Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull updates of sound stuff from Takashi Iwai:
 "Here is the first big update chunk of sound stuff for 3.4-rc1.

  In the common sound infrastructure, there are a few changes for
  dynamic PCM support (used in ASoC) and a few clean-ups.  Majority of
  changes are found, as usual, in HD-audio and ASoC.

  Some highlights of HD-audio changes:

   - All the long-standing static quirk codes for Realtek codec were
     finally removed by fixing and extending the Realtek auto-parser.

   - The mute-LED control is standardized over all HD-audio codec
     drivers using the extended vmaster hook.

   - The vmaster slave mixer elements are initialized to 0dB as default
     so that the user won't be annoyed by the silent output after
     updates, e.g.  due to the additions of new elements.

   - Other many fix-ups for the misc HD-audio devices.

  In the ASoC side, this is a very active release, including a quite a
  few framework enhancements.  Some highlights:

   - Support for widgets not associated with a CODEC, an important part
     of the dynamic PCM framework.

   - A library factoring out the common code shared by dmaengine based
     DMA drivers contributed by Lars-Peter Clausen.  This will save a
     lot of code and make it much easier to deploy enhancements to
     dmaengine.

   - Support for binary controls, used for providing runtime
     configuration of algorithm coefficients.

   - A new DAPM widget type for regulator supplies allowing drivers for
     devices that can power down unused supplies while active to do
     without any per-driver code.

   - DAPM widgets for DAIs, initially giving a speed boost for playback
     startup and shutdown and also the basis for CODEC<->CODEC DAI link
     support.

   - Support for specifying the number of significant bits on audio
     interfaces, useful for allowing applications to know how much
     effort to put into generating data for a larger sample format.

   - Conversion of the FSI driver used on some SH processors to
     DMAEngine.

   - Conversion of EP93xx drivers to DMAEngine.

   - New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
     WM2200.

   - Move audmux driver from arc/arm to sound/soc

   - McBSP move from arch/ to sound/ and updates

  Also, a few small updates and fixes for other drivers like au88x0,
  ymfpci, USB 6fire, USB usx2yaudio are included."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits)
  ASoC: wm8994: Provide VMID mode control and fix default sequence
  ASoC: wm8994: Add missing break in resume
  ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
  ASoC: pxa-ssp: atomically set stream active masks
  ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
  ASoC: Samsung: Added to support mono recording
  ALSA: hda - Fix build with CONFIG_PM=n
  ALSA: au88x0 - Avoid possible Oops at unbinding
  ALSA: usb-audio - Fix build error by consitification of rate list
  ASoC: core: Fix obscure leak of runtime array
  ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link()
  ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
  ASoC: wm8996: Add 44.1kHz support
  ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE
  ASoC: mx27vis-aic32x4: Convert it to platform driver
  ALSA: hda - fix printing of high HDMI sample rates
  ALSA: ymfpci - Fix legacy registers on S3/S4 resume
  ALSA: control - Fixe a trailing white space error
  ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook()
  ALSA: hda - Add "Mute-LED Mode" enum control
  ...

1 files changed, 136 insertions, 61 deletions

diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 390e437d7c5..e2111e0ccad 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -145,15 +145,14 @@ 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);
 /* 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 */
@@ -176,26 +175,32 @@ 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_TLV("Phone Capture Volume",
+	SOC_SINGLE("Voice DAC Playback Switch",
+			ALC5632_VOICE_DAC_VOL, 12, 1, 1),
+	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),
 	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),
-	SOC_SINGLE_TLV("Digital Boost Volume",
+			ALC5632_MIC_CTRL, 8, 3, 0, boost_tlv),
+	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),
+	SOC_SINGLE("DMIC PreFilter Capture Switch",
+			ALC5632_DIGI_BOOST_CTRL, 12, 1, 0),
 };
 
 /*
@@ -244,36 +249,48 @@ 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[] = {
+/* 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 *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 =
@@ -312,6 +329,17 @@ 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 */
@@ -325,6 +353,10 @@ 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", ALC5632_PWR_MANAG_ADD1, 11, 0,
+	&alc5632_i2s_out_controls),
 
 /* output mixers */
 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
@@ -343,6 +375,12 @@ 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,
@@ -352,20 +390,28 @@ 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),
@@ -393,10 +439,12 @@ 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"),
@@ -404,6 +452,10 @@ 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"},
@@ -426,9 +478,12 @@ 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"},
+	{"HPOut Mix", NULL, "HP Mix"},
+	{"HPOut Mix", NULL, "HPR Mix"},
+	{"HPOut Mix", NULL, "HPL Mix"},
 
 	/* speaker mixer */
 	{"Speaker Mix", "LI2SPK Playback Switch",	"Line Mix"},
@@ -436,35 +491,34 @@ 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"},
 	{"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"},
+	{"Mono Mix", "VOICE2MONO Playback Switch",	"Voice Mix"},
 
 	/* 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"},
@@ -504,10 +558,30 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
 
 	/* left ADC */
 	{"Left ADC", NULL,				"Left Capture Mix"},
+	{"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, "Right Capture Mix"},
+	{"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"},
+	{"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"},
@@ -714,6 +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:
@@ -994,7 +1069,7 @@ static int alc5632_probe(struct snd_soc_codec *codec)
 
 	switch (alc5632->id) {
 	case 0x5c:
-		snd_soc_add_controls(codec, alc5632_vol_snd_controls,
+		snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls,
 			ARRAY_SIZE(alc5632_vol_snd_controls));
 		break;
 	default:
@@ -1109,7 +1184,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
 	return ret;
 }
 
-static int alc5632_i2c_remove(struct i2c_client *client)
+static __devexit int alc5632_i2c_remove(struct i2c_client *client)
 {
 	struct alc5632_priv *alc5632 = i2c_get_clientdata(client);
 	snd_soc_unregister_codec(&client->dev);