[PATCH v2 00/12] ASoC: ALC5632: Capture path improvements

[PATCH v2 00/12] ASoC: ALC5632: Capture path improvements

[Leon Romanovsky]

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(-)

-- 
1.7.3.4

[PATCH v2 01/12] ASoC: alc5632: Coding style. Remove two extra empty lines.

[Leon Romanovsky]

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"},
-- 
1.7.3.4

[PATCH v2 02/12] ASoC: alc5632: Fixed voice DAC volume step.

[Leon Romanovsky]

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 */
-- 
1.7.3.4

[PATCH v2 03/12] ASoC: alc5632: Add voice DAC playback switch

[Leon Romanovsky]

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",
-- 
1.7.3.4

[PATCH v2 04/12] ASoC: alc5632: Rename capture switches to common scheme XXX2REC

[Leon Romanovsky]

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"},
-- 
1.7.3.4

[PATCH v2 05/12] ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix

[Leon Romanovsky]

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"},
-- 
1.7.3.4

[PATCH v2 06/12] ASoC: alc5632: Refactored DAPM routes to add voice support

[Leon Romanovsky]

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 */
-- 
1.7.3.4

[PATCH v2 07/12] ASoC: alc5632: Add DMIC switches and controls

[Leon Romanovsky]

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"},
-- 
1.7.3.4

[PATCH v2 08/12] ASoC: alc5632: Fix I2S digital interface power for recording

[Leon Romanovsky]

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 */
-- 
1.7.3.4

[PATCH v2 09/12] ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix

[Leon Romanovsky]

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"},
-- 
1.7.3.4

[PATCH v2 10/12] ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones

[Leon Romanovsky]

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",
-- 
1.7.3.4

[PATCH v2 11/12] ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs

[Leon Romanovsky]

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),
-- 
1.7.3.4

[PATCH v2 12/12] ASoC: alc5632: Allow 8kHz stream support.

[Leon Romanovsky]

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:
-- 
1.7.3.4

Re: [PATCH v2 01/12] ASoC: alc5632: Coding style. Remove two extra empty lines.

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 232 bytes --]

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.

[-- Attachment #1.2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[-- Attachment #2: Type: text/plain, Size: 0 bytes --]

Re: [PATCH v2 02/12] ASoC: alc5632: Fixed voice DAC volume step.

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 305 bytes --]

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.

[-- Attachment #1.2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[-- Attachment #2: Type: text/plain, Size: 0 bytes --]

Re: [PATCH v2 03/12] ASoC: alc5632: Add voice DAC playback switch

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 270 bytes --]

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.

[-- Attachment #1.2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[-- Attachment #2: Type: text/plain, Size: 0 bytes --]

Re: [PATCH v2 04/12] ASoC: alc5632: Rename capture switches to common scheme XXX2REC

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 289 bytes --]

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.

[-- Attachment #1.2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[-- Attachment #2: Type: text/plain, Size: 0 bytes --]

Re: [PATCH v2 05/12] ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 672 bytes --]

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?

[-- Attachment #1.2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[-- Attachment #2: Type: text/plain, Size: 0 bytes --]

Re: [PATCH v2 05/12] ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 477 bytes --]

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.

[-- Attachment #1.2: Digital signature --]
[-- Type: application/pgp-signature, Size: 836 bytes --]

[-- Attachment #2: Type: text/plain, Size: 0 bytes --]