD63733-203 BugLink: https://launchpad.net/bugs/459083 The OR has verified with 2.6.32.11 and the latest alsa-driver stable daily snapshot that position_fix=1 is necessary for the external mic to work and for PulseAudio not to crash constantly. This patch is necessary also for 2.6.32.11 and 2.6.33.2. Reported-by: Tested-by: Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f669442b7c82..cec68152dcb1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2273,6 +2273,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; -- cgit v1.2.2 From 5c1bccf645d4ab65e4c7502acb42e8b9afdb5bdc Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Thu, 22 Apr 2010 17:54:45 -0400 Subject: ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio 1558 BugLink: https://launchpad.net/bugs/568600 The OR has verified that the dell-m6 model quirk is necessary for audio to be audible by default on the Dell Studio XPS 1645. This change is necessary for 2.6.32.11 and 2.6.33.2 alike. Reported-by: Andy Ross Tested-by: Andy Ross Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 81ecd9388a80..7fb7d017a347 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1609,6 +1609,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "Dell Studio 1557", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe, "Dell Studio XPS 1645", STAC_DELL_M6_BOTH), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413, + "Dell Studio 1558", STAC_DELL_M6_BOTH), {} /* terminator */ }; -- cgit v1.2.2 From 867f1845c53f52e6b9822bea387c7b16740ba2f8 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 25 Apr 2010 13:12:45 +0200 Subject: ALSA: es968: fix wrong PnP dma index There is only one dma for the ESS ES968 based board. Its index is 0 and not 1. This make the es968 card working. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sb/es968.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index cafc3a7316a8..ff18286fef9d 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -93,7 +93,7 @@ static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard, return err; } port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 1); + dma8[dev] = pnp_dma(pdev, 0); irq[dev] = pnp_irq(pdev, 0); return 0; -- cgit v1.2.2 From b0b4ce38a535ed3de5ec6fdd4f3c34435a1c1d1e Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Thu, 8 Apr 2010 20:52:00 +0200 Subject: MIPS: TXx9: Add missing MODULE_ALIAS definitions for TXx9 platform devices This enables autoloading of the TXx9 sound driver on RBTX4927. Signed-off-by: Geert Uytterhoeven To: Atsushi Nemoto Cc: Linux MIPS Mailing List Patchwork: http://patchwork.linux-mips.org/patch/1101/ Signed-off-by: Ralf Baechle --- sound/soc/txx9/txx9aclc-ac97.c | 1 + sound/soc/txx9/txx9aclc-generic.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 612e18b4bf4e..0ec20b68e8cb 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -254,3 +254,4 @@ module_exit(txx9aclc_ac97_exit); MODULE_AUTHOR("Atsushi Nemoto "); MODULE_DESCRIPTION("TXx9 ACLC AC97 driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:txx9aclc-ac97"); diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c index 3175de9a92cb..95b17f731aec 100644 --- a/sound/soc/txx9/txx9aclc-generic.c +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -96,3 +96,4 @@ module_exit(txx9aclc_generic_exit); MODULE_AUTHOR("Atsushi Nemoto "); MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:txx9aclc-generic"); -- cgit v1.2.2 From 8dd34ab111dc6ccb35a1a7a59222cb9bb0160e6f Mon Sep 17 00:00:00 2001 From: "Brian J. Tarricone" Date: Sun, 2 May 2010 17:32:10 -0700 Subject: ALSA: hda - fix array indexing while creating inputs for Cirrus codecs This fixes a problem where cards show up as only having a single mixer element, suppressing all sound output. Signed-off-by: Brian J. Tarricone Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7de782a5b8f4..350ee8ac4153 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -766,7 +766,7 @@ static int build_input(struct hda_codec *codec) for (n = 0; n < AUTO_PIN_LAST; n++) { if (!spec->adc_nid[n]) continue; - err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]); + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); if (err < 0) return err; } -- cgit v1.2.2 From 4442dd4613fe3795b4c8a5f42fc96b7ffb90d01a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Mon, 3 May 2010 20:39:31 -0400 Subject: ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F BugLink: https://launchpad.net/bugs/573284 The OR verified that using the olpc-xo-1_5 model quirk allows the headphones to be audible when inserted into the jack. Capture was also verified to work correctly. Reported-by: Andy Couldrake Tested-by: Andy Couldrake Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 61682e1d09da..e1323e45f124 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2842,6 +2842,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} }; -- cgit v1.2.2 From c53666813813a0ea3d0391e1911eefc05a5e6b4f Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Tue, 4 May 2010 22:07:58 -0400 Subject: ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T BugLink: https://launchpad.net/bugs/549267 The OR verified that using the olpc-xo-1_5 model quirk allows the headphones to be audible when inserted into the jack. Capture was also verified to work correctly. Reported-by: Richard Gagne Tested-by: Richard Gagne Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e1323e45f124..924c122f16fa 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2842,6 +2842,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} -- cgit v1.2.2 From bfe70783ca8e61f1fc3588cd59c4f1b755e9d3cf Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 28 Apr 2010 10:29:14 +0200 Subject: ALSA: take tu->qlock with irqs disabled We should disable irqs when we take the tu->qlock because it is used in the irq handler. The only place that doesn't is snd_timer_user_ccallback(). Most of the time snd_timer_user_ccallback() is called with interrupts disabled but the the first ti->ccallback() call in snd_timer_notify1() has interrupts enabled. This was caught by lockdep which generates the following message: > ================================= > [ INFO: inconsistent lock state ] > 2.6.34-rc5 #5 > --------------------------------- > inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage. > dolphin/4003 [HC1[1]:SC0[0]:HE0:SE1] takes: > (&(&tu->qlock)->rlock){?.+...}, at: [] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] > {HARDIRQ-ON-W} state was registered at: > [] __lock_acquire+0x654/0x1482 > [] lock_acquire+0x5c/0x73 > [] _raw_spin_lock+0x25/0x34 > [] snd_timer_user_ccallback+0x55/0x95 [snd_timer] > [] snd_timer_notify1+0x53/0xca [snd_timer] Reported-by: Stefan Richter Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/timer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 73943651caed..5040c7b862fe 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1160,6 +1160,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, { struct snd_timer_user *tu = timeri->callback_data; struct snd_timer_tread r1; + unsigned long flags; if (event >= SNDRV_TIMER_EVENT_START && event <= SNDRV_TIMER_EVENT_PAUSE) @@ -1169,9 +1170,9 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, r1.event = event; r1.tstamp = *tstamp; r1.val = resolution; - spin_lock(&tu->qlock); + spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); - spin_unlock(&tu->qlock); + spin_unlock_irqrestore(&tu->qlock, flags); kill_fasync(&tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } -- cgit v1.2.2 From 231f50bc0e9735fd1b3fd376a8d3b6a14aee0694 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Wed, 28 Apr 2010 18:05:06 +0200 Subject: ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582 Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper HP and Mic support. Signed-off-by: Anisse Astier Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 924c122f16fa..e2b698b721db 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2842,6 +2842,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), -- cgit v1.2.2 From 8f0f5ff6777104084b4b2e1ae079541c2a6ed6d9 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 28 Apr 2010 18:00:11 -0400 Subject: ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice) BugLink: https://launchpad.net/bugs/541802 The OR's hardware distorts at PCM 100% because it does not correspond to 0 dB. Fix this in patch_cxt5045() for all Packard Bell models. Reported-by: Valombre Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e2b698b721db..56e52071c769 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1195,9 +1195,10 @@ static int patch_cxt5045(struct hda_codec *codec) switch (codec->subsystem_id >> 16) { case 0x103c: + case 0x1631: case 0x1734: - /* HP & Fujitsu-Siemens laptops have really bad sound over 0dB - * on NID 0x17. Fix max PCM level to 0 dB + /* HP, Packard Bell, & Fujitsu-Siemens laptops have really bad + * sound over 0dB on NID 0x17. Fix max PCM level to 0 dB * (originally it has 0x2b steps with 0dB offset 0x14) */ snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, -- cgit v1.2.2 From 4d26f44657915f082806abfe3624aeded4c121fa Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 7 May 2010 08:47:54 +0800 Subject: ALSA: hda - fix DG45ID SPDIF output MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts part of commit 52dc438606d1e, in order to fix a regression: broken SPDIF output on Intel DG45FC motherboard (IDT 92HD73E1X5 codec). --- DG45FC-IDT-codec-2.6.32 (SPDIF OK) +++ DG45FC-IDT-codec-2.6.33 (SPDIF broken) Node 0x22 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT - Pin Default 0x40f000f0: [N/A] Other at Ext N/A - Conn = Unknown, Color = Unknown - DefAssociation = 0xf, Sequence = 0x0 - Pin-ctls: 0x00: + Pin Default 0x014510a0: [Jack] SPDIF Out at Ext Rear + Conn = Optical, Color = Black + DefAssociation = 0xa, Sequence = 0x0 + Pin-ctls: 0x40: OUT Connection: 3 0x25* 0x20 0x21 Node 0x23 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT - Pin Default 0x01451140: [Jack] SPDIF Out at Ext Rear + Pin Default 0x074510b0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Black - DefAssociation = 0x4, Sequence = 0x0 - Misc = NO_PRESENCE - Pin-ctls: 0x40: OUT + DefAssociation = 0xb, Sequence = 0x0 + Pin-ctls: 0x00: Connection: 3 0x26* 0x20 0x21 Cc: Cc: Alexey Fisher Tested-by: David Härdeman Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7fb7d017a347..12825aa03106 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1544,11 +1544,9 @@ static unsigned int alienware_m17x_pin_configs[13] = { 0x904601b0, }; -static unsigned int intel_dg45id_pin_configs[14] = { +static unsigned int intel_dg45id_pin_configs[13] = { 0x02214230, 0x02A19240, 0x01013214, 0x01014210, - 0x01A19250, 0x01011212, 0x01016211, 0x40f000f0, - 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x014510A0, - 0x074510B0, 0x40f000f0 + 0x01A19250, 0x01011212, 0x01016211 }; static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { -- cgit v1.2.2 From 0217f1499cf880d93c64579b2943e9382d8c2c21 Mon Sep 17 00:00:00 2001 From: Andrej Gelenberg Date: Sun, 9 May 2010 22:10:41 +0200 Subject: ALSA: hda - add support for Lenovo ThinkPad X100e in conexant codec Ideapad quirks working for my ThinkPad X100e (microphone is not tested). Signed-off-by: Andrej Gelenberg Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56e52071c769..d8213e2231a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2846,6 +2846,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} }; -- cgit v1.2.2 From 482c45331519524e4aeaf8a9084a445500822b85 Mon Sep 17 00:00:00 2001 From: Stefan Lippers-Hollmann Date: Mon, 10 May 2010 17:14:34 +0200 Subject: ALSA: Revert "ALSA: hda/realtek: quirk for D945GCLF2 mainboard" This reverts commit 7aee67466536bbf8bb44a95712c848a61c5a0acd. As it doesn't seem to be universally valid for all mainboard revisions of the D945GCLF2 and breaks snd-hda-intel/ snd-hda-codec-realtek on the Intel Corporation "D945GCLF2" (LF94510J.86A.0229.2009.0729.0209) mainboard. 00:1b.0 Audio device [0403]: Intel Corporation N10/ICH 7 Family High Definition Audio Controller [8086:27d8] (rev 01) Signed-off-by: Stefan Lippers-Hollmann Cc: [2.6.33] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7404dba16f83..886d8e46bb37 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17871,7 +17871,6 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x8086, 0xd604, "Intel mobo", ALC662_3ST_2ch_DIG), {} }; -- cgit v1.2.2 From 0ebf9e3692d640917fb792a7494d05e1f5b1058f Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Mon, 10 May 2010 21:50:04 +0200 Subject: ALSA: hda: Fix 0 dB for Lenovo models using Conexant CX20549 (Venice) Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-May/027525.html As reported on the mailing list, we also need to cap to the 0 dB offset for Lenovo models, else the sound will be distorted. Reported-and-Tested-by: Tim Starling Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d8213e2231a6..feabb44c7ca4 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1197,9 +1197,10 @@ static int patch_cxt5045(struct hda_codec *codec) case 0x103c: case 0x1631: case 0x1734: - /* HP, Packard Bell, & Fujitsu-Siemens laptops have really bad - * sound over 0dB on NID 0x17. Fix max PCM level to 0 dB - * (originally it has 0x2b steps with 0dB offset 0x14) + case 0x17aa: + /* HP, Packard Bell, Fujitsu-Siemens & Lenovo laptops have + * really bad sound over 0dB on NID 0x17. Fix max PCM level to + * 0 dB (originally it has 0x2b steps with 0dB offset 0x14) */ snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, (0x14 << AC_AMPCAP_OFFSET_SHIFT) | -- cgit v1.2.2 From 26ebe0a28986f4845b2c5bea43ac5cc0b9f27f0a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 May 2010 08:36:29 +0200 Subject: ALSA: hda - Fix mute-LED GPIO pin for HP dv series Old HP dv series seem to use the GPIO pin 0 for controlling the mute LED although the pin is a large package, where the newer models use GPIO 3 in such a case. For fixing the regression from the previous kernels, set spec->gpio_led statically for these model quirks. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 12825aa03106..eb4ea3df5d84 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4766,6 +4766,9 @@ static void set_hp_led_gpio(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; unsigned int gpio; + if (spec->gpio_led) + return; + gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); gpio &= AC_GPIO_IO_COUNT; if (gpio > 3) @@ -5683,11 +5686,13 @@ again: * detection. */ spec->hp_detect = 1; + spec->gpio_led = 0x01; break; case STAC_HP_HDX: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; + spec->gpio_led = 0x08; break; } -- cgit v1.2.2 From 2a6ce6e5fda4721b35f309acedf4cac61ecbfb04 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 May 2010 10:16:20 +0200 Subject: ALSA: hda - Add hp-dv4 model for IDT 92HD71bx It turned out that HP dv series have inconsistent the mute-LED GPIO mapping among various models. dv4/7 seem to use GPIO 0 while dv 5/6 seem to use GPIO 3. The previous commit 26ebe0a28986f4845b2c5bea43ac5cc0b9f27f0a ALSA: hda - Fix mute-LED GPIO pin for HP dv series breaks dv5/6. This patch adds the new quirk model, hp-dv4, to handle HP dv4/7 separately from HP dv5/6. Tested-by: Kunal Gangakhedkar (for dv6-1110ax) Acked-by: Kunal Gangakhedkar Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index eb4ea3df5d84..a0e06d82da1f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -104,6 +104,7 @@ enum { STAC_DELL_M4_2, STAC_DELL_M4_3, STAC_HP_M4, + STAC_HP_DV4, STAC_HP_DV5, STAC_HP_HDX, STAC_HP_DV4_1222NR, @@ -1691,6 +1692,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_2] = dell_m4_2_pin_configs, [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, + [STAC_HP_DV4] = NULL, [STAC_HP_DV5] = NULL, [STAC_HP_HDX] = NULL, [STAC_HP_DV4_1222NR] = NULL, @@ -1703,6 +1705,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_2] = "dell-m4-2", [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", + [STAC_HP_DV4] = "hp-dv4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", @@ -1721,7 +1724,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, - "HP dv4-7", STAC_HP_DV5), + "HP dv4-7", STAC_HP_DV4), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, @@ -5678,6 +5681,9 @@ again: spec->num_smuxes = 1; spec->num_dmuxes = 1; /* fallthrough */ + case STAC_HP_DV4: + spec->gpio_led = 0x01; + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); @@ -5686,7 +5692,6 @@ again: * detection. */ spec->hp_detect = 1; - spec->gpio_led = 0x01; break; case STAC_HP_HDX: spec->num_dmics = 1; @@ -5749,7 +5754,8 @@ again: } /* enable bass on HP dv7 */ - if (spec->board_config == STAC_HP_DV5) { + if (spec->board_config == STAC_HP_DV4 || + spec->board_config == STAC_HP_DV5) { unsigned int cap; cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP); cap &= AC_GPIO_IO_COUNT; -- cgit v1.2.2 From 6a45f7822544c54a2cf070d84f4e85f2fb32ec02 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 11 May 2010 16:34:39 +0200 Subject: ALSA: virtuoso: fix Xonar D1/DX front panel microphone Commit 65c3ac885ce9852852b895a4a62212f62cb5f2e9 in 2.6.33 accidentally left out the initialization of the AC97 codec FMIC2MIC bit, which broke recording from the front panel microphone. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_cs43xx.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 16c226bfcd2b..7c4986b27f2b 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -56,6 +56,7 @@ #include #include #include "xonar.h" +#include "cm9780.h" #include "cs4398.h" #include "cs4362a.h" @@ -172,6 +173,8 @@ static void xonar_d1_init(struct oxygen *chip) oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); + oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); + xonar_init_cs53x1(chip); xonar_enable_output(chip); -- cgit v1.2.2 From 9fe17b5d47d3d3c85b35623dea8f571a184134c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 May 2010 10:32:42 +0200 Subject: ALSA: pcm - Use pgprot_noncached() for MIPS non-coherent archs MIPS non-coherent archs need the noncached pgprot in mmap of PCM buffers. But, since the coherency needs to be checked dynamically via plat_device_is_coherent(), we need an ugly check dependent on MIPS in ALSA core code. This should be cleaned up in MIPS arch side (e.g. creating dma_mmap_coherent()) in near future. Tested-by: Wu Zhangjin Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 872887624030..20b5982c996b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -36,6 +36,9 @@ #include #include #include +#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) +#include +#endif /* * Compatibility @@ -3184,6 +3187,10 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); +#elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV && + !plat_device_is_coherent(substream->dma_buffer.dev.dev)) + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); #endif /* ARCH_HAS_DMA_MMAP_COHERENT */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; -- cgit v1.2.2 From 8213466596bf10b75887754773ee13c10cf86f5c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 May 2010 16:43:32 +0200 Subject: ALSA: ice1724 - Fix ESI Maya44 capture source control The capture source control of maya44 was wrongly coded with the bit shift instead of the bit mask. Also, the slot for line-in was wrongly assigned (slot 5 instead of 4). Reported-by: Alex Chernyshoff Cc: Signed-off-by: Takashi Iwai --- sound/pci/ice1712/maya44.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/maya44.c b/sound/pci/ice1712/maya44.c index 3e1c20ae2f1c..726fd4b92e19 100644 --- a/sound/pci/ice1712/maya44.c +++ b/sound/pci/ice1712/maya44.c @@ -347,7 +347,7 @@ static int maya_gpio_sw_put(struct snd_kcontrol *kcontrol, /* known working input slots (0-4) */ #define MAYA_LINE_IN 1 /* in-2 */ -#define MAYA_MIC_IN 4 /* in-5 */ +#define MAYA_MIC_IN 3 /* in-4 */ static void wm8776_select_input(struct snd_maya44 *chip, int idx, int line) { @@ -393,8 +393,8 @@ static int maya_rec_src_put(struct snd_kcontrol *kcontrol, int changed; mutex_lock(&chip->mutex); - changed = maya_set_gpio_bits(chip->ice, GPIO_MIC_RELAY, - sel ? GPIO_MIC_RELAY : 0); + changed = maya_set_gpio_bits(chip->ice, 1 << GPIO_MIC_RELAY, + sel ? (1 << GPIO_MIC_RELAY) : 0); wm8776_select_input(chip, 0, sel ? MAYA_MIC_IN : MAYA_LINE_IN); mutex_unlock(&chip->mutex); return changed; -- cgit v1.2.2