Merge remote branch 'tiwai/topic/asoc' into for-2.6.38

include/sound/alc5623.h

+#ifndef _INCLUDE_SOUND_ALC5623_H
+#define _INCLUDE_SOUND_ALC5623_H
+struct alc5623_platform_data {
+	/* configure :                              */
+	/* Lineout/Speaker Amps Vmid ratio control  */
+	/* enable/disable adc/dac high pass filters */
+	unsigned int add_ctrl;
+	/* configure :                              */
+	/* output to enable when jack is low        */
+	/* output to enable when jack is high       */
+	/* jack detect (gpio/nc/jack detect [12]    */
+	unsigned int jack_det_ctrl;
+};
+#endif
+

sound/soc/codecs/Kconfig

@@ -22,11 +22,13 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
+	select SND_SOC_ALC5623 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS42L51 if I2C
 	select SND_SOC_CS4270 if I2C
+	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
-	select SND_SOC_JZ4740 if SOC_JZ4740
+	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
@@ -129,6 +131,9 @@ config SND_SOC_AK4642
 config SND_SOC_AK4671
 	tristate
 
+config SND_SOC_ALC5623
+       tristate
+
 config SND_SOC_CQ0093VC
 	tristate
 
@@ -317,3 +322,4 @@ config SND_SOC_WM2000
 
 config SND_SOC_WM9090
 	tristate
+

sound/soc/codecs/Makefile

@@ -17,6 +17,7 @@ snd-soc-da7210-objs := da7210.o
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-alc5623-objs := alc5623.o
 snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-stac9766-objs := stac9766.o
@@ -92,6 +93,7 @@ obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o

sound/soc/codecs/alc5623.c

+/*
+ * alc5623.c  --  alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
+ *
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ *
+ * Based on WM8753.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/alc5623.h>
+
+#include "alc5623.h"
+
+static int caps_charge = 2000;
+module_param(caps_charge, int, 0);
+MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
+
+/* codec private data */
+struct alc5623_priv {
+	enum snd_soc_control_type control_type;
+	void *control_data;
+	struct mutex mutex;
+	u8 id;
+	unsigned int sysclk;
+	u16 reg_cache[ALC5623_VENDOR_ID2+2];
+	unsigned int add_ctrl;
+	unsigned int jack_det_ctrl;
+};
+
+static void alc5623_fill_cache(struct snd_soc_codec *codec)
+{
+	int i, step = codec->driver->reg_cache_step;
+	u16 *cache = codec->reg_cache;
+
+	/* not really efficient ... */
+	for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
+		cache[i] = codec->hw_read(codec, i);
+}
+
+static inline int alc5623_reset(struct snd_soc_codec *codec)
+{
+	return snd_soc_write(codec, ALC5623_RESET, 0);
+}
+
+static int amp_mixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	/* to power-on/off class-d amp generators/speaker */
+	/* need to write to 'index-46h' register :        */
+	/* so write index num (here 0x46) to reg 0x6a     */
+	/* and then 0xffff/0 to reg 0x6c                  */
+	snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
+		break;
+	}
+
+	return 0;
+}
+
+/*
+ * ALC5623 Controls
+ */
+
+static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
+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),
+};
+static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = {
+	SOC_DOUBLE_TLV("Speaker Playback Volume",
+			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Speaker Playback Switch",
+			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Playback Volume",
+			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Headphone Playback Switch",
+			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = {
+	SOC_DOUBLE_TLV("Speaker Playback Volume",
+			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Speaker Playback Switch",
+			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("Line Playback Volume",
+			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Line Playback Switch",
+			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
+	SOC_DOUBLE_TLV("Line Playback Volume",
+			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Line Playback Switch",
+			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Playback Volume",
+			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Headphone Playback Switch",
+			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_snd_controls[] = {
+	SOC_DOUBLE_TLV("Auxout Playback Volume",
+			ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Auxout Playback Switch",
+			ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("PCM Playback Volume",
+			ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
+	SOC_DOUBLE_TLV("AuxI Capture Volume",
+			ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
+	SOC_DOUBLE_TLV("LineIn Capture Volume",
+			ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
+	SOC_SINGLE_TLV("Mic1 Capture Volume",
+			ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
+	SOC_SINGLE_TLV("Mic2 Capture Volume",
+			ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
+	SOC_DOUBLE_TLV("Rec Capture Volume",
+			ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
+	SOC_SINGLE_TLV("Mic 1 Boost Volume",
+			ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
+	SOC_SINGLE_TLV("Mic 2 Boost Volume",
+			ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
+	SOC_SINGLE_TLV("Digital Boost Volume",
+			ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
+SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
+SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
+SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
+SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
+SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
+};
+
+/* Left Record Mixer */
+static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
+SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
+SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
+SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
+};
+
+/* Right Record Mixer */
+static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
+SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
+SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
+SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
+};
+
+static const char *alc5623_spk_n_sour_sel[] = {
+		"RN/-R", "RP/+R", "LN/-R", "Vmid" };
+static const char *alc5623_hpl_out_input_sel[] = {
+		"Vmid", "HP Left Mix"};
+static const char *alc5623_hpr_out_input_sel[] = {
+		"Vmid", "HP Right Mix"};
+static const char *alc5623_spkout_input_sel[] = {
+		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+static const char *alc5623_aux_out_input_sel[] = {
+		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+
+/* auxout output mux */
+static const struct soc_enum alc5623_aux_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
+static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
+
+/* speaker output mux */
+static const struct soc_enum alc5623_spkout_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
+static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
+
+/* headphone left output mux */
+static const struct soc_enum alc5623_hpl_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
+static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
+
+/* headphone right output mux */
+static const struct soc_enum alc5623_hpr_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
+static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
+
+/* speaker output N select */
+static const struct soc_enum alc5623_spk_n_sour_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
+static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
+
+static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
+/* Muxes */
+SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_auxout_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_spkout_mux_controls),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_hpl_out_mux_controls),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_hpr_out_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_spkoutn_mux_controls),
+
+/* output mixers */
+SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
+	&alc5623_hp_mixer_controls[0],
+	ARRAY_SIZE(alc5623_hp_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
+	&alc5623_hpr_mixer_controls[0],
+	ARRAY_SIZE(alc5623_hpr_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
+	&alc5623_hpl_mixer_controls[0],
+	ARRAY_SIZE(alc5623_hpl_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
+	&alc5623_mono_mixer_controls[0],
+	ARRAY_SIZE(alc5623_mono_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
+	&alc5623_speaker_mixer_controls[0],
+	ARRAY_SIZE(alc5623_speaker_mixer_controls)),
+
+/* input mixers */
+SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
+	&alc5623_captureL_mixer_controls[0],
+	ARRAY_SIZE(alc5623_captureL_mixer_controls)),
+SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
+	&alc5623_captureR_mixer_controls[0],
+	ARRAY_SIZE(alc5623_captureR_mixer_controls)),
+
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+	ALC5623_PWR_MANAG_ADD2, 9, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+	ALC5623_PWR_MANAG_ADD2, 8, 0),
+SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("AuxI 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", "Left HiFi Capture",
+	ALC5623_PWR_MANAG_ADD2, 7, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
+	ALC5623_PWR_MANAG_ADD2, 6, 0),
+SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
+
+SND_SOC_DAPM_OUTPUT("AUXOUTL"),
+SND_SOC_DAPM_OUTPUT("AUXOUTR"),
+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("AUXINL"),
+SND_SOC_DAPM_INPUT("AUXINR"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_VMID("Vmid"),
+};
+
+static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
+static const struct soc_enum alc5623_amp_enum =
+	SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
+static const struct snd_kcontrol_new alc5623_amp_mux_controls =
+	SOC_DAPM_ENUM("Route", alc5623_amp_enum);
+
+static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
+SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
+	amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
+SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_amp_mux_controls),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* virtual mixer - mixes left & right channels */
+	{"I2S Mix", NULL,				"Left DAC"},
+	{"I2S Mix", NULL,				"Right DAC"},
+	{"Line Mix", NULL,				"Right LineIn"},
+	{"Line Mix", NULL,				"Left LineIn"},
+	{"AuxI Mix", NULL,				"Left AuxI"},
+	{"AuxI Mix", NULL,				"Right AuxI"},
+	{"AUXOUTL", NULL,				"Left AuxOut"},
+	{"AUXOUTR", NULL,				"Right AuxOut"},
+
+	/* HP mixer */
+	{"HPL Mix", "ADC2HP_L Playback Switch",		"Left Capture Mix"},
+	{"HPL Mix", NULL,				"HP Mix"},
+	{"HPR Mix", "ADC2HP_R Playback Switch",		"Right Capture Mix"},
+	{"HPR Mix", NULL,				"HP Mix"},
+	{"HP Mix", "LI2HP Playback Switch",		"Line Mix"},
+	{"HP Mix", "AUXI2HP Playback Switch",		"AuxI Mix"},
+	{"HP Mix", "MIC12HP Playback Switch",		"MIC1 PGA"},
+	{"HP Mix", "MIC22HP Playback Switch",		"MIC2 PGA"},
+	{"HP Mix", "DAC2HP Playback Switch",		"I2S Mix"},
+
+	/* speaker mixer */
+	{"Speaker Mix", "LI2SPK Playback Switch",	"Line Mix"},
+	{"Speaker Mix", "AUXI2SPK Playback Switch",	"AuxI Mix"},
+	{"Speaker Mix", "MIC12SPK Playback Switch",	"MIC1 PGA"},
+	{"Speaker Mix", "MIC22SPK Playback Switch",	"MIC2 PGA"},
+	{"Speaker Mix", "DAC2SPK Playback Switch",	"I2S 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", "AUXI2MONO Playback Switch",	"AuxI Mix"},
+	{"Mono Mix", "MIC12MONO Playback Switch",	"MIC1 PGA"},
+	{"Mono Mix", "MIC22MONO Playback Switch",	"MIC2 PGA"},
+	{"Mono Mix", "DAC2MONO Playback Switch",	"I2S Mix"},
+
+	/* Left record mixer */
+	{"Left Capture Mix", "LineInL Capture Switch",	"LINEINL"},
+	{"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
+	{"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"},
+
+	/*Right record mixer */
+	{"Right Capture Mix", "LineInR Capture Switch",	"LINEINR"},
+	{"Right Capture Mix", "Right AuxI Capture Switch",	"AUXINR"},
+	{"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"},
+
+	/* headphone left mux */
+	{"Left Headphone Mux", "HP Left Mix",		"HPL Mix"},
+	{"Left Headphone Mux", "Vmid",			"Vmid"},
+
+	/* headphone right mux */
+	{"Right Headphone Mux", "HP Right Mix",		"HPR Mix"},
+	{"Right Headphone Mux", "Vmid",			"Vmid"},
+
+	/* speaker out mux */
+	{"SpeakerOut Mux", "Vmid",			"Vmid"},
+	{"SpeakerOut Mux", "HPOut Mix",			"HPOut Mix"},
+	{"SpeakerOut Mux", "Speaker Mix",		"Speaker Mix"},
+	{"SpeakerOut Mux", "Mono Mix",			"Mono Mix"},
+
+	/* Mono/Aux Out mux */
+	{"AuxOut Mux", "Vmid",				"Vmid"},
+	{"AuxOut Mux", "HPOut Mix",			"HPOut Mix"},
+	{"AuxOut Mux", "Speaker Mix",			"Speaker Mix"},
+	{"AuxOut Mux", "Mono Mix",			"Mono Mix"},
+
+	/* output pga */
+	{"HPL", NULL,					"Left Headphone"},
+	{"Left Headphone", NULL,			"Left Headphone Mux"},
+	{"HPR", NULL,					"Right Headphone"},
+	{"Right Headphone", NULL,			"Right Headphone Mux"},
+	{"Left AuxOut", NULL,				"AuxOut Mux"},
+	{"Right AuxOut", NULL,				"AuxOut Mux"},
+
+	/* input pga */
+	{"Left LineIn", NULL,				"LINEINL"},
+	{"Right LineIn", NULL,				"LINEINR"},
+	{"Left AuxI", NULL,				"AUXINL"},
+	{"Right AuxI", NULL,				"AUXINR"},
+	{"MIC1 Pre Amp", NULL,				"MIC1"},
+	{"MIC2 Pre Amp", NULL,				"MIC2"},
+	{"MIC1 PGA", NULL,				"MIC1 Pre Amp"},
+	{"MIC2 PGA", NULL,				"MIC2 Pre Amp"},
+
+	/* left ADC */
+	{"Left ADC", NULL,				"Left Capture Mix"},
+
+	/* right ADC */
+	{"Right ADC", NULL,				"Right Capture Mix"},
+
+	{"SpeakerOut N Mux", "RN/-R",			"SpeakerOut"},
+	{"SpeakerOut N Mux", "RP/+R",			"SpeakerOut"},
+	{"SpeakerOut N Mux", "LN/-R",			"SpeakerOut"},
+	{"SpeakerOut N Mux", "Vmid",			"Vmid"},
+
+	{"SPKOUT", NULL,				"SpeakerOut"},
+	{"SPKOUTN", NULL,				"SpeakerOut N Mux"},
+};
+
+static const struct snd_soc_dapm_route intercon_spk[] = {
+	{"SpeakerOut", NULL,				"SpeakerOut Mux"},
+};
+
+static const struct snd_soc_dapm_route intercon_amp_spk[] = {
+	{"AB Amp", NULL,				"SpeakerOut Mux"},
+	{"D Amp", NULL,					"SpeakerOut Mux"},
+	{"AB-D Amp Mux", "AB Amp",			"AB Amp"},
+	{"AB-D Amp Mux", "D Amp",			"D Amp"},
+	{"SpeakerOut", NULL,				"AB-D Amp Mux"},
+};
+
+/* PLL divisors */
+struct _pll_div {
+	u32 pll_in;
+	u32 pll_out;
+	u16 regvalue;
+};
+
+/* Note : pll code from original alc5623 driver. Not sure of how good it is */
+/* usefull only for master mode */
+static const struct _pll_div codec_master_pll_div[] = {
+
+	{  2048000,  8192000,	0x0ea0},
+	{  3686400,  8192000,	0x4e27},
+	{ 12000000,  8192000,	0x456b},
+	{ 13000000,  8192000,	0x495f},
+	{ 13100000,  8192000,	0x0320},
+	{  2048000,  11289600,	0xf637},
+	{  3686400,  11289600,	0x2f22},
+	{ 12000000,  11289600,	0x3e2f},
+	{ 13000000,  11289600,	0x4d5b},
+	{ 13100000,  11289600,	0x363b},
+	{  2048000,  16384000,	0x1ea0},
+	{  3686400,  16384000,	0x9e27},
+	{ 12000000,  16384000,	0x452b},
+	{ 13000000,  16384000,	0x542f},
+	{ 13100000,  16384000,	0x03a0},
+	{  2048000,  16934400,	0xe625},
+	{  3686400,  16934400,	0x9126},
+	{ 12000000,  16934400,	0x4d2c},
+	{ 13000000,  16934400,	0x742f},
+	{ 13100000,  16934400,	0x3c27},
+	{  2048000,  22579200,	0x2aa0},
+	{  3686400,  22579200,	0x2f20},
+	{ 12000000,  22579200,	0x7e2f},
+	{ 13000000,  22579200,	0x742f},
+	{ 13100000,  22579200,	0x3c27},
+	{  2048000,  24576000,	0x2ea0},
+	{  3686400,  24576000,	0xee27},
+	{ 12000000,  24576000,	0x2915},
+	{ 13000000,  24576000,	0x772e},
+	{ 13100000,  24576000,	0x0d20},
+};
+
+static const struct _pll_div codec_slave_pll_div[] = {
+
+	{  1024000,  16384000,  0x3ea0},
+	{  1411200,  22579200,	0x3ea0},
+	{  1536000,  24576000,	0x3ea0},
+	{  2048000,  16384000,  0x1ea0},
+	{  2822400,  22579200,	0x1ea0},
+	{  3072000,  24576000,	0x1ea0},
+
+};
+
+static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
+{
+	int i;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int gbl_clk = 0, pll_div = 0;
+	u16 reg;
+
+	if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
+		return -ENODEV;
+
+	/* Disable PLL power */
+	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
+				ALC5623_PWR_ADD2_PLL,
+				0);
+
+	/* pll is not used in slave mode */
+	reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
+	if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
+		return 0;
+
+	if (!freq_in || !freq_out)
+		return 0;
+
+	switch (pll_id) {
+	case ALC5623_PLL_FR_MCLK:
+		for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
+			if (codec_master_pll_div[i].pll_in == freq_in
+			   && codec_master_pll_div[i].pll_out == freq_out) {
+				/* PLL source from MCLK */
+				pll_div  = codec_master_pll_div[i].regvalue;
+				break;
+			}
+		}
+		break;
+	case ALC5623_PLL_FR_BCK:
+		for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
+			if (codec_slave_pll_div[i].pll_in == freq_in
+			   && codec_slave_pll_div[i].pll_out == freq_out) {
+				/* PLL source from Bitclk */
+				gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
+				pll_div = codec_slave_pll_div[i].regvalue;
+				break;
+			}
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (!pll_div)
+		return -EINVAL;
+
+	snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
+	snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
+	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
+				ALC5623_PWR_ADD2_PLL,
+				ALC5623_PWR_ADD2_PLL);
+	gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
+	snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
+
+	return 0;
+}
+
+struct _coeff_div {
+	u16 fs;
+	u16 regvalue;
+};
+
+/* codec hifi mclk (after PLL) clock divider coefficients */
+/* values inspired from column BCLK=32Fs of Appendix A table */
+static const struct _coeff_div coeff_div[] = {
+	{256*8, 0x3a69},
+	{384*8, 0x3c6b},
+	{256*4, 0x2a69},
+	{384*4, 0x2c6b},
+	{256*2, 0x1a69},
+	{384*2, 0x1c6b},
+	{256*1, 0x0a69},
+	{384*1, 0x0c6b},
+};
+
+static int get_coeff(struct snd_soc_codec *codec, int rate)
+{
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].fs * rate == alc5623->sysclk)
+			return i;
+	}
+	return -EINVAL;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
+	switch (freq) {
+	case  8192000:
+	case 11289600:
+	case 12288000:
+	case 16384000:
+	case 16934400:
+	case 18432000:
+	case 22579200:
+	case 24576000:
+		alc5623->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface = ALC5623_DAI_SDP_MASTER_MODE;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		iface = ALC5623_DAI_SDP_SLAVE_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= ALC5623_DAI_I2S_DF_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface |= ALC5623_DAI_I2S_DF_RIGHT;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= ALC5623_DAI_I2S_DF_LEFT;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= ALC5623_DAI_I2S_DF_PCM;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
+}
+
+static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+	int coeff, rate;
+	u16 iface;
+
+	iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
+	iface &= ~ALC5623_DAI_I2S_DL_MASK;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		iface |= ALC5623_DAI_I2S_DL_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= ALC5623_DAI_I2S_DL_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= ALC5623_DAI_I2S_DL_24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= ALC5623_DAI_I2S_DL_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set iface & srate */
+	snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
+	rate = params_rate(params);
+	coeff = get_coeff(codec, rate);
+	if (coeff < 0)
+		return -EINVAL;
+
+	coeff = coeff_div[coeff].regvalue;
+	dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
+		__func__, alc5623->sysclk, rate, coeff);
+	snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
+
+	return 0;
+}
+
+static int alc5623_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
+	u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
+
+	if (mute)
+		mute_reg |= hp_mute;
+
+	return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
+}
+
+#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
+	| ALC5623_PWR_ADD2_DAC_REF_CIR)
+
+#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
+	| ALC5623_PWR_ADD3_MIC1_BOOST_AD)
+
+#define ALC5623_ADD1_POWER_EN \
+	(ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
+	| ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
+	| ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
+
+#define ALC5623_ADD1_POWER_EN_5622 \
+	(ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
+	| ALC5623_PWR_ADD1_HP_OUT_AMP)
+
+static void enable_power_depop(struct snd_soc_codec *codec)
+{
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
+	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
+				ALC5623_PWR_ADD1_SOFTGEN_EN,
+				ALC5623_PWR_ADD1_SOFTGEN_EN);
+
+	snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
+
+	snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
+				ALC5623_MISC_HP_DEPOP_MODE2_EN,
+				ALC5623_MISC_HP_DEPOP_MODE2_EN);
+
+	msleep(500);
+
+	snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
+
+	/* avoid writing '1' into 5622 reserved bits */
+	if (alc5623->id == 0x22)
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
+			ALC5623_ADD1_POWER_EN_5622);
+	else
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
+			ALC5623_ADD1_POWER_EN);
+
+	/* disable HP Depop2 */
+	snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
+				ALC5623_MISC_HP_DEPOP_MODE2_EN,
+				0);
+
+}
+
+static int alc5623_set_bias_level(struct snd_soc_codec *codec,
+				      enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		enable_power_depop(codec);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* everything off except vref/vmid, */
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
+				ALC5623_PWR_ADD2_VREF);
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
+				ALC5623_PWR_ADD3_MAIN_BIAS);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define ALC5623_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE \
+			| SNDRV_PCM_FMTBIT_S24_LE \
+			| SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops alc5623_dai_ops = {
+		.hw_params = alc5623_pcm_hw_params,
+		.digital_mute = alc5623_mute,
+		.set_fmt = alc5623_set_dai_fmt,
+		.set_sysclk = alc5623_set_dai_sysclk,
+		.set_pll = alc5623_set_dai_pll,
+};
+
+static struct snd_soc_dai_driver alc5623_dai = {
+	.name = "alc5623-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rate_min =	8000,
+		.rate_max =	48000,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = ALC5623_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rate_min =	8000,
+		.rate_max =	48000,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = ALC5623_FORMATS,},
+
+	.ops = &alc5623_dai_ops,
+};
+
+static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+{
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int alc5623_resume(struct snd_soc_codec *codec)
+{
+	int i, step = codec->driver->reg_cache_step;
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
+		snd_soc_write(codec, i, cache[i]);
+
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* charge alc5623 caps */
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+		alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+		codec->bias_level = SND_SOC_BIAS_ON;
+		alc5623_set_bias_level(codec, codec->bias_level);
+	}
+
+	return 0;
+}
+
+static int alc5623_probe(struct snd_soc_codec *codec)
+{
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	alc5623_reset(codec);
+	alc5623_fill_cache(codec);
+
+	/* power on device */
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	if (alc5623->add_ctrl) {
+		snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
+				alc5623->add_ctrl);
+	}
+
+	if (alc5623->jack_det_ctrl) {
+		snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
+				alc5623->jack_det_ctrl);
+	}
+
+	switch (alc5623->id) {
+	default:
+	case 0x21:
+		snd_soc_add_controls(codec, rt5621_vol_snd_controls,
+			ARRAY_SIZE(rt5621_vol_snd_controls));
+		break;
+	case 0x22:
+		snd_soc_add_controls(codec, rt5622_vol_snd_controls,
+			ARRAY_SIZE(rt5622_vol_snd_controls));
+		break;
+	case 0x23:
+		snd_soc_add_controls(codec, alc5623_vol_snd_controls,
+			ARRAY_SIZE(alc5623_vol_snd_controls));
+		break;
+	}
+
+	snd_soc_add_controls(codec, alc5623_snd_controls,
+			ARRAY_SIZE(alc5623_snd_controls));
+
+	snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets,
+					ARRAY_SIZE(alc5623_dapm_widgets));
+
+	/* set up audio path interconnects */
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	switch (alc5623->id) {
+	default:
+	case 0x21:
+	case 0x22:
+		snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets,
+					ARRAY_SIZE(alc5623_dapm_amp_widgets));
+		snd_soc_dapm_add_routes(codec, intercon_amp_spk,
+						ARRAY_SIZE(intercon_amp_spk));
+		break;
+	case 0x23:
+		snd_soc_dapm_add_routes(codec, intercon_spk,
+						ARRAY_SIZE(intercon_spk));
+		break;
+	}
+
+	return ret;
+}
+
+/* power down chip */
+static int alc5623_remove(struct snd_soc_codec *codec)
+{
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
+	.probe = alc5623_probe,
+	.remove = alc5623_remove,
+	.suspend = alc5623_suspend,
+	.resume = alc5623_resume,
+	.set_bias_level = alc5623_set_bias_level,
+	.reg_cache_size = ALC5623_VENDOR_ID2+2,
+	.reg_word_size = sizeof(u16),
+	.reg_cache_step = 2,
+};
+
+/*
+ * ALC5623 2 wire address is determined by A1 pin
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+static int alc5623_i2c_probe(struct i2c_client *client,
+				const struct i2c_device_id *id)
+{
+	struct alc5623_platform_data *pdata;
+	struct alc5623_priv *alc5623;
+	int ret, vid1, vid2;
+
+	vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
+	if (vid1 < 0) {
+		dev_err(&client->dev, "failed to read I2C\n");
+		return -EIO;
+	}
+	vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
+
+	vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
+	if (vid2 < 0) {
+		dev_err(&client->dev, "failed to read I2C\n");
+		return -EIO;
+	}
+
+	if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
+		dev_err(&client->dev, "unknown or wrong codec\n");
+		dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
+				0x10ec, id->driver_data,
+				vid1, vid2);
+		return -ENODEV;
+	}
+
+	dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
+
+	alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL);
+	if (alc5623 == NULL) {
+		ret = -ENOMEM;
+		goto err;
+	}
+
+	pdata = client->dev.platform_data;
+	if (pdata) {
+		alc5623->add_ctrl = pdata->add_ctrl;
+		alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
+	}
+
+	alc5623->id = vid2;
+	switch (alc5623->id) {
+	case 0x21:
+		alc5623_dai.name = "alc5621-hifi";
+		break;
+	case 0x22:
+		alc5623_dai.name = "alc5622-hifi";
+		break;
+	default:
+	case 0x23:
+		alc5623_dai.name = "alc5623-hifi";
+		break;
+	}
+
+	i2c_set_clientdata(client, alc5623);
+	alc5623->control_data = client;
+	alc5623->control_type = SND_SOC_I2C;
+	mutex_init(&alc5623->mutex);
+
+	ret =  snd_soc_register_codec(&client->dev,
+		&soc_codec_device_alc5623, &alc5623_dai, 1);
+	if (ret != 0) {
+		dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	return 0;
+
+err:
+	return ret;
+}
+
+static int alc5623_i2c_remove(struct i2c_client *client)
+{
+	struct alc5623_priv *alc5623 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+	kfree(alc5623);
+	return 0;
+}
+
+static const struct i2c_device_id alc5623_i2c_table[] = {
+	{"alc5621", 0x21},
+	{"alc5622", 0x22},
+	{"alc5623", 0x23},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
+
+/*  i2c codec control layer */
+static struct i2c_driver alc5623_i2c_driver = {
+	.driver = {
+		.name = "alc562x-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = alc5623_i2c_probe,
+	.remove =  __devexit_p(alc5623_i2c_remove),
+	.id_table = alc5623_i2c_table,
+};
+
+static int __init alc5623_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&alc5623_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "%s: can't add i2c driver", __func__);
+		return ret;
+	}
+
+	return ret;
+}
+module_init(alc5623_modinit);
+
+static void __exit alc5623_modexit(void)
+{
+	i2c_del_driver(&alc5623_i2c_driver);
+}
+module_exit(alc5623_modexit);
+
+MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
+MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/alc5623.h

+/*
+ * alc5623.h  --  alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * Author: flove <flove@realtek.com>
+ * Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _ALC5623_H
+#define _ALC5623_H
+
+#define ALC5623_RESET				0x00
+/*				5621 5622 5623  */
+/* speaker output vol		   2    2       */
+/* line output vol                      4    2  */
+/* HP output vol		   4    0    4  */
+#define ALC5623_SPK_OUT_VOL			0x02
+#define ALC5623_HP_OUT_VOL			0x04
+#define ALC5623_MONO_AUX_OUT_VOL		0x06
+#define ALC5623_AUXIN_VOL			0x08
+#define ALC5623_LINE_IN_VOL			0x0A
+#define ALC5623_STEREO_DAC_VOL			0x0C
+#define ALC5623_MIC_VOL				0x0E
+#define ALC5623_MIC_ROUTING_CTRL		0x10
+#define ALC5623_ADC_REC_GAIN			0x12
+#define ALC5623_ADC_REC_MIXER			0x14
+#define ALC5623_SOFT_VOL_CTRL_TIME		0x16
+/* ALC5623_OUTPUT_MIXER_CTRL :			*/
+/* same remark as for reg 2 line vs speaker	*/
+#define ALC5623_OUTPUT_MIXER_CTRL		0x1C
+#define ALC5623_MIC_CTRL			0x22
+
+#define	ALC5623_DAI_CONTROL			0x34
+#define ALC5623_DAI_SDP_MASTER_MODE		(0 << 15)
+#define ALC5623_DAI_SDP_SLAVE_MODE		(1 << 15)
+#define ALC5623_DAI_I2S_PCM_MODE		(1 << 14)
+#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL	(1 <<  7)
+#define ALC5623_DAI_ADC_DATA_L_R_SWAP		(1 <<  5)
+#define ALC5623_DAI_DAC_DATA_L_R_SWAP		(1 <<  4)
+#define ALC5623_DAI_I2S_DL_MASK			(3 <<  2)
+#define ALC5623_DAI_I2S_DL_32			(3 <<  2)
+#define	ALC5623_DAI_I2S_DL_24			(2 <<  2)
+#define ALC5623_DAI_I2S_DL_20			(1 <<  2)
+#define ALC5623_DAI_I2S_DL_16			(0 <<  2)
+#define ALC5623_DAI_I2S_DF_PCM			(3 <<  0)
+#define	ALC5623_DAI_I2S_DF_LEFT			(2 <<  0)
+#define ALC5623_DAI_I2S_DF_RIGHT		(1 <<  0)
+#define ALC5623_DAI_I2S_DF_I2S			(0 <<  0)
+
+#define ALC5623_STEREO_AD_DA_CLK_CTRL		0x36
+#define	ALC5623_COMPANDING_CTRL			0x38
+
+#define	ALC5623_PWR_MANAG_ADD1			0x3A
+#define ALC5623_PWR_ADD1_MAIN_I2S_EN		(1 << 15)
+#define ALC5623_PWR_ADD1_ZC_DET_PD_EN		(1 << 14)
+#define ALC5623_PWR_ADD1_MIC1_BIAS_EN		(1 << 11)
+#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN	(1 << 10)
+#define ALC5623_PWR_ADD1_SOFTGEN_EN		(1 <<  8) /* rsvd on 5622 */
+#define	ALC5623_PWR_ADD1_DEPOP_BUF_HP		(1 <<  6) /* rsvd on 5622 */
+#define	ALC5623_PWR_ADD1_HP_OUT_AMP		(1 <<  5)
+#define	ALC5623_PWR_ADD1_HP_OUT_ENH_AMP		(1 <<  4) /* rsvd on 5622 */
+#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX		(1 <<  2)
+#define ALC5623_PWR_ADD1_AUX_OUT_AMP		(1 <<  1)
+#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP	(1 <<  0) /* rsvd on 5622 */
+
+#define ALC5623_PWR_MANAG_ADD2			0x3C
+#define ALC5623_PWR_ADD2_LINEOUT		(1 << 15) /* rt5623 */
+#define ALC5623_PWR_ADD2_CLASS_AB		(1 << 15) /* rt5621 */
+#define ALC5623_PWR_ADD2_CLASS_D		(1 << 14) /* rt5621 */
+#define ALC5623_PWR_ADD2_VREF			(1 << 13)
+#define ALC5623_PWR_ADD2_PLL			(1 << 12)
+#define ALC5623_PWR_ADD2_DAC_REF_CIR		(1 << 10)
+#define ALC5623_PWR_ADD2_L_DAC_CLK		(1 <<  9)
+#define ALC5623_PWR_ADD2_R_DAC_CLK		(1 <<  8)
+#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN		(1 <<  7)
+#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN		(1 <<  6)
+#define ALC5623_PWR_ADD2_L_HP_MIXER		(1 <<  5)
+#define ALC5623_PWR_ADD2_R_HP_MIXER		(1 <<  4)
+#define ALC5623_PWR_ADD2_SPK_MIXER		(1 <<  3)
+#define ALC5623_PWR_ADD2_MONO_MIXER		(1 <<  2)
+#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER	(1 <<  1)
+#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER	(1 <<  0)
+
+#define ALC5623_PWR_MANAG_ADD3			0x3E
+#define ALC5623_PWR_ADD3_MAIN_BIAS		(1 << 15)
+#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP	(1 << 14)
+#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP	(1 << 13)
+#define ALC5623_PWR_ADD3_SPK_OUT		(1 << 12)
+#define ALC5623_PWR_ADD3_HP_L_OUT_VOL		(1 << 10)
+#define ALC5623_PWR_ADD3_HP_R_OUT_VOL		(1 <<  9)
+#define ALC5623_PWR_ADD3_LINEIN_L_VOL		(1 <<  7)
+#define ALC5623_PWR_ADD3_LINEIN_R_VOL		(1 <<  6)
+#define ALC5623_PWR_ADD3_AUXIN_L_VOL		(1 <<  5)
+#define ALC5623_PWR_ADD3_AUXIN_R_VOL		(1 <<  4)
+#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL		(1 <<  3)
+#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL		(1 <<  2)
+#define ALC5623_PWR_ADD3_MIC1_BOOST_AD		(1 <<  1)
+#define ALC5623_PWR_ADD3_MIC2_BOOST_AD		(1 <<  0)
+
+#define ALC5623_ADD_CTRL_REG			0x40
+
+#define	ALC5623_GLOBAL_CLK_CTRL_REG		0x42
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL	(1 << 15)
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK	(0 << 15)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK	(1 << 14)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK	(0 << 14)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8	(3 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4	(2 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2	(1 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1	(0 <<  1)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV2		(1 <<  0)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV1		(0 <<  0)
+
+#define ALC5623_PLL_CTRL			0x44
+#define ALC5623_PLL_CTRL_N_VAL(n)		(((n)&0xff) << 8)
+#define ALC5623_PLL_CTRL_K_VAL(k)		(((k)&0x7)  << 4)
+#define ALC5623_PLL_CTRL_M_VAL(m)		((m)&0xf)
+
+#define ALC5623_GPIO_OUTPUT_PIN_CTRL		0x4A
+#define ALC5623_GPIO_PIN_CONFIG			0x4C
+#define ALC5623_GPIO_PIN_POLARITY		0x4E
+#define ALC5623_GPIO_PIN_STICKY			0x50
+#define ALC5623_GPIO_PIN_WAKEUP			0x52
+#define ALC5623_GPIO_PIN_STATUS			0x54
+#define ALC5623_GPIO_PIN_SHARING		0x56
+#define	ALC5623_OVER_CURR_STATUS		0x58
+#define ALC5623_JACK_DET_CTRL			0x5A
+
+#define ALC5623_MISC_CTRL			0x5E
+#define ALC5623_MISC_DISABLE_FAST_VREG		(1 << 15)
+#define ALC5623_MISC_SPK_CLASS_AB_OC_PD		(1 << 13) /* 5621 */
+#define ALC5623_MISC_SPK_CLASS_AB_OC_DET	(1 << 12) /* 5621 */
+#define ALC5623_MISC_HP_DEPOP_MODE3_EN		(1 << 10)
+#define ALC5623_MISC_HP_DEPOP_MODE2_EN		(1 <<  9)
+#define ALC5623_MISC_HP_DEPOP_MODE1_EN		(1 <<  8)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN	(1 <<  6)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN	(1 <<  5)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN	(1 <<  4)
+#define ALC5623_MISC_M_DAC_L_INPUT		(1 <<  3)
+#define ALC5623_MISC_M_DAC_R_INPUT		(1 <<  2)
+#define ALC5623_MISC_IRQOUT_INV_CTRL		(1 <<  0)
+
+#define	ALC5623_PSEDUEO_SPATIAL_CTRL		0x60
+#define ALC5623_EQ_CTRL				0x62
+#define ALC5623_EQ_MODE_ENABLE			0x66
+#define ALC5623_AVC_CTRL			0x68
+#define ALC5623_HID_CTRL_INDEX			0x6A
+#define ALC5623_HID_CTRL_DATA			0x6C
+#define ALC5623_VENDOR_ID1			0x7C
+#define ALC5623_VENDOR_ID2			0x7E
+
+#define ALC5623_PLL_FR_MCLK			0
+#define ALC5623_PLL_FR_BCK			1
+#endif

sound/soc/codecs/wm8900.c

@@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg)
 {
 	switch (reg) {
 	case WM8900_REG_ID:
-	case WM8900_REG_POWER1:
 		return 1;
 	default:
 		return 0;
@@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec)
 		return -ENODEV;
 	}
 
-	/* Read back from the chip */
-	reg = snd_soc_read(codec, WM8900_REG_POWER1);
-	reg = (reg >> 12) & 0xf;
-	dev_info(codec->dev, "WM8900 revision %d\n", reg);
-
 	wm8900_reset(codec);
 
 	/* Turn the chip on */

sound/soc/codecs/wm8993.c

@@ -735,6 +735,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol,
 					    0);
 		}
 		wm8993->class_w_users++;
+		wm8993->hubs_data.class_w = true;
 	}
 
 	/* Implement the change */
@@ -751,6 +752,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol,
 					    WM8993_CP_DYN_V);
 		}
 		wm8993->class_w_users--;
+		wm8993->hubs_data.class_w = false;
 	}
 
 	dev_dbg(codec->dev, "Indirect DAC use count now %d\n",

sound/soc/codecs/wm8994.c

@@ -2228,6 +2228,7 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
 
 static void wm8994_update_class_w(struct snd_soc_codec *codec)
 {
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int enable = 1;
 	int source = 0;  /* GCC flow analysis can't track enable */
 	int reg, reg_r;
@@ -2278,11 +2279,13 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
 				    WM8994_CP_DYN_PWR |
 				    WM8994_CP_DYN_SRC_SEL_MASK,
 				    source | WM8994_CP_DYN_PWR);
+		wm8994->hubs.class_w = true;
 		
 	} else {
 		dev_dbg(codec->dev, "Class W disabled\n");
 		snd_soc_update_bits(codec, WM8994_CLASS_W_1,
 				    WM8994_CP_DYN_PWR, 0);
+		wm8994->hubs.class_w = false;
 	}
 }
 

sound/soc/codecs/wm_hubs.c

@@ -94,6 +94,18 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	u16 reg, reg_l, reg_r, dcs_cfg;
 
+	/* If we're using a digital only path and have a previously
+	 * callibrated DC servo offset stored then use that. */
+	if (hubs->class_w && hubs->class_w_dcs) {
+		dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
+			hubs->class_w_dcs);
+		snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+		wait_for_dc_servo(codec,
+				  WM8993_DCS_TRIG_DAC_WR_0 |
+				  WM8993_DCS_TRIG_DAC_WR_1);
+		return;
+	}
+
 	/* Set for 32 series updates */
 	snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
 			    WM8993_DCS_SERIES_NO_01_MASK,
@@ -101,34 +113,34 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 	wait_for_dc_servo(codec,
 			  WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1);
 
+	/* Different chips in the family support different readback
+	 * methods.
+	 */
+	switch (hubs->dcs_readback_mode) {
+	case 0:
+		reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+			& WM8993_DCS_INTEG_CHAN_0_MASK;;
+		reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
+			& WM8993_DCS_INTEG_CHAN_1_MASK;
+		break;
+	case 1:
+		reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+		reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+			>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+		break;
+	default:
+		WARN(1, "Unknown DCS readback method\n");
+		break;
+	}
+
+	dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
+
 	/* Apply correction to DC servo result */
 	if (hubs->dcs_codes) {
 		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
 			hubs->dcs_codes);
 
-		/* Different chips in the family support different
-		 * readback methods.
-		 */
-		switch (hubs->dcs_readback_mode) {
-		case 0:
-			reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
-				& WM8993_DCS_INTEG_CHAN_0_MASK;;
-			reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
-				& WM8993_DCS_INTEG_CHAN_1_MASK;
-			break;
-		case 1:
-			reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
-			reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
-				>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
-			reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
-			break;
-		default:
-			WARN(1, "Unknown DCS readback method");
-			break;
-		}
-
-		dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
-
 		/* HPOUT1L */
 		if (reg_l + hubs->dcs_codes > 0 &&
 		    reg_l + hubs->dcs_codes < 0xff)
@@ -148,7 +160,15 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 		wait_for_dc_servo(codec,
 				  WM8993_DCS_TRIG_DAC_WR_0 |
 				  WM8993_DCS_TRIG_DAC_WR_1);
+	} else {
+		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		dcs_cfg |= reg_r;
 	}
+
+	/* Save the callibrated offset if we're in class W mode and
+	 * therefore don't have any analogue signal mixed in. */
+	if (hubs->class_w)
+		hubs->class_w_dcs = dcs_cfg;
 }
 
 /*
@@ -163,6 +183,9 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 
 	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
 
+	/* Updating the analogue gains invalidates the DC servo cache */
+	hubs->class_w_dcs = 0;
+
 	/* If we're applying an offset correction then updating the
 	 * callibration would be likely to introduce further offsets. */
 	if (hubs->dcs_codes)

sound/soc/codecs/wm_hubs.h

@@ -23,6 +23,9 @@ struct wm_hubs_data {
 	int dcs_codes;
 	int dcs_readback_mode;
 	int hp_startup_mode;
+
+	bool class_w;
+	u16 class_w_dcs;
 };
 
 extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);

sound/soc/kirkwood/Kconfig

@@ -18,3 +18,12 @@ config SND_KIRKWOOD_SOC_OPENRD
 	  Say Y if you want to add support for SoC audio on
 	  Openrd Client.
 
+config SND_KIRKWOOD_SOC_T5325
+	tristate "SoC Audio support for HP t5325"
+	depends on SND_KIRKWOOD_SOC && MACH_T5325
+	select SND_KIRKWOOD_SOC_I2S
+	select SND_SOC_ALC5623
+	help
+	  Say Y if you want to add support for SoC audio on
+	  the HP t5325 thin client.
+

sound/soc/kirkwood/Makefile

@@ -5,5 +5,7 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
 obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
 
 snd-soc-openrd-objs := kirkwood-openrd.o
+snd-soc-t5325-objs := kirkwood-t5325.o
 
 obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
+obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o

sound/soc/kirkwood/kirkwood-t5325.c

+/*
+ * kirkwood-t5325.c
+ *
+ * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <mach/kirkwood.h>
+#include <plat/audio.h>
+#include <asm/mach-types.h>
+#include "../codecs/alc5623.h"
+
+static int t5325_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+	unsigned int freq, fmt;
+
+	fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	freq = params_rate(params) * 256;
+
+	return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
+
+}
+
+static struct snd_soc_ops t5325_ops = {
+	.hw_params = t5325_hw_params,
+};
+
+static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route t5325_route[] = {
+	{ "Headphone Jack",	NULL,	"HPL" },
+	{ "Headphone Jack",	NULL,	"HPR" },
+
+	{"Speaker",		NULL,	"SPKOUT"},
+	{"Speaker",		NULL,	"SPKOUTN"},
+
+	{ "MIC1",		NULL,	"Mic Jack" },
+	{ "MIC2",		NULL,	"Mic Jack" },
+};
+
+static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+
+	snd_soc_dapm_new_controls(codec, t5325_dapm_widgets,
+				ARRAY_SIZE(t5325_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, t5325_route, ARRAY_SIZE(t5325_route));
+
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Speaker");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link t5325_dai[] = {
+{
+	.name = "ALC5621",
+	.stream_name = "ALC5621 HiFi",
+	.cpu_dai_name = "kirkwood-i2s",
+	.platform_name = "kirkwood-pcm-audio",
+	.codec_dai_name = "alc5621-hifi",
+	.codec_name = "alc562x-codec.0-001a",
+	.ops = &t5325_ops,
+	.init = t5325_dai_init,
+},
+};
+
+
+static struct snd_soc_card t5325 = {
+	.name = "t5325",
+	.dai_link = t5325_dai,
+	.num_links = ARRAY_SIZE(t5325_dai),
+};
+
+static struct platform_device *t5325_snd_device;
+
+static int __init t5325_init(void)
+{
+	int ret;
+
+	if (!machine_is_t5325())
+		return 0;
+
+	t5325_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!t5325_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(t5325_snd_device,
+			&t5325);
+
+	ret = platform_device_add(t5325_snd_device);
+	if (ret) {
+		printk(KERN_ERR "%s: platform_device_add failed\n", __func__);
+		platform_device_put(t5325_snd_device);
+	}
+
+	return ret;
+}
+module_init(t5325_init);
+
+static void __exit t5325_exit(void)
+{
+	platform_device_unregister(t5325_snd_device);
+}
+module_exit(t5325_exit);
+
+MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
+MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
+MODULE_LICENSE("GPL");

sound/soc/pxa/tosa.c

@@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);

sound/soc/soc-core.c

@@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev,
 {
 	struct snd_soc_pcm_runtime *rtd =
 			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	int ret;
 
-	strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	if (ret)
+		return ret;
 
 	return count;
 }