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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc/codecs/tlv320aic23.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc/codecs/wm9713.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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 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/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 +- 5 files changed, 9 insertions(+), 9 deletions(-) (limited to 'sound/soc') 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 */); -- 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec.h | 2 +- sound/soc/soc-core.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc/codecs/wm8990.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') 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/soc/fsl/efika-audio-fabric.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc/au1x/dbdma2.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') 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/soc/au1x/dbdma2.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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 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/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 + 82 files changed, 81 insertions(+), 4 deletions(-) (limited to 'sound/soc') 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 -- 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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/soc') 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/soc') 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/soc') 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 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/soc') 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 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/soc') 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 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/soc') 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