Loading sound/soc/codecs/ak4642.c +45 −24 Original line number Diff line number Diff line Loading @@ -80,6 +80,17 @@ #define AK4642_CACHEREGNUM 0x25 /* 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 /* MD_CTL1 */ #define PLL3 (1 << 7) #define PLL2 (1 << 6) Loading @@ -87,6 +98,9 @@ #define PLL0 (1 << 4) #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) #define BCKO_MASK (1 << 3) #define BCKO_64 BCKO_MASK struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ Loading Loading @@ -188,9 +202,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. * * Example code use 0x39, 0x79 value for 0x01 address, * But we need MCKO (0x02) bit now */ ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x0f, 0x09); Loading @@ -200,8 +211,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0a, 0x28); ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* * start stereo input Loading Loading @@ -238,8 +249,8 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, if (is_play) { /* stop headphone output */ ak4642_write(codec, 0x01, 0x3b); ak4642_write(codec, 0x01, 0x0b); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0); ak4642_write(codec, 0x00, 0x40); ak4642_write(codec, 0x0e, 0x11); ak4642_write(codec, 0x0f, 0x08); Loading Loading @@ -284,10 +295,37 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, 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, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); 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, }; struct snd_soc_dai ak4642_dai = { Loading Loading @@ -366,23 +404,6 @@ static int ak4642_init(struct ak4642_priv *ak4642) goto reg_cache_err; } /* * clock setting * * Audio I/F Format: MSB justified (ADC & DAC) * BICK frequency at Master Mode: 64fs * MCKO: Enable * Sampling Frequency: 44.1kHz * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p89. * * please fix-me */ ak4642_write(codec, 0x01, 0x08); ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x04, 0x0a); return ret; reg_cache_err: Loading sound/soc/sh/fsi-ak4642.c +4 −0 Original line number Diff line number Diff line Loading @@ -26,6 +26,10 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) { int ret; ret = snd_soc_dai_set_fmt(&ak4642_dai, SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0); return ret; Loading Loading
sound/soc/codecs/ak4642.c +45 −24 Original line number Diff line number Diff line Loading @@ -80,6 +80,17 @@ #define AK4642_CACHEREGNUM 0x25 /* 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 /* MD_CTL1 */ #define PLL3 (1 << 7) #define PLL2 (1 << 6) Loading @@ -87,6 +98,9 @@ #define PLL0 (1 << 4) #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) #define BCKO_MASK (1 << 3) #define BCKO_64 BCKO_MASK struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ Loading Loading @@ -188,9 +202,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. * * Example code use 0x39, 0x79 value for 0x01 address, * But we need MCKO (0x02) bit now */ ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x0f, 0x09); Loading @@ -200,8 +211,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0a, 0x28); ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* * start stereo input Loading Loading @@ -238,8 +249,8 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, if (is_play) { /* stop headphone output */ ak4642_write(codec, 0x01, 0x3b); ak4642_write(codec, 0x01, 0x0b); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0); ak4642_write(codec, 0x00, 0x40); ak4642_write(codec, 0x0e, 0x11); ak4642_write(codec, 0x0f, 0x08); Loading Loading @@ -284,10 +295,37 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, 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, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); 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, }; struct snd_soc_dai ak4642_dai = { Loading Loading @@ -366,23 +404,6 @@ static int ak4642_init(struct ak4642_priv *ak4642) goto reg_cache_err; } /* * clock setting * * Audio I/F Format: MSB justified (ADC & DAC) * BICK frequency at Master Mode: 64fs * MCKO: Enable * Sampling Frequency: 44.1kHz * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p89. * * please fix-me */ ak4642_write(codec, 0x01, 0x08); ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x04, 0x0a); return ret; reg_cache_err: Loading
sound/soc/sh/fsi-ak4642.c +4 −0 Original line number Diff line number Diff line Loading @@ -26,6 +26,10 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) { int ret; ret = snd_soc_dai_set_fmt(&ak4642_dai, SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0); return ret; Loading