Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
  ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
  ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
  ALSA: usb-audio: add Yamaha MOX6/MOX8 support
  ALSA: virtuoso: add S/PDIF input support for all Xonars
  ALSA: ice1724 - Support for ooAoo SQ210a
  ALSA: ice1724 - Allow card info based on model only
  ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
  ALSA: hdspm - Provide unique driver id based on card serial
  ASoC: Dynamically allocate the rtd device for a non-empty release()
  ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
  ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
  ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
  ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
  ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
  ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
  ALSA: hda/cirrus - support for iMac12,2 model
  ASoC: cx20442: add bias control over a platform provided regulator
  ALSA: usb-audio - Avoid flood of frame-active debug messages
  ALSA: snd-usb-us122l: Delete calls to preempt_disable
  mfd: Put WM8994 into cache only mode when suspending
  ...

Fix up trivial conflicts in:
 - arch/arm/mach-s3c64xx/mach-crag6410.c:
	renamed speyside_wm8962 to tobermory, added littlemill right
	next to it
 - drivers/base/regmap/{regcache.c,regmap.c}:
	duplicate diff that had already come in with other changes in
	the regmap tree

1 files changed, 1159 insertions, 0 deletions

diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
new file mode 100644
index 000000000000..390e437d7c5e
--- /dev/null
+++ b/sound/soc/codecs/alc5632.c
@@ -0,0 +1,1159 @@
+/*
+* alc5632.c  --  ALC5632 ALSA SoC Audio Codec
+*
+* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
+*
+* Authors:  Leon Romanovsky <leon@leon.nu>
+*           Andrey Danin <danindrey@mail.ru>
+*           Ilya Petrov <ilya.muromec@gmail.com>
+*           Marc Dietrich <marvin24@gmx.de>
+*
+* Based on alc5623.c by Arnaud Patard
+*
+* 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/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#include "alc5632.h"
+
+/*
+ * ALC5632 register cache
+ */
+static struct reg_default  alc5632_reg_defaults[] = {
+	{   2, 0x8080 },	/* R2   - Speaker Output Volume */
+	{   4, 0x8080 },	/* R4   - Headphone Output Volume */
+	{   6, 0x8080 },	/* R6   - AUXOUT Volume */
+	{   8, 0xC800 },	/* R8   - Phone Input */
+	{  10, 0xE808 },	/* R10  - LINE_IN Volume */
+	{  12, 0x1010 },	/* R12  - STEREO DAC Input Volume */
+	{  14, 0x0808 },	/* R14  - MIC Input Volume */
+	{  16, 0xEE0F },	/* R16  - Stereo DAC and MIC Routing Control */
+	{  18, 0xCBCB },	/* R18  - ADC Record Gain */
+	{  20, 0x7F7F },	/* R20  - ADC Record Mixer Control */
+	{  24, 0xE010 },	/* R24  - Voice DAC Volume */
+	{  28, 0x8008 },	/* R28  - Output Mixer Control */
+	{  34, 0x0000 },	/* R34  - Microphone Control */
+	{  36, 0x00C0 },    /* R36  - Codec Digital MIC/Digital Boost
+						   Control */
+	{  46, 0x0000 },	/* R46  - Stereo DAC/Voice DAC/Stereo ADC
+						   Function Select */
+	{  52, 0x8000 },	/* R52  - Main Serial Data Port Control
+						   (Stereo I2S) */
+	{  54, 0x0000 },	/* R54  - Extend Serial Data Port Control
+						   (VoDAC_I2S/PCM) */
+	{  58, 0x0000 },	/* R58  - Power Management Addition 1 */
+	{  60, 0x0000 },	/* R60  - Power Management Addition 2 */
+	{  62, 0x8000 },	/* R62  - Power Management Addition 3 */
+	{  64, 0x0C0A },	/* R64  - General Purpose Control Register 1 */
+	{  66, 0x0000 },	/* R66  - General Purpose Control Register 2 */
+	{  68, 0x0000 },	/* R68  - PLL1 Control */
+	{  70, 0x0000 },	/* R70  - PLL2 Control */
+	{  76, 0xBE3E },	/* R76  - GPIO Pin Configuration */
+	{  78, 0xBE3E },	/* R78  - GPIO Pin Polarity */
+	{  80, 0x0000 },	/* R80  - GPIO Pin Sticky */
+	{  82, 0x0000 },	/* R82  - GPIO Pin Wake Up */
+	{  86, 0x0000 },	/* R86  - Pin Sharing */
+	{  90, 0x0009 },	/* R90  - Soft Volume Control Setting */
+	{  92, 0x0000 },	/* R92  - GPIO_Output Pin Control */
+	{  94, 0x3000 },	/* R94  - MISC Control */
+	{  96, 0x3075 },	/* R96  - Stereo DAC Clock Control_1 */
+	{  98, 0x1010 },	/* R98  - Stereo DAC Clock Control_2 */
+	{ 100, 0x3110 },	/* R100 - VoDAC_PCM Clock Control_1 */
+	{ 104, 0x0553 },	/* R104 - Pseudo Stereo and Spatial Effect
+						   Block Control */
+	{ 106, 0x0000 },	/* R106 - Private Register Address */
+};
+
+/* codec private data */
+struct alc5632_priv {
+	struct regmap *regmap;
+	u8 id;
+	unsigned int sysclk;
+};
+
+static bool alc5632_volatile_register(struct device *dev,
+							unsigned int reg)
+{
+	switch (reg) {
+	case ALC5632_RESET:
+	case ALC5632_PWR_DOWN_CTRL_STATUS:
+	case ALC5632_GPIO_PIN_STATUS:
+	case ALC5632_OVER_CURR_STATUS:
+	case ALC5632_HID_CTRL_DATA:
+	case ALC5632_EQ_CTRL:
+	case ALC5632_VENDOR_ID1:
+	case ALC5632_VENDOR_ID2:
+		return true;
+
+	default:
+		break;
+	}
+
+	return false;
+}
+
+static inline int alc5632_reset(struct regmap *map)
+{
+	return regmap_write(map, ALC5632_RESET, 0x59B4);
+}
+
+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, ALC5632_HID_CTRL_INDEX, 0x46);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0);
+		break;
+	}
+
+	return 0;
+}
+
+/*
+ * ALC5632 Controls
+ */
+
+/* -34.5db min scale, 1.5db steps, no mute */
+static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
+/* -46.5db min scale, 1.5db steps, no mute */
+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),
+};
+/* 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 struct snd_kcontrol_new alc5632_vol_snd_controls[] = {
+	/* left starts at bit 8, right at bit 0 */
+	/* 31 steps (5 bit), -46.5db scale */
+	SOC_DOUBLE_TLV("Speaker Playback Volume",
+			ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	/* bit 15 mutes left, bit 7 right */
+	SOC_DOUBLE("Speaker Playback Switch",
+			ALC5632_SPK_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Playback Volume",
+			ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Headphone Playback Switch",
+			ALC5632_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_snd_controls[] = {
+	SOC_DOUBLE_TLV("Auxout Playback Volume",
+			ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Auxout Playback Switch",
+			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",
+			ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv),
+	SOC_DOUBLE_TLV("LineIn Capture 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",
+			ALC5632_MIC_VOL, 8, 31, 1, vol_tlv),
+	SOC_SINGLE_TLV("Mic2 Capture 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),
+	SOC_SINGLE_TLV("Mic 2 Boost Volume",
+			ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv),
+	SOC_SINGLE_TLV("Digital Boost Volume",
+			ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1),
+SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1),
+SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1),
+SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1),
+SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1),
+SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC12MONO Playback Switch",
+					ALC5632_MIC_ROUTING_CTRL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC22MONO Playback Switch",
+					ALC5632_MIC_ROUTING_CTRL, 9, 1, 1),
+SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1),
+SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC12SPK Playback Switch",
+					ALC5632_MIC_ROUTING_CTRL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC22SPK Playback Switch",
+					ALC5632_MIC_ROUTING_CTRL, 10, 1, 1),
+SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1),
+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),
+};
+
+/* 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),
+};
+
+static const char *alc5632_spk_n_sour_sel[] = {
+		"RN/-R", "RP/+R", "LN/-R", "Mute"};
+static const char *alc5632_hpl_out_input_sel[] = {
+		"Vmid", "HP Left Mix"};
+static const char *alc5632_hpr_out_input_sel[] = {
+		"Vmid", "HP Right Mix"};
+static const char *alc5632_spkout_input_sel[] = {
+		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+static const char *alc5632_aux_out_input_sel[] = {
+		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+
+/* auxout output mux */
+static const struct soc_enum alc5632_aux_out_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel);
+static const struct snd_kcontrol_new alc5632_auxout_mux_controls =
+SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum);
+
+/* speaker output mux */
+static const struct soc_enum alc5632_spkout_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel);
+static const struct snd_kcontrol_new alc5632_spkout_mux_controls =
+SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum);
+
+/* headphone left output mux */
+static const struct soc_enum alc5632_hpl_out_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel);
+static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls =
+SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum);
+
+/* headphone right output mux */
+static const struct soc_enum alc5632_hpr_out_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel);
+static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls =
+SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum);
+
+/* speaker output N select */
+static const struct soc_enum alc5632_spk_n_sour_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel);
+static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls =
+SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum);
+
+/* speaker amplifier */
+static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"};
+static const struct soc_enum alc5632_amp_enum =
+	SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names);
+static const struct snd_kcontrol_new alc5632_amp_mux_controls =
+	SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum);
+
+
+static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = {
+/* Muxes */
+SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
+	&alc5632_auxout_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
+	&alc5632_spkout_mux_controls),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
+	&alc5632_hpl_out_mux_controls),
+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),
+
+/* output mixers */
+SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
+	&alc5632_hp_mixer_controls[0],
+	ARRAY_SIZE(alc5632_hp_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0,
+	&alc5632_hpr_mixer_controls[0],
+	ARRAY_SIZE(alc5632_hpr_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0,
+	&alc5632_hpl_mixer_controls[0],
+	ARRAY_SIZE(alc5632_hpl_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0,
+	&alc5632_mono_mixer_controls[0],
+	ARRAY_SIZE(alc5632_mono_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0,
+	&alc5632_speaker_mixer_controls[0],
+	ARRAY_SIZE(alc5632_speaker_mixer_controls)),
+
+/* input mixers */
+SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0,
+	&alc5632_captureL_mixer_controls[0],
+	ARRAY_SIZE(alc5632_captureL_mixer_controls)),
+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_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_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),
+SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_E("D Amp", ALC5632_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", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0),
+SND_SOC_DAPM_MUX("AB-D Amp Mux", ALC5632_PWR_MANAG_ADD1, 10, 0,
+	&alc5632_amp_mux_controls),
+
+SND_SOC_DAPM_OUTPUT("AUXOUT"),
+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("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_VMID("Vmid"),
+};
+
+
+static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
+	/* 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"},
+	{"Phone Mix",	NULL,	"Phone"},
+	{"Phone Mix",	NULL,	"Phone ADMix"},
+	{"AUXOUT",		NULL,	"Aux Out"},
+
+	/* DAC */
+	{"DAC Right Channel",	NULL,	"I2S Mix"},
+	{"DAC Left Channel",	NULL,   "I2S Mix"},
+
+	/* 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",	"PHONE2HP Playback Switch",	"Phone Mix"},
+	{"HP Mix",	"MIC12HP Playback Switch",	"MIC1 PGA"},
+	{"HP Mix",	"MIC22HP Playback Switch",	"MIC2 PGA"},
+
+	{"HPR Mix", "DACR2HP Playback Switch",	"DAC Right Channel"},
+	{"HPL Mix", "DACL2HP Playback Switch",	"DAC Left Channel"},
+
+	/* speaker mixer */
+	{"Speaker Mix", "LI2SPK Playback Switch",	"Line Mix"},
+	{"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"},
+	{"Speaker Mix", "MIC12SPK Playback Switch",	"MIC1 PGA"},
+	{"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"},
+	{"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"},
+
+	/* 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"},
+
+	/*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"},
+
+	/* 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"},
+	{"Aux Out", NULL,				"AuxOut Mux"},
+
+	/* input pga */
+	{"Left LineIn", NULL,				"LINEINL"},
+	{"Right LineIn", NULL,				"LINEINR"},
+	{"Phone", NULL,				"PHONEP"},
+	{"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",			"Left Speaker"},
+	{"SpeakerOut N Mux", "RP/+R",			"Left Speaker"},
+	{"SpeakerOut N Mux", "LN/-R",			"Left Speaker"},
+	{"SpeakerOut N Mux", "Mute",			"Vmid"},
+
+	{"SpeakerOut N Mux", "RN/-R",			"Right Speaker"},
+	{"SpeakerOut N Mux", "RP/+R",			"Right Speaker"},
+	{"SpeakerOut N Mux", "LN/-R",			"Right Speaker"},
+	{"SpeakerOut N Mux", "Mute",			"Vmid"},
+
+	{"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"},
+	{"Left Speaker", NULL,				"AB-D Amp Mux"},
+	{"Right Speaker", NULL,				"AB-D Amp Mux"},
+
+	{"SPKOUT", NULL,				"Left Speaker"},
+	{"SPKOUT", NULL,				"Right Speaker"},
+
+	{"SPKOUTN", NULL,				"SpeakerOut N Mux"},
+
+};
+
+/* PLL divisors */
+struct _pll_div {
+	u32 pll_in;
+	u32 pll_out;
+	u16 regvalue;
+};
+
+/* Note : pll code from original alc5632 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},
+};
+
+/* FOUT = MCLK*(N+2)/((M+2)*(K+2))
+   N: bit 15:8 (div 2 .. div 257)
+   K: bit  6:4 typical 2
+   M: bit  3:0 (div 2 .. div 17)
+
+   same as for 5623 - thanks!
+*/
+
+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 alc5632_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 < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK)
+		return -EINVAL;
+
+	/* Disable PLL power */
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+				ALC5632_PWR_ADD2_PLL1,
+				0);
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+				ALC5632_PWR_ADD2_PLL2,
+				0);
+
+	/* pll is not used in slave mode */
+	reg = snd_soc_read(codec, ALC5632_DAI_CONTROL);
+	if (reg & ALC5632_DAI_SDP_SLAVE_MODE)
+		return 0;
+
+	if (!freq_in || !freq_out)
+		return 0;
+
+	switch (pll_id) {
+	case ALC5632_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 ALC5632_PLL_FR_BCLK:
+		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 = ALC5632_PLL_FR_BCLK;
+				pll_div = codec_slave_pll_div[i].regvalue;
+				break;
+			}
+		}
+		break;
+	case ALC5632_PLL_FR_VBCLK:
+		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 voice clock */
+				gbl_clk = ALC5632_PLL_FR_VBCLK;
+				pll_div = codec_slave_pll_div[i].regvalue;
+				break;
+			}
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (!pll_div)
+		return -EINVAL;
+
+	/* choose MCLK/BCLK/VBCLK */
+	snd_soc_write(codec, ALC5632_GPCR2, gbl_clk);
+	/* choose PLL1 clock rate */
+	snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div);
+	/* enable PLL1 */
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+				ALC5632_PWR_ADD2_PLL1,
+				ALC5632_PWR_ADD2_PLL1);
+	/* enable PLL2 */
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+				ALC5632_PWR_ADD2_PLL2,
+				ALC5632_PWR_ADD2_PLL2);
+	/* use PLL1 as main SYSCLK */
+	snd_soc_update_bits(codec, ALC5632_GPCR1,
+			ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1,
+			ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1);
+
+	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[] = {
+	{512*1, 0x3075},
+};
+
+static int get_coeff(struct snd_soc_codec *codec, int rate)
+{
+	struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].fs * rate == alc5632->sysclk)
+			return i;
+	}
+	return -EINVAL;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int alc5632_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 alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
+
+	switch (freq) {
+	case  8192000:
+	case 11289600:
+	case 12288000:
+	case 16384000:
+	case 16934400:
+	case 18432000:
+	case 22579200:
+	case 24576000:
+		alc5632->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int alc5632_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 = ALC5632_DAI_SDP_MASTER_MODE;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		iface = ALC5632_DAI_SDP_SLAVE_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= ALC5632_DAI_I2S_DF_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= ALC5632_DAI_I2S_DF_LEFT;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= ALC5632_DAI_I2S_DF_PCM_A;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= ALC5632_DAI_I2S_DF_PCM_B;
+		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 |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface);
+}
+
+static int alc5632_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;
+	int coeff, rate;
+	u16 iface;
+
+	iface = snd_soc_read(codec, ALC5632_DAI_CONTROL);
+	iface &= ~ALC5632_DAI_I2S_DL_MASK;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		iface |= ALC5632_DAI_I2S_DL_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= ALC5632_DAI_I2S_DL_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= ALC5632_DAI_I2S_DL_24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set iface & srate */
+	snd_soc_write(codec, ALC5632_DAI_CONTROL, iface);
+	rate = params_rate(params);
+	coeff = get_coeff(codec, rate);
+	if (coeff < 0)
+		return -EINVAL;
+
+	coeff = coeff_div[coeff].regvalue;
+	snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff);
+
+	return 0;
+}
+
+static int alc5632_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L
+						|ALC5632_MISC_HP_DEPOP_MUTE_R;
+	u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute;
+
+	if (mute)
+		mute_reg |= hp_mute;
+
+	return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg);
+}
+
+#define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF)
+
+#define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD)
+
+#define ALC5632_ADD1_POWER_EN \
+		(ALC5632_PWR_ADD1_DAC_REF \
+		| ALC5632_PWR_ADD1_SOFTGEN_EN \
+		| ALC5632_PWR_ADD1_HP_OUT_AMP \
+		| ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \
+		| ALC5632_PWR_ADD1_MAIN_BIAS)
+
+static void enable_power_depop(struct snd_soc_codec *codec)
+{
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+				ALC5632_PWR_ADD1_SOFTGEN_EN,
+				ALC5632_PWR_ADD1_SOFTGEN_EN);
+
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3,
+				ALC5632_ADD3_POWER_EN,
+				ALC5632_ADD3_POWER_EN);
+
+	snd_soc_update_bits(codec, ALC5632_MISC_CTRL,
+				ALC5632_MISC_HP_DEPOP_MODE2_EN,
+				ALC5632_MISC_HP_DEPOP_MODE2_EN);
+
+	/* "normal" mode: 0 @ 26 */
+	/* set all PR0-7 mixers to 0 */
+	snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS,
+				ALC5632_PWR_DOWN_CTRL_STATUS_MASK,
+				0);
+
+	msleep(500);
+
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+				ALC5632_ADD2_POWER_EN,
+				ALC5632_ADD2_POWER_EN);
+
+	snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+				ALC5632_ADD1_POWER_EN,
+				ALC5632_ADD1_POWER_EN);
+
+	/* disable HP Depop2 */
+	snd_soc_update_bits(codec, ALC5632_MISC_CTRL,
+				ALC5632_MISC_HP_DEPOP_MODE2_EN,
+				0);
+
+}
+
+static int alc5632_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_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+				ALC5632_PWR_MANAG_ADD1_MASK,
+				ALC5632_PWR_ADD1_MAIN_BIAS);
+		snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+				ALC5632_PWR_MANAG_ADD2_MASK,
+				ALC5632_PWR_ADD2_VREF);
+		/* "normal" mode: 0 @ 26 */
+		snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS,
+				ALC5632_PWR_DOWN_CTRL_STATUS_MASK,
+				0xffff ^ (ALC5632_PWR_VREF_PR3
+				| ALC5632_PWR_VREF_PR2));
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+				ALC5632_PWR_MANAG_ADD2_MASK, 0);
+		snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3,
+				ALC5632_PWR_MANAG_ADD3_MASK, 0);
+		snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+				ALC5632_PWR_MANAG_ADD1_MASK, 0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define ALC5632_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE \
+			| SNDRV_PCM_FMTBIT_S24_LE \
+			| SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops alc5632_dai_ops = {
+		.hw_params = alc5632_pcm_hw_params,
+		.digital_mute = alc5632_mute,
+		.set_fmt = alc5632_set_dai_fmt,
+		.set_sysclk = alc5632_set_dai_sysclk,
+		.set_pll = alc5632_set_dai_pll,
+};
+
+static struct snd_soc_dai_driver alc5632_dai = {
+	.name = "alc5632-hifi",
+	.playback = {
+		.stream_name = "HiFi Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rate_min =	8000,
+		.rate_max =	48000,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = ALC5632_FORMATS,},
+	.capture = {
+		.stream_name = "HiFi Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rate_min =	8000,
+		.rate_max =	48000,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = ALC5632_FORMATS,},
+
+	.ops = &alc5632_dai_ops,
+	.symmetric_rates = 1,
+};
+
+#ifdef CONFIG_PM
+static int alc5632_suspend(struct snd_soc_codec *codec)
+{
+	alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int alc5632_resume(struct snd_soc_codec *codec)
+{
+	struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
+
+	regcache_sync(alc5632->regmap);
+
+	alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+#else
+#define	alc5632_suspend	NULL
+#define	alc5632_resume	NULL
+#endif
+
+static int alc5632_probe(struct snd_soc_codec *codec)
+{
+	struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	codec->control_data = alc5632->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	/* power on device  */
+	alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	switch (alc5632->id) {
+	case 0x5c:
+		snd_soc_add_controls(codec, alc5632_vol_snd_controls,
+			ARRAY_SIZE(alc5632_vol_snd_controls));
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return ret;
+}
+
+/* power down chip */
+static int alc5632_remove(struct snd_soc_codec *codec)
+{
+	alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_alc5632 = {
+	.probe = alc5632_probe,
+	.remove = alc5632_remove,
+	.suspend = alc5632_suspend,
+	.resume = alc5632_resume,
+	.set_bias_level = alc5632_set_bias_level,
+	.controls = alc5632_snd_controls,
+	.num_controls = ARRAY_SIZE(alc5632_snd_controls),
+	.dapm_widgets = alc5632_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(alc5632_dapm_widgets),
+	.dapm_routes = alc5632_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes),
+};
+
+static struct regmap_config alc5632_regmap = {
+	.reg_bits = 8,
+	.val_bits = 16,
+
+	.max_register = ALC5632_MAX_REGISTER,
+	.reg_defaults = alc5632_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(alc5632_reg_defaults),
+	.volatile_reg = alc5632_volatile_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+/*
+ * alc5632 2 wire address is determined by A1 pin
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+static __devinit int alc5632_i2c_probe(struct i2c_client *client,
+				const struct i2c_device_id *id)
+{
+	struct alc5632_priv *alc5632;
+	int ret, ret1, ret2;
+	unsigned int vid1, vid2;
+
+	alc5632 = devm_kzalloc(&client->dev,
+			 sizeof(struct alc5632_priv), GFP_KERNEL);
+	if (alc5632 == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(client, alc5632);
+
+	alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap);
+	if (IS_ERR(alc5632->regmap)) {
+		ret = PTR_ERR(alc5632->regmap);
+		dev_err(&client->dev, "regmap_init() failed: %d\n", ret);
+		return ret;
+	}
+
+	ret1 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID1, &vid1);
+	ret2 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID2, &vid2);
+	if (ret1 != 0 || ret2 != 0) {
+		dev_err(&client->dev,
+		"Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2);
+		regmap_exit(alc5632->regmap);
+		return -EIO;
+	}
+
+	vid2 >>= 8;
+
+	if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) {
+		dev_err(&client->dev,
+		"Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2);
+		regmap_exit(alc5632->regmap);
+		return -EINVAL;
+	}
+
+	ret = alc5632_reset(alc5632->regmap);
+	if (ret < 0) {
+		dev_err(&client->dev, "Failed to issue reset\n");
+		regmap_exit(alc5632->regmap);
+		return ret;
+	}
+
+	alc5632->id = vid2;
+	switch (alc5632->id) {
+	case 0x5c:
+		alc5632_dai.name = "alc5632-hifi";
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ret = snd_soc_register_codec(&client->dev,
+		&soc_codec_device_alc5632, &alc5632_dai, 1);
+
+	if (ret < 0) {
+		dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+		regmap_exit(alc5632->regmap);
+		return ret;
+	}
+
+	return ret;
+}
+
+static int alc5632_i2c_remove(struct i2c_client *client)
+{
+	struct alc5632_priv *alc5632 = i2c_get_clientdata(client);
+	snd_soc_unregister_codec(&client->dev);
+	regmap_exit(alc5632->regmap);
+	return 0;
+}
+
+static const struct i2c_device_id alc5632_i2c_table[] = {
+	{"alc5632", 0x5c},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table);
+
+/* i2c codec control layer */
+static struct i2c_driver alc5632_i2c_driver = {
+	.driver = {
+		.name = "alc5632",
+		.owner = THIS_MODULE,
+	},
+	.probe = alc5632_i2c_probe,
+	.remove =  __devexit_p(alc5632_i2c_remove),
+	.id_table = alc5632_i2c_table,
+};
+
+static int __init alc5632_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&alc5632_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "%s: can't add i2c driver", __func__);
+		return ret;
+	}
+
+	return ret;
+}
+module_init(alc5632_modinit);
+
+static void __exit alc5632_modexit(void)
+{
+	i2c_del_driver(&alc5632_i2c_driver);
+}
+module_exit(alc5632_modexit);
+
+MODULE_DESCRIPTION("ASoC ALC5632 driver");
+MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>");
+MODULE_LICENSE("GPL");