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

[alsa-devel] [PATCH 8/9] ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
by Ashish Chavan 12 Oct '11

12 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. Tested on Samsung SMDK6410 board with DA7210 evaluation board. 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 9cab62a..2c6e47c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -274,6 +274,28 @@ 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) @@ -426,6 +448,16 @@ 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 */ @@ -497,7 +529,6 @@ 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 */ @@ -528,68 +559,45 @@ 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: @@ -599,7 +607,6 @@ 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 5/9] ASoC: da7210: Add support for other DAI word lengths, format and mode
by Ashish Chavan 12 Oct '11

12 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 1b4554a..60f403b 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) @@ -401,9 +406,15 @@ 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; } @@ -508,6 +519,9 @@ 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; } @@ -520,6 +534,12 @@ 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; } @@ -536,7 +556,8 @@ 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 4/9] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 12 Oct '11

12 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 Tested on Samsung SMDK6410 board with DA7210 evaluation board. 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 | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index b48a8af..1b4554a 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) + /* Default gain/vol values */ #define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ #define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ @@ -227,6 +232,13 @@ static const char *da7210_mic_bias_voltage_txt[] = { static const struct soc_enum da7210_mic_bias = SOC_ENUM_SINGLE(DA7210_MIC_L, 4, 4, da7210_mic_bias_voltage_txt); +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", @@ -284,6 +296,24 @@ 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_SINGLE("Mic Left Mute Switch", DA7210_MIC_L, 3, 1, 0), + SOC_SINGLE("Mic Right Mute Switch", 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 */ @@ -629,6 +659,8 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Set default In PGA gain */ snd_soc_write(codec, DA7210_IN_GAIN, DA7210_DFLT_INPGA_VOL | (DA7210_DFLT_INPGA_VOL << 4)); + /* 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 00/09] ASoC: da7210: Add support for all IOs and other features
by Ashish Chavan 12 Oct '11

12 Oct '11

Hi, Existing da7210 driver is very minimalistic in terms of functionality. It supports only one output i.e. head phone. da7210 codec hardware supports many more IOs and features. Here is a patch series that is "feature update" in nature. This adds support for all inputs(MIC L-R, AUX1 L-R, AUX2) and outputs (OUT1 L-R, OUT2) as well as following features, - ADC and DAC equalizers - Mute - Zero cross - Other DAI word lengths, format and mode - ALC - Noise suppression - High pass and voice filters - DAPM This series also contains a small bug fix for head phone volume control. Thanks, -- Ashish (GNU FAN) --- Ashish Chavan (9): ASoC: da7210: Add support for line out and DAC ASoC: da7210: Add support for line input and mic ASoC: da7210: Add support for ADC & DAC equalizers ASoC: da7210: Add support for mute and zero cross controls ASoC: da7210: Add support for other DAI length, format and mode ASoC: da7210: bugfix for head phone volume control ASoC: da7210: Add support for ALC and Noise suppression ASoC: da7210: Add support for High pass and Voice filters ASoC: da7210: Add support for DAPM sound/soc/codecs/da7210.c | 642 +++++++++++++++++++++++++++++++++++--- 1 files changed, 586 insertions(+), 56 deletions(-) -- 1.7.1

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[alsa-devel] [PATCH v2] ASoC: twl6040: Workaround for headset DC offset caused pop noise
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Both Headset DAC need to be turned on/off at the same time before any of the output drivers are enabled (HS Left/Right, Earpiece). Move the HS DAC enable code to sequenced DAPM_SUPPLY, and attach it to the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Changes since v1: - Use DAPM_SUPPLY_S for the HS DAC enable code Regards, Peter include/linux/mfd/twl6040.h | 1 + sound/soc/codecs/twl6040.c | 36 ++++++++++++++++++++++++++++-------- 2 files changed, 29 insertions(+), 8 deletions(-) diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index 87a4778..2463c261 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -122,6 +122,7 @@ /* HSLCTL/R (0x10/0x11) fields */ +#define TWL6040_HSDACENA (1 << 0) #define TWL6040_HSDACMODE (1 << 1) #define TWL6040_HSDRVMODE (1 << 3) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8648498..6369230 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -654,6 +654,26 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + u8 hslctl, hsrctl; + + /* + * Workaround for Headset DC offset caused pop noise: + * Both HS DAC need to be turned on (before the HS driver) and off at + * the same time. + */ + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + if (SND_SOC_DAPM_EVENT_ON(event)) { + hslctl |= TWL6040_HSDACENA; + hsrctl |= TWL6040_HSDACENA; + } else { + hslctl &= ~TWL6040_HSDACENA; + hsrctl &= ~TWL6040_HSDACENA; + } + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + msleep(1); return 0; } @@ -1103,14 +1123,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -1175,6 +1189,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { NULL, 0), SND_SOC_DAPM_SUPPLY("Vibra Right Control", TWL6040_REG_VIBCTLR, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("HSDAC Power", 1, SND_SOC_NOPM, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HF Left PGA", @@ -1204,6 +1221,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"AFMAmpL", NULL, "AFML"}, {"AFMAmpR", NULL, "AFMR"}, + {"HSDAC Left", NULL, "HSDAC Power"}, + {"HSDAC Right", NULL, "HSDAC Power"}, + {"Headset Left Playback", "HS DAC", "HSDAC Left"}, {"Headset Left Playback", "Line-In amp", "AFMAmpL"}, -- 1.7.7

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[alsa-devel] [PATCH] ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 depend on MFD_88PM860X
by Axel Lin 12 Oct '11

12 Oct '11

In saarb_pm860x_init() and evb3_pm860x_init(), we call pm860x_hs_jack_detect() and pm860x_mic_jack_detect() which in turn calls pm860x_set_bits(). Thus make SND_SOC_SAARB and SND_SOC_TAVOREVB3 depend on MFD_88PM860X This patch fixes below build error if CONFIG_MFD_88PM860X is not configured. LD .tmp_vmlinux1 sound/built-in.o: In function `pm860x_write_reg_cache': last.c:(.text+0x29e9c): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_set_bias_level': last.c:(.text+0x29ecc): undefined reference to `pm860x_set_bits' last.c:(.text+0x29f00): undefined reference to `pm860x_reg_write' last.c:(.text+0x29f18): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_read_reg_cache': last.c:(.text+0x29f40): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_probe': last.c:(.text+0x2a034): undefined reference to `pm860x_bulk_read' sound/built-in.o: In function `pm860x_codec_handler': last.c:(.text+0x2a344): undefined reference to `pm860x_reg_read' last.c:(.text+0x2a354): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_mic_jack_detect': last.c:(.text+0x2a450): undefined reference to `pm860x_set_bits' sound/built-in.o: In function `pm860x_hs_jack_detect': last.c:(.text+0x2a4d0): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a4f8): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a510): undefined reference to `pm860x_set_bits' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/Kconfig | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46b..b12b4af 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -121,6 +121,7 @@ config SND_PXA2XX_SOC_PALM27X config SND_SOC_SAARB tristate "SoC Audio support for Marvell Saarb" depends on SND_PXA2XX_SOC && MACH_SAARB + depends on MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help @@ -130,6 +131,7 @@ config SND_SOC_SAARB config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + depends on MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help -- 1.7.4.1

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[alsa-devel] [PATCH v3 0/5] ASoC/MFD/Input: twl6040: Support for audio driven vibra
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Hello, Changes since v2: - Rebased on git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch - Acked-by lines added to patches. Intro mail form v1: The following series adds support for the soc driver to route audio (PCM stream) to the vibra ports of the twl6040. We have ALSA controls for the vibra source selection (PDM, or Input/FF). The route can not be changed from Input/FF to PDM during ongoing effect coming from Input/FF. The Input/FF driver can not execute new effect, while the vibra source is configured to be PDM. TO get things working in legacy mode the omap-mcpdm, and twl6040 soc codec driver's channel count needed to be fixed. To minimize the need for reads through the I2C bus, the MFD driver will cache the vibra control registers. With this series we can route PCM audio to drive the twl6040 vibra output. In order to that we need to play 5 channel audio, and place the vibra sample at the 5th channel. This will change with ABE support (change == simplified). The series has been created on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch If there are no objections, it would be good if this series goes via audio. Regards, Peter --- Peter Ujfalusi (5): Input: twl6040: Simplify vibra regsiter definitions MFD: twl6040: Cache the vibra control registers MFD: twl6040: function to query the vibra status for clients Input: twl6040-vibra: Check the selected path for vibra ASoC: twl6040: Support for vibra output paths drivers/input/misc/twl6040-vibra.c | 19 ++++++--- drivers/mfd/twl6040-core.c | 31 +++++++++++++-- include/linux/mfd/twl6040.h | 24 +++++------ sound/soc/codecs/twl6040.c | 72 ++++++++++++++++++++++++++++++++++++ 4 files changed, 123 insertions(+), 23 deletions(-) -- 1.7.7

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[alsa-devel] [PATCH] ASoC: twl6040: Workaround for headset DC offset caused pop noise
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Both Headset DAC need to be enabled at the same time, before any of the output drivers are enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- include/linux/mfd/twl6040.h | 1 + sound/soc/codecs/twl6040.c | 42 ++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 41 insertions(+), 2 deletions(-) diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index d9e05ea..0bbb1e7 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -122,6 +122,7 @@ /* HSLCTL/R (0x10/0x11) fields */ +#define TWL6040_HSDACENA (1 << 0) #define TWL6040_HSDACMODE (1 << 1) #define TWL6040_HSDRVMODE (1 << 3) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 93f8a59..924eaf3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -91,6 +91,7 @@ struct twl6040_data { int pll_power_mode; int hs_power_mode; int hs_power_mode_locked; + int hs_dac_enabled; unsigned int clk_in; unsigned int sysclk; u16 hs_left_step; @@ -654,6 +655,43 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + u8 hslctl, hsrctl; + + /* + * Workaround for Headset DC offset caused pop noise: + * Both HS DAC need to be turned on (before the HS driver) and off at + * the same time. + */ + if (SND_SOC_DAPM_EVENT_ON(event)) { + if (!priv->hs_dac_enabled++) { + hslctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSLCTL); + hslctl |= TWL6040_HSDACENA; + + hsrctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSRCTL); + hsrctl |= TWL6040_HSDACENA; + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + } + } else { + if (!--priv->hs_dac_enabled) { + hslctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSLCTL); + hslctl &= ~TWL6040_HSDACENA; + + hsrctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSRCTL); + hsrctl &= ~TWL6040_HSDACENA; + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + } + } + msleep(1); return 0; } @@ -1061,11 +1099,11 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0, + SND_SOC_NOPM, 0, 0, twl6040_hs_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0, + SND_SOC_NOPM, 0, 0, twl6040_hs_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", -- 1.7.7

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[alsa-devel] [PATCH] ASoC: da7210: convert to soc-cache
by Axel Lin 12 Oct '11

12 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi Kuninori, This patch converts da7210 to use soc-cache APIs. I don't have this hardware, I'd appreciate if you can test this patch. Thanks, Axel sound/soc/codecs/da7210.c | 123 ++++++++++++++++---------------------------- 1 files changed, 45 insertions(+), 78 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 33b4823..1f30ef9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -162,7 +162,6 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -189,50 +188,16 @@ static const u8 da7210_reg[] = { 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->driver->volatile_register); - - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (reg >= codec->driver->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) +static int da7210_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { - if (DA7210_STATUS == reg) - return i2c_smbus_read_byte_data(codec->control_data, reg); - - return da7210_read_reg_cache(codec, reg); + switch (reg) { + case DA7210_STATUS: + return 1; + default: + return 0; + } } - static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -271,13 +236,13 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, u32 fs, bypass; /* set DAI source to Left and Right ADC */ - da7210_write(codec, DA7210_DAI_SRC_SEL, + snd_soc_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); + snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); - dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -290,7 +255,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? DA7210_DAC_HPF : DA7210_ADC_HPF; @@ -383,8 +348,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) u32 dai_cfg1; u32 dai_cfg3; - dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); - dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -412,8 +377,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) */ dai_cfg1 |= DA7210_DAI_FLEN_64BIT; - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); return 0; } @@ -452,11 +417,15 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* FIXME * @@ -473,8 +442,8 @@ static int da7210_probe(struct snd_soc_codec *codec) /* * make sure that DA7210 use bypass mode before start up */ - da7210_write(codec, DA7210_STARTUP1, 0); - da7210_write(codec, DA7210_PLL_DIV3, + snd_soc_write(codec, DA7210_STARTUP1, 0); + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* @@ -482,36 +451,36 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* 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); + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + snd_soc_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); + snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + snd_soc_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); + snd_soc_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, + snd_soc_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); + snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); /* Enable Left and Right HeadPhone PGA */ - da7210_write(codec, DA7210_HP_CFG, + snd_soc_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); + snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); /* * If 48kHz sound came, it use bypass mode, @@ -522,22 +491,22 @@ static int da7210_probe(struct snd_soc_codec *codec) * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. * see da7210_hw_params */ - da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - da7210_write(codec, DA7210_PLL_DIV2, 0x99); - da7210_write(codec, DA7210_PLL_DIV3, 0x0A | + snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); + snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); - da7210_write(codec, DA7210_A_PLL1, 0x01); - da7210_write(codec, DA7210_A_CP_MODE, 0x7C); - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + snd_soc_write(codec, DA7210_A_PLL1, 0x01); + snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ - da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); snd_soc_add_controls(codec, da7210_snd_controls, ARRAY_SIZE(da7210_snd_controls)); @@ -549,11 +518,10 @@ static int da7210_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, - .read = da7210_read, - .write = da7210_write, .reg_cache_size = ARRAY_SIZE(da7210_reg), .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, + .volatile_register = da7210_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -568,7 +536,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, da7210); - da7210->control_data = i2c; da7210->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

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[alsa-devel] [PATCH v2] ASoC: Remove impossible case from wm8994_hw_params
by Axel Lin 12 Oct '11

12 Oct '11

We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params. Thus no need to check wm8994-aif3 in wm8994_hw_params. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: original patch does not apply after commit 81204c84c "ASoC: Add WM1811 support". sound/soc/codecs/wm8994.c | 10 ---------- 1 files changed, 0 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 16542de..68e769e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2235,7 +2235,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; int aif2_reg; @@ -2278,15 +2277,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "AIF2 using split LRCLK\n"); } break; - case 3: - switch (control->type) { - case WM1811: - case WM8958: - aif1_reg = WM8958_AIF3_CONTROL_1; - break; - default: - return 0; - } default: return -EINVAL; } -- 1.7.4.1

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