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

[alsa-devel] [PATCH v2 2/9] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 14 Oct '11

14 Oct '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 54 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 47 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 1ef058b..87b9ae5 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,10 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -147,6 +151,15 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_EN (1 << 7) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) + #define DA7210_VERSION "0.0.1" /* @@ -165,6 +178,9 @@ */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); 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); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -180,6 +196,13 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -193,6 +216,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + + /* MIC related controls */ + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + + /* AUX related controls */ + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), }; /* Codec private data */ @@ -246,13 +284,9 @@ static int da7210_startup(struct snd_pcm_substream *substream, 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); + /* Enable Mic,AUX1 and AUX2 */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD); } return 0; @@ -521,6 +555,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_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] ASoC: WM8903: Implement DMIC support
by Stephen Warren 14 Oct '11

14 Oct '11

In addition to the currently supported analog capture path, the WM8903 also supports digital mics. The analog and digital capture paths are exclusive; a mux is present to select the capture source. Logically, the mux exists to select the decimator's input, from either the ADC or DMIC block outputs. However, the ADC power domain also includes the DMIC interface. Consequently, this change represents the mux as existing immediately before the ADC, and selecting between the Input PGA and DMIC block outputs. An alternative might be to represent the mux in its correct location, and associate the ADC power enable controls with both the real ADC, and a fake ADC for the DMIC? Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d53f206..f3cab84 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -634,6 +634,13 @@ static const struct soc_enum lsidetone_enum = static const struct soc_enum rsidetone_enum = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static const char *adcinput_text[] = { + "ADC", "DMIC" +}; + +static const struct soc_enum adcinput_enum = + SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text); + static const char *aif_text[] = { "Left", "Right" }; @@ -767,6 +774,9 @@ static const struct snd_kcontrol_new lsidetone_mux = static const struct snd_kcontrol_new rsidetone_mux = SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); +static const struct snd_kcontrol_new adcinput_mux = + SOC_DAPM_ENUM("ADC Input", adcinput_enum); + static const struct snd_kcontrol_new lcapture_mux = SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum); @@ -817,6 +827,7 @@ SND_SOC_DAPM_INPUT("IN2L"), SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), @@ -842,6 +853,9 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux), SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), +SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), +SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), + SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0), @@ -979,6 +993,11 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Left Input PGA", NULL, "Left Input Mode Mux" }, { "Right Input PGA", NULL, "Right Input Mode Mux" }, + { "Left ADC Input", "ADC", "Left Input PGA" }, + { "Left ADC Input", "DMIC", "DMICDAT" }, + { "Right ADC Input", "ADC", "Right Input PGA" }, + { "Right ADC Input", "DMIC", "DMICDAT" }, + { "Left Capture Mux", "Left", "ADCL" }, { "Left Capture Mux", "Right", "ADCR" }, @@ -988,9 +1007,9 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "AIFTXL", NULL, "Left Capture Mux" }, { "AIFTXR", NULL, "Right Capture Mux" }, - { "ADCL", NULL, "Left Input PGA" }, + { "ADCL", NULL, "Left ADC Input" }, { "ADCL", NULL, "CLK_DSP" }, - { "ADCR", NULL, "Right Input PGA" }, + { "ADCR", NULL, "Right ADC Input" }, { "ADCR", NULL, "CLK_DSP" }, { "Left Playback Mux", "Left", "AIFRXL" }, -- 1.7.1

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Re: [alsa-devel] [pulseaudio-discuss] alsa ucm in pulseaudio
by Mark Brown 14 Oct '11

14 Oct '11

On Fri, Oct 14, 2011 at 12:19:53PM +0200, David Henningsson wrote: > On 10/14/2011 11:42 AM, Feng Wei wrote: > >Actually we must map the concepts because ucm is designed for abstract > >the complicated kcontrol. In my mind, if we use ucm in pa, the > >original profile configuration files and mixer configuration files > >will be replaced by ucm configuration files. > Removing the original profile configuration files might be possible > in an embedded scenario, but for all hundreds of different > snd-hda-intel/AC'97/etc configurations out there, writing UCM > configurations to fit all of them is just not scalable. (If so, one > would have to write an auto-UCM tool that autogenerates UCM profiles > based on present hardware.) Yes, writing UCM files for PCs which have reasonably regular and discoverable audio is silly - if UCM is used at all on most desktops it should be with automatically generated configuration.

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Re: [alsa-devel] [pulseaudio-discuss] alsa ucm in pulseaudio
by Mark Brown 14 Oct '11

14 Oct '11

On Fri, Oct 14, 2011 at 10:57:08AM +0200, David Henningsson wrote: > There is also the "Use Case" concept in UCM, Is there always exactly > one verb for every use case? If not, one might wonder if "Use Case" > or "Verb" is what should correspond to the "Profile" concept. Yes, pretty much. > As for ports, this again depends on what is mutually exclusive and > what could be used in parallel, I vaguely remember "Modifiers" as > something that could be used in parallel with existing streams, if > so, they need to be separate sinks/sources. Modifiers and Devices can both be used concurrently. > Hmm, for every sink/source, there can only be one active port at a > time. Also, ports on the same sink/source can be changed on the fly > without restarting the stream. Does this fit with the description of > a UCM "Device"? Pretty much.

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[alsa-devel] [PATCH v3 8/8] ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
by Ashish Chavan 14 Oct '11

14 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 f2e154a..ecb4cb7 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -260,6 +260,28 @@ 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) @@ -407,6 +429,16 @@ 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), }; /* @@ -638,7 +670,6 @@ 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 */ @@ -669,68 +700,45 @@ 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: @@ -740,7 +748,6 @@ 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 v3 7/8] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 14 Oct '11

14 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 | 102 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 102 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ef67a4d..f2e154a 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_STARTUP2 0x04 @@ -67,6 +68,12 @@ #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 @@ -89,6 +96,7 @@ #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) @@ -167,9 +175,11 @@ #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 */ @@ -178,6 +188,23 @@ /* 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" /* @@ -233,6 +260,68 @@ 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", @@ -305,6 +394,19 @@ 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), }; /* -- 1.7.1

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[alsa-devel] [PATCH v3 6/8] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 14 Oct '11

14 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8dad77e..ef67a4d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -47,6 +47,7 @@ #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 @@ -58,6 +59,7 @@ #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 @@ -173,6 +175,9 @@ /* AUX2 bit fields */ #define DA7210_AUX2_EN (1 << 3) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -221,6 +226,13 @@ 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", @@ -277,6 +289,22 @@ 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 Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Capture Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Capture Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital 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), }; /* @@ -674,6 +702,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -681,6 +721,7 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) static struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -782,6 +823,9 @@ 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] busy loop in snd_pcm_plugin_status()
by Thierry Reding 14 Oct '11

14 Oct '11

Hi, I'm seeing a strange issue when using linphone (3.4.3) with alsa-lib (1.0.24.1) on an Atom-based platform. I was able to trace the problem down to the snd_pcm_plugin_status() function. The problem is that the call to the snd_atomic_read_ok() function fails because snd_pcm_plugin_avail_update() has already called snd_atomic_write_{begin,end}() in the meantime. The result is that I'm seeing calls to snd_pcm_plugin_status() where the code jumps back to the _again between 200 and 1800 times before succeeding. This in turn leads to delays in the linphone of 20 to 180 milliseconds, which results in rather absurdly high CPU usage (usually 100%) and correspondingly bad audio playback and side-effects (choppiness, linphone no longer terminates calls properly, ...). I'm using the attached patch as a workaround, but I'm pretty sure it's not a proper fix. The strange thing is that I see that behaviour only on the Atom platform (Intel HDA, Sigmatel 92HD83 codec). Another ARM-based platform does not exhibit the same issue. Can anybody shed some light on the situation? Cheers, Thierry

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[alsa-devel] [PATCH v3 4/8] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 14 Oct '11

14 Oct '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a7ea07d..bc40fd8 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -32,6 +32,10 @@ #define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -154,6 +158,15 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_EN (1 << 7) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) + #define DA7210_VERSION "0.0.1" /* @@ -175,6 +188,9 @@ static const unsigned int hp_out_tlv[] = { }; 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); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -190,6 +206,13 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -203,6 +226,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + + /* MIC related controls */ + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + + /* AUX related controls */ + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), }; /* @@ -702,6 +740,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_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] [RFC 0/2] ASoC: core: Ignoring pmdown_time for playback stream power down
by Peter Ujfalusi 14 Oct '11

14 Oct '11

Hello Mark, Add support in core to ignore the pmdown_time when stopping the playback stream. This will inline the DAPM stream stop event, so it is going to be executed without delay at stream close time (for playback stream). Backround: with the McPDM protocol we are sendning not only the pure audio stream, but OMAP McPDM also transmits additional information (for example offset cancellation). If McPDM is stopped prior to the DAC this information will be not sent to the codec, which can result noise rendered by the twl6040 codec. Is this something that you were suggesting? Regards, Peter --- Peter Ujfalusi (2): ASoC: soc-pcm: Add flag to ignore pmdown_time at pcm_close ASoC: sdp4430: Request core to inline the DAPM sequence include/sound/soc.h | 3 +++ sound/soc/omap/sdp4430.c | 1 + sound/soc/soc-pcm.c | 15 +++++++++++---- 3 files changed, 15 insertions(+), 4 deletions(-) -- 1.7.7

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