[alsa-devel] [PATCH v2 00/12] ASoC: ALC5632: Capture path improvements
This patches is bringing improved support of capture path to ALC5632 codec.
--- Sorry for resending it, I rebased first patchset on the wrong branch.
Andrey Danin (10): ASoC: alc5632: Coding style. Remove two extra empty lines. ASoC: alc5632: Fixed voice DAC volume step. ASoC: alc5632: Add voice DAC playback switch ASoC: alc5632: Rename capture switches to common scheme XXX2REC ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix ASoC: alc5632: Refactored DAPM routes to add voice support ASoC: alc5632: Add DMIC switches and controls ASoC: alc5632: Fix I2S digital interface power for recording ASoC: alc5632: Allow 8kHz stream support. ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix
Paul Fertser (2): ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs
sound/soc/codecs/alc5632.c | 183 +++++++++++++++++++++++++++++++------------- sound/soc/codecs/alc5632.h | 1 + 2 files changed, 130 insertions(+), 54 deletions(-)
From: Andrey Danin danindrey@mail.ru
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index f69fb42..0d574aa 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -437,8 +437,6 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"},
- - /* mono mixer */ {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
On Sat, Feb 11, 2012 at 11:26:27PM +0200, Leon Romanovsky wrote:
From: Andrey Danin danindrey@mail.ru
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu
Applied, thanks.
From: Andrey Danin danindrey@mail.ru
Remove extra zero from volume step in DECLARE_TLV_DB_SCALE macro.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 0d574aa..fce5e76 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -153,7 +153,7 @@ static const unsigned int boost_tlv[] = { /* 0db min scale, 6 db steps, no mute */ static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); /* 0db min scalem 0.75db steps, no mute */ -static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 75, 0);
static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { /* left starts at bit 8, right at bit 0 */
On Sat, Feb 11, 2012 at 11:26:28PM +0200, Leon Romanovsky wrote:
From: Andrey Danin danindrey@mail.ru
Remove extra zero from volume step in DECLARE_TLV_DB_SCALE macro.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu
Applied, thanks.
From: Andrey Danin danindrey@mail.ru
Add voice DAC playback switch.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index fce5e76..6bb6075 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -176,6 +176,8 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), SOC_SINGLE_TLV("Voice DAC Playback Volume", ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE("Voice DAC Playback Switch", + ALC5632_VOICE_DAC_VOL, 12, 1, 1), SOC_SINGLE_TLV("Phone Capture Volume", ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), SOC_DOUBLE_TLV("LineIn Capture Volume",
On Sat, Feb 11, 2012 at 11:26:29PM +0200, Leon Romanovsky wrote:
From: Andrey Danin danindrey@mail.ru
Add voice DAC playback switch.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu
Applied, thanks.
From: Andrey Danin danindrey@mail.ru
Rename capture switches to common scheme XXX2REC.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 56 ++++++++++++++++++++++---------------------- 1 files changed, 28 insertions(+), 28 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 6bb6075..d56218f 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -246,24 +246,24 @@ SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1),
/* Left Record Mixer */ static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { -SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), -SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), -SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), -SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), -SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), -SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), -SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +SOC_DAPM_SINGLE("MIC12REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LIL2REC Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("PH2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPL2REC Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPK2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MONO2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), };
/* Right Record Mixer */ static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { -SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), -SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), -SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), -SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), -SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), -SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), -SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +SOC_DAPM_SINGLE("MIC12REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("MIC22REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LIR2REC Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("PH2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPR2REC Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPK2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MONO2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), };
static const char *alc5632_spk_n_sour_sel[] = { @@ -449,22 +449,22 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"},
/* Left record mixer */ - {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, - {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, - {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, - {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, - {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, - {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, - {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + {"Left Capture Mix", "LIL2REC Capture Switch", "LINEINL"}, + {"Left Capture Mix", "PH2REC_L Capture Switch", "PHONEN"}, + {"Left Capture Mix", "MIC12REC_L Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "MIC22REC_L Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPL2REC Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPK2REC_L Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MONO2REC_L Capture Switch", "Mono Mix"},
/*Right record mixer */ - {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, - {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, - {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, - {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, - {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, - {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, - {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + {"Right Capture Mix", "LIR2REC Capture Switch", "LINEINR"}, + {"Right Capture Mix", "PH2REC_R Capture Switch", "PHONEP"}, + {"Right Capture Mix", "MIC12REC_R Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "MIC22REC_R Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPR2REC Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPK2REC_R Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MONO2REC_R Capture Switch", "Mono Mix"},
/* headphone left mux */ {"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
On Sat, Feb 11, 2012 at 11:26:30PM +0200, Leon Romanovsky wrote:
From: Andrey Danin danindrey@mail.ru
Rename capture switches to common scheme XXX2REC.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu
Applied, thanks.
From: Andrey Danin danindrey@mail.ru
There is no Phone Mix<->Mono Mix route in datasheet.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d56218f..11da806 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -443,7 +443,6 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, - {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"},
On Sat, Feb 11, 2012 at 11:26:31PM +0200, Leon Romanovsky wrote:
@@ -443,7 +443,6 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
- {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"},
I'd expect the corresponding switch in the mixer control array to be removed?
On Sat, Feb 11, 2012 at 11:26:31PM +0200, Leon Romanovsky wrote:
From: Andrey Danin danindrey@mail.ru
There is no Phone Mix<->Mono Mix route in datasheet.
I'll apply anyway as looking at the subsequent patches it appears that the changelog is just misleading here and actually a path does exist, it's just not the one currently routed out. In general quite a few of the patches here are confusingly changelogged like this.
I've applied all the rest too.
From: Andrey Danin danindrey@mail.ru
Refactored DAPM routes to add voice support.
- Added undocumented register - Used AIF in/out - Added missed voice items and routes - Added DMIC input - Romoved unrelevant items
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 85 +++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/alc5632.h | 1 + 2 files changed, 70 insertions(+), 16 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 11da806..b1e9af4 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -266,16 +266,20 @@ SOC_DAPM_SINGLE("SPK2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), SOC_DAPM_SINGLE("MONO2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), };
-static const char *alc5632_spk_n_sour_sel[] = { +static const char * const alc5632_spk_n_sour_sel[] = { "RN/-R", "RP/+R", "LN/-R", "Mute"}; -static const char *alc5632_hpl_out_input_sel[] = { +static const char * const alc5632_hpl_out_input_sel[] = { "Vmid", "HP Left Mix"}; -static const char *alc5632_hpr_out_input_sel[] = { +static const char * const alc5632_hpr_out_input_sel[] = { "Vmid", "HP Right Mix"}; -static const char *alc5632_spkout_input_sel[] = { +static const char * const alc5632_spkout_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; -static const char *alc5632_aux_out_input_sel[] = { +static const char * const alc5632_aux_out_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char * const alc5632_adcr_func_sel[] = { + "Stereo ADC", "Voice ADC"}; +static const char * const alc5632_i2s_out_sel[] = { + "ADC LR", "Voice Stereo Digital"};
/* auxout output mux */ static const struct soc_enum alc5632_aux_out_input_enum = @@ -314,6 +318,17 @@ static const struct soc_enum alc5632_amp_enum = static const struct snd_kcontrol_new alc5632_amp_mux_controls = SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum);
+/* ADC output select */ +static const struct soc_enum alc5632_adcr_func_enum = + SOC_ENUM_SINGLE(ALC5632_DAC_FUNC_SELECT, 5, 2, alc5632_adcr_func_sel); +static const struct snd_kcontrol_new alc5632_adcr_func_controls = + SOC_DAPM_ENUM("ADCR Mux", alc5632_adcr_func_enum); + +/* I2S out select */ +static const struct soc_enum alc5632_i2s_out_enum = + SOC_ENUM_SINGLE(ALC5632_I2S_OUT_CTL, 5, 2, alc5632_i2s_out_sel); +static const struct snd_kcontrol_new alc5632_i2s_out_controls = + SOC_DAPM_ENUM("I2SOut Mux", alc5632_i2s_out_enum);
static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { /* Muxes */ @@ -327,6 +342,10 @@ SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &alc5632_hpr_out_mux_controls), SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, &alc5632_spkoutn_mux_controls), +SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &alc5632_adcr_func_controls), +SND_SOC_DAPM_MUX("I2SOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_i2s_out_controls),
/* output mixers */ SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, @@ -354,20 +373,28 @@ SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, &alc5632_captureR_mixer_controls[0], ARRAY_SIZE(alc5632_captureR_mixer_controls)),
-SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback", - ALC5632_PWR_MANAG_ADD2, 9, 0), -SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback", - ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_AIF_IN("AIFRXL", "Left HiFi Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFRXR", "Right HiFi Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("VAIFRX", "Voice Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("VAIFTX", "Voice Capture", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_DAC("Voice DAC", NULL, ALC5632_PWR_MANAG_ADD2, 10, 0), +SND_SOC_DAPM_DAC("Left DAC", NULL, ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", NULL, ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_ADC("Left ADC", NULL, ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", NULL, ALC5632_PWR_MANAG_ADD2, 6, 0), + SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0), SND_SOC_DAPM_MIXER("DAC Right Channel", ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0), SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", - ALC5632_PWR_MANAG_ADD2, 7, 0), -SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", - ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_MIXER("Voice Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("ADCLR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), @@ -395,10 +422,12 @@ SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), SND_SOC_DAPM_OUTPUT("SPKOUT"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), + SND_SOC_DAPM_INPUT("LINEINL"), SND_SOC_DAPM_INPUT("LINEINR"), SND_SOC_DAPM_INPUT("PHONEP"), SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("Vmid"), @@ -406,6 +435,10 @@ SND_SOC_DAPM_VMID("Vmid"),
static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { + /* Playback streams */ + {"Left DAC", NULL, "AIFRXL"}, + {"Right DAC", NULL, "AIFRXR"}, + /* virtual mixer - mixes left & right channels */ {"I2S Mix", NULL, "Left DAC"}, {"I2S Mix", NULL, "Right DAC"}, @@ -428,7 +461,7 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, - + {"HP Mix", "VOICE2HP Playback Switch", "Voice Mix"}, {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"},
@@ -438,6 +471,7 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, + {"Speaker Mix", "VOICE2SPK Playback Switch", "Voice Mix"},
/* mono mixer */ {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, @@ -446,6 +480,7 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Voice Mix"},
/* Left record mixer */ {"Left Capture Mix", "LIL2REC Capture Switch", "LINEINL"}, @@ -503,10 +538,28 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
/* left ADC */ {"Left ADC", NULL, "Left Capture Mix"}, + {"Left ADC", NULL, "DMICDAT"}, + {"ADCLR", NULL, "Left ADC"},
/* right ADC */ - {"Right ADC", NULL, "Right Capture Mix"}, - + {"Right ADC", NULL, "Right Capture Mix"}, + {"Right ADC", NULL, "DMICDAT"}, + {"ADCR Mux", "Stereo ADC", "Right ADC"}, + {"ADCR Mux", "Voice ADC", "Right ADC"}, + {"ADCLR", NULL, "ADCR Mux"}, + {"VAIFTX", NULL, "ADCR Mux"}, + + /* Digital I2S out */ + {"I2SOut Mux", "ADC LR", "ADCLR"}, + {"I2SOut Mux", "Voice Stereo Digital", "VAIFRX"}, + {"AIFTXL", NULL, "I2SOut Mux"}, + {"AIFTXR", NULL, "I2SOut Mux"}, + + /* Voice Mix */ + {"Voice DAC", NULL, "VAIFRX"}, + {"Voice Mix", NULL, "Voice DAC"}, + + /* Speaker Output */ {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h index 357651e..1b5bda5 100644 --- a/sound/soc/codecs/alc5632.h +++ b/sound/soc/codecs/alc5632.h @@ -51,6 +51,7 @@ #define ALC5632_ADC_REC_MONOMIX (1 << 0)
#define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */ +#define ALC5632_I2S_OUT_CTL 0x1A /* undocumented reg. found in path scheme */ /* ALC5632_OUTPUT_MIXER_CTRL : */ /* same remark as for reg 2 line vs speaker */ #define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */
From: Andrey Danin danindrey@mail.ru
Add DMIC switches and controls to ALC5632 codec.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 26 +++++++++++++++++++++++--- 1 files changed, 23 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b1e9af4..b6b95c8 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -196,8 +196,12 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), - SOC_SINGLE_TLV("Digital Boost Volume", + SOC_SINGLE_TLV("DMIC Boost Volume", ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), + SOC_SINGLE("DMIC En Capture Switch", + ALC5632_DIGI_BOOST_CTRL, 15, 1, 0), + SOC_SINGLE("DMIC PreFilter Capture Switch", + ALC5632_DIGI_BOOST_CTRL, 12, 1, 0), };
/* @@ -266,6 +270,14 @@ SOC_DAPM_SINGLE("SPK2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), SOC_DAPM_SINGLE("MONO2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), };
+/* Dmic Mixer */ +static const struct snd_kcontrol_new alc5632_dmicl_mixer_controls[] = { +SOC_DAPM_SINGLE("DMICL2ADC Capture Switch", ALC5632_DIGI_BOOST_CTRL, 7, 1, 1), +}; +static const struct snd_kcontrol_new alc5632_dmicr_mixer_controls[] = { +SOC_DAPM_SINGLE("DMICR2ADC Capture Switch", ALC5632_DIGI_BOOST_CTRL, 6, 1, 1), +}; + static const char * const alc5632_spk_n_sour_sel[] = { "RN/-R", "RP/+R", "LN/-R", "Mute"}; static const char * const alc5632_hpl_out_input_sel[] = { @@ -364,6 +376,12 @@ SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, &alc5632_speaker_mixer_controls[0], ARRAY_SIZE(alc5632_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("DMICL Mix", SND_SOC_NOPM, 0, 0, + &alc5632_dmicl_mixer_controls[0], + ARRAY_SIZE(alc5632_dmicl_mixer_controls)), +SND_SOC_DAPM_MIXER("DMICR Mix", SND_SOC_NOPM, 0, 0, + &alc5632_dmicr_mixer_controls[0], + ARRAY_SIZE(alc5632_dmicr_mixer_controls)),
/* input mixers */ SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, @@ -538,12 +556,14 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
/* left ADC */ {"Left ADC", NULL, "Left Capture Mix"}, - {"Left ADC", NULL, "DMICDAT"}, + {"DMICL Mix", "DMICL2ADC Capture Switch", "DMICDAT"}, + {"Left ADC", NULL, "DMICL Mix"}, {"ADCLR", NULL, "Left ADC"},
/* right ADC */ {"Right ADC", NULL, "Right Capture Mix"}, - {"Right ADC", NULL, "DMICDAT"}, + {"DMICR Mix", "DMICR2ADC Capture Switch", "DMICDAT"}, + {"Right ADC", NULL, "DMICR Mix"}, {"ADCR Mux", "Stereo ADC", "Right ADC"}, {"ADCR Mux", "Voice ADC", "Right ADC"}, {"ADCLR", NULL, "ADCR Mux"},
From: Andrey Danin danindrey@mail.ru
Fix I2S digital interface power for recording.
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b6b95c8..f965560 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -356,7 +356,7 @@ SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, &alc5632_spkoutn_mux_controls), SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, &alc5632_adcr_func_controls), -SND_SOC_DAPM_MUX("I2SOut Mux", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("I2SOut Mux", ALC5632_PWR_MANAG_ADD1, 11, 0, &alc5632_i2s_out_controls),
/* output mixers */
From: Marc Dietrich marvin24@gmx.de
This patch should fix output through speakers using HP mixer.
Signed-off-by: Andrey Danin danindrey@mail.ru --- sound/soc/codecs/alc5632.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index f965560..69c652c 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -482,6 +482,9 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"HP Mix", "VOICE2HP Playback Switch", "Voice Mix"}, {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, + {"HPOut Mix", NULL, "HP Mix"}, + {"HPOut Mix", NULL, "HPR Mix"}, + {"HPOut Mix", NULL, "HPL Mix"},
/* speaker mixer */ {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
From: Paul Fertser fercerpav@gmail.com
This brings the TLVs in sync with the documentation and allows to properly manipulate mic boost controls with alsamixer.
Signed-off-by: Paul Fertser fercerpav@gmail.com Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 11 +++++------ 1 files changed, 5 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 69c652c..b77f4e7 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -145,10 +145,9 @@ static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); /* -16.5db min scale, 1.5db steps, no mute */ static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); static const unsigned int boost_tlv[] = { - TLV_DB_RANGE_HEAD(3), - 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), - 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), }; /* 0db min scale, 6 db steps, no mute */ static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); @@ -193,9 +192,9 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { SOC_DOUBLE_TLV("Rec Capture Volume", ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), SOC_SINGLE_TLV("Mic 1 Boost Volume", - ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + ALC5632_MIC_CTRL, 10, 3, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", - ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + ALC5632_MIC_CTRL, 8, 3, 0, boost_tlv), SOC_SINGLE_TLV("DMIC Boost Volume", ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), SOC_SINGLE("DMIC En Capture Switch",
From: Paul Fertser fercerpav@gmail.com
According to the mixer path diagram input sources' attenuators logically belong to the playback path and DMIC boost only affects capture.
Signed-off-by: Paul Fertser fercerpav@gmail.com Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b77f4e7..b347611 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -177,17 +177,17 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), SOC_SINGLE("Voice DAC Playback Switch", ALC5632_VOICE_DAC_VOL, 12, 1, 1), - SOC_SINGLE_TLV("Phone Capture Volume", + SOC_SINGLE_TLV("Phone Playback Volume", ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), - SOC_DOUBLE_TLV("LineIn Capture Volume", + SOC_DOUBLE_TLV("LineIn Playback Volume", ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("Master Playback Volume", ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), SOC_DOUBLE("Master Playback Switch", ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), - SOC_SINGLE_TLV("Mic1 Capture Volume", + SOC_SINGLE_TLV("Mic1 Playback Volume", ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), - SOC_SINGLE_TLV("Mic2 Capture Volume", + SOC_SINGLE_TLV("Mic2 Playback Volume", ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("Rec Capture Volume", ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), @@ -195,7 +195,7 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_MIC_CTRL, 10, 3, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", ALC5632_MIC_CTRL, 8, 3, 0, boost_tlv), - SOC_SINGLE_TLV("DMIC Boost Volume", + SOC_SINGLE_TLV("DMIC Boost Capture Volume", ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), SOC_SINGLE("DMIC En Capture Switch", ALC5632_DIGI_BOOST_CTRL, 15, 1, 0),
From: Andrey Danin danindrey@mail.ru
Signed-off-by: Andrey Danin danindrey@mail.ru Signed-off-by: Leon Romanovsky leon@leon.nu --- sound/soc/codecs/alc5632.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b347611..e2111e0 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -788,6 +788,7 @@ static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
switch (freq) { + case 4096000: case 8192000: case 11289600: case 12288000:
participants (2)
-
Leon Romanovsky -
Mark Brown