/* * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver * * Copyright (C) 2009 Renesas Solutions Corp. * Kuninori Morimoto * * Based on wm8731.c by Richard Purdie * Based on ak4535.c by Richard Purdie * Based on wm8753.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. */ /* ** CAUTION ** * * This is very simple driver. * It can use headphone output / stereo input only * * AK4642 is tested. * AK4643 is tested. */ #include #include #include #include #include #include #include #include #define AK4642_VERSION "0.0.1" #define PW_MGMT1 0x00 #define PW_MGMT2 0x01 #define SG_SL1 0x02 #define SG_SL2 0x03 #define MD_CTL1 0x04 #define MD_CTL2 0x05 #define TIMER 0x06 #define ALC_CTL1 0x07 #define ALC_CTL2 0x08 #define L_IVC 0x09 #define L_DVC 0x0a #define ALC_CTL3 0x0b #define R_IVC 0x0c #define R_DVC 0x0d #define MD_CTL3 0x0e #define MD_CTL4 0x0f #define PW_MGMT3 0x10 #define DF_S 0x11 #define FIL3_0 0x12 #define FIL3_1 0x13 #define FIL3_2 0x14 #define FIL3_3 0x15 #define EQ_0 0x16 #define EQ_1 0x17 #define EQ_2 0x18 #define EQ_3 0x19 #define EQ_4 0x1a #define EQ_5 0x1b #define FIL1_0 0x1c #define FIL1_1 0x1d #define FIL1_2 0x1e #define FIL1_3 0x1f #define PW_MGMT4 0x20 #define MD_CTL5 0x21 #define LO_MS 0x22 #define HP_MS 0x23 #define SPK_MS 0x24 #define AK4642_CACHEREGNUM 0x25 /* PW_MGMT1*/ #define PMVCM (1 << 6) /* VCOM Power Management */ #define PMMIN (1 << 5) /* MIN Input Power Management */ #define PMDAC (1 << 2) /* DAC Power Management */ #define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */ /* PW_MGMT2 */ #define HPMTN (1 << 6) #define PMHPL (1 << 5) #define PMHPR (1 << 4) #define MS (1 << 3) /* master/slave select */ #define MCKO (1 << 1) #define PMPLL (1 << 0) #define PMHP_MASK (PMHPL | PMHPR) #define PMHP PMHP_MASK /* PW_MGMT3 */ #define PMADR (1 << 0) /* MIC L / ADC R Power Management */ /* SG_SL1 */ #define MINS (1 << 6) /* Switch from MIN to Speaker */ #define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */ #define PMMP (1 << 2) /* MPWR pin Power Management */ #define MGAIN0 (1 << 0) /* MIC amp gain*/ /* TIMER */ #define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */ #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) /* ALC_CTL1 */ #define ALC (1 << 5) /* ALC Enable */ #define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */ /* MD_CTL1 */ #define PLL3 (1 << 7) #define PLL2 (1 << 6) #define PLL1 (1 << 5) #define PLL0 (1 << 4) #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) #define BCKO_MASK (1 << 3) #define BCKO_64 BCKO_MASK #define DIF_MASK (3 << 0) #define DSP (0 << 0) #define RIGHT_J (1 << 0) #define LEFT_J (2 << 0) #define I2S (3 << 0) /* MD_CTL2 */ #define FS0 (1 << 0) #define FS1 (1 << 1) #define FS2 (1 << 2) #define FS3 (1 << 5) #define FS_MASK (FS0 | FS1 | FS2 | FS3) /* MD_CTL3 */ #define BST1 (1 << 3) /* MD_CTL4 */ #define DACH (1 << 0) /* * Playback Volume (table 39) * * max : 0x00 : +12.0 dB * ( 0.5 dB step ) * min : 0xFE : -115.0 dB * mute: 0xFF */ static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), }; static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), }; static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { /* Outputs */ SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, &ak4642_hpout_mixer_controls[0], ARRAY_SIZE(ak4642_hpout_mixer_controls)), SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, &ak4642_hpout_mixer_controls[0], ARRAY_SIZE(ak4642_hpout_mixer_controls)), /* DAC */ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), }; static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ {"HPOUTL", NULL, "HPOUTL Mixer"}, {"HPOUTR", NULL, "HPOUTR Mixer"}, {"HPOUTL Mixer", "DACH", "DAC"}, {"HPOUTR Mixer", "DACH", "DAC"}, }; /* codec private data */ struct ak4642_priv { unsigned int sysclk; enum snd_soc_control_type control_type; }; /* * ak4642 register cache */ static const u8 ak4642_reg[AK4642_CACHEREGNUM] = { 0x00, 0x00, 0x01, 0x00, 0x02, 0x00, 0x00, 0x00, 0xe1, 0xe1, 0x18, 0x00, 0xe1, 0x18, 0x11, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, }; static int ak4642_dai_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) { /* * start headphone output * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) * Bass Boost Level : Middle * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. */ snd_soc_write(codec, L_IVC, 0x91); /* volume */ snd_soc_write(codec, R_IVC, 0x91); /* volume */ snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* * start stereo input * * PLL Master Mode * Audio I/F Format:MSB justified (ADC & DAC) * Pre MIC AMP:+20dB * MIC Power On * ALC setting:Refer to Table 35 * ALC bit=“1” * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); } return 0; } static void ak4642_dai_shutdown(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) { /* stop headphone output */ snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); } else { /* stop stereo input */ snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); } } static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; u8 pll; switch (freq) { case 11289600: pll = PLL2; break; case 12288000: pll = PLL2 | PLL0; break; case 12000000: pll = PLL2 | PLL1; break; case 24000000: pll = PLL2 | PLL1 | PLL0; break; case 13500000: pll = PLL3 | PLL2; break; case 27000000: pll = PLL3 | PLL2 | PLL0; break; default: return -EINVAL; } snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); return 0; } static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; u8 data; u8 bcko; data = MCKO | PMPLL; /* use MCKO */ bcko = 0; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: data |= MS; bcko = BCKO_64; break; case SND_SOC_DAIFMT_CBS_CFS: break; default: return -EINVAL; } snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ data = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: data = LEFT_J; break; case SND_SOC_DAIFMT_I2S: data = I2S; break; /* FIXME * Please add RIGHT_J / DSP support here */ default: return -EINVAL; break; } snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); return 0; } static int ak4642_dai_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 rate; switch (params_rate(params)) { case 7350: rate = FS2; break; case 8000: rate = 0; break; case 11025: rate = FS2 | FS0; break; case 12000: rate = FS0; break; case 14700: rate = FS2 | FS1; break; case 16000: rate = FS1; break; case 22050: rate = FS2 | FS1 | FS0; break; case 24000: rate = FS1 | FS0; break; case 29400: rate = FS3 | FS2 | FS1; break; case 32000: rate = FS3 | FS1; break; case 44100: rate = FS3 | FS2 | FS1 | FS0; break; case 48000: rate = FS3 | FS1 | FS0; break; default: return -EINVAL; break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); return 0; } static int ak4642_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_OFF: snd_soc_write(codec, PW_MGMT1, 0x00); break; default: snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); break; } codec->dapm.bias_level = level; return 0; } static struct snd_soc_dai_ops ak4642_dai_ops = { .startup = ak4642_dai_startup, .shutdown = ak4642_dai_shutdown, .set_sysclk = ak4642_dai_set_sysclk, .set_fmt = ak4642_dai_set_fmt, .hw_params = ak4642_dai_hw_params, }; static struct snd_soc_dai_driver ak4642_dai = { .name = "ak4642-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .ops = &ak4642_dai_ops, .symmetric_rates = 1, }; static int ak4642_resume(struct snd_soc_codec *codec) { snd_soc_cache_sync(codec); return 0; } static int ak4642_probe(struct snd_soc_codec *codec) { struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); int ret; dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } snd_soc_add_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } static int ak4642_remove(struct snd_soc_codec *codec) { ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, .remove = ak4642_remove, .resume = ak4642_resume, .set_bias_level = ak4642_set_bias_level, .reg_cache_size = ARRAY_SIZE(ak4642_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, .dapm_widgets = ak4642_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), .dapm_routes = ak4642_intercon, .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct ak4642_priv *ak4642; int ret; ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); if (!ak4642) return -ENOMEM; i2c_set_clientdata(i2c, ak4642); ak4642->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4642, &ak4642_dai, 1); if (ret < 0) kfree(ak4642); return ret; } static __devexit int ak4642_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); kfree(i2c_get_clientdata(client)); return 0; } static const struct i2c_device_id ak4642_i2c_id[] = { { "ak4642", 0 }, { "ak4643", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", .owner = THIS_MODULE, }, .probe = ak4642_i2c_probe, .remove = __devexit_p(ak4642_i2c_remove), .id_table = ak4642_i2c_id, }; #endif static int __init ak4642_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ak4642_i2c_driver); #endif return ret; } module_init(ak4642_modinit); static void __exit ak4642_exit(void) { #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&ak4642_i2c_driver); #endif } module_exit(ak4642_exit); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto "); MODULE_LICENSE("GPL");