October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC: mixing PCI card and SoC codec
by Matteo Fortini 14 Oct '11

14 Oct '11

I have a Geode board with integrated PCI CS5535 sound card, and I'll be using an external ADAU1761 codec as an aux_dev (mixer and probably DSP). Until now I defined the ADAU as a dummy sound card, just to be able to use its mixer. I guess there's a cleaner solution to that, which I suspect involves bridging SoC and PCI codecs (I think avoiding to call snd_device_new in the PCI codec and moving it to a SOC card would be a good start...). Should I try that, or stick to the dummy soc implementation? Thank you.

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[alsa-devel] [PATCH 1/3] ASoC: alc5623: Convert codec->hw_read to regmap_read
by Axel Lin 14 Oct '11

14 Oct '11

codec->hw_read is broken now, let's covert to regmap_read. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 557b3af..8c156cb 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -51,10 +51,13 @@ static void alc5623_fill_cache(struct snd_soc_codec *codec) { int i, step = codec->driver->reg_cache_step; u16 *cache = codec->reg_cache; + unsigned int val; /* not really efficient ... */ - for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = codec->hw_read(codec, i); + for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) { + regmap_read(codec->control_data, i, &val); + cache[i] = val; + } } static inline int alc5623_reset(struct snd_soc_codec *codec) -- 1.7.4.1

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Re: [alsa-devel] linux-next: Tree for Oct 12 (sound/soc/codecs/wm1250-ev1)
by Randy Dunlap 13 Oct '11

13 Oct '11

On 10/11/2011 11:54 PM, Stephen Rothwell wrote: > Hi all, > > The linux-next tree is now available from > git://github.com/sfrothwell/linux-next.git as a temporary measure while > the kernel.org servers are unavailable. > > It may also turn up on git.kernel.org (depending on the mirroring). The > patch is still absent, however. When no GPIO kconfig options are enabled: (not GPIOLIB, not GENERIC_GPIO): sound/soc/codecs/wm1250-ev1.c:32:14: error: array type has incomplete element type sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:126:18: error: negative width in bit-field '<anonymous>' sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:134:42: error: negative width in bit-field '<anonymous>' sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:155:34: error: negative width in bit-field '<anonymous>' This was on an i386 build, but that shouldn't matter. -- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Use snd_soc_update_bits for read-modify-write
by Axel Lin 13 Oct '11

13 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch also fixes the mix usage of AIC32X4_MDACEN/AIC32X4_NDACEN/AIC32X4_NADCEN/AIC32X4_MADCEN in current code. (They are all defined as (0x01 << 7), so it is not a bug, but not good for readability.) Axel sound/soc/codecs/tlv320aic32x4.c | 61 +++++++++++++++----------------------- 1 files changed, 24 insertions(+), 37 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index a68982e..b21c610 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -528,40 +528,33 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - u8 value; switch (level) { case SND_SOC_BIAS_ON: if (aic32x4->master) { /* Switch on PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value | AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); /* Switch on NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value | AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); /* Switch on MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); /* Switch on NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); /* Switch on MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); /* Switch on BCLK_N Divider */ - value = snd_soc_read(codec, AIC32X4_BCLKN); - snd_soc_write(codec, AIC32X4_BCLKN, - value | AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); } break; case SND_SOC_BIAS_PREPARE: @@ -569,34 +562,28 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (aic32x4->master) { /* Switch off PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value & ~AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); /* Switch off NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); /* Switch off MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value & ~AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); /* Switch off NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); /* Switch off MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value & ~AIC32X4_MDACEN); - value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); /* Switch off BCLK_N Divider */ - snd_soc_write(codec, AIC32X4_BCLKN, - value & ~AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); } break; case SND_SOC_BIAS_OFF: -- 1.7.4.1

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[alsa-devel] [PATCH v2 5/9] ASoC: da7210: Add support for other DAI word lengths, format and mode
by Ashish Chavan 13 Oct '11

13 Oct '11

This patchs adds support for following, (1) DAI 20 and 32 bit word sizes (2) DAI left and right justified formats (3) DAI slave mode Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 23 ++++++++++++++++++++++- 1 files changed, 22 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cb00935..8cbeb6c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -120,12 +120,17 @@ /* DAI_CFG1 bit fields */ #define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S20_3LE (1 << 0) #define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_WORD_S32_LE (3 << 0) #define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_SLAVE (0 << 7) #define DA7210_DAI_MODE_MASTER (1 << 7) /* DAI_CFG3 bit fields */ #define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_FORMAT_LEFT_J (1 << 0) +#define DA7210_DAI_FORMAT_RIGHT_J (2 << 0) #define DA7210_DAI_OE (1 << 3) #define DA7210_DAI_EN (1 << 7) @@ -383,9 +388,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; + break; case SNDRV_PCM_FORMAT_S24_LE: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_cfg1 |= DA7210_DAI_WORD_S32_LE; + break; default: return -EINVAL; } @@ -490,6 +501,9 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_CBM_CFM: dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_cfg1 |= DA7210_DAI_MODE_SLAVE; + break; default: return -EINVAL; } @@ -502,6 +516,12 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_I2S: dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; break; + case SND_SOC_DAIFMT_LEFT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J; + break; default: return -EINVAL; } @@ -518,7 +538,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } -#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { -- 1.7.1

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[alsa-devel] [PATCH v2 4/9] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX, ADC and DAC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7e36ed6..cb00935 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -45,6 +45,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -56,6 +57,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -166,6 +168,9 @@ /* AUX2 bit fields */ #define DA7210_AUX2_EN (1 << 3) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -211,6 +216,13 @@ static const unsigned int aux1_vol_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) }; +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -267,6 +279,23 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Mute Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Mute Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Mute Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Mute Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("DAC Mute Switch", DA7210_DAC_HPF, 2, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("DAC Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -597,6 +626,9 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] [PATCH 0/4] ASoC: twl6040: Remove PLL restrictions
by Peter Ujfalusi 13 Oct '11

13 Oct '11

Hello, twl6040 has no restrictions on the selected PLL, and sysclk. All input, and output paths can be used with any PLL configuration. Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: Remove Capture restriction for 17.64MHz sysclk ASoC: twl6040: Remove PLL usage restrictions ASoC: twl6040: Change event ordering for Earphone driver ASoC: twl6040: Rename the Earphone Driver event handler sound/soc/codecs/twl6040.c | 53 ++++++++++---------------------------------- 1 files changed, 12 insertions(+), 41 deletions(-) -- 1.7.7

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[alsa-devel] [PATCH v2 8/9] ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch add controls for setting cut-off for high pass and voice filters of ADC and DAC. There are also switches to enable/disable these filters. Also removed hard coded, fixed values of these parameters used by previous version of driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 57 +++++++++++++++++++++++++------------------- 1 files changed, 32 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8df7d84..fcbe557 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -258,6 +258,28 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -406,6 +428,16 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), }; /* Codec private data */ @@ -477,7 +509,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ @@ -508,68 +539,45 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; - switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -579,7 +587,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); -- 1.7.1

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[alsa-devel] [PATCH v2 7/9] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Reading from register instead of cache - Made debug message more specific about error - Changed "Gain" to "Volume" in names of ALC controls --- sound/soc/codecs/da7210.c | 113 +++++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 109 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8cbeb6c..8df7d84 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 @@ -65,6 +66,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -87,6 +94,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -165,9 +173,11 @@ #define DA7210_OUT2_EN (1 << 7) /* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) #define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) #define DA7210_AUX1_R_EN (1 << 7) /* AUX2 bit fields */ @@ -176,6 +186,23 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -187,12 +214,15 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); @@ -228,6 +258,68 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EPERM; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != 0xc3) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EPERM; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -301,6 +393,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

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[alsa-devel] [PATCH v2 6/9] ASoC: da7210: bugfix for head phone volume control
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch takes care of reserved bits of headphone volume register by using correct volume range. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8cbeb6c..21ef6e0 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -187,12 +187,15 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); -- 1.7.1

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