Merge branch 'master' of git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6 into for-linus-1

* 'master' of git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6: (9356 commits)
  [media] rc: update for bitop name changes
  fs: simplify iget & friends
  fs: pull inode->i_lock up out of writeback_single_inode
  fs: rename inode_lock to inode_hash_lock
  fs: move i_wb_list out from under inode_lock
  fs: move i_sb_list out from under inode_lock
  fs: remove inode_lock from iput_final and prune_icache
  fs: Lock the inode LRU list separately
  fs: factor inode disposal
  fs: protect inode->i_state with inode->i_lock
  lib, arch: add filter argument to show_mem and fix private implementations
  SLUB: Write to per cpu data when allocating it
  slub: Fix debugobjects with lockless fastpath
  autofs4: Do not potentially dereference NULL pointer returned by fget() in autofs_dev_ioctl_setpipefd()
  autofs4 - remove autofs4_lock
  autofs4 - fix d_manage() return on rcu-walk
  autofs4 - fix autofs4_expire_indirect() traversal
  autofs4 - fix dentry leak in autofs4_expire_direct()
  autofs4 - reinstate last used update on access
  vfs - check non-mountpoint dentry might block in __follow_mount_rcu()
  ...

NOTE!

This merge commit was created to fix compilation error. The block
tree was merged upstream and removed the 'elv_queue_empty()'
function which the new 'mtdswap' driver is using. So a simple
merge of the mtd tree with upstream does not compile. And the
mtd tree has already be published, so re-basing it is not an option.

To fix this unfortunate situation, I had to merge upstream into the
mtd-2.6.git tree without committing, put the fixup patch on top of
this, and then commit this. The result is that we do not have commits
which do not compile.

In other words, this merge commit "merges" 3 things: the MTD tree, the
upstream tree, and the fixup patch.

148 files changed, 14120 insertions, 2731 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index a3efc52a34d..8224db5f043 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -50,10 +50,12 @@ source "sound/soc/jz4740/Kconfig"
 source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/kirkwood/Kconfig"
+source "sound/soc/mid-x86/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/samsung/Kconfig"
 source "sound/soc/s6000/Kconfig"
 source "sound/soc/sh/Kconfig"
+source "sound/soc/tegra/Kconfig"
 source "sound/soc/txx9/Kconfig"
 
 # Supported codecs
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index ce913bf5213..1ed61c5df2c 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -10,6 +10,7 @@ obj-$(CONFIG_SND_SOC)	+= ep93xx/
 obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)   += imx/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
+obj-$(CONFIG_SND_SOC)	+= mid-x86/
 obj-$(CONFIG_SND_SOC)	+= nuc900/
 obj-$(CONFIG_SND_SOC)	+= omap/
 obj-$(CONFIG_SND_SOC)	+= kirkwood/
@@ -17,4 +18,5 @@ obj-$(CONFIG_SND_SOC)	+= pxa/
 obj-$(CONFIG_SND_SOC)	+= samsung/
 obj-$(CONFIG_SND_SOC)	+= s6000/
 obj-$(CONFIG_SND_SOC)	+= sh/
+obj-$(CONFIG_SND_SOC)	+= tegra/
 obj-$(CONFIG_SND_SOC)	+= txx9/
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e06b0..5e4d499d843 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
 	.cpu_dai_name = "atmel-ssc-dai.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.platform_name = "atmel_pcm-audio",
-	.codec_name = "tlv320aic23-codec.0-0x1a",
+	.codec_name = "tlv320aic23-codec.0-001a",
 	.init = afeb9260_tlv320aic23_init,
 	.ops = &afeb9260_ops,
 };
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24c185..ad28663f5bb 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
 	.cpu_dai_name = "bf5xx-i2s",
 	.codec_dai_name = "ssm2602-hifi",
 	.platform_name = "bf5xx-pcm-audio",
-	.codec_name = "ssm2602-codec.0-0x1b",
+	.codec_name = "ssm2602-codec.0-001b",
 	.ops = &bf5xx_ssm2602_ops,
 };
 
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index c48b23c1d4f..6943e24a74a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -26,17 +26,24 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS42L51 if I2C
 	select SND_SOC_CS4270 if I2C
+	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
+	select SND_SOC_DFBMCS320
 	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
+	select SND_SOC_LM4857 if I2C
 	select SND_SOC_MAX98088 if I2C
+	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
+	select SND_SOC_SGTL5000 if I2C
+	select SND_SOC_SN95031 if INTEL_SCU_IPC
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
+	select SND_SOC_TVL320AIC32X4 if I2C
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
@@ -44,7 +51,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TWL6040 if TWL4030_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
-	select SND_SOC_WL1273 if RADIO_WL1273
+	select SND_SOC_WL1273 if MFD_WL1273_CORE
 	select SND_SOC_WM2000 if I2C
 	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8400 if MFD_WM8400
@@ -76,6 +83,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8990 if I2C
+	select SND_SOC_WM8991 if I2C
 	select SND_SOC_WM8993 if I2C
 	select SND_SOC_WM8994 if MFD_WM8994
 	select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
@@ -155,6 +163,9 @@ config SND_SOC_CS4270_VD33_ERRATA
 	bool
 	depends on SND_SOC_CS4270
 
+config SND_SOC_CS4271
+	tristate
+
 config SND_SOC_CX20442
 	tristate
 
@@ -167,15 +178,28 @@ config SND_SOC_L3
 config SND_SOC_DA7210
         tristate
 
+config SND_SOC_DFBMCS320
+	tristate
+
 config SND_SOC_DMIC
 	tristate
 
 config SND_SOC_MAX98088
        tristate
 
+config SND_SOC_MAX9850
+	tristate
+
 config SND_SOC_PCM3008
        tristate
 
+#Freescale sgtl5000 codec
+config SND_SOC_SGTL5000
+	tristate
+
+config SND_SOC_SN95031
+	tristate
+
 config SND_SOC_SPDIF
 	tristate
 
@@ -192,6 +216,9 @@ config SND_SOC_TLV320AIC26
 	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
 	depends on SPI
 
+config SND_SOC_TVL320AIC32X4
+	tristate
+
 config SND_SOC_TLV320AIC3X
 	tristate
 
@@ -304,6 +331,9 @@ config SND_SOC_WM8988
 config SND_SOC_WM8990
 	tristate
 
+config SND_SOC_WM8991
+	tristate
+
 config SND_SOC_WM8993
 	tristate
 
@@ -326,6 +356,9 @@ config SND_SOC_WM9713
 	tristate
 
 # Amp
+config SND_SOC_LM4857
+	tristate
+
 config SND_SOC_MAX9877
 	tristate
 
@@ -337,4 +370,3 @@ config SND_SOC_WM2000
 
 config SND_SOC_WM9090
 	tristate
-
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 579af9c4f12..379bc55f072 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -12,19 +12,25 @@ snd-soc-ak4671-objs := ak4671.o
 snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs42l51-objs := cs42l51.o
 snd-soc-cs4270-objs := cs4270.o
+snd-soc-cs4271-objs := cs4271.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
+snd-soc-dfbmcs320-objs := dfbmcs320.o
 snd-soc-dmic-objs := dmic.o
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
+snd-soc-max9850-objs := max9850.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-sgtl5000-objs := sgtl5000.o
 snd-soc-alc5623-objs := alc5623.o
+snd-soc-sn95031-objs := sn95031.o
 snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
 snd-soc-twl4030-objs := twl4030.o
 snd-soc-twl6040-objs := twl6040.o
@@ -61,6 +67,7 @@ snd-soc-wm8978-objs := wm8978.o
 snd-soc-wm8985-objs := wm8985.o
 snd-soc-wm8988-objs := wm8988.o
 snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm8991-objs := wm8991.o
 snd-soc-wm8993-objs := wm8993.o
 snd-soc-wm8994-objs := wm8994.o wm8994-tables.o
 snd-soc-wm8995-objs := wm8995.o
@@ -72,6 +79,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o
 snd-soc-jz4740-codec-objs := jz4740.o
 
 # Amp
+snd-soc-lm4857-objs := lm4857.o
 snd-soc-max9877-objs := max9877.o
 snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-wm2000-objs := wm2000.o
@@ -88,23 +96,29 @@ obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
+obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_CS4271)	+= snd-soc-cs4271.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
+obj-$(CONFIG_SND_SOC_DFBMCS320)	+= snd-soc-dfbmcs320.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(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-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
-obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
+obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
+obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
 obj-$(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
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TVL320AIC32X4)     += snd-soc-tlv320aic32x4.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
@@ -141,6 +155,7 @@ obj-$(CONFIG_SND_SOC_WM8978)	+= snd-soc-wm8978.o
 obj-$(CONFIG_SND_SOC_WM8985)	+= snd-soc-wm8985.o
 obj-$(CONFIG_SND_SOC_WM8988)	+= snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
 obj-$(CONFIG_SND_SOC_WM8993)	+= snd-soc-wm8993.o
 obj-$(CONFIG_SND_SOC_WM8994)	+= snd-soc-wm8994.o
 obj-$(CONFIG_SND_SOC_WM8995)	+= snd-soc-wm8995.o
@@ -151,6 +166,7 @@ obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
 obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 
 # Amp
+obj-$(CONFIG_SND_SOC_LM4857)	+= snd-soc-lm4857.o
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
 obj-$(CONFIG_SND_SOC_WM2000)	+= snd-soc-wm2000.o
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index c27f8f59dc6..cbf0b6d400b 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -294,7 +294,6 @@ static struct spi_driver ak4104_spi_driver = {
 
 static int __init ak4104_init(void)
 {
-	pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n");
 	return spi_register_driver(&ak4104_spi_driver);
 }
 module_init(ak4104_init);
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index f00eba313df..4be0570e3f1 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -116,6 +116,12 @@
 #define BCKO_MASK	(1 << 3)
 #define BCKO_64		BCKO_MASK
 
+#define DIF_MASK	(3 << 0)
+#define DSP		(0 << 0)
+#define RIGHT_J		(1 << 0)
+#define LEFT_J		(2 << 0)
+#define I2S		(3 << 0)
+
 /* MD_CTL2 */
 #define FS0		(1 << 0)
 #define FS1		(1 << 1)
@@ -354,6 +360,24 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	snd_soc_update_bits(codec, PW_MGMT2, MS, data);
 	snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
 
+	/* format type */
+	data = 0;
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_LEFT_J:
+		data = LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		data = I2S;
+		break;
+	/* FIXME
+	 * Please add RIGHT_J / DSP support here
+	 */
+	default:
+		return -EINVAL;
+		break;
+	}
+	snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd067f7..b8066ef10bb 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,8 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
 
 static int cq93vc_probe(struct snd_soc_codec *codec)
 {
-	struct davinci_vc *davinci_vc = codec->dev->platform_data;
+	struct davinci_vc *davinci_vc =
+			mfd_get_data(to_platform_device(codec->dev));
 
 	davinci_vc->cq93vc.codec = codec;
 	codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 8b51245f231..0206a17d728 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -193,12 +193,12 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
 /* The number of MCLK/LRCK ratios supported by the CS4270 */
 #define NUM_MCLK_RATIOS		ARRAY_SIZE(cs4270_mode_ratios)
 
-static int cs4270_reg_is_readable(unsigned int reg)
+static int cs4270_reg_is_readable(struct snd_soc_codec *codec, unsigned int reg)
 {
 	return (reg >= CS4270_FIRSTREG) && (reg <= CS4270_LASTREG);
 }
 
-static int cs4270_reg_is_volatile(unsigned int reg)
+static int cs4270_reg_is_volatile(struct snd_soc_codec *codec, unsigned int reg)
 {
 	/* Unreadable registers are considered volatile */
 	if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
@@ -719,7 +719,7 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 /*
  * cs4270_id - I2C device IDs supported by this driver
  */
-static struct i2c_device_id cs4270_id[] = {
+static const struct i2c_device_id cs4270_id[] = {
 	{"cs4270", 0},
 	{}
 };
@@ -743,8 +743,6 @@ static struct i2c_driver cs4270_i2c_driver = {
 
 static int __init cs4270_init(void)
 {
-	pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
-
 	return i2c_add_driver(&cs4270_i2c_driver);
 }
 module_init(cs4270_init);
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
new file mode 100644
index 00000000000..083aab96ca8
--- /dev/null
+++ b/sound/soc/codecs/cs4271.c
@@ -0,0 +1,667 @@
+/*
+ * CS4271 ASoC codec driver
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * This driver support CS4271 codec being master or slave, working
+ * in control port mode, connected either via SPI or I2C.
+ * The data format accepted is I2S or left-justified.
+ * DAPM support not implemented.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/cs4271.h>
+
+#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+			    SNDRV_PCM_FMTBIT_S24_LE | \
+			    SNDRV_PCM_FMTBIT_S32_LE)
+#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000
+
+/*
+ * CS4271 registers
+ * High byte represents SPI chip address (0x10) + write command (0)
+ * Low byte - codec register address
+ */
+#define CS4271_MODE1	0x2001	/* Mode Control 1 */
+#define CS4271_DACCTL	0x2002	/* DAC Control */
+#define CS4271_DACVOL	0x2003	/* DAC Volume & Mixing Control */
+#define CS4271_VOLA	0x2004	/* DAC Channel A Volume Control */
+#define CS4271_VOLB	0x2005	/* DAC Channel B Volume Control */
+#define CS4271_ADCCTL	0x2006	/* ADC Control */
+#define CS4271_MODE2	0x2007	/* Mode Control 2 */
+#define CS4271_CHIPID	0x2008	/* Chip ID */
+
+#define CS4271_FIRSTREG	CS4271_MODE1
+#define CS4271_LASTREG	CS4271_MODE2
+#define CS4271_NR_REGS	((CS4271_LASTREG & 0xFF) + 1)
+
+/* Bit masks for the CS4271 registers */
+#define CS4271_MODE1_MODE_MASK	0xC0
+#define CS4271_MODE1_MODE_1X	0x00
+#define CS4271_MODE1_MODE_2X	0x80
+#define CS4271_MODE1_MODE_4X	0xC0
+
+#define CS4271_MODE1_DIV_MASK	0x30
+#define CS4271_MODE1_DIV_1	0x00
+#define CS4271_MODE1_DIV_15	0x10
+#define CS4271_MODE1_DIV_2	0x20
+#define CS4271_MODE1_DIV_3	0x30
+
+#define CS4271_MODE1_MASTER	0x08
+
+#define CS4271_MODE1_DAC_DIF_MASK	0x07
+#define CS4271_MODE1_DAC_DIF_LJ		0x00
+#define CS4271_MODE1_DAC_DIF_I2S	0x01
+#define CS4271_MODE1_DAC_DIF_RJ16	0x02
+#define CS4271_MODE1_DAC_DIF_RJ24	0x03
+#define CS4271_MODE1_DAC_DIF_RJ20	0x04
+#define CS4271_MODE1_DAC_DIF_RJ18	0x05
+
+#define CS4271_DACCTL_AMUTE	0x80
+#define CS4271_DACCTL_IF_SLOW	0x40
+
+#define CS4271_DACCTL_DEM_MASK	0x30
+#define CS4271_DACCTL_DEM_DIS	0x00
+#define CS4271_DACCTL_DEM_441	0x10
+#define CS4271_DACCTL_DEM_48	0x20
+#define CS4271_DACCTL_DEM_32	0x30
+
+#define CS4271_DACCTL_SVRU	0x08
+#define CS4271_DACCTL_SRD	0x04
+#define CS4271_DACCTL_INVA	0x02
+#define CS4271_DACCTL_INVB	0x01
+
+#define CS4271_DACVOL_BEQUA	0x40
+#define CS4271_DACVOL_SOFT	0x20
+#define CS4271_DACVOL_ZEROC	0x10
+
+#define CS4271_DACVOL_ATAPI_MASK	0x0F
+#define CS4271_DACVOL_ATAPI_M_M		0x00
+#define CS4271_DACVOL_ATAPI_M_BR	0x01
+#define CS4271_DACVOL_ATAPI_M_BL	0x02
+#define CS4271_DACVOL_ATAPI_M_BLR2	0x03
+#define CS4271_DACVOL_ATAPI_AR_M	0x04
+#define CS4271_DACVOL_ATAPI_AR_BR	0x05
+#define CS4271_DACVOL_ATAPI_AR_BL	0x06
+#define CS4271_DACVOL_ATAPI_AR_BLR2	0x07
+#define CS4271_DACVOL_ATAPI_AL_M	0x08
+#define CS4271_DACVOL_ATAPI_AL_BR	0x09
+#define CS4271_DACVOL_ATAPI_AL_BL	0x0A
+#define CS4271_DACVOL_ATAPI_AL_BLR2	0x0B
+#define CS4271_DACVOL_ATAPI_ALR2_M	0x0C
+#define CS4271_DACVOL_ATAPI_ALR2_BR	0x0D
+#define CS4271_DACVOL_ATAPI_ALR2_BL	0x0E
+#define CS4271_DACVOL_ATAPI_ALR2_BLR2	0x0F
+
+#define CS4271_VOLA_MUTE	0x80
+#define CS4271_VOLA_VOL_MASK	0x7F
+#define CS4271_VOLB_MUTE	0x80
+#define CS4271_VOLB_VOL_MASK	0x7F
+
+#define CS4271_ADCCTL_DITHER16	0x20
+
+#define CS4271_ADCCTL_ADC_DIF_MASK	0x10
+#define CS4271_ADCCTL_ADC_DIF_LJ	0x00
+#define CS4271_ADCCTL_ADC_DIF_I2S	0x10
+
+#define CS4271_ADCCTL_MUTEA	0x08
+#define CS4271_ADCCTL_MUTEB	0x04
+#define CS4271_ADCCTL_HPFDA	0x02
+#define CS4271_ADCCTL_HPFDB	0x01
+
+#define CS4271_MODE2_LOOP	0x10
+#define CS4271_MODE2_MUTECAEQUB	0x08
+#define CS4271_MODE2_FREEZE	0x04
+#define CS4271_MODE2_CPEN	0x02
+#define CS4271_MODE2_PDN	0x01
+
+#define CS4271_CHIPID_PART_MASK	0xF0
+#define CS4271_CHIPID_REV_MASK	0x0F
+
+/*
+ * Default CS4271 power-up configuration
+ * Array contains non-existing in hw register at address 0
+ * Array do not include Chip ID, as codec driver does not use
+ * registers read operations at all
+ */
+static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = {
+	0,
+	0,
+	CS4271_DACCTL_AMUTE,
+	CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR,
+	0,
+	0,
+	0,
+	0,
+};
+
+struct cs4271_private {
+	/* SND_SOC_I2C or SND_SOC_SPI */
+	enum snd_soc_control_type	bus_type;
+	void				*control_data;
+	unsigned int			mclk;
+	bool				master;
+	bool				deemph;
+	/* Current sample rate for de-emphasis control */
+	int				rate;
+	/* GPIO driving Reset pin, if any */
+	int				gpio_nreset;
+	/* GPIO that disable serial bus, if any */
+	int				gpio_disable;
+};
+
+/*
+ * @freq is the desired MCLK rate
+ * MCLK rate should (c) be the sample rate, multiplied by one of the
+ * ratios listed in cs4271_mclk_fs_ratios table
+ */
+static int cs4271_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 cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
+	cs4271->mclk = freq;
+	return 0;
+}
+
+static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val = 0;
+	int ret;
+
+	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs4271->master = 0;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs4271->master = 1;
+		val |= CS4271_MODE1_MASTER;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_LEFT_J:
+		val |= CS4271_MODE1_DAC_DIF_LJ;
+		ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+			CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ);
+		if (ret < 0)
+			return ret;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		val |= CS4271_MODE1_DAC_DIF_I2S;
+		ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+			CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S);
+		if (ret < 0)
+			return ret;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	ret = snd_soc_update_bits(codec, CS4271_MODE1,
+		CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val);
+	if (ret < 0)
+		return ret;
+	return 0;
+}
+
+static int cs4271_deemph[] = {0, 44100, 48000, 32000};
+
+static int cs4271_set_deemph(struct snd_soc_codec *codec)
+{
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+	int i, ret;
+	int val = CS4271_DACCTL_DEM_DIS;
+
+	if (cs4271->deemph) {
+		/* Find closest de-emphasis freq */
+		val = 1;
+		for (i = 2; i < ARRAY_SIZE(cs4271_deemph); i++)
+			if (abs(cs4271_deemph[i] - cs4271->rate) <
+			    abs(cs4271_deemph[val] - cs4271->rate))
+				val = i;
+		val <<= 4;
+	}
+
+	ret = snd_soc_update_bits(codec, CS4271_DACCTL,
+		CS4271_DACCTL_DEM_MASK, val);
+	if (ret < 0)
+		return ret;
+	return 0;
+}
+
+static int cs4271_get_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = cs4271->deemph;
+	return 0;
+}
+
+static int cs4271_put_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
+	cs4271->deemph = ucontrol->value.enumerated.item[0];
+	return cs4271_set_deemph(codec);
+}
+
+struct cs4271_clk_cfg {
+	bool		master;		/* codec mode */
+	u8		speed_mode;	/* codec speed mode: 1x, 2x, 4x */
+	unsigned short	ratio;		/* MCLK / sample rate */
+	u8		ratio_mask;	/* ratio bit mask for Master mode */
+};
+
+static struct cs4271_clk_cfg cs4271_clk_tab[] = {
+	{1, CS4271_MODE1_MODE_1X, 256,  CS4271_MODE1_DIV_1},
+	{1, CS4271_MODE1_MODE_1X, 384,  CS4271_MODE1_DIV_15},
+	{1, CS4271_MODE1_MODE_1X, 512,  CS4271_MODE1_DIV_2},
+	{1, CS4271_MODE1_MODE_1X, 768,  CS4271_MODE1_DIV_3},
+	{1, CS4271_MODE1_MODE_2X, 128,  CS4271_MODE1_DIV_1},
+	{1, CS4271_MODE1_MODE_2X, 192,  CS4271_MODE1_DIV_15},
+	{1, CS4271_MODE1_MODE_2X, 256,  CS4271_MODE1_DIV_2},
+	{1, CS4271_MODE1_MODE_2X, 384,  CS4271_MODE1_DIV_3},
+	{1, CS4271_MODE1_MODE_4X, 64,   CS4271_MODE1_DIV_1},
+	{1, CS4271_MODE1_MODE_4X, 96,   CS4271_MODE1_DIV_15},
+	{1, CS4271_MODE1_MODE_4X, 128,  CS4271_MODE1_DIV_2},
+	{1, CS4271_MODE1_MODE_4X, 192,  CS4271_MODE1_DIV_3},
+	{0, CS4271_MODE1_MODE_1X, 256,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_1X, 384,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_1X, 512,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_1X, 768,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_2X, 128,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_2X, 192,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_2X, 256,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_2X, 384,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_2X, 512,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_4X, 64,   CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_4X, 96,   CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_4X, 128,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_4X, 192,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_4X, 256,  CS4271_MODE1_DIV_2},
+};
+
+#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
+
+static int cs4271_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 cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+	int i, ret;
+	unsigned int ratio, val;
+
+	cs4271->rate = params_rate(params);
+
+	/* Configure DAC */
+	if (cs4271->rate < 50000)
+		val = CS4271_MODE1_MODE_1X;
+	else if (cs4271->rate < 100000)
+		val = CS4271_MODE1_MODE_2X;
+	else
+		val = CS4271_MODE1_MODE_4X;
+
+	ratio = cs4271->mclk / cs4271->rate;
+	for (i = 0; i < CS4171_NR_RATIOS; i++)
+		if ((cs4271_clk_tab[i].master == cs4271->master) &&
+		    (cs4271_clk_tab[i].speed_mode == val) &&
+		    (cs4271_clk_tab[i].ratio == ratio))
+			break;
+
+	if (i == CS4171_NR_RATIOS) {
+		dev_err(codec->dev, "Invalid sample rate\n");
+		return -EINVAL;
+	}
+
+	val |= cs4271_clk_tab[i].ratio_mask;
+
+	ret = snd_soc_update_bits(codec, CS4271_MODE1,
+		CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);
+	if (ret < 0)
+		return ret;
+
+	return cs4271_set_deemph(codec);
+}
+
+static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int ret;
+	int val_a = 0;
+	int val_b = 0;
+
+	if (mute) {
+		val_a = CS4271_VOLA_MUTE;
+		val_b = CS4271_VOLB_MUTE;
+	}
+
+	ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a);
+	if (ret < 0)
+		return ret;
+	ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+/* CS4271 controls */
+static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0);
+
+static const struct snd_kcontrol_new cs4271_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB,
+		0, 0x7F, 1, cs4271_dac_tlv),
+	SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0),
+	SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0),
+	SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0),
+	SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0,
+		cs4271_get_deemph, cs4271_put_deemph),
+	SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0),
+	SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0),
+	SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0),
+	SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0),
+	SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0),
+	SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0),
+	SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1),
+	SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0),
+	SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1),
+	SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB,
+		7, 1, 1),
+};
+
+static struct snd_soc_dai_ops cs4271_dai_ops = {
+	.hw_params	= cs4271_hw_params,
+	.set_sysclk	= cs4271_set_dai_sysclk,
+	.set_fmt	= cs4271_set_dai_fmt,
+	.digital_mute	= cs4271_digital_mute,
+};
+
+static struct snd_soc_dai_driver cs4271_dai = {
+	.name = "cs4271-hifi",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= CS4271_PCM_RATES,
+		.formats	= CS4271_PCM_FORMATS,
+	},
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= CS4271_PCM_RATES,
+		.formats	= CS4271_PCM_FORMATS,
+	},
+	.ops = &cs4271_dai_ops,
+	.symmetric_rates = 1,
+};
+
+#ifdef CONFIG_PM
+static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+{
+	int ret;
+	/* Set power-down bit */
+	ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN);
+	if (ret < 0)
+		return ret;
+	return 0;
+}
+
+static int cs4271_soc_resume(struct snd_soc_codec *codec)
+{
+	int ret;
+	/* Restore codec state */
+	ret = snd_soc_cache_sync(codec);
+	if (ret < 0)
+		return ret;
+	/* then disable the power-down bit */
+	ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+	if (ret < 0)
+		return ret;
+	return 0;
+}
+#else
+#define cs4271_soc_suspend	NULL
+#define cs4271_soc_resume	NULL
+#endif /* CONFIG_PM */
+
+static int cs4271_probe(struct snd_soc_codec *codec)
+{
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+	struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
+	int ret;
+	int gpio_nreset = -EINVAL;
+
+	codec->control_data = cs4271->control_data;
+
+	if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset))
+		gpio_nreset = cs4271plat->gpio_nreset;
+
+	if (gpio_nreset >= 0)
+		if (gpio_request(gpio_nreset, "CS4271 Reset"))
+			gpio_nreset = -EINVAL;
+	if (gpio_nreset >= 0) {
+		/* Reset codec */
+		gpio_direction_output(gpio_nreset, 0);
+		udelay(1);
+		gpio_set_value(gpio_nreset, 1);
+		/* Give the codec time to wake up */
+		udelay(1);
+	}
+
+	cs4271->gpio_nreset = gpio_nreset;
+
+	/*
+	 * In case of I2C, chip address specified in board data.
+	 * So cache IO operations use 8 bit codec register address.
+	 * In case of SPI, chip address and register address
+	 * passed together as 16 bit value.
+	 * Anyway, register address is masked with 0xFF inside
+	 * soc-cache code.
+	 */
+	if (cs4271->bus_type == SND_SOC_SPI)
+		ret = snd_soc_codec_set_cache_io(codec, 16, 8,
+			cs4271->bus_type);
+	else
+		ret = snd_soc_codec_set_cache_io(codec, 8, 8,
+			cs4271->bus_type);
+	if (ret) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_update_bits(codec, CS4271_MODE2, 0,
+		CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+	if (ret < 0)
+		return ret;
+	ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+	if (ret < 0)
+		return ret;
+	/* Power-up sequence requires 85 uS */
+	udelay(85);
+
+	return snd_soc_add_controls(codec, cs4271_snd_controls,
+		ARRAY_SIZE(cs4271_snd_controls));
+}
+
+static int cs4271_remove(struct snd_soc_codec *codec)
+{
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+	int gpio_nreset;
+
+	gpio_nreset = cs4271->gpio_nreset;
+
+	if (gpio_is_valid(gpio_nreset)) {
+		/* Set codec to the reset state */
+		gpio_set_value(gpio_nreset, 0);
+		gpio_free(gpio_nreset);
+	}
+
+	return 0;
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
+	.probe			= cs4271_probe,
+	.remove			= cs4271_remove,
+	.suspend		= cs4271_soc_suspend,
+	.resume			= cs4271_soc_resume,
+	.reg_cache_default	= cs4271_dflt_reg,
+	.reg_cache_size		= ARRAY_SIZE(cs4271_dflt_reg),
+	.reg_word_size		= sizeof(cs4271_dflt_reg[0]),
+	.compress_type		= SND_SOC_FLAT_COMPRESSION,
+};
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit cs4271_spi_probe(struct spi_device *spi)
+{
+	struct cs4271_private *cs4271;
+
+	cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL);
+	if (!cs4271)
+		return -ENOMEM;
+
+	spi_set_drvdata(spi, cs4271);
+	cs4271->control_data = spi;
+	cs4271->bus_type = SND_SOC_SPI;
+
+	return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271,
+		&cs4271_dai, 1);
+}
+
+static int __devexit cs4271_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	return 0;
+}
+
+static struct spi_driver cs4271_spi_driver = {
+	.driver = {
+		.name	= "cs4271",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= cs4271_spi_probe,
+	.remove		= __devexit_p(cs4271_spi_remove),
+};
+#endif /* defined(CONFIG_SPI_MASTER) */
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct i2c_device_id cs4271_i2c_id[] = {
+	{"cs4271", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id);
+
+static int __devinit cs4271_i2c_probe(struct i2c_client *client,
+				      const struct i2c_device_id *id)
+{
+	struct cs4271_private *cs4271;
+
+	cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL);
+	if (!cs4271)
+		return -ENOMEM;
+
+	i2c_set_clientdata(client, cs4271);
+	cs4271->control_data = client;
+	cs4271->bus_type = SND_SOC_I2C;
+
+	return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271,
+		&cs4271_dai, 1);
+}
+
+static int __devexit cs4271_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static struct i2c_driver cs4271_i2c_driver = {
+	.driver = {
+		.name	= "cs4271",
+		.owner	= THIS_MODULE,
+	},
+	.id_table	= cs4271_i2c_id,
+	.probe		= cs4271_i2c_probe,
+	.remove		= __devexit_p(cs4271_i2c_remove),
+};
+#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */
+
+/*
+ * We only register our serial bus driver here without
+ * assignment to particular chip. So if any of the below
+ * fails, there is some problem with I2C or SPI subsystem.
+ * In most cases this module will be compiled with support
+ * of only one serial bus.
+ */
+static int __init cs4271_modinit(void)
+{
+	int ret;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&cs4271_i2c_driver);
+	if (ret) {
+		pr_err("Failed to register CS4271 I2C driver: %d\n", ret);
+		return ret;
+	}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&cs4271_spi_driver);
+	if (ret) {
+		pr_err("Failed to register CS4271 SPI driver: %d\n", ret);
+		return ret;
+	}
+#endif
+
+	return 0;
+}
+module_init(cs4271_modinit);
+
+static void __exit cs4271_modexit(void)
+{
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&cs4271_spi_driver);
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&cs4271_i2c_driver);
+#endif
+}
+module_exit(cs4271_modexit);
+
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e860d22..0bb424af956 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static const u8 cx20442_reg;
+
 static struct snd_soc_codec_driver cx20442_codec_dev = {
 	.probe = 	cx20442_codec_probe,
 	.remove = 	cx20442_codec_remove,
+	.reg_cache_default = &cx20442_reg,
 	.reg_cache_size = 1,
 	.reg_word_size = sizeof(u8),
 	.read = cx20442_read_reg_cache,
diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c
new file mode 100644
index 00000000000..704bbde6573
--- /dev/null
+++ b/sound/soc/codecs/dfbmcs320.c
@@ -0,0 +1,72 @@
+/*
+ * Driver for the DFBM-CS320 bluetooth module
+ * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de>
+ *
+ *  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/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver dfbmcs320_dai = {
+	.name = "dfbmcs320-pcm",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320;
+
+static int __devinit dfbmcs320_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320,
+			&dfbmcs320_dai, 1);
+}
+
+static int __devexit dfbmcs320_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver dfmcs320_driver = {
+	.driver = {
+		.name = "dfbmcs320",
+		.owner = THIS_MODULE,
+	},
+	.probe = dfbmcs320_probe,
+	.remove = __devexit_p(dfbmcs320_remove),
+};
+
+static int __init dfbmcs320_init(void)
+{
+	return platform_driver_register(&dfmcs320_driver);
+}
+module_init(dfbmcs320_init);
+
+static void __exit dfbmcs320_exit(void)
+{
+	platform_driver_unregister(&dfmcs320_driver);
+}
+module_exit(dfbmcs320_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
new file mode 100644
index 00000000000..72de47e5d04
--- /dev/null
+++ b/sound/soc/codecs/lm4857.c
@@ -0,0 +1,276 @@
+/*
+ * LM4857 AMP driver
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de>
+ *
+ *  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/init.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+struct lm4857 {
+	struct i2c_client *i2c;
+	uint8_t mode;
+};
+
+static const uint8_t lm4857_default_regs[] = {
+	0x00, 0x00, 0x00, 0x00,
+};
+
+/* The register offsets in the cache array */
+#define LM4857_MVOL 0
+#define LM4857_LVOL 1
+#define LM4857_RVOL 2
+#define LM4857_CTRL 3
+
+/* the shifts required to set these bits */
+#define LM4857_3D 5
+#define LM4857_WAKEUP 5
+#define LM4857_EPGAIN 4
+
+static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg,
+		unsigned int value)
+{
+	uint8_t data;
+	int ret;
+
+	ret = snd_soc_cache_write(codec, reg, value);
+	if (ret < 0)
+		return ret;
+
+	data = (reg << 6) | value;
+	ret = i2c_master_send(codec->control_data, &data, 1);
+	if (ret != 1) {
+		dev_err(codec->dev, "Failed to write register: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static unsigned int lm4857_read(struct snd_soc_codec *codec,
+		unsigned int reg)
+{
+	unsigned int val;
+	int ret;
+
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret)
+		return -1;
+
+	return val;
+}
+
+static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = lm4857->mode;
+
+	return 0;
+}
+
+static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+	uint8_t value = ucontrol->value.integer.value[0];
+
+	lm4857->mode = value;
+
+	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6);
+
+	return 1;
+}
+
+static int lm4857_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0);
+		break;
+	default:
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static const char *lm4857_mode[] = {
+	"Earpiece",
+	"Loudspeaker",
+	"Loudspeaker + Headphone",
+	"Headphone",
+};
+
+static const struct soc_enum lm4857_mode_enum =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode);
+
+static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("IN"),
+
+	SND_SOC_DAPM_OUTPUT("LS"),
+	SND_SOC_DAPM_OUTPUT("HP"),
+	SND_SOC_DAPM_OUTPUT("EP"),
+};
+
+static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
+
+static const struct snd_kcontrol_new lm4857_controls[] = {
+	SOC_SINGLE_TLV("Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+		stereo_tlv),
+	SOC_SINGLE_TLV("Right Playback Volume", LM4857_RVOL, 0, 31, 0,
+		stereo_tlv),
+	SOC_SINGLE_TLV("Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
+		mono_tlv),
+	SOC_SINGLE("Spk 3D Playback Switch", LM4857_LVOL, LM4857_3D, 1, 0),
+	SOC_SINGLE("HP 3D Playback Switch", LM4857_RVOL, LM4857_3D, 1, 0),
+	SOC_SINGLE("Fast Wakeup Playback Switch", LM4857_CTRL,
+		LM4857_WAKEUP, 1, 0),
+	SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL,
+		LM4857_EPGAIN, 1, 0),
+
+	SOC_ENUM_EXT("Mode", lm4857_mode_enum,
+		lm4857_get_mode, lm4857_set_mode),
+};
+
+/* There is a demux inbetween the the input signal and the output signals.
+ * Currently there is no easy way to model it in ASoC and since it does not make
+ * much of a difference in practice simply connect the input direclty to the
+ * outputs. */
+static const struct snd_soc_dapm_route lm4857_routes[] = {
+	{"LS", NULL, "IN"},
+	{"HP", NULL, "IN"},
+	{"EP", NULL, "IN"},
+};
+
+static int lm4857_probe(struct snd_soc_codec *codec)
+{
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	codec->control_data = lm4857->i2c;
+
+	ret = snd_soc_add_controls(codec, lm4857_controls,
+			ARRAY_SIZE(lm4857_controls));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets,
+			ARRAY_SIZE(lm4857_dapm_widgets));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_add_routes(dapm, lm4857_routes,
+			ARRAY_SIZE(lm4857_routes));
+	if (ret)
+		return ret;
+
+	snd_soc_dapm_new_widgets(dapm);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
+	.write = lm4857_write,
+	.read = lm4857_read,
+	.probe = lm4857_probe,
+	.reg_cache_size = ARRAY_SIZE(lm4857_default_regs),
+	.reg_word_size = sizeof(uint8_t),
+	.reg_cache_default = lm4857_default_regs,
+	.set_bias_level = lm4857_set_bias_level,
+};
+
+static int __devinit lm4857_i2c_probe(struct i2c_client *i2c,
+	const struct i2c_device_id *id)
+{
+	struct lm4857 *lm4857;
+	int ret;
+
+	lm4857 = kzalloc(sizeof(*lm4857), GFP_KERNEL);
+	if (!lm4857)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, lm4857);
+
+	lm4857->i2c = i2c;
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
+
+	if (ret) {
+		kfree(lm4857);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit lm4857_i2c_remove(struct i2c_client *i2c)
+{
+	struct lm4857 *lm4857 = i2c_get_clientdata(i2c);
+
+	snd_soc_unregister_codec(&i2c->dev);
+	kfree(lm4857);
+
+	return 0;
+}
+
+static const struct i2c_device_id lm4857_i2c_id[] = {
+	{ "lm4857", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id);
+
+static struct i2c_driver lm4857_i2c_driver = {
+	.driver = {
+		.name = "lm4857",
+		.owner = THIS_MODULE,
+	},
+	.probe = lm4857_i2c_probe,
+	.remove = __devexit_p(lm4857_i2c_remove),
+	.id_table = lm4857_i2c_id,
+};
+
+static int __init lm4857_init(void)
+{
+	return i2c_add_driver(&lm4857_i2c_driver);
+}
+module_init(lm4857_init);
+
+static void __exit lm4857_exit(void)
+{
+	i2c_del_driver(&lm4857_i2c_driver);
+}
+module_exit(lm4857_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("LM4857 amplifier driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 89498f9ad2e..bd0517cb798 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -608,7 +608,7 @@ static struct {
        { 0xFF, 0x00, 1 }, /* FF */
 };
 
-static int max98088_volatile_register(unsigned int reg)
+static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
        return max98088_access[reg].vol;
 }
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
new file mode 100644
index 00000000000..208d2ee6185
--- /dev/null
+++ b/sound/soc/codecs/max9850.c
@@ -0,0 +1,389 @@
+/*
+ * max9850.c  --  codec driver for max9850
+ *
+ * Copyright (C) 2011 taskit GmbH
+ *
+ * Author: Christian Glindkamp <christian.glindkamp@taskit.de>
+ *
+ * Initial development of this code was funded by
+ * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/
+ *
+ * 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/init.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "max9850.h"
+
+struct max9850_priv {
+	unsigned int sysclk;
+};
+
+/* max9850 register cache */
+static const u8 max9850_reg[MAX9850_CACHEREGNUM] = {
+	0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+};
+
+/* these registers are not used at the moment but provided for the sake of
+ * completeness */
+static int max9850_volatile_register(struct snd_soc_codec *codec,
+		unsigned int reg)
+{
+	switch (reg) {
+	case MAX9850_STATUSA:
+	case MAX9850_STATUSB:
+		return 1;
+	default:
+		return 0;
+	}
+}
+
+static const unsigned int max9850_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0),
+	0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0),
+	0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0),
+	0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0),
+};
+
+static const struct snd_kcontrol_new max9850_controls[] = {
+SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv),
+SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1),
+SOC_SINGLE("Mono Switch", MAX9850_GENERAL_PURPOSE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new max9850_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("Charge Pump 1", MAX9850_ENABLE, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("Charge Pump 2", MAX9850_ENABLE, 5, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("SHDN", MAX9850_ENABLE, 7, 0, NULL, 0),
+SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0,
+		&max9850_mixer_controls[0],
+		ARRAY_SIZE(max9850_mixer_controls)),
+SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0),
+SND_SOC_DAPM_OUTPUT("OUTL"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("OUTR"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_INPUT("INL"),
+SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* output mixer */
+	{"Output Mixer", NULL, "DAC"},
+	{"Output Mixer", "Line In Switch", "Line Input"},
+
+	/* outputs */
+	{"Headphone Output", NULL, "Output Mixer"},
+	{"HPL", NULL, "Headphone Output"},
+	{"HPR", NULL, "Headphone Output"},
+	{"OUTL", NULL, "Output Mixer"},
+	{"OUTR", NULL, "Output Mixer"},
+
+	/* inputs */
+	{"Line Input", NULL, "INL"},
+	{"Line Input", NULL, "INR"},
+
+	/* supplies */
+	{"Output Mixer", NULL, "Charge Pump 1"},
+	{"Output Mixer", NULL, "Charge Pump 2"},
+	{"Output Mixer", NULL, "SHDN"},
+	{"DAC", NULL, "MCLK"},
+};
+
+static int max9850_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
+	u64 lrclk_div;
+	u8 sf, da;
+
+	if (!max9850->sysclk)
+		return -EINVAL;
+
+	/* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */
+	sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1;
+	lrclk_div = (1 << 22);
+	lrclk_div *= params_rate(params);
+	lrclk_div *= sf;
+	do_div(lrclk_div, max9850->sysclk);
+
+	snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f);
+	snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		da = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		da = 0x2;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		da = 0x3;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, MAX9850_DIGITAL_AUDIO, 0x3, da);
+
+	return 0;
+}
+
+static int max9850_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 max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
+
+	/* calculate mclk -> iclk divider */
+	if (freq <= 13000000)
+		snd_soc_write(codec, MAX9850_CLOCK, 0x0);
+	else if (freq <= 26000000)
+		snd_soc_write(codec, MAX9850_CLOCK, 0x4);
+	else if (freq <= 40000000)
+		snd_soc_write(codec, MAX9850_CLOCK, 0x8);
+	else
+		return -EINVAL;
+
+	max9850->sysclk = freq;
+	return 0;
+}
+
+static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 da = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		da |= MAX9850_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		da |= MAX9850_DLY;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		da |= MAX9850_RTJ;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		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:
+		da |= MAX9850_BCINV | MAX9850_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		da |= MAX9850_BCINV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		da |= MAX9850_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set da */
+	snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da);
+
+	return 0;
+}
+
+static int max9850_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = snd_soc_cache_sync(codec);
+			if (ret) {
+				dev_err(codec->dev,
+					"Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000
+
+#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops max9850_dai_ops = {
+	.hw_params	= max9850_hw_params,
+	.set_sysclk	= max9850_set_dai_sysclk,
+	.set_fmt	= max9850_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver max9850_dai = {
+	.name = "max9850-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = MAX9850_RATES,
+		.formats = MAX9850_FORMATS
+	},
+	.ops = &max9850_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	max9850_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int max9850_resume(struct snd_soc_codec *codec)
+{
+	max9850_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+#else
+#define max9850_suspend NULL
+#define max9850_resume NULL
+#endif
+
+static int max9850_probe(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	/* enable zero-detect */
+	snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1);
+	/* enable slew-rate control */
+	snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40);
+	/* set slew-rate 125ms */
+	snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0);
+
+	snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets,
+				  ARRAY_SIZE(max9850_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_add_controls(codec, max9850_controls,
+			ARRAY_SIZE(max9850_controls));
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_max9850 = {
+	.probe =	max9850_probe,
+	.suspend =	max9850_suspend,
+	.resume =	max9850_resume,
+	.set_bias_level = max9850_set_bias_level,
+	.reg_cache_size = ARRAY_SIZE(max9850_reg),
+	.reg_word_size = sizeof(u8),
+	.reg_cache_default = max9850_reg,
+	.volatile_register = max9850_volatile_register,
+};
+
+static int __devinit max9850_i2c_probe(struct i2c_client *i2c,
+		const struct i2c_device_id *id)
+{
+	struct max9850_priv *max9850;
+	int ret;
+
+	max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL);
+	if (max9850 == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, max9850);
+
+	ret = snd_soc_register_codec(&i2c->dev,
+			&soc_codec_dev_max9850, &max9850_dai, 1);
+	if (ret < 0)
+		kfree(max9850);
+	return ret;
+}
+
+static __devexit int max9850_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(i2c_get_clientdata(client));
+	return 0;
+}
+
+static const struct i2c_device_id max9850_i2c_id[] = {
+	{ "max9850", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, max9850_i2c_id);
+
+static struct i2c_driver max9850_i2c_driver = {
+	.driver = {
+		.name = "max9850",
+		.owner = THIS_MODULE,
+	},
+	.probe = max9850_i2c_probe,
+	.remove = __devexit_p(max9850_i2c_remove),
+	.id_table = max9850_i2c_id,
+};
+
+static int __init max9850_init(void)
+{
+	return i2c_add_driver(&max9850_i2c_driver);
+}
+module_init(max9850_init);
+
+static void __exit max9850_exit(void)
+{
+	i2c_del_driver(&max9850_i2c_driver);
+}
+module_exit(max9850_exit);
+
+MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp@taskit.de>");
+MODULE_DESCRIPTION("ASoC MAX9850 codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h
new file mode 100644
index 00000000000..72b1ddb04b0
--- /dev/null
+++ b/sound/soc/codecs/max9850.h
@@ -0,0 +1,38 @@
+/*
+ * max9850.h  --  codec driver for max9850
+ *
+ * Copyright (C) 2011 taskit GmbH
+ * Author: Christian Glindkamp <christian.glindkamp@taskit.de>
+ *
+ * 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.
+ *
+ */
+
+#ifndef _MAX9850_H
+#define _MAX9850_H
+
+#define MAX9850_STATUSA			0x00
+#define MAX9850_STATUSB			0x01
+#define MAX9850_VOLUME			0x02
+#define MAX9850_GENERAL_PURPOSE		0x03
+#define MAX9850_INTERRUPT		0x04
+#define MAX9850_ENABLE			0x05
+#define MAX9850_CLOCK			0x06
+#define MAX9850_CHARGE_PUMP		0x07
+#define MAX9850_LRCLK_MSB		0x08
+#define MAX9850_LRCLK_LSB		0x09
+#define MAX9850_DIGITAL_AUDIO		0x0a
+
+#define MAX9850_CACHEREGNUM 11
+
+/* MAX9850_DIGITAL_AUDIO */
+#define MAX9850_MASTER			(1<<7)
+#define MAX9850_INV			(1<<6)
+#define MAX9850_BCINV			(1<<5)
+#define MAX9850_DLY			(1<<3)
+#define MAX9850_RTJ			(1<<2)
+
+#endif
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
new file mode 100644
index 00000000000..ff29380c9ed
--- /dev/null
+++ b/sound/soc/codecs/sgtl5000.c
@@ -0,0 +1,1527 @@
+/*
+ * sgtl5000.c  --  SGTL5000 ALSA SoC Audio driver
+ *
+ * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * 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/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/driver.h>
+#include <linux/regulator/machine.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "sgtl5000.h"
+
+#define SGTL5000_DAP_REG_OFFSET	0x0100
+#define SGTL5000_MAX_REG_OFFSET	0x013A
+
+/* default value of sgtl5000 registers except DAP */
+static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] =  {
+	0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */
+	0x0000, /* 0x0002, CHIP_DIG_POWER. */
+	0x0008, /* 0x0004, CHIP_CKL_CTRL */
+	0x0010, /* 0x0006, CHIP_I2S_CTRL */
+	0x0000, /* 0x0008, reserved */
+	0x0008, /* 0x000A, CHIP_SSS_CTRL */
+	0x0000, /* 0x000C, reserved */
+	0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */
+	0x3c3c, /* 0x0010, CHIP_DAC_VOL */
+	0x0000, /* 0x0012, reserved */
+	0x015f, /* 0x0014, CHIP_PAD_STRENGTH */
+	0x0000, /* 0x0016, reserved */
+	0x0000, /* 0x0018, reserved */
+	0x0000, /* 0x001A, reserved */
+	0x0000, /* 0x001E, reserved */
+	0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */
+	0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */
+	0x0111, /* 0x0024, CHIP_ANN_CTRL */
+	0x0000, /* 0x0026, CHIP_LINREG_CTRL */
+	0x0000, /* 0x0028, CHIP_REF_CTRL */
+	0x0000, /* 0x002A, CHIP_MIC_CTRL */
+	0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */
+	0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */
+	0x7060, /* 0x0030, CHIP_ANA_POWER */
+	0x5000, /* 0x0032, CHIP_PLL_CTRL */
+	0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */
+	0x0000, /* 0x0036, CHIP_ANA_STATUS */
+	0x0000, /* 0x0038, reserved */
+	0x0000, /* 0x003A, CHIP_ANA_TEST2 */
+	0x0000, /* 0x003C, CHIP_SHORT_CTRL */
+	0x0000, /* reserved */
+};
+
+/* default value of dap registers */
+static const u16 sgtl5000_dap_regs[] = {
+	0x0000, /* 0x0100, DAP_CONTROL */
+	0x0000, /* 0x0102, DAP_PEQ */
+	0x0040, /* 0x0104, DAP_BASS_ENHANCE */
+	0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */
+	0x0000, /* 0x0108, DAP_AUDIO_EQ */
+	0x0040, /* 0x010A, DAP_SGTL_SURROUND */
+	0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */
+	0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */
+	0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */
+	0x0000, /* 0x0112, reserved */
+	0x0000, /* 0x0114, reserved */
+	0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */
+	0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */
+	0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */
+	0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */
+	0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */
+	0x8000, /* 0x0120, DAP_MAIN_CHAN */
+	0x0000, /* 0x0122, DAP_MIX_CHAN */
+	0x0510, /* 0x0124, DAP_AVC_CTRL */
+	0x1473, /* 0x0126, DAP_AVC_THRESHOLD */
+	0x0028, /* 0x0128, DAP_AVC_ATTACK */
+	0x0050, /* 0x012A, DAP_AVC_DECAY */
+	0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */
+	0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */
+	0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */
+	0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */
+	0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */
+	0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */
+	0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */
+	0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */
+};
+
+/* regulator supplies for sgtl5000, VDDD is an optional external supply */
+enum sgtl5000_regulator_supplies {
+	VDDA,
+	VDDIO,
+	VDDD,
+	SGTL5000_SUPPLY_NUM
+};
+
+/* vddd is optional supply */
+static const char *supply_names[SGTL5000_SUPPLY_NUM] = {
+	"VDDA",
+	"VDDIO",
+	"VDDD"
+};
+
+#define LDO_CONSUMER_NAME	"VDDD_LDO"
+#define LDO_VOLTAGE		1200000
+
+static struct regulator_consumer_supply ldo_consumer[] = {
+	REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL),
+};
+
+static struct regulator_init_data ldo_init_data = {
+	.constraints = {
+		.min_uV                 = 850000,
+		.max_uV                 = 1600000,
+		.valid_modes_mask       = REGULATOR_MODE_NORMAL,
+		.valid_ops_mask         = REGULATOR_CHANGE_STATUS,
+	},
+	.num_consumer_supplies = 1,
+	.consumer_supplies = &ldo_consumer[0],
+};
+
+/*
+ * sgtl5000 internal ldo regulator,
+ * enabled when VDDD not provided
+ */
+struct ldo_regulator {
+	struct regulator_desc desc;
+	struct regulator_dev *dev;
+	int voltage;
+	void *codec_data;
+	bool enabled;
+};
+
+/* sgtl5000 private structure in codec */
+struct sgtl5000_priv {
+	int sysclk;	/* sysclk rate */
+	int master;	/* i2s master or not */
+	int fmt;	/* i2s data format */
+	struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM];
+	struct ldo_regulator *ldo;
+};
+
+/*
+ * mic_bias power on/off share the same register bits with
+ * output impedance of mic bias, when power on mic bias, we
+ * need reclaim it to impedance value.
+ * 0x0 = Powered off
+ * 0x1 = 2Kohm
+ * 0x2 = 4Kohm
+ * 0x3 = 8Kohm
+ */
+static int mic_bias_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* change mic bias resistor to 4Kohm */
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+				SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		/*
+		 * SGTL5000_BIAS_R_8k as mask to clean the two bits
+		 * of mic bias and output impedance
+		 */
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+				SGTL5000_BIAS_R_8k, 0);
+		break;
+	}
+	return 0;
+}
+
+/*
+ * using codec assist to small pop, hp_powerup or lineout_powerup
+ * should stay setting until vag_powerup is fully ramped down,
+ * vag fully ramped down require 400ms.
+ */
+static int small_pop_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_VAG_POWERUP, 0);
+		msleep(400);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+/* input sources for ADC */
+static const char *adc_mux_text[] = {
+	"MIC_IN", "LINE_IN"
+};
+
+static const struct soc_enum adc_enum =
+SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text);
+
+static const struct snd_kcontrol_new adc_mux =
+SOC_DAPM_ENUM("Capture Mux", adc_enum);
+
+/* input sources for DAC */
+static const char *dac_mux_text[] = {
+	"DAC", "LINE_IN"
+};
+
+static const struct soc_enum dac_enum =
+SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text);
+
+static const struct snd_kcontrol_new dac_mux =
+SOC_DAPM_ENUM("Headphone Mux", dac_enum);
+
+static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("LINE_IN"),
+	SND_SOC_DAPM_INPUT("MIC_IN"),
+
+	SND_SOC_DAPM_OUTPUT("HP_OUT"),
+	SND_SOC_DAPM_OUTPUT("LINE_OUT"),
+
+	SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
+				mic_bias_event,
+				SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
+			small_pop_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0,
+			small_pop_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
+	SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux),
+
+	/* aif for i2s input */
+	SND_SOC_DAPM_AIF_IN("AIFIN", "Playback",
+				0, SGTL5000_CHIP_DIG_POWER,
+				0, 0),
+
+	/* aif for i2s output */
+	SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture",
+				0, SGTL5000_CHIP_DIG_POWER,
+				1, 0),
+
+	SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
+
+	SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
+};
+
+/* routes for sgtl5000 */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Capture Mux", "LINE_IN", "LINE_IN"},	/* line_in --> adc_mux */
+	{"Capture Mux", "MIC_IN", "MIC_IN"},	/* mic_in --> adc_mux */
+
+	{"ADC", NULL, "Capture Mux"},		/* adc_mux --> adc */
+	{"AIFOUT", NULL, "ADC"},		/* adc --> i2s_out */
+
+	{"DAC", NULL, "AIFIN"},			/* i2s-->dac,skip audio mux */
+	{"Headphone Mux", "DAC", "DAC"},	/* dac --> hp_mux */
+	{"LO", NULL, "DAC"},			/* dac --> line_out */
+
+	{"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */
+	{"HP", NULL, "Headphone Mux"},		/* hp_mux --> hp */
+
+	{"LINE_OUT", NULL, "LO"},
+	{"HP_OUT", NULL, "HP"},
+};
+
+/* custom function to fetch info of PCM playback volume */
+static int dac_info_volsw(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 0xfc - 0x3c;
+	return 0;
+}
+
+/*
+ * custom function to get of PCM playback volume
+ *
+ * dac volume register
+ * 15-------------8-7--------------0
+ * | R channel vol | L channel vol |
+ *  -------------------------------
+ *
+ * PCM volume with 0.5017 dB steps from 0 to -90 dB
+ *
+ * register values map to dB
+ * 0x3B and less = Reserved
+ * 0x3C = 0 dB
+ * 0x3D = -0.5 dB
+ * 0xF0 = -90 dB
+ * 0xFC and greater = Muted
+ *
+ * register value map to userspace value
+ *
+ * register value	0x3c(0dB)	  0xf0(-90dB)0xfc
+ *			------------------------------
+ * userspace value	0xc0			     0
+ */
+static int dac_get_volsw(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg;
+	int l;
+	int r;
+
+	reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL);
+
+	/* get left channel volume */
+	l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT;
+
+	/* get right channel volume */
+	r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT;
+
+	/* make sure value fall in (0x3c,0xfc) */
+	l = clamp(l, 0x3c, 0xfc);
+	r = clamp(r, 0x3c, 0xfc);
+
+	/* invert it and map to userspace value */
+	l = 0xfc - l;
+	r = 0xfc - r;
+
+	ucontrol->value.integer.value[0] = l;
+	ucontrol->value.integer.value[1] = r;
+
+	return 0;
+}
+
+/*
+ * custom function to put of PCM playback volume
+ *
+ * dac volume register
+ * 15-------------8-7--------------0
+ * | R channel vol | L channel vol |
+ *  -------------------------------
+ *
+ * PCM volume with 0.5017 dB steps from 0 to -90 dB
+ *
+ * register values map to dB
+ * 0x3B and less = Reserved
+ * 0x3C = 0 dB
+ * 0x3D = -0.5 dB
+ * 0xF0 = -90 dB
+ * 0xFC and greater = Muted
+ *
+ * userspace value map to register value
+ *
+ * userspace value	0xc0			     0
+ *			------------------------------
+ * register value	0x3c(0dB)	0xf0(-90dB)0xfc
+ */
+static int dac_put_volsw(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg;
+	int l;
+	int r;
+
+	l = ucontrol->value.integer.value[0];
+	r = ucontrol->value.integer.value[1];
+
+	/* make sure userspace volume fall in (0, 0xfc-0x3c) */
+	l = clamp(l, 0, 0xfc - 0x3c);
+	r = clamp(r, 0, 0xfc - 0x3c);
+
+	/* invert it, get the value can be set to register */
+	l = 0xfc - l;
+	r = 0xfc - r;
+
+	/* shift to get the register value */
+	reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT |
+		r << SGTL5000_DAC_VOL_RIGHT_SHIFT;
+
+	snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg);
+
+	return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
+
+/* tlv for mic gain, 0db 20db 30db 40db */
+static const unsigned int mic_gain_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
+};
+
+/* tlv for hp volume, -51.5db to 12.0db, step .5db */
+static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0);
+
+static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
+	/* SOC_DOUBLE_S8_TLV with invert */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "PCM Playback Volume",
+		.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+			SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info = dac_info_volsw,
+		.get = dac_get_volsw,
+		.put = dac_put_volsw,
+	},
+
+	SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),
+	SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",
+			SGTL5000_CHIP_ANA_ADC_CTRL,
+			8, 2, 0, capture_6db_attenuate),
+	SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
+
+	SOC_DOUBLE_TLV("Headphone Playback Volume",
+			SGTL5000_CHIP_ANA_HP_CTRL,
+			0, 8,
+			0x7f, 1,
+			headphone_volume),
+	SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL,
+			5, 1, 0),
+
+	SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
+			0, 4, 0, mic_gain_tlv),
+};
+
+/* mute the codec used by alsa core */
+static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL,
+			adcdac_ctrl, mute ? adcdac_ctrl : 0);
+
+	return 0;
+}
+
+/* set codec format */
+static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	u16 i2sctl = 0;
+
+	sgtl5000->master = 0;
+	/*
+	 * i2s clock and frame master setting.
+	 * ONLY support:
+	 *  - clock and frame slave,
+	 *  - clock and frame master
+	 */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		i2sctl |= SGTL5000_I2S_MASTER;
+		sgtl5000->master = 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* setting i2s data format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		i2sctl |= SGTL5000_I2S_MODE_PCM;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		i2sctl |= SGTL5000_I2S_MODE_PCM;
+		i2sctl |= SGTL5000_I2S_LRALIGN;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		i2sctl |= SGTL5000_I2S_MODE_RJ;
+		i2sctl |= SGTL5000_I2S_LRPOL;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+		i2sctl |= SGTL5000_I2S_LRALIGN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+	/* Clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		i2sctl |= SGTL5000_I2S_SCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl);
+
+	return 0;
+}
+
+/* set codec sysclk */
+static int sgtl5000_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 sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	switch (clk_id) {
+	case SGTL5000_SYSCLK:
+		sgtl5000->sysclk = freq;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * set clock according to i2s frame clock,
+ * sgtl5000 provide 2 clock sources.
+ * 1. sys_mclk. sample freq can only configure to
+ *	1/256, 1/384, 1/512 of sys_mclk.
+ * 2. pll. can derive any audio clocks.
+ *
+ * clock setting rules:
+ * 1. in slave mode, only sys_mclk can use.
+ * 2. as constraint by sys_mclk, sample freq should
+ *	set to 32k, 44.1k and above.
+ * 3. using sys_mclk prefer to pll to save power.
+ */
+static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
+{
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	int clk_ctl = 0;
+	int sys_fs;	/* sample freq */
+
+	/*
+	 * sample freq should be divided by frame clock,
+	 * if frame clock lower than 44.1khz, sample feq should set to
+	 * 32khz or 44.1khz.
+	 */
+	switch (frame_rate) {
+	case 8000:
+	case 16000:
+		sys_fs = 32000;
+		break;
+	case 11025:
+	case 22050:
+		sys_fs = 44100;
+		break;
+	default:
+		sys_fs = frame_rate;
+		break;
+	}
+
+	/* set divided factor of frame clock */
+	switch (sys_fs / frame_rate) {
+	case 4:
+		clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT;
+		break;
+	case 2:
+		clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT;
+		break;
+	case 1:
+		clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set the sys_fs according to frame rate */
+	switch (sys_fs) {
+	case 32000:
+		clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	case 44100:
+		clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	case 48000:
+		clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	case 96000:
+		clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	default:
+		dev_err(codec->dev, "frame rate %d not supported\n",
+			frame_rate);
+		return -EINVAL;
+	}
+
+	/*
+	 * calculate the divider of mclk/sample_freq,
+	 * factor of freq =96k can only be 256, since mclk in range (12m,27m)
+	 */
+	switch (sgtl5000->sysclk / sys_fs) {
+	case 256:
+		clk_ctl |= SGTL5000_MCLK_FREQ_256FS <<
+			SGTL5000_MCLK_FREQ_SHIFT;
+		break;
+	case 384:
+		clk_ctl |= SGTL5000_MCLK_FREQ_384FS <<
+			SGTL5000_MCLK_FREQ_SHIFT;
+		break;
+	case 512:
+		clk_ctl |= SGTL5000_MCLK_FREQ_512FS <<
+			SGTL5000_MCLK_FREQ_SHIFT;
+		break;
+	default:
+		/* if mclk not satisify the divider, use pll */
+		if (sgtl5000->master) {
+			clk_ctl |= SGTL5000_MCLK_FREQ_PLL <<
+				SGTL5000_MCLK_FREQ_SHIFT;
+		} else {
+			dev_err(codec->dev,
+				"PLL not supported in slave mode\n");
+			return -EINVAL;
+		}
+	}
+
+	/* if using pll, please check manual 6.4.2 for detail */
+	if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) {
+		u64 out, t;
+		int div2;
+		int pll_ctl;
+		unsigned int in, int_div, frac_div;
+
+		if (sgtl5000->sysclk > 17000000) {
+			div2 = 1;
+			in = sgtl5000->sysclk / 2;
+		} else {
+			div2 = 0;
+			in = sgtl5000->sysclk;
+		}
+		if (sys_fs == 44100)
+			out = 180633600;
+		else
+			out = 196608000;
+		t = do_div(out, in);
+		int_div = out;
+		t *= 2048;
+		do_div(t, in);
+		frac_div = t;
+		pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT |
+		    frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT;
+
+		snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl);
+		if (div2)
+			snd_soc_update_bits(codec,
+				SGTL5000_CHIP_CLK_TOP_CTRL,
+				SGTL5000_INPUT_FREQ_DIV2,
+				SGTL5000_INPUT_FREQ_DIV2);
+		else
+			snd_soc_update_bits(codec,
+				SGTL5000_CHIP_CLK_TOP_CTRL,
+				SGTL5000_INPUT_FREQ_DIV2,
+				0);
+
+		/* power up pll */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
+			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP);
+	} else {
+		/* power down pll */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
+			0);
+	}
+
+	/* if using pll, clk_ctrl must be set after pll power up */
+	snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
+
+	return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ * input: params_rate, params_fmt
+ */
+static int sgtl5000_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 sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	int channels = params_channels(params);
+	int i2s_ctl = 0;
+	int stereo;
+	int ret;
+
+	/* sysclk should already set */
+	if (!sgtl5000->sysclk) {
+		dev_err(codec->dev, "%s: set sysclk first!\n", __func__);
+		return -EFAULT;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		stereo = SGTL5000_DAC_STEREO;
+	else
+		stereo = SGTL5000_ADC_STEREO;
+
+	/* set mono to save power */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo,
+			channels == 1 ? 0 : stereo);
+
+	/* set codec clock base on lrclk */
+	ret = sgtl5000_set_clock(codec, params_rate(params));
+	if (ret)
+		return ret;
+
+	/* set i2s data format */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
+			return -EINVAL;
+		i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
+			return -EINVAL;
+		i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl);
+
+	return 0;
+}
+
+#ifdef CONFIG_REGULATOR
+static int ldo_regulator_is_enabled(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+
+	return ldo->enabled;
+}
+
+static int ldo_regulator_enable(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+	struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data;
+	int reg;
+
+	if (ldo_regulator_is_enabled(dev))
+		return 0;
+
+	/* set regulator value firstly */
+	reg = (1600 - ldo->voltage / 1000) / 50;
+	reg = clamp(reg, 0x0, 0xf);
+
+	/* amend the voltage value, unit: uV */
+	ldo->voltage = (1600 - reg * 50) * 1000;
+
+	/* set voltage to register */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+				(0x1 << 4) - 1, reg);
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINEREG_D_POWERUP,
+				SGTL5000_LINEREG_D_POWERUP);
+
+	/* when internal ldo enabled, simple digital power can be disabled */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINREG_SIMPLE_POWERUP,
+				0);
+
+	ldo->enabled = 1;
+	return 0;
+}
+
+static int ldo_regulator_disable(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+	struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINEREG_D_POWERUP,
+				0);
+
+	/* clear voltage info */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+				(0x1 << 4) - 1, 0);
+
+	ldo->enabled = 0;
+
+	return 0;
+}
+
+static int ldo_regulator_get_voltage(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+
+	return ldo->voltage;
+}
+
+static struct regulator_ops ldo_regulator_ops = {
+	.is_enabled = ldo_regulator_is_enabled,
+	.enable = ldo_regulator_enable,
+	.disable = ldo_regulator_disable,
+	.get_voltage = ldo_regulator_get_voltage,
+};
+
+static int ldo_regulator_register(struct snd_soc_codec *codec,
+				struct regulator_init_data *init_data,
+				int voltage)
+{
+	struct ldo_regulator *ldo;
+
+	ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
+
+	if (!ldo) {
+		dev_err(codec->dev, "failed to allocate ldo_regulator\n");
+		return -ENOMEM;
+	}
+
+	ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL);
+	if (!ldo->desc.name) {
+		kfree(ldo);
+		dev_err(codec->dev, "failed to allocate decs name memory\n");
+		return -ENOMEM;
+	}
+
+	ldo->desc.type  = REGULATOR_VOLTAGE;
+	ldo->desc.owner = THIS_MODULE;
+	ldo->desc.ops   = &ldo_regulator_ops;
+	ldo->desc.n_voltages = 1;
+
+	ldo->codec_data = codec;
+	ldo->voltage = voltage;
+
+	ldo->dev = regulator_register(&ldo->desc, codec->dev,
+					  init_data, ldo);
+	if (IS_ERR(ldo->dev)) {
+		int ret = PTR_ERR(ldo->dev);
+
+		dev_err(codec->dev, "failed to register regulator\n");
+		kfree(ldo->desc.name);
+		kfree(ldo);
+
+		return ret;
+	}
+
+	return 0;
+}
+
+static int ldo_regulator_remove(struct snd_soc_codec *codec)
+{
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	struct ldo_regulator *ldo = sgtl5000->ldo;
+
+	if (!ldo)
+		return 0;
+
+	regulator_unregister(ldo->dev);
+	kfree(ldo->desc.name);
+	kfree(ldo);
+
+	return 0;
+}
+#else
+static int ldo_regulator_register(struct snd_soc_codec *codec,
+				struct regulator_init_data *init_data,
+				int voltage)
+{
+	return -EINVAL;
+}
+
+static int ldo_regulator_remove(struct snd_soc_codec *codec)
+{
+	return 0;
+}
+#endif
+
+/*
+ * set dac bias
+ * common state changes:
+ * startup:
+ * off --> standby --> prepare --> on
+ * standby --> prepare --> on
+ *
+ * stop:
+ * on --> prepare --> standby
+ */
+static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
+				   enum snd_soc_bias_level level)
+{
+	int ret;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(
+						ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+			if (ret)
+				return ret;
+			udelay(10);
+		}
+
+		break;
+	case SND_SOC_BIAS_OFF:
+		regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE |\
+			SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops sgtl5000_ops = {
+	.hw_params = sgtl5000_pcm_hw_params,
+	.digital_mute = sgtl5000_digital_mute,
+	.set_fmt = sgtl5000_set_dai_fmt,
+	.set_sysclk = sgtl5000_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver sgtl5000_dai = {
+	.name = "sgtl5000",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		/*
+		 * only support 8~48K + 96K,
+		 * TODO modify hw_param to support more
+		 */
+		.rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000,
+		.formats = SGTL5000_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000,
+		.formats = SGTL5000_FORMATS,
+	},
+	.ops = &sgtl5000_ops,
+	.symmetric_rates = 1,
+};
+
+static int sgtl5000_volatile_register(struct snd_soc_codec *codec,
+					unsigned int reg)
+{
+	switch (reg) {
+	case SGTL5000_CHIP_ID:
+	case SGTL5000_CHIP_ADCDAC_CTRL:
+	case SGTL5000_CHIP_ANA_STATUS:
+		return 1;
+	}
+
+	return 0;
+}
+
+#ifdef CONFIG_SUSPEND
+static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+/*
+ * restore all sgtl5000 registers,
+ * since a big hole between dap and regular registers,
+ * we will restore them respectively.
+ */
+static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
+{
+	u16 *cache = codec->reg_cache;
+	int i;
+	int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1;
+
+	/* restore regular registers */
+	for (i = 0; i < regular_regs; i++) {
+		int reg = i << 1;
+
+		/* this regs depends on the others */
+		if (reg == SGTL5000_CHIP_ANA_POWER ||
+			reg == SGTL5000_CHIP_CLK_CTRL ||
+			reg == SGTL5000_CHIP_LINREG_CTRL ||
+			reg == SGTL5000_CHIP_LINE_OUT_CTRL ||
+			reg == SGTL5000_CHIP_CLK_CTRL)
+			continue;
+
+		snd_soc_write(codec, reg, cache[i]);
+	}
+
+	/* restore dap registers */
+	for (i = SGTL5000_DAP_REG_OFFSET >> 1;
+			i < SGTL5000_MAX_REG_OFFSET >> 1; i++) {
+		int reg = i << 1;
+
+		snd_soc_write(codec, reg, cache[i]);
+	}
+
+	/*
+	 * restore power and other regs according
+	 * to set_power() and set_clock()
+	 */
+	snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
+			cache[SGTL5000_CHIP_LINREG_CTRL >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER,
+			cache[SGTL5000_CHIP_ANA_POWER >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL,
+			cache[SGTL5000_CHIP_CLK_CTRL >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL,
+			cache[SGTL5000_CHIP_REF_CTRL >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
+			cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]);
+	return 0;
+}
+
+static int sgtl5000_resume(struct snd_soc_codec *codec)
+{
+	/* Bring the codec back up to standby to enable regulators */
+	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* Restore registers by cached in memory */
+	sgtl5000_restore_regs(codec);
+	return 0;
+}
+#else
+#define sgtl5000_suspend NULL
+#define sgtl5000_resume  NULL
+#endif	/* CONFIG_SUSPEND */
+
+/*
+ * sgtl5000 has 3 internal power supplies:
+ * 1. VAG, normally set to vdda/2
+ * 2. chargepump, set to different value
+ *	according to voltage of vdda and vddio
+ * 3. line out VAG, normally set to vddio/2
+ *
+ * and should be set according to:
+ * 1. vddd provided by external or not
+ * 2. vdda and vddio voltage value. > 3.1v or not
+ * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd.
+ */
+static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
+{
+	int vddd;
+	int vdda;
+	int vddio;
+	u16 ana_pwr;
+	u16 lreg_ctrl;
+	int vag;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	vdda  = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer);
+	vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer);
+	vddd  = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer);
+
+	vdda  = vdda / 1000;
+	vddio = vddio / 1000;
+	vddd  = vddd / 1000;
+
+	if (vdda <= 0 || vddio <= 0 || vddd < 0) {
+		dev_err(codec->dev, "regulator voltage not set correctly\n");
+
+		return -EINVAL;
+	}
+
+	/* according to datasheet, maximum voltage of supplies */
+	if (vdda > 3600 || vddio > 3600 || vddd > 1980) {
+		dev_err(codec->dev,
+			"exceed max voltage vdda %dmv vddio %dma vddd %dma\n",
+			vdda, vddio, vddd);
+
+		return -EINVAL;
+	}
+
+	/* reset value */
+	ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER);
+	ana_pwr |= SGTL5000_DAC_STEREO |
+			SGTL5000_ADC_STEREO |
+			SGTL5000_REFTOP_POWERUP;
+	lreg_ctrl = snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL);
+
+	if (vddio < 3100 && vdda < 3100) {
+		/* enable internal oscillator used for charge pump */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL,
+					SGTL5000_INT_OSC_EN,
+					SGTL5000_INT_OSC_EN);
+		/* Enable VDDC charge pump */
+		ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
+	} else if (vddio >= 3100 && vdda >= 3100) {
+		/*
+		 * if vddio and vddd > 3.1v,
+		 * charge pump should be clean before set ana_pwr
+		 */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_VDDC_CHRGPMP_POWERUP, 0);
+
+		/* VDDC use VDDIO rail */
+		lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
+		lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
+			    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+	}
+
+	snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
+
+	snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr);
+
+	/* set voltage to register */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+				(0x1 << 4) - 1, 0x8);
+
+	/*
+	 * if vddd linear reg has been enabled,
+	 * simple digital supply should be clear to get
+	 * proper VDDD voltage.
+	 */
+	if (ana_pwr & SGTL5000_LINEREG_D_POWERUP)
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINREG_SIMPLE_POWERUP,
+				0);
+	else
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINREG_SIMPLE_POWERUP |
+				SGTL5000_STARTUP_POWERUP,
+				0);
+
+	/*
+	 * set ADC/DAC VAG to vdda / 2,
+	 * should stay in range (0.8v, 1.575v)
+	 */
+	vag = vdda / 2;
+	if (vag <= SGTL5000_ANA_GND_BASE)
+		vag = 0;
+	else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP *
+		 (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT))
+		vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT;
+	else
+		vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
+			vag << SGTL5000_ANA_GND_SHIFT,
+			vag << SGTL5000_ANA_GND_SHIFT);
+
+	/* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */
+	vag = vddio / 2;
+	if (vag <= SGTL5000_LINE_OUT_GND_BASE)
+		vag = 0;
+	else if (vag >= SGTL5000_LINE_OUT_GND_BASE +
+		SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX)
+		vag = SGTL5000_LINE_OUT_GND_MAX;
+	else
+		vag = (vag - SGTL5000_LINE_OUT_GND_BASE) /
+		    SGTL5000_LINE_OUT_GND_STP;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
+			vag << SGTL5000_LINE_OUT_GND_SHIFT |
+			SGTL5000_LINE_OUT_CURRENT_360u <<
+				SGTL5000_LINE_OUT_CURRENT_SHIFT,
+			vag << SGTL5000_LINE_OUT_GND_SHIFT |
+			SGTL5000_LINE_OUT_CURRENT_360u <<
+				SGTL5000_LINE_OUT_CURRENT_SHIFT);
+
+	return 0;
+}
+
+static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
+{
+	u16 reg;
+	int ret;
+	int rev;
+	int i;
+	int external_vddd = 0;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++)
+		sgtl5000->supplies[i].supply = supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	if (!ret)
+		external_vddd = 1;
+	else {
+		/* set internal ldo to 1.2v */
+		int voltage = LDO_VOLTAGE;
+
+		ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
+		if (ret) {
+			dev_err(codec->dev,
+			"Failed to register vddd internal supplies: %d\n",
+				ret);
+			return ret;
+		}
+
+		sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
+
+		ret = regulator_bulk_get(codec->dev,
+				ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+
+		if (ret) {
+			ldo_regulator_remove(codec);
+			dev_err(codec->dev,
+				"Failed to request supplies: %d\n", ret);
+
+			return ret;
+		}
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+	if (ret)
+		goto err_regulator_free;
+
+	/* wait for all power rails bring up */
+	udelay(10);
+
+	/* read chip information */
+	reg = snd_soc_read(codec, SGTL5000_CHIP_ID);
+	if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) !=
+	    SGTL5000_PARTID_PART_ID) {
+		dev_err(codec->dev,
+			"Device with ID register %x is not a sgtl5000\n", reg);
+		ret = -ENODEV;
+		goto err_regulator_disable;
+	}
+
+	rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT;
+	dev_info(codec->dev, "sgtl5000 revision %d\n", rev);
+
+	/*
+	 * workaround for revision 0x11 and later,
+	 * roll back to use internal LDO
+	 */
+	if (external_vddd && rev >= 0x11) {
+		int voltage = LDO_VOLTAGE;
+		/* disable all regulator first */
+		regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+		/* free VDDD regulator */
+		regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+
+		ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
+		if (ret)
+			return ret;
+
+		sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
+
+		ret = regulator_bulk_get(codec->dev,
+				ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+		if (ret) {
+			ldo_regulator_remove(codec);
+			dev_err(codec->dev,
+				"Failed to request supplies: %d\n", ret);
+
+			return ret;
+		}
+
+		ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+		if (ret)
+			goto err_regulator_free;
+
+		/* wait for all power rails bring up */
+		udelay(10);
+	}
+
+	return 0;
+
+err_regulator_disable:
+	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+err_regulator_free:
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	if (external_vddd)
+		ldo_regulator_remove(codec);
+	return ret;
+
+}
+
+static int sgtl5000_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	/* setup i2c data ops */
+	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	ret = sgtl5000_enable_regulators(codec);
+	if (ret)
+		return ret;
+
+	/* power up sgtl5000 */
+	ret = sgtl5000_set_power_regs(codec);
+	if (ret)
+		goto err;
+
+	/* enable small pop, introduce 400ms delay in turning off */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
+				SGTL5000_SMALL_POP,
+				SGTL5000_SMALL_POP);
+
+	/* disable short cut detector */
+	snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0);
+
+	/*
+	 * set i2s as default input of sound switch
+	 * TODO: add sound switch to control and dapm widge.
+	 */
+	snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL,
+			SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT);
+	snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER,
+			SGTL5000_ADC_EN | SGTL5000_DAC_EN);
+
+	/* enable dac volume ramp by default */
+	snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL,
+			SGTL5000_DAC_VOL_RAMP_EN |
+			SGTL5000_DAC_MUTE_RIGHT |
+			SGTL5000_DAC_MUTE_LEFT);
+
+	snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f);
+
+	snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL,
+			SGTL5000_HP_ZCD_EN |
+			SGTL5000_ADC_ZCD_EN);
+
+	snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+
+	/*
+	 * disable DAP
+	 * TODO:
+	 * Enable DAP in kcontrol and dapm.
+	 */
+	snd_soc_write(codec, SGTL5000_DAP_CTRL, 0);
+
+	/* leading to standby state */
+	ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	if (ret)
+		goto err;
+
+	snd_soc_add_controls(codec, sgtl5000_snd_controls,
+			     ARRAY_SIZE(sgtl5000_snd_controls));
+
+	snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets,
+				  ARRAY_SIZE(sgtl5000_dapm_widgets));
+
+	snd_soc_dapm_add_routes(&codec->dapm, audio_map,
+				ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(&codec->dapm);
+
+	return 0;
+
+err:
+	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	ldo_regulator_remove(codec);
+
+	return ret;
+}
+
+static int sgtl5000_remove(struct snd_soc_codec *codec)
+{
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	ldo_regulator_remove(codec);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver sgtl5000_driver = {
+	.probe = sgtl5000_probe,
+	.remove = sgtl5000_remove,
+	.suspend = sgtl5000_suspend,
+	.resume = sgtl5000_resume,
+	.set_bias_level = sgtl5000_set_bias_level,
+	.reg_cache_size = ARRAY_SIZE(sgtl5000_regs),
+	.reg_word_size = sizeof(u16),
+	.reg_cache_step = 2,
+	.reg_cache_default = sgtl5000_regs,
+	.volatile_register = sgtl5000_volatile_register,
+};
+
+static __devinit int sgtl5000_i2c_probe(struct i2c_client *client,
+					const struct i2c_device_id *id)
+{
+	struct sgtl5000_priv *sgtl5000;
+	int ret;
+
+	sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL);
+	if (!sgtl5000)
+		return -ENOMEM;
+
+	/*
+	 * copy DAP default values to default value array.
+	 * sgtl5000 register space has a big hole, merge it
+	 * at init phase makes life easy.
+	 * FIXME: should we drop 'const' of sgtl5000_regs?
+	 */
+	memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)),
+			sgtl5000_dap_regs,
+			SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET);
+
+	i2c_set_clientdata(client, sgtl5000);
+
+	ret = snd_soc_register_codec(&client->dev,
+			&sgtl5000_driver, &sgtl5000_dai, 1);
+	if (ret) {
+		dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+		kfree(sgtl5000);
+		return ret;
+	}
+
+	return 0;
+}
+
+static __devexit int sgtl5000_i2c_remove(struct i2c_client *client)
+{
+	struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+
+	kfree(sgtl5000);
+	return 0;
+}
+
+static const struct i2c_device_id sgtl5000_id[] = {
+	{"sgtl5000", 0},
+	{},
+};
+
+MODULE_DEVICE_TABLE(i2c, sgtl5000_id);
+
+static struct i2c_driver sgtl5000_i2c_driver = {
+	.driver = {
+		   .name = "sgtl5000",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = sgtl5000_i2c_probe,
+	.remove = __devexit_p(sgtl5000_i2c_remove),
+	.id_table = sgtl5000_id,
+};
+
+static int __init sgtl5000_modinit(void)
+{
+	return i2c_add_driver(&sgtl5000_i2c_driver);
+}
+module_init(sgtl5000_modinit);
+
+static void __exit sgtl5000_exit(void)
+{
+	i2c_del_driver(&sgtl5000_i2c_driver);
+}
+module_exit(sgtl5000_exit);
+
+MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
+MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng@freescale.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
new file mode 100644
index 00000000000..eec3ab368f3
--- /dev/null
+++ b/sound/soc/codecs/sgtl5000.h
@@ -0,0 +1,400 @@
+/*
+ * sgtl5000.h - SGTL5000 audio codec interface
+ *
+ * Copyright 2010-2011 Freescale Semiconductor, Inc.
+ *
+ * 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 _SGTL5000_H
+#define _SGTL5000_H
+
+/*
+ * Register values.
+ */
+#define SGTL5000_CHIP_ID			0x0000
+#define SGTL5000_CHIP_DIG_POWER			0x0002
+#define SGTL5000_CHIP_CLK_CTRL			0x0004
+#define SGTL5000_CHIP_I2S_CTRL			0x0006
+#define SGTL5000_CHIP_SSS_CTRL			0x000a
+#define SGTL5000_CHIP_ADCDAC_CTRL		0x000e
+#define SGTL5000_CHIP_DAC_VOL			0x0010
+#define SGTL5000_CHIP_PAD_STRENGTH		0x0014
+#define SGTL5000_CHIP_ANA_ADC_CTRL		0x0020
+#define SGTL5000_CHIP_ANA_HP_CTRL		0x0022
+#define SGTL5000_CHIP_ANA_CTRL			0x0024
+#define SGTL5000_CHIP_LINREG_CTRL		0x0026
+#define SGTL5000_CHIP_REF_CTRL			0x0028
+#define SGTL5000_CHIP_MIC_CTRL			0x002a
+#define SGTL5000_CHIP_LINE_OUT_CTRL		0x002c
+#define SGTL5000_CHIP_LINE_OUT_VOL		0x002e
+#define SGTL5000_CHIP_ANA_POWER			0x0030
+#define SGTL5000_CHIP_PLL_CTRL			0x0032
+#define SGTL5000_CHIP_CLK_TOP_CTRL		0x0034
+#define SGTL5000_CHIP_ANA_STATUS		0x0036
+#define SGTL5000_CHIP_SHORT_CTRL		0x003c
+#define SGTL5000_CHIP_ANA_TEST2			0x003a
+#define SGTL5000_DAP_CTRL			0x0100
+#define SGTL5000_DAP_PEQ			0x0102
+#define SGTL5000_DAP_BASS_ENHANCE		0x0104
+#define SGTL5000_DAP_BASS_ENHANCE_CTRL		0x0106
+#define SGTL5000_DAP_AUDIO_EQ			0x0108
+#define SGTL5000_DAP_SURROUND			0x010a
+#define SGTL5000_DAP_FLT_COEF_ACCESS		0x010c
+#define SGTL5000_DAP_COEF_WR_B0_MSB		0x010e
+#define SGTL5000_DAP_COEF_WR_B0_LSB		0x0110
+#define SGTL5000_DAP_EQ_BASS_BAND0		0x0116
+#define SGTL5000_DAP_EQ_BASS_BAND1		0x0118
+#define SGTL5000_DAP_EQ_BASS_BAND2		0x011a
+#define SGTL5000_DAP_EQ_BASS_BAND3		0x011c
+#define SGTL5000_DAP_EQ_BASS_BAND4		0x011e
+#define SGTL5000_DAP_MAIN_CHAN			0x0120
+#define SGTL5000_DAP_MIX_CHAN			0x0122
+#define SGTL5000_DAP_AVC_CTRL			0x0124
+#define SGTL5000_DAP_AVC_THRESHOLD		0x0126
+#define SGTL5000_DAP_AVC_ATTACK			0x0128
+#define SGTL5000_DAP_AVC_DECAY			0x012a
+#define SGTL5000_DAP_COEF_WR_B1_MSB		0x012c
+#define SGTL5000_DAP_COEF_WR_B1_LSB		0x012e
+#define SGTL5000_DAP_COEF_WR_B2_MSB		0x0130
+#define SGTL5000_DAP_COEF_WR_B2_LSB		0x0132
+#define SGTL5000_DAP_COEF_WR_A1_MSB		0x0134
+#define SGTL5000_DAP_COEF_WR_A1_LSB		0x0136
+#define SGTL5000_DAP_COEF_WR_A2_MSB		0x0138
+#define SGTL5000_DAP_COEF_WR_A2_LSB		0x013a
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * SGTL5000_CHIP_ID
+ */
+#define SGTL5000_PARTID_MASK			0xff00
+#define SGTL5000_PARTID_SHIFT			8
+#define SGTL5000_PARTID_WIDTH			8
+#define SGTL5000_PARTID_PART_ID			0xa0
+#define SGTL5000_REVID_MASK			0x00ff
+#define SGTL5000_REVID_SHIFT			0
+#define SGTL5000_REVID_WIDTH			8
+
+/*
+ * SGTL5000_CHIP_DIG_POWER
+ */
+#define SGTL5000_ADC_EN				0x0040
+#define SGTL5000_DAC_EN				0x0020
+#define SGTL5000_DAP_POWERUP			0x0010
+#define SGTL5000_I2S_OUT_POWERUP		0x0002
+#define SGTL5000_I2S_IN_POWERUP			0x0001
+
+/*
+ * SGTL5000_CHIP_CLK_CTRL
+ */
+#define SGTL5000_RATE_MODE_MASK			0x0030
+#define SGTL5000_RATE_MODE_SHIFT		4
+#define SGTL5000_RATE_MODE_WIDTH		2
+#define SGTL5000_RATE_MODE_DIV_1		0
+#define SGTL5000_RATE_MODE_DIV_2		1
+#define SGTL5000_RATE_MODE_DIV_4		2
+#define SGTL5000_RATE_MODE_DIV_6		3
+#define SGTL5000_SYS_FS_MASK			0x000c
+#define SGTL5000_SYS_FS_SHIFT			2
+#define SGTL5000_SYS_FS_WIDTH			2
+#define SGTL5000_SYS_FS_32k			0x0
+#define SGTL5000_SYS_FS_44_1k			0x1
+#define SGTL5000_SYS_FS_48k			0x2
+#define SGTL5000_SYS_FS_96k			0x3
+#define SGTL5000_MCLK_FREQ_MASK			0x0003
+#define SGTL5000_MCLK_FREQ_SHIFT		0
+#define SGTL5000_MCLK_FREQ_WIDTH		2
+#define SGTL5000_MCLK_FREQ_256FS		0x0
+#define SGTL5000_MCLK_FREQ_384FS		0x1
+#define SGTL5000_MCLK_FREQ_512FS		0x2
+#define SGTL5000_MCLK_FREQ_PLL			0x3
+
+/*
+ * SGTL5000_CHIP_I2S_CTRL
+ */
+#define SGTL5000_I2S_SCLKFREQ_MASK		0x0100
+#define SGTL5000_I2S_SCLKFREQ_SHIFT		8
+#define SGTL5000_I2S_SCLKFREQ_WIDTH		1
+#define SGTL5000_I2S_SCLKFREQ_64FS		0x0
+#define SGTL5000_I2S_SCLKFREQ_32FS		0x1	/* Not for RJ mode */
+#define SGTL5000_I2S_MASTER			0x0080
+#define SGTL5000_I2S_SCLK_INV			0x0040
+#define SGTL5000_I2S_DLEN_MASK			0x0030
+#define SGTL5000_I2S_DLEN_SHIFT			4
+#define SGTL5000_I2S_DLEN_WIDTH			2
+#define SGTL5000_I2S_DLEN_32			0x0
+#define SGTL5000_I2S_DLEN_24			0x1
+#define SGTL5000_I2S_DLEN_20			0x2
+#define SGTL5000_I2S_DLEN_16			0x3
+#define SGTL5000_I2S_MODE_MASK			0x000c
+#define SGTL5000_I2S_MODE_SHIFT			2
+#define SGTL5000_I2S_MODE_WIDTH			2
+#define SGTL5000_I2S_MODE_I2S_LJ		0x0
+#define SGTL5000_I2S_MODE_RJ			0x1
+#define SGTL5000_I2S_MODE_PCM			0x2
+#define SGTL5000_I2S_LRALIGN			0x0002
+#define SGTL5000_I2S_LRPOL			0x0001	/* set for which mode */
+
+/*
+ * SGTL5000_CHIP_SSS_CTRL
+ */
+#define SGTL5000_DAP_MIX_LRSWAP			0x4000
+#define SGTL5000_DAP_LRSWAP			0x2000
+#define SGTL5000_DAC_LRSWAP			0x1000
+#define SGTL5000_I2S_OUT_LRSWAP			0x0400
+#define SGTL5000_DAP_MIX_SEL_MASK		0x0300
+#define SGTL5000_DAP_MIX_SEL_SHIFT		8
+#define SGTL5000_DAP_MIX_SEL_WIDTH		2
+#define SGTL5000_DAP_MIX_SEL_ADC		0x0
+#define SGTL5000_DAP_MIX_SEL_I2S_IN		0x1
+#define SGTL5000_DAP_SEL_MASK			0x00c0
+#define SGTL5000_DAP_SEL_SHIFT			6
+#define SGTL5000_DAP_SEL_WIDTH			2
+#define SGTL5000_DAP_SEL_ADC			0x0
+#define SGTL5000_DAP_SEL_I2S_IN			0x1
+#define SGTL5000_DAC_SEL_MASK			0x0030
+#define SGTL5000_DAC_SEL_SHIFT			4
+#define SGTL5000_DAC_SEL_WIDTH			2
+#define SGTL5000_DAC_SEL_ADC			0x0
+#define SGTL5000_DAC_SEL_I2S_IN			0x1
+#define SGTL5000_DAC_SEL_DAP			0x3
+#define SGTL5000_I2S_OUT_SEL_MASK		0x0003
+#define SGTL5000_I2S_OUT_SEL_SHIFT		0
+#define SGTL5000_I2S_OUT_SEL_WIDTH		2
+#define SGTL5000_I2S_OUT_SEL_ADC		0x0
+#define SGTL5000_I2S_OUT_SEL_I2S_IN		0x1
+#define SGTL5000_I2S_OUT_SEL_DAP		0x3
+
+/*
+ * SGTL5000_CHIP_ADCDAC_CTRL
+ */
+#define SGTL5000_VOL_BUSY_DAC_RIGHT		0x2000
+#define SGTL5000_VOL_BUSY_DAC_LEFT		0x1000
+#define SGTL5000_DAC_VOL_RAMP_EN		0x0200
+#define SGTL5000_DAC_VOL_RAMP_EXPO		0x0100
+#define SGTL5000_DAC_MUTE_RIGHT			0x0008
+#define SGTL5000_DAC_MUTE_LEFT			0x0004
+#define SGTL5000_ADC_HPF_FREEZE			0x0002
+#define SGTL5000_ADC_HPF_BYPASS			0x0001
+
+/*
+ * SGTL5000_CHIP_DAC_VOL
+ */
+#define SGTL5000_DAC_VOL_RIGHT_MASK		0xff00
+#define SGTL5000_DAC_VOL_RIGHT_SHIFT		8
+#define SGTL5000_DAC_VOL_RIGHT_WIDTH		8
+#define SGTL5000_DAC_VOL_LEFT_MASK		0x00ff
+#define SGTL5000_DAC_VOL_LEFT_SHIFT		0
+#define SGTL5000_DAC_VOL_LEFT_WIDTH		8
+
+/*
+ * SGTL5000_CHIP_PAD_STRENGTH
+ */
+#define SGTL5000_PAD_I2S_LRCLK_MASK		0x0300
+#define SGTL5000_PAD_I2S_LRCLK_SHIFT		8
+#define SGTL5000_PAD_I2S_LRCLK_WIDTH		2
+#define SGTL5000_PAD_I2S_SCLK_MASK		0x00c0
+#define SGTL5000_PAD_I2S_SCLK_SHIFT		6
+#define SGTL5000_PAD_I2S_SCLK_WIDTH		2
+#define SGTL5000_PAD_I2S_DOUT_MASK		0x0030
+#define SGTL5000_PAD_I2S_DOUT_SHIFT		4
+#define SGTL5000_PAD_I2S_DOUT_WIDTH		2
+#define SGTL5000_PAD_I2C_SDA_MASK		0x000c
+#define SGTL5000_PAD_I2C_SDA_SHIFT		2
+#define SGTL5000_PAD_I2C_SDA_WIDTH		2
+#define SGTL5000_PAD_I2C_SCL_MASK		0x0003
+#define SGTL5000_PAD_I2C_SCL_SHIFT		0
+#define SGTL5000_PAD_I2C_SCL_WIDTH		2
+
+/*
+ * SGTL5000_CHIP_ANA_ADC_CTRL
+ */
+#define SGTL5000_ADC_VOL_M6DB			0x0100
+#define SGTL5000_ADC_VOL_RIGHT_MASK		0x00f0
+#define SGTL5000_ADC_VOL_RIGHT_SHIFT		4
+#define SGTL5000_ADC_VOL_RIGHT_WIDTH		4
+#define SGTL5000_ADC_VOL_LEFT_MASK		0x000f
+#define SGTL5000_ADC_VOL_LEFT_SHIFT		0
+#define SGTL5000_ADC_VOL_LEFT_WIDTH		4
+
+/*
+ * SGTL5000_CHIP_ANA_HP_CTRL
+ */
+#define SGTL5000_HP_VOL_RIGHT_MASK		0x7f00
+#define SGTL5000_HP_VOL_RIGHT_SHIFT		8
+#define SGTL5000_HP_VOL_RIGHT_WIDTH		7
+#define SGTL5000_HP_VOL_LEFT_MASK		0x007f
+#define SGTL5000_HP_VOL_LEFT_SHIFT		0
+#define SGTL5000_HP_VOL_LEFT_WIDTH		7
+
+/*
+ * SGTL5000_CHIP_ANA_CTRL
+ */
+#define SGTL5000_LINE_OUT_MUTE			0x0100
+#define SGTL5000_HP_SEL_MASK			0x0040
+#define SGTL5000_HP_SEL_SHIFT			6
+#define SGTL5000_HP_SEL_WIDTH			1
+#define SGTL5000_HP_SEL_DAC			0x0
+#define SGTL5000_HP_SEL_LINE_IN			0x1
+#define SGTL5000_HP_ZCD_EN			0x0020
+#define SGTL5000_HP_MUTE			0x0010
+#define SGTL5000_ADC_SEL_MASK			0x0004
+#define SGTL5000_ADC_SEL_SHIFT			2
+#define SGTL5000_ADC_SEL_WIDTH			1
+#define SGTL5000_ADC_SEL_MIC			0x0
+#define SGTL5000_ADC_SEL_LINE_IN		0x1
+#define SGTL5000_ADC_ZCD_EN			0x0002
+#define SGTL5000_ADC_MUTE			0x0001
+
+/*
+ * SGTL5000_CHIP_LINREG_CTRL
+ */
+#define SGTL5000_VDDC_MAN_ASSN_MASK		0x0040
+#define SGTL5000_VDDC_MAN_ASSN_SHIFT		6
+#define SGTL5000_VDDC_MAN_ASSN_WIDTH		1
+#define SGTL5000_VDDC_MAN_ASSN_VDDA		0x0
+#define SGTL5000_VDDC_MAN_ASSN_VDDIO		0x1
+#define SGTL5000_VDDC_ASSN_OVRD			0x0020
+#define SGTL5000_LINREG_VDDD_MASK		0x000f
+#define SGTL5000_LINREG_VDDD_SHIFT		0
+#define SGTL5000_LINREG_VDDD_WIDTH		4
+
+/*
+ * SGTL5000_CHIP_REF_CTRL
+ */
+#define SGTL5000_ANA_GND_MASK			0x01f0
+#define SGTL5000_ANA_GND_SHIFT			4
+#define SGTL5000_ANA_GND_WIDTH			5
+#define SGTL5000_ANA_GND_BASE			800	/* mv */
+#define SGTL5000_ANA_GND_STP			25	/*mv */
+#define SGTL5000_BIAS_CTRL_MASK			0x000e
+#define SGTL5000_BIAS_CTRL_SHIFT		1
+#define SGTL5000_BIAS_CTRL_WIDTH		3
+#define SGTL5000_SMALL_POP			0x0001
+
+/*
+ * SGTL5000_CHIP_MIC_CTRL
+ */
+#define SGTL5000_BIAS_R_MASK			0x0200
+#define SGTL5000_BIAS_R_SHIFT			8
+#define SGTL5000_BIAS_R_WIDTH			2
+#define SGTL5000_BIAS_R_off			0x0
+#define SGTL5000_BIAS_R_2K			0x1
+#define SGTL5000_BIAS_R_4k			0x2
+#define SGTL5000_BIAS_R_8k			0x3
+#define SGTL5000_BIAS_VOLT_MASK			0x0070
+#define SGTL5000_BIAS_VOLT_SHIFT		4
+#define SGTL5000_BIAS_VOLT_WIDTH		3
+#define SGTL5000_MIC_GAIN_MASK			0x0003
+#define SGTL5000_MIC_GAIN_SHIFT			0
+#define SGTL5000_MIC_GAIN_WIDTH			2
+
+/*
+ * SGTL5000_CHIP_LINE_OUT_CTRL
+ */
+#define SGTL5000_LINE_OUT_CURRENT_MASK		0x0f00
+#define SGTL5000_LINE_OUT_CURRENT_SHIFT		8
+#define SGTL5000_LINE_OUT_CURRENT_WIDTH		4
+#define SGTL5000_LINE_OUT_CURRENT_180u		0x0
+#define SGTL5000_LINE_OUT_CURRENT_270u		0x1
+#define SGTL5000_LINE_OUT_CURRENT_360u		0x3
+#define SGTL5000_LINE_OUT_CURRENT_450u		0x7
+#define SGTL5000_LINE_OUT_CURRENT_540u		0xf
+#define SGTL5000_LINE_OUT_GND_MASK		0x003f
+#define SGTL5000_LINE_OUT_GND_SHIFT		0
+#define SGTL5000_LINE_OUT_GND_WIDTH		6
+#define SGTL5000_LINE_OUT_GND_BASE		800	/* mv */
+#define SGTL5000_LINE_OUT_GND_STP		25
+#define SGTL5000_LINE_OUT_GND_MAX		0x23
+
+/*
+ * SGTL5000_CHIP_LINE_OUT_VOL
+ */
+#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK	0x1f00
+#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT	8
+#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH	5
+#define SGTL5000_LINE_OUT_VOL_LEFT_MASK		0x001f
+#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT	0
+#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH	5
+
+/*
+ * SGTL5000_CHIP_ANA_POWER
+ */
+#define SGTL5000_DAC_STEREO			0x4000
+#define SGTL5000_LINREG_SIMPLE_POWERUP		0x2000
+#define SGTL5000_STARTUP_POWERUP		0x1000
+#define SGTL5000_VDDC_CHRGPMP_POWERUP		0x0800
+#define SGTL5000_PLL_POWERUP			0x0400
+#define SGTL5000_LINEREG_D_POWERUP		0x0200
+#define SGTL5000_VCOAMP_POWERUP			0x0100
+#define SGTL5000_VAG_POWERUP			0x0080
+#define SGTL5000_ADC_STEREO			0x0040
+#define SGTL5000_REFTOP_POWERUP			0x0020
+#define SGTL5000_HP_POWERUP			0x0010
+#define SGTL5000_DAC_POWERUP			0x0008
+#define SGTL5000_CAPLESS_HP_POWERUP		0x0004
+#define SGTL5000_ADC_POWERUP			0x0002
+#define SGTL5000_LINE_OUT_POWERUP		0x0001
+
+/*
+ * SGTL5000_CHIP_PLL_CTRL
+ */
+#define SGTL5000_PLL_INT_DIV_MASK		0xf800
+#define SGTL5000_PLL_INT_DIV_SHIFT		11
+#define SGTL5000_PLL_INT_DIV_WIDTH		5
+#define SGTL5000_PLL_FRAC_DIV_MASK		0x0700
+#define SGTL5000_PLL_FRAC_DIV_SHIFT		0
+#define SGTL5000_PLL_FRAC_DIV_WIDTH		11
+
+/*
+ * SGTL5000_CHIP_CLK_TOP_CTRL
+ */
+#define SGTL5000_INT_OSC_EN			0x0800
+#define SGTL5000_INPUT_FREQ_DIV2		0x0008
+
+/*
+ * SGTL5000_CHIP_ANA_STATUS
+ */
+#define SGTL5000_HP_LRSHORT			0x0200
+#define SGTL5000_CAPLESS_SHORT			0x0100
+#define SGTL5000_PLL_LOCKED			0x0010
+
+/*
+ * SGTL5000_CHIP_SHORT_CTRL
+ */
+#define SGTL5000_LVLADJR_MASK			0x7000
+#define SGTL5000_LVLADJR_SHIFT			12
+#define SGTL5000_LVLADJR_WIDTH			3
+#define SGTL5000_LVLADJL_MASK			0x0700
+#define SGTL5000_LVLADJL_SHIFT			8
+#define SGTL5000_LVLADJL_WIDTH			3
+#define SGTL5000_LVLADJC_MASK			0x0070
+#define SGTL5000_LVLADJC_SHIFT			4
+#define SGTL5000_LVLADJC_WIDTH			3
+#define SGTL5000_LR_SHORT_MOD_MASK		0x000c
+#define SGTL5000_LR_SHORT_MOD_SHIFT		2
+#define SGTL5000_LR_SHORT_MOD_WIDTH		2
+#define SGTL5000_CM_SHORT_MOD_MASK		0x0003
+#define SGTL5000_CM_SHORT_MOD_SHIFT		0
+#define SGTL5000_CM_SHORT_MOD_WIDTH		2
+
+/*
+ *SGTL5000_CHIP_ANA_TEST2
+ */
+#define SGTL5000_MONO_DAC			0x1000
+
+/*
+ * SGTL5000_DAP_CTRL
+ */
+#define SGTL5000_DAP_MIX_EN			0x0010
+#define SGTL5000_DAP_EN				0x0001
+
+#define SGTL5000_SYSCLK				0x00
+#define SGTL5000_LRCLK				0x01
+
+#endif
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
new file mode 100644
index 00000000000..2a30eae1881
--- /dev/null
+++ b/sound/soc/codecs/sn95031.c
@@ -0,0 +1,949 @@
+/*
+ *  sn95031.c -  TI sn95031 Codec driver
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *
+ */
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <asm/intel_scu_ipc.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include "sn95031.h"
+
+#define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100)
+#define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+
+/* adc helper functions */
+
+/* enables mic bias voltage */
+static void sn95031_enable_mic_bias(struct snd_soc_codec *codec)
+{
+	snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0));
+	snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2));
+}
+
+/* Enable/Disable the ADC depending on the argument */
+static void configure_adc(struct snd_soc_codec *sn95031_codec, int val)
+{
+	int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
+
+	if (val) {
+		/* Enable and start the ADC */
+		value |= (SN95031_ADC_ENBL | SN95031_ADC_START);
+		value &= (~SN95031_ADC_NO_LOOP);
+	} else {
+		/* Just stop the ADC */
+		value &= (~SN95031_ADC_START);
+	}
+	snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value);
+}
+
+/*
+ * finds an empty channel for conversion
+ * If the ADC is not enabled then start using 0th channel
+ * itself. Otherwise find an empty channel by looking for a
+ * channel in which the stopbit is set to 1. returns the index
+ * of the first free channel if succeeds or an error code.
+ *
+ * Context: can sleep
+ *
+ */
+static int find_free_channel(struct snd_soc_codec *sn95031_codec)
+{
+	int ret = 0, i, value;
+
+	/* check whether ADC is enabled */
+	value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
+
+	if ((value & SN95031_ADC_ENBL) == 0)
+		return 0;
+
+	/* ADC is already enabled; Looking for an empty channel */
+	for (i = 0; i <	SN95031_ADC_CHANLS_MAX; i++) {
+		value = snd_soc_read(sn95031_codec,
+				SN95031_ADC_CHNL_START_ADDR + i);
+		if (value & SN95031_STOPBIT_MASK) {
+			ret = i;
+			break;
+		}
+	}
+	return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret;
+}
+
+/* Initialize the ADC for reading micbias values. Can sleep. */
+static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec)
+{
+	int base_addr, chnl_addr;
+	int value;
+	static int channel_index;
+
+	/* Index of the first channel in which the stop bit is set */
+	channel_index = find_free_channel(sn95031_codec);
+	if (channel_index < 0) {
+		pr_err("No free ADC channels");
+		return channel_index;
+	}
+
+	base_addr = SN95031_ADC_CHNL_START_ADDR + channel_index;
+
+	if (!(channel_index == 0 || channel_index ==  SN95031_ADC_LOOP_MAX)) {
+		/* Reset stop bit for channels other than 0 and 12 */
+		value = snd_soc_read(sn95031_codec, base_addr);
+		/* Set the stop bit to zero */
+		snd_soc_write(sn95031_codec, base_addr, value & 0xEF);
+		/* Index of the first free channel */
+		base_addr++;
+		channel_index++;
+	}
+
+	/* Since this is the last channel, set the stop bit
+	   to 1 by ORing the DIE_SENSOR_CODE with 0x10 */
+	snd_soc_write(sn95031_codec, base_addr,
+				SN95031_AUDIO_DETECT_CODE | 0x10);
+
+	chnl_addr = SN95031_ADC_DATA_START_ADDR + 2 * channel_index;
+	pr_debug("mid_initialize : %x", chnl_addr);
+	configure_adc(sn95031_codec, 1);
+	return chnl_addr;
+}
+
+
+/* reads the ADC registers and gets the mic bias value in mV. */
+static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec)
+{
+	u16 adc_adr = sn95031_initialize_adc(codec);
+	u16 adc_val1, adc_val2;
+	unsigned int mic_bias;
+
+	sn95031_enable_mic_bias(codec);
+
+	/* Enable the sound card for conversion before reading */
+	snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05);
+	/* Re-toggle the RRDATARD bit */
+	snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04);
+
+	/* Read the higher bits of data */
+	msleep(1000);
+	adc_val1 = snd_soc_read(codec, adc_adr);
+	adc_adr++;
+	adc_val2 = snd_soc_read(codec, adc_adr);
+
+	/* Adding lower two bits to the higher bits */
+	mic_bias = (adc_val1 << 2) + (adc_val2 & 3);
+	mic_bias = (mic_bias * SN95031_ADC_ONE_LSB_MULTIPLIER) / 1000;
+	pr_debug("mic bias = %dmV\n", mic_bias);
+	return mic_bias;
+}
+EXPORT_SYMBOL_GPL(sn95031_get_mic_bias);
+/*end - adc helper functions */
+
+static inline unsigned int sn95031_read(struct snd_soc_codec *codec,
+			unsigned int reg)
+{
+	u8 value = 0;
+	int ret;
+
+	ret = intel_scu_ipc_ioread8(reg, &value);
+	if (ret)
+		pr_err("read of %x failed, err %d\n", reg, ret);
+	return value;
+
+}
+
+static inline int sn95031_write(struct snd_soc_codec *codec,
+			unsigned int reg, unsigned int value)
+{
+	int ret;
+
+	ret = intel_scu_ipc_iowrite8(reg, value);
+	if (ret)
+		pr_err("write of %x failed, err %d\n", reg, ret);
+	return ret;
+}
+
+static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+			pr_debug("vaud_bias powering up pll\n");
+			/* power up the pll */
+			snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5));
+			/* enable pcm 2 */
+			snd_soc_update_bits(codec, SN95031_PCM2C2,
+					BIT(0), BIT(0));
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			pr_debug("vaud_bias power up rail\n");
+			/* power up the rail */
+			snd_soc_write(codec, SN95031_VAUD,
+					BIT(2)|BIT(1)|BIT(0));
+			msleep(1);
+		} else if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
+			/* turn off pcm */
+			pr_debug("vaud_bias power dn pcm\n");
+			snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0);
+			snd_soc_write(codec, SN95031_AUDPLLCTRL, 0);
+		}
+		break;
+
+
+	case SND_SOC_BIAS_OFF:
+		pr_debug("vaud_bias _OFF doing rail shutdown\n");
+		snd_soc_write(codec, SN95031_VAUD, BIT(3));
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static int sn95031_vhs_event(struct snd_soc_dapm_widget *w,
+		    struct snd_kcontrol *kcontrol, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n");
+		/* power up the rail */
+		snd_soc_write(w->codec, SN95031_VHSP, 0x3D);
+		snd_soc_write(w->codec, SN95031_VHSN, 0x3F);
+		msleep(1);
+	} else if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n");
+		snd_soc_write(w->codec, SN95031_VHSP, 0xC4);
+		snd_soc_write(w->codec, SN95031_VHSN, 0x04);
+	}
+	return 0;
+}
+
+static int sn95031_vihf_event(struct snd_soc_dapm_widget *w,
+		    struct snd_kcontrol *kcontrol, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n");
+		/* power up the rail */
+		snd_soc_write(w->codec, SN95031_VIHF, 0x27);
+		msleep(1);
+	} else if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n");
+		snd_soc_write(w->codec, SN95031_VIHF, 0x24);
+	}
+	return 0;
+}
+
+static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
+{
+	unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
+
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ldo = BIT(5)|BIT(4);
+		clk_dir = BIT(0);
+		data_dir = BIT(7);
+	}
+	/* program DMIC LDO, clock and set clock */
+	snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
+	snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(0), clk_dir);
+	snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(7), data_dir);
+	return 0;
+}
+
+static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
+{
+	unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
+
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ldo = BIT(5)|BIT(4);
+		clk_dir = BIT(2);
+		data_dir = BIT(1);
+	}
+	/* program DMIC LDO, clock and set clock */
+	snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
+	snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(2), clk_dir);
+	snd_soc_update_bits(w->codec, SN95031_DMICBUF45, BIT(1), data_dir);
+	return 0;
+}
+
+static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
+{
+	unsigned int ldo = 0;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		ldo = BIT(7)|BIT(6);
+
+	/* program DMIC LDO */
+	snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo);
+	return 0;
+}
+
+/* mux controls */
+static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" };
+
+static const struct soc_enum sn95031_micl_enum =
+	SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts);
+
+static const struct snd_kcontrol_new sn95031_micl_mux_control =
+	SOC_DAPM_ENUM("Route", sn95031_micl_enum);
+
+static const struct soc_enum sn95031_micr_enum =
+	SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts);
+
+static const struct snd_kcontrol_new sn95031_micr_mux_control =
+	SOC_DAPM_ENUM("Route", sn95031_micr_enum);
+
+static const char *sn95031_input_texts[] = {	"DMIC1", "DMIC2", "DMIC3",
+						"DMIC4", "DMIC5", "DMIC6",
+						"ADC Left", "ADC Right" };
+
+static const struct soc_enum sn95031_input1_enum =
+	SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input1_mux_control =
+	SOC_DAPM_ENUM("Route", sn95031_input1_enum);
+
+static const struct soc_enum sn95031_input2_enum =
+	SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input2_mux_control =
+	SOC_DAPM_ENUM("Route", sn95031_input2_enum);
+
+static const struct soc_enum sn95031_input3_enum =
+	SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input3_mux_control =
+	SOC_DAPM_ENUM("Route", sn95031_input3_enum);
+
+static const struct soc_enum sn95031_input4_enum =
+	SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input4_mux_control =
+	SOC_DAPM_ENUM("Route", sn95031_input4_enum);
+
+/* capture path controls */
+
+static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"};
+
+/* 0dB to 30dB in 10dB steps */
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0);
+
+static const struct soc_enum sn95031_micmode1_enum =
+	SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text);
+static const struct soc_enum sn95031_micmode2_enum =
+	SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text);
+
+static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"};
+
+static const struct soc_enum sn95031_dmic12_cfg_enum =
+	SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text);
+static const struct soc_enum sn95031_dmic34_cfg_enum =
+	SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text);
+static const struct soc_enum sn95031_dmic56_cfg_enum =
+	SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text);
+
+static const struct snd_kcontrol_new sn95031_snd_controls[] = {
+	SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum),
+	SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum),
+	SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum),
+	SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum),
+	SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum),
+	SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1,
+			2, 4, 0, mic_tlv),
+	SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2,
+			2, 4, 0, mic_tlv),
+};
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = {
+
+	/* all end points mic, hs etc */
+	SND_SOC_DAPM_OUTPUT("HPOUTL"),
+	SND_SOC_DAPM_OUTPUT("HPOUTR"),
+	SND_SOC_DAPM_OUTPUT("EPOUT"),
+	SND_SOC_DAPM_OUTPUT("IHFOUTL"),
+	SND_SOC_DAPM_OUTPUT("IHFOUTR"),
+	SND_SOC_DAPM_OUTPUT("LINEOUTL"),
+	SND_SOC_DAPM_OUTPUT("LINEOUTR"),
+	SND_SOC_DAPM_OUTPUT("VIB1OUT"),
+	SND_SOC_DAPM_OUTPUT("VIB2OUT"),
+
+	SND_SOC_DAPM_INPUT("AMIC1"), /* headset mic */
+	SND_SOC_DAPM_INPUT("AMIC2"),
+	SND_SOC_DAPM_INPUT("DMIC1"),
+	SND_SOC_DAPM_INPUT("DMIC2"),
+	SND_SOC_DAPM_INPUT("DMIC3"),
+	SND_SOC_DAPM_INPUT("DMIC4"),
+	SND_SOC_DAPM_INPUT("DMIC5"),
+	SND_SOC_DAPM_INPUT("DMIC6"),
+	SND_SOC_DAPM_INPUT("LINEINL"),
+	SND_SOC_DAPM_INPUT("LINEINR"),
+
+	SND_SOC_DAPM_MICBIAS("AMIC1Bias", SN95031_MICBIAS, 2, 0),
+	SND_SOC_DAPM_MICBIAS("AMIC2Bias", SN95031_MICBIAS, 3, 0),
+	SND_SOC_DAPM_MICBIAS("DMIC12Bias", SN95031_DMICMUX, 3, 0),
+	SND_SOC_DAPM_MICBIAS("DMIC34Bias", SN95031_DMICMUX, 4, 0),
+	SND_SOC_DAPM_MICBIAS("DMIC56Bias", SN95031_DMICMUX, 5, 0),
+
+	SND_SOC_DAPM_SUPPLY("DMIC12supply", SN95031_DMICLK, 0, 0,
+				sn95031_dmic12_event,
+				SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("DMIC34supply", SN95031_DMICLK, 1, 0,
+				sn95031_dmic34_event,
+				SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("DMIC56supply", SN95031_DMICLK, 2, 0,
+				sn95031_dmic56_event,
+				SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0,
+			SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0,
+			sn95031_vhs_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0,
+			sn95031_vihf_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* playback path driver enables */
+	SND_SOC_DAPM_PGA("Headset Left Playback",
+			SN95031_DRIVEREN, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Headset Right Playback",
+			SN95031_DRIVEREN, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Speaker Left Playback",
+			SN95031_DRIVEREN, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Speaker Right Playback",
+			SN95031_DRIVEREN, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Vibra1 Playback",
+			SN95031_DRIVEREN, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Vibra2 Playback",
+			SN95031_DRIVEREN, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Earpiece Playback",
+			SN95031_DRIVEREN, 6, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Lineout Left Playback",
+			SN95031_LOCTL, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Lineout Right Playback",
+			SN95031_LOCTL, 4, 0, NULL, 0),
+
+	/* playback path filter enable */
+	SND_SOC_DAPM_PGA("Headset Left Filter",
+			SN95031_HSEPRXCTRL, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Headset Right Filter",
+			SN95031_HSEPRXCTRL, 5, 0,  NULL, 0),
+	SND_SOC_DAPM_PGA("Speaker Left Filter",
+			SN95031_IHFRXCTRL, 0, 0,  NULL, 0),
+	SND_SOC_DAPM_PGA("Speaker Right Filter",
+			SN95031_IHFRXCTRL, 1, 0,  NULL, 0),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("HSDAC Left", "Headset",
+			SN95031_DACCONFIG, 0, 0),
+	SND_SOC_DAPM_DAC("HSDAC Right", "Headset",
+			SN95031_DACCONFIG, 1, 0),
+	SND_SOC_DAPM_DAC("IHFDAC Left", "Speaker",
+			SN95031_DACCONFIG, 2, 0),
+	SND_SOC_DAPM_DAC("IHFDAC Right", "Speaker",
+			SN95031_DACCONFIG, 3, 0),
+	SND_SOC_DAPM_DAC("Vibra1 DAC", "Vibra1",
+			SN95031_VIB1C5, 1, 0),
+	SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2",
+			SN95031_VIB2C5, 1, 0),
+
+	/* capture widgets */
+	SND_SOC_DAPM_PGA("LineIn Enable Left", SN95031_MICAMP1,
+				7, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("LineIn Enable Right", SN95031_MICAMP2,
+				7, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("MIC1 Enable", SN95031_MICAMP1, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC2 Enable", SN95031_MICAMP2, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("TX1 Enable", SN95031_AUDIOTXEN, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("TX2 Enable", SN95031_AUDIOTXEN, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("TX3 Enable", SN95031_AUDIOTXEN, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("TX4 Enable", SN95031_AUDIOTXEN, 5, 0, NULL, 0),
+
+	/* ADC have null stream as they will be turned ON by TX path */
+	SND_SOC_DAPM_ADC("ADC Left", NULL,
+			SN95031_ADCCONFIG, 0, 0),
+	SND_SOC_DAPM_ADC("ADC Right", NULL,
+			SN95031_ADCCONFIG, 2, 0),
+
+	SND_SOC_DAPM_MUX("Mic_InputL Capture Route",
+			SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control),
+	SND_SOC_DAPM_MUX("Mic_InputR Capture Route",
+			SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control),
+
+	SND_SOC_DAPM_MUX("Txpath1 Capture Route",
+			SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control),
+	SND_SOC_DAPM_MUX("Txpath2 Capture Route",
+			SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control),
+	SND_SOC_DAPM_MUX("Txpath3 Capture Route",
+			SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control),
+	SND_SOC_DAPM_MUX("Txpath4 Capture Route",
+			SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control),
+
+};
+
+static const struct snd_soc_dapm_route sn95031_audio_map[] = {
+	/* headset and earpiece map */
+	{ "HPOUTL", NULL, "Headset Rail"},
+	{ "HPOUTR", NULL, "Headset Rail"},
+	{ "HPOUTL", NULL, "Headset Left Playback" },
+	{ "HPOUTR", NULL, "Headset Right Playback" },
+	{ "EPOUT", NULL, "Earpiece Playback" },
+	{ "Headset Left Playback", NULL, "Headset Left Filter"},
+	{ "Headset Right Playback", NULL, "Headset Right Filter"},
+	{ "Earpiece Playback", NULL, "Headset Left Filter"},
+	{ "Headset Left Filter", NULL, "HSDAC Left"},
+	{ "Headset Right Filter", NULL, "HSDAC Right"},
+
+	/* speaker map */
+	{ "IHFOUTL", NULL, "Speaker Rail"},
+	{ "IHFOUTR", NULL, "Speaker Rail"},
+	{ "IHFOUTL", "NULL", "Speaker Left Playback"},
+	{ "IHFOUTR", "NULL", "Speaker Right Playback"},
+	{ "Speaker Left Playback", NULL, "Speaker Left Filter"},
+	{ "Speaker Right Playback", NULL, "Speaker Right Filter"},
+	{ "Speaker Left Filter", NULL, "IHFDAC Left"},
+	{ "Speaker Right Filter", NULL, "IHFDAC Right"},
+
+	/* vibra map */
+	{ "VIB1OUT", NULL, "Vibra1 Playback"},
+	{ "Vibra1 Playback", NULL, "Vibra1 DAC"},
+
+	{ "VIB2OUT", NULL, "Vibra2 Playback"},
+	{ "Vibra2 Playback", NULL, "Vibra2 DAC"},
+
+	/* lineout */
+	{ "LINEOUTL", NULL, "Lineout Left Playback"},
+	{ "LINEOUTR", NULL, "Lineout Right Playback"},
+	{ "Lineout Left Playback", NULL, "Headset Left Filter"},
+	{ "Lineout Left Playback", NULL, "Speaker Left Filter"},
+	{ "Lineout Left Playback", NULL, "Vibra1 DAC"},
+	{ "Lineout Right Playback", NULL, "Headset Right Filter"},
+	{ "Lineout Right Playback", NULL, "Speaker Right Filter"},
+	{ "Lineout Right Playback", NULL, "Vibra2 DAC"},
+
+	/* Headset (AMIC1) mic */
+	{ "AMIC1Bias", NULL, "AMIC1"},
+	{ "MIC1 Enable", NULL, "AMIC1Bias"},
+	{ "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"},
+
+	/* AMIC2 */
+	{ "AMIC2Bias", NULL, "AMIC2"},
+	{ "MIC2 Enable", NULL, "AMIC2Bias"},
+	{ "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"},
+
+
+	/* Linein */
+	{ "LineIn Enable Left", NULL, "LINEINL"},
+	{ "LineIn Enable Right", NULL, "LINEINR"},
+	{ "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"},
+	{ "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"},
+
+	/* ADC connection */
+	{ "ADC Left", NULL, "Mic_InputL Capture Route"},
+	{ "ADC Right", NULL, "Mic_InputR Capture Route"},
+
+	/*DMIC connections */
+	{ "DMIC1", NULL, "DMIC12supply"},
+	{ "DMIC2", NULL, "DMIC12supply"},
+	{ "DMIC3", NULL, "DMIC34supply"},
+	{ "DMIC4", NULL, "DMIC34supply"},
+	{ "DMIC5", NULL, "DMIC56supply"},
+	{ "DMIC6", NULL, "DMIC56supply"},
+
+	{ "DMIC12Bias", NULL, "DMIC1"},
+	{ "DMIC12Bias", NULL, "DMIC2"},
+	{ "DMIC34Bias", NULL, "DMIC3"},
+	{ "DMIC34Bias", NULL, "DMIC4"},
+	{ "DMIC56Bias", NULL, "DMIC5"},
+	{ "DMIC56Bias", NULL, "DMIC6"},
+
+	/*TX path inputs*/
+	{ "Txpath1 Capture Route", "ADC Left", "ADC Left"},
+	{ "Txpath2 Capture Route", "ADC Left", "ADC Left"},
+	{ "Txpath3 Capture Route", "ADC Left", "ADC Left"},
+	{ "Txpath4 Capture Route", "ADC Left", "ADC Left"},
+	{ "Txpath1 Capture Route", "ADC Right", "ADC Right"},
+	{ "Txpath2 Capture Route", "ADC Right", "ADC Right"},
+	{ "Txpath3 Capture Route", "ADC Right", "ADC Right"},
+	{ "Txpath4 Capture Route", "ADC Right", "ADC Right"},
+	{ "Txpath1 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath2 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath3 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath4 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath1 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath2 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath3 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath4 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath1 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath2 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath3 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath4 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath1 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath2 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath3 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath4 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath1 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath2 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath3 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath4 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath1 Capture Route", "DMIC6", "DMIC6"},
+	{ "Txpath2 Capture Route", "DMIC6", "DMIC6"},
+	{ "Txpath3 Capture Route", "DMIC6", "DMIC6"},
+	{ "Txpath4 Capture Route", "DMIC6", "DMIC6"},
+
+	/* tx path */
+	{ "TX1 Enable", NULL, "Txpath1 Capture Route"},
+	{ "TX2 Enable", NULL, "Txpath2 Capture Route"},
+	{ "TX3 Enable", NULL, "Txpath3 Capture Route"},
+	{ "TX4 Enable", NULL, "Txpath4 Capture Route"},
+	{ "PCM_Out", NULL, "TX1 Enable"},
+	{ "PCM_Out", NULL, "TX2 Enable"},
+	{ "PCM_Out", NULL, "TX3 Enable"},
+	{ "PCM_Out", NULL, "TX4 Enable"},
+
+};
+
+/* speaker and headset mutes, for audio pops and clicks */
+static int sn95031_pcm_hs_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec,
+			SN95031_HSLVOLCTRL, BIT(7), (!mute << 7));
+	snd_soc_update_bits(dai->codec,
+			SN95031_HSRVOLCTRL, BIT(7), (!mute << 7));
+	return 0;
+}
+
+static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec,
+			SN95031_IHFLVOLCTRL, BIT(7), (!mute << 7));
+	snd_soc_update_bits(dai->codec,
+			SN95031_IHFRVOLCTRL, BIT(7), (!mute << 7));
+	return 0;
+}
+
+int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	unsigned int format, rate;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		format = BIT(4)|BIT(5);
+		break;
+
+	case SNDRV_PCM_FORMAT_S24_LE:
+		format = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(dai->codec, SN95031_PCM2C2,
+			BIT(4)|BIT(5), format);
+
+	switch (params_rate(params)) {
+	case 48000:
+		pr_debug("RATE_48000\n");
+		rate = 0;
+		break;
+
+	case 44100:
+		pr_debug("RATE_44100\n");
+		rate = BIT(7);
+		break;
+
+	default:
+		pr_err("ERR rate %d\n", params_rate(params));
+		return -EINVAL;
+	}
+	snd_soc_update_bits(dai->codec, SN95031_PCM1C1, BIT(7), rate);
+
+	return 0;
+}
+
+/* Codec DAI section */
+static struct snd_soc_dai_ops sn95031_headset_dai_ops = {
+	.digital_mute	= sn95031_pcm_hs_mute,
+	.hw_params	= sn95031_pcm_hw_params,
+};
+
+static struct snd_soc_dai_ops sn95031_speaker_dai_ops = {
+	.digital_mute	= sn95031_pcm_spkr_mute,
+	.hw_params	= sn95031_pcm_hw_params,
+};
+
+static struct snd_soc_dai_ops sn95031_vib1_dai_ops = {
+	.hw_params	= sn95031_pcm_hw_params,
+};
+
+static struct snd_soc_dai_ops sn95031_vib2_dai_ops = {
+	.hw_params	= sn95031_pcm_hw_params,
+};
+
+struct snd_soc_dai_driver sn95031_dais[] = {
+{
+	.name = "SN95031 Headset",
+	.playback = {
+		.stream_name = "Headset",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SN95031_RATES,
+		.formats = SN95031_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 5,
+		.rates = SN95031_RATES,
+		.formats = SN95031_FORMATS,
+	},
+	.ops = &sn95031_headset_dai_ops,
+},
+{	.name = "SN95031 Speaker",
+	.playback = {
+		.stream_name = "Speaker",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SN95031_RATES,
+		.formats = SN95031_FORMATS,
+	},
+	.ops = &sn95031_speaker_dai_ops,
+},
+{	.name = "SN95031 Vibra1",
+	.playback = {
+		.stream_name = "Vibra1",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SN95031_RATES,
+		.formats = SN95031_FORMATS,
+	},
+	.ops = &sn95031_vib1_dai_ops,
+},
+{	.name = "SN95031 Vibra2",
+	.playback = {
+		.stream_name = "Vibra2",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SN95031_RATES,
+		.formats = SN95031_FORMATS,
+	},
+	.ops = &sn95031_vib2_dai_ops,
+},
+};
+
+static inline void sn95031_disable_jack_btn(struct snd_soc_codec *codec)
+{
+	snd_soc_write(codec, SN95031_BTNCTRL2, 0x00);
+}
+
+static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec)
+{
+	snd_soc_write(codec, SN95031_BTNCTRL1, 0x77);
+	snd_soc_write(codec, SN95031_BTNCTRL2, 0x01);
+}
+
+static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack)
+{
+	int micbias = sn95031_get_mic_bias(mfld_jack->codec);
+
+	int jack_type = snd_soc_jack_get_type(mfld_jack, micbias);
+
+	pr_debug("jack type detected = %d\n", jack_type);
+	if (jack_type == SND_JACK_HEADSET)
+		sn95031_enable_jack_btn(mfld_jack->codec);
+	return jack_type;
+}
+
+void sn95031_jack_detection(struct mfld_jack_data *jack_data)
+{
+	unsigned int status;
+	unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET;
+
+	pr_debug("interrupt id read in sram = 0x%x\n", jack_data->intr_id);
+	if (jack_data->intr_id & 0x1) {
+		pr_debug("short_push detected\n");
+		status = SND_JACK_HEADSET | SND_JACK_BTN_0;
+	} else if (jack_data->intr_id & 0x2) {
+		pr_debug("long_push detected\n");
+		status = SND_JACK_HEADSET | SND_JACK_BTN_1;
+	} else if (jack_data->intr_id & 0x4) {
+		pr_debug("headset or headphones inserted\n");
+		status = sn95031_get_headset_state(jack_data->mfld_jack);
+	} else if (jack_data->intr_id & 0x8) {
+		pr_debug("headset or headphones removed\n");
+		status = 0;
+		sn95031_disable_jack_btn(jack_data->mfld_jack->codec);
+	} else {
+		pr_err("unidentified interrupt\n");
+		return;
+	}
+
+	snd_soc_jack_report(jack_data->mfld_jack, status, mask);
+	/*button pressed and released so we send explicit button release */
+	if ((status & SND_JACK_BTN_0) | (status & SND_JACK_BTN_1))
+		snd_soc_jack_report(jack_data->mfld_jack,
+				SND_JACK_HEADSET, mask);
+}
+EXPORT_SYMBOL_GPL(sn95031_jack_detection);
+
+/* codec registration */
+static int sn95031_codec_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	pr_debug("codec_probe called\n");
+
+	codec->dapm.bias_level = SND_SOC_BIAS_OFF;
+	codec->dapm.idle_bias_off = 1;
+
+	/* PCM interface config
+	 * This sets the pcm rx slot conguration to max 6 slots
+	 * for max 4 dais (2 stereo and 2 mono)
+	 */
+	snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10);
+	snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32);
+	snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54);
+	snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10);
+	snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32);
+	/* pcm port setting
+	 * This sets the pcm port to slave and clock at 19.2Mhz which
+	 * can support 6slots, sampling rate set per stream in hw-params
+	 */
+	snd_soc_write(codec, SN95031_PCM1C1, 0x00);
+	snd_soc_write(codec, SN95031_PCM2C1, 0x01);
+	snd_soc_write(codec, SN95031_PCM2C2, 0x0A);
+	snd_soc_write(codec, SN95031_HSMIXER, BIT(0)|BIT(4));
+	/* vendor vibra workround, the vibras are muted by
+	 * custom register so unmute them
+	 */
+	snd_soc_write(codec, SN95031_SSR5, 0x80);
+	snd_soc_write(codec, SN95031_SSR6, 0x80);
+	snd_soc_write(codec, SN95031_VIB1C5, 0x00);
+	snd_soc_write(codec, SN95031_VIB2C5, 0x00);
+	/* configure vibras for pcm port */
+	snd_soc_write(codec, SN95031_VIB1C3, 0x00);
+	snd_soc_write(codec, SN95031_VIB2C3, 0x00);
+
+	/* soft mute ramp time */
+	snd_soc_write(codec, SN95031_SOFTMUTE, 0x3);
+	/* fix the initial volume at 1dB,
+	 * default in +9dB,
+	 * 1dB give optimal swing on DAC, amps
+	 */
+	snd_soc_write(codec, SN95031_HSLVOLCTRL, 0x08);
+	snd_soc_write(codec, SN95031_HSRVOLCTRL, 0x08);
+	snd_soc_write(codec, SN95031_IHFLVOLCTRL, 0x08);
+	snd_soc_write(codec, SN95031_IHFRVOLCTRL, 0x08);
+	/* dac mode and lineout workaround */
+	snd_soc_write(codec, SN95031_SSR2, 0x10);
+	snd_soc_write(codec, SN95031_SSR3, 0x40);
+
+	snd_soc_add_controls(codec, sn95031_snd_controls,
+			     ARRAY_SIZE(sn95031_snd_controls));
+
+	ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets,
+				ARRAY_SIZE(sn95031_dapm_widgets));
+	if (ret)
+		pr_err("soc_dapm_new_control failed %d", ret);
+	ret = snd_soc_dapm_add_routes(&codec->dapm, sn95031_audio_map,
+				ARRAY_SIZE(sn95031_audio_map));
+	if (ret)
+		pr_err("soc_dapm_add_routes failed %d", ret);
+
+	return ret;
+}
+
+static int sn95031_codec_remove(struct snd_soc_codec *codec)
+{
+	pr_debug("codec_remove called\n");
+	sn95031_set_vaud_bias(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+struct snd_soc_codec_driver sn95031_codec = {
+	.probe		= sn95031_codec_probe,
+	.remove		= sn95031_codec_remove,
+	.read		= sn95031_read,
+	.write		= sn95031_write,
+	.set_bias_level	= sn95031_set_vaud_bias,
+};
+
+static int __devinit sn95031_device_probe(struct platform_device *pdev)
+{
+	pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev));
+	return snd_soc_register_codec(&pdev->dev, &sn95031_codec,
+			sn95031_dais, ARRAY_SIZE(sn95031_dais));
+}
+
+static int __devexit sn95031_device_remove(struct platform_device *pdev)
+{
+	pr_debug("codec device remove called\n");
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver sn95031_codec_driver = {
+	.driver		= {
+		.name		= "sn95031",
+		.owner		= THIS_MODULE,
+	},
+	.probe		= sn95031_device_probe,
+	.remove		= sn95031_device_remove,
+};
+
+static int __init sn95031_init(void)
+{
+	pr_debug("driver init called\n");
+	return platform_driver_register(&sn95031_codec_driver);
+}
+module_init(sn95031_init);
+
+static void __exit sn95031_exit(void)
+{
+	pr_debug("driver exit called\n");
+	platform_driver_unregister(&sn95031_codec_driver);
+}
+module_exit(sn95031_exit);
+
+MODULE_DESCRIPTION("ASoC TI SN95031 codec driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sn95031");
diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h
new file mode 100644
index 00000000000..20376d234fb
--- /dev/null
+++ b/sound/soc/codecs/sn95031.h
@@ -0,0 +1,132 @@
+/*
+ *  sn95031.h - TI sn95031 Codec driver
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *
+ */
+#ifndef _SN95031_H
+#define _SN95031_H
+
+/*register map*/
+#define SN95031_VAUD			0xDB
+#define SN95031_VHSP			0xDC
+#define SN95031_VHSN			0xDD
+#define SN95031_VIHF			0xC9
+
+#define SN95031_AUDPLLCTRL		0x240
+#define SN95031_DMICBUF0123		0x241
+#define SN95031_DMICBUF45		0x242
+#define SN95031_DMICGPO			0x244
+#define SN95031_DMICMUX			0x245
+#define SN95031_DMICLK			0x246
+#define SN95031_MICBIAS			0x247
+#define SN95031_ADCCONFIG		0x248
+#define SN95031_MICAMP1			0x249
+#define SN95031_MICAMP2			0x24A
+#define SN95031_NOISEMUX		0x24B
+#define SN95031_AUDIOMUX12		0x24C
+#define SN95031_AUDIOMUX34		0x24D
+#define SN95031_AUDIOSINC		0x24E
+#define SN95031_AUDIOTXEN		0x24F
+#define SN95031_HSEPRXCTRL		0x250
+#define SN95031_IHFRXCTRL		0x251
+#define SN95031_HSMIXER			0x256
+#define SN95031_DACCONFIG		0x257
+#define SN95031_SOFTMUTE		0x258
+#define SN95031_HSLVOLCTRL		0x259
+#define SN95031_HSRVOLCTRL		0x25A
+#define SN95031_IHFLVOLCTRL		0x25B
+#define SN95031_IHFRVOLCTRL		0x25C
+#define SN95031_DRIVEREN		0x25D
+#define SN95031_LOCTL			0x25E
+#define SN95031_VIB1C1			0x25F
+#define SN95031_VIB1C2			0x260
+#define SN95031_VIB1C3			0x261
+#define SN95031_VIB1SPIPCM1		0x262
+#define SN95031_VIB1SPIPCM2		0x263
+#define SN95031_VIB1C5			0x264
+#define SN95031_VIB2C1			0x265
+#define SN95031_VIB2C2			0x266
+#define SN95031_VIB2C3			0x267
+#define SN95031_VIB2SPIPCM1		0x268
+#define SN95031_VIB2SPIPCM2		0x269
+#define SN95031_VIB2C5			0x26A
+#define SN95031_BTNCTRL1		0x26B
+#define SN95031_BTNCTRL2		0x26C
+#define SN95031_PCM1TXSLOT01		0x26D
+#define SN95031_PCM1TXSLOT23		0x26E
+#define SN95031_PCM1TXSLOT45		0x26F
+#define SN95031_PCM1RXSLOT0_3		0x270
+#define SN95031_PCM1RXSLOT45		0x271
+#define SN95031_PCM2TXSLOT01		0x272
+#define SN95031_PCM2TXSLOT23		0x273
+#define SN95031_PCM2TXSLOT45		0x274
+#define SN95031_PCM2RXSLOT01		0x275
+#define SN95031_PCM2RXSLOT23		0x276
+#define SN95031_PCM2RXSLOT45		0x277
+#define SN95031_PCM1C1			0x278
+#define SN95031_PCM1C2			0x279
+#define SN95031_PCM1C3			0x27A
+#define SN95031_PCM2C1			0x27B
+#define SN95031_PCM2C2			0x27C
+/*end codec register defn*/
+
+/*vendor defn these are not part of avp*/
+#define SN95031_SSR2			0x381
+#define SN95031_SSR3			0x382
+#define SN95031_SSR5			0x384
+#define SN95031_SSR6			0x385
+
+/* ADC registers */
+
+#define SN95031_ADC1CNTL1 0x1C0
+#define SN95031_ADC_ENBL 0x10
+#define SN95031_ADC_START 0x08
+#define SN95031_ADC1CNTL3 0x1C2
+#define SN95031_ADCTHERM_ENBL 0x04
+#define SN95031_ADCRRDATA_ENBL 0x05
+#define SN95031_STOPBIT_MASK 16
+#define SN95031_ADCTHERM_MASK 4
+#define SN95031_ADC_CHANLS_MAX 15 /* Number of ADC channels */
+#define SN95031_ADC_LOOP_MAX (SN95031_ADC_CHANLS_MAX - 1)
+#define SN95031_ADC_NO_LOOP 0x07
+#define SN95031_AUDIO_GPIO_CTRL 0x070
+
+/* ADC channel code values */
+#define SN95031_AUDIO_DETECT_CODE 0x06
+
+/* ADC base addresses */
+#define SN95031_ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */
+#define SN95031_ADC_DATA_START_ADDR 0x1D4  /* increments by 2 */
+/* multipier to convert to mV */
+#define SN95031_ADC_ONE_LSB_MULTIPLIER 2346
+
+
+struct mfld_jack_data {
+	int intr_id;
+	int micbias_vol;
+	struct snd_soc_jack *mfld_jack;
+};
+
+extern void sn95031_jack_detection(struct mfld_jack_data *jack_data);
+
+#endif
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
new file mode 100644
index 00000000000..e93b9d1ae1d
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -0,0 +1,794 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4.c
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/cdev.h>
+#include <linux/slab.h>
+
+#include <sound/tlv320aic32x4.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "tlv320aic32x4.h"
+
+struct aic32x4_rate_divs {
+	u32 mclk;
+	u32 rate;
+	u8 p_val;
+	u8 pll_j;
+	u16 pll_d;
+	u16 dosr;
+	u8 ndac;
+	u8 mdac;
+	u8 aosr;
+	u8 nadc;
+	u8 madc;
+	u8 blck_N;
+};
+
+struct aic32x4_priv {
+	u32 sysclk;
+	s32 master;
+	u8 page_no;
+	void *control_data;
+	u32 power_cfg;
+	u32 micpga_routing;
+	bool swapdacs;
+};
+
+/* 0dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0);
+/* 0dB min, 0.5dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0);
+
+static const struct snd_kcontrol_new aic32x4_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
+			AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5),
+	SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN,
+			AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1),
+	SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN,
+			AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1),
+	SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
+			AIC32X4_HPRGAIN, 6, 0x01, 1),
+	SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN,
+			AIC32X4_LORGAIN, 6, 0x01, 1),
+	SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL,
+			AIC32X4_RMICPGAVOL, 7, 0x01, 1),
+
+	SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0),
+	SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0),
+
+	SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL,
+			AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5),
+	SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL,
+			AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5),
+
+	SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+
+	SOC_SINGLE("AGC Left Switch", AIC32X4_LAGC1, 7, 1, 0),
+	SOC_SINGLE("AGC Right Switch", AIC32X4_RAGC1, 7, 1, 0),
+	SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1,
+			4, 0x07, 0),
+	SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1,
+			0, 0x03, 0),
+	SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2,
+			6, 0x03, 0),
+	SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2,
+			1, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3,
+			0, 0x7F, 0),
+	SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4,
+			3, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5,
+			3, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6,
+			0, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7,
+			0, 0x0F, 0),
+};
+
+static const struct aic32x4_rate_divs aic32x4_divs[] = {
+	/* 8k rate */
+	{AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24},
+	{AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24},
+	{AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24},
+	/* 11.025k rate */
+	{AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16},
+	{AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16},
+	/* 16k rate */
+	{AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12},
+	{AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12},
+	{AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12},
+	/* 22.05k rate */
+	{AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8},
+	{AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8},
+	{AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8},
+	/* 32k rate */
+	{AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6},
+	{AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6},
+	/* 44.1k rate */
+	{AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4},
+	{AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4},
+	{AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4},
+	/* 48k rate */
+	{AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4},
+	{AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4},
+	{AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}
+};
+
+static const struct snd_kcontrol_new hpl_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0),
+	SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new hpr_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_HPRROUTE, 3, 1, 0),
+	SOC_DAPM_SINGLE("IN1_R Switch", AIC32X4_HPRROUTE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new lol_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_LOLROUTE, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new lor_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_input_mixer_controls[] = {
+	SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0),
+	SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0),
+	SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_input_mixer_controls[] = {
+	SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0),
+	SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0),
+	SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0),
+	SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &hpl_output_mixer_controls[0],
+			   ARRAY_SIZE(hpl_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0),
+
+	SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &lol_output_mixer_controls[0],
+			   ARRAY_SIZE(lol_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0),
+
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0),
+	SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &hpr_output_mixer_controls[0],
+			   ARRAY_SIZE(hpr_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &lor_output_mixer_controls[0],
+			   ARRAY_SIZE(lor_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
+			   &left_input_mixer_controls[0],
+			   ARRAY_SIZE(left_input_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
+			   &right_input_mixer_controls[0],
+			   ARRAY_SIZE(right_input_mixer_controls)),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0),
+
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+	SND_SOC_DAPM_OUTPUT("LOL"),
+	SND_SOC_DAPM_OUTPUT("LOR"),
+	SND_SOC_DAPM_INPUT("IN1_L"),
+	SND_SOC_DAPM_INPUT("IN1_R"),
+	SND_SOC_DAPM_INPUT("IN2_L"),
+	SND_SOC_DAPM_INPUT("IN2_R"),
+	SND_SOC_DAPM_INPUT("IN3_L"),
+	SND_SOC_DAPM_INPUT("IN3_R"),
+};
+
+static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
+	/* Left Output */
+	{"HPL Output Mixer", "L_DAC Switch", "Left DAC"},
+	{"HPL Output Mixer", "IN1_L Switch", "IN1_L"},
+
+	{"HPL Power", NULL, "HPL Output Mixer"},
+	{"HPL", NULL, "HPL Power"},
+
+	{"LOL Output Mixer", "L_DAC Switch", "Left DAC"},
+
+	{"LOL Power", NULL, "LOL Output Mixer"},
+	{"LOL", NULL, "LOL Power"},
+
+	/* Right Output */
+	{"HPR Output Mixer", "R_DAC Switch", "Right DAC"},
+	{"HPR Output Mixer", "IN1_R Switch", "IN1_R"},
+
+	{"HPR Power", NULL, "HPR Output Mixer"},
+	{"HPR", NULL, "HPR Power"},
+
+	{"LOR Output Mixer", "R_DAC Switch", "Right DAC"},
+
+	{"LOR Power", NULL, "LOR Output Mixer"},
+	{"LOR", NULL, "LOR Power"},
+
+	/* Left input */
+	{"Left Input Mixer", "IN1_L P Switch", "IN1_L"},
+	{"Left Input Mixer", "IN2_L P Switch", "IN2_L"},
+	{"Left Input Mixer", "IN3_L P Switch", "IN3_L"},
+
+	{"Left ADC", NULL, "Left Input Mixer"},
+
+	/* Right Input */
+	{"Right Input Mixer", "IN1_R P Switch", "IN1_R"},
+	{"Right Input Mixer", "IN2_R P Switch", "IN2_R"},
+	{"Right Input Mixer", "IN3_R P Switch", "IN3_R"},
+
+	{"Right ADC", NULL, "Right Input Mixer"},
+};
+
+static inline int aic32x4_change_page(struct snd_soc_codec *codec,
+					unsigned int new_page)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 data[2];
+	int ret;
+
+	data[0] = 0x00;
+	data[1] = new_page & 0xff;
+
+	ret = codec->hw_write(codec->control_data, data, 2);
+	if (ret == 2) {
+		aic32x4->page_no = new_page;
+		return 0;
+	} else {
+		return ret;
+	}
+}
+
+static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg,
+				unsigned int val)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	unsigned int page = reg / 128;
+	unsigned int fixed_reg = reg % 128;
+	u8 data[2];
+	int ret;
+
+	/* A write to AIC32X4_PSEL is really a non-explicit page change */
+	if (reg == AIC32X4_PSEL)
+		return aic32x4_change_page(codec, val);
+
+	if (aic32x4->page_no != page) {
+		ret = aic32x4_change_page(codec, page);
+		if (ret != 0)
+			return ret;
+	}
+
+	data[0] = fixed_reg & 0xff;
+	data[1] = val & 0xff;
+
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	unsigned int page = reg / 128;
+	unsigned int fixed_reg = reg % 128;
+	int ret;
+
+	if (aic32x4->page_no != page) {
+		ret = aic32x4_change_page(codec, page);
+		if (ret != 0)
+			return ret;
+	}
+	return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff);
+}
+
+static inline int aic32x4_get_divs(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) {
+		if ((aic32x4_divs[i].rate == rate)
+		    && (aic32x4_divs[i].mclk == mclk)) {
+			return i;
+		}
+	}
+	printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n");
+	return -EINVAL;
+}
+
+static int aic32x4_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
+				  ARRAY_SIZE(aic32x4_dapm_widgets));
+
+	snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
+				ARRAY_SIZE(aic32x4_dapm_routes));
+
+	snd_soc_dapm_new_widgets(&codec->dapm);
+	return 0;
+}
+
+static int aic32x4_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 aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+
+	switch (freq) {
+	case AIC32X4_FREQ_12000000:
+	case AIC32X4_FREQ_24000000:
+	case AIC32X4_FREQ_25000000:
+		aic32x4->sysclk = freq;
+		return 0;
+	}
+	printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n");
+	return -EINVAL;
+}
+
+static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 iface_reg_1;
+	u8 iface_reg_2;
+	u8 iface_reg_3;
+
+	iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1);
+	iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2);
+	iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2);
+	iface_reg_2 = 0;
+	iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3);
+	iface_reg_3 = iface_reg_3 & ~(1 << 3);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aic32x4->master = 1;
+		iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		aic32x4->master = 0;
+		break;
+	default:
+		printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n");
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT);
+		iface_reg_3 |= (1 << 3); /* invert bit clock */
+		iface_reg_2 = 0x01; /* add offset 1 */
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT);
+		iface_reg_3 |= (1 << 3); /* invert bit clock */
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface_reg_1 |=
+			(AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg_1 |=
+			(AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT);
+		break;
+	default:
+		printk(KERN_ERR "aic32x4: invalid DAI interface format\n");
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1);
+	snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2);
+	snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3);
+	return 0;
+}
+
+static int aic32x4_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 data;
+	int i;
+
+	i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params));
+	if (i < 0) {
+		printk(KERN_ERR "aic32x4: sampling rate not supported\n");
+		return i;
+	}
+
+	/* Use PLL as CODEC_CLKIN and DAC_MOD_CLK as BDIV_CLKIN */
+	snd_soc_write(codec, AIC32X4_CLKMUX, AIC32X4_PLLCLKIN);
+	snd_soc_write(codec, AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK);
+
+	/* We will fix R value to 1 and will make P & J=K.D as varialble */
+	data = snd_soc_read(codec, AIC32X4_PLLPR);
+	data &= ~(7 << 4);
+	snd_soc_write(codec, AIC32X4_PLLPR,
+		      (data | (aic32x4_divs[i].p_val << 4) | 0x01));
+
+	snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j);
+
+	snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8));
+	snd_soc_write(codec, AIC32X4_PLLDLSB,
+		      (aic32x4_divs[i].pll_d & 0xff));
+
+	/* NDAC divider value */
+	data = snd_soc_read(codec, AIC32X4_NDAC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac);
+
+	/* MDAC divider value */
+	data = snd_soc_read(codec, AIC32X4_MDAC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac);
+
+	/* DOSR MSB & LSB values */
+	snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8);
+	snd_soc_write(codec, AIC32X4_DOSRLSB,
+		      (aic32x4_divs[i].dosr & 0xff));
+
+	/* NADC divider value */
+	data = snd_soc_read(codec, AIC32X4_NADC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc);
+
+	/* MADC divider value */
+	data = snd_soc_read(codec, AIC32X4_MADC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc);
+
+	/* AOSR value */
+	snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr);
+
+	/* BCLK N divider */
+	data = snd_soc_read(codec, AIC32X4_BCLKN);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N);
+
+	data = snd_soc_read(codec, AIC32X4_IFACE1);
+	data = data & ~(3 << 4);
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT);
+		break;
+	}
+	snd_soc_write(codec, AIC32X4_IFACE1, data);
+
+	return 0;
+}
+
+static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 dac_reg;
+
+	dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON;
+	if (mute)
+		snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON);
+	else
+		snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg);
+	return 0;
+}
+
+static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 value;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		if (aic32x4->master) {
+			/* Switch on PLL */
+			value = snd_soc_read(codec, AIC32X4_PLLPR);
+			snd_soc_write(codec, AIC32X4_PLLPR,
+				      (value | AIC32X4_PLLEN));
+
+			/* Switch on NDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_NDAC);
+			snd_soc_write(codec, AIC32X4_NDAC,
+				      value | AIC32X4_NDACEN);
+
+			/* Switch on MDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_MDAC);
+			snd_soc_write(codec, AIC32X4_MDAC,
+				      value | AIC32X4_MDACEN);
+
+			/* Switch on NADC Divider */
+			value = snd_soc_read(codec, AIC32X4_NADC);
+			snd_soc_write(codec, AIC32X4_NADC,
+				      value | AIC32X4_MDACEN);
+
+			/* Switch on MADC Divider */
+			value = snd_soc_read(codec, AIC32X4_MADC);
+			snd_soc_write(codec, AIC32X4_MADC,
+				      value | AIC32X4_MDACEN);
+
+			/* Switch on BCLK_N Divider */
+			value = snd_soc_read(codec, AIC32X4_BCLKN);
+			snd_soc_write(codec, AIC32X4_BCLKN,
+				      value | AIC32X4_BCLKEN);
+		}
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (aic32x4->master) {
+			/* Switch off PLL */
+			value = snd_soc_read(codec, AIC32X4_PLLPR);
+			snd_soc_write(codec, AIC32X4_PLLPR,
+				      (value & ~AIC32X4_PLLEN));
+
+			/* Switch off NDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_NDAC);
+			snd_soc_write(codec, AIC32X4_NDAC,
+				      value & ~AIC32X4_NDACEN);
+
+			/* Switch off MDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_MDAC);
+			snd_soc_write(codec, AIC32X4_MDAC,
+				      value & ~AIC32X4_MDACEN);
+
+			/* Switch off NADC Divider */
+			value = snd_soc_read(codec, AIC32X4_NADC);
+			snd_soc_write(codec, AIC32X4_NADC,
+				      value & ~AIC32X4_NDACEN);
+
+			/* Switch off MADC Divider */
+			value = snd_soc_read(codec, AIC32X4_MADC);
+			snd_soc_write(codec, AIC32X4_MADC,
+				      value & ~AIC32X4_MDACEN);
+			value = snd_soc_read(codec, AIC32X4_BCLKN);
+
+			/* Switch off BCLK_N Divider */
+			snd_soc_write(codec, AIC32X4_BCLKN,
+				      value & ~AIC32X4_BCLKEN);
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define AIC32X4_RATES	SNDRV_PCM_RATE_8000_48000
+#define AIC32X4_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops aic32x4_ops = {
+	.hw_params = aic32x4_hw_params,
+	.digital_mute = aic32x4_mute,
+	.set_fmt = aic32x4_set_dai_fmt,
+	.set_sysclk = aic32x4_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver aic32x4_dai = {
+	.name = "tlv320aic32x4-hifi",
+	.playback = {
+		     .stream_name = "Playback",
+		     .channels_min = 1,
+		     .channels_max = 2,
+		     .rates = AIC32X4_RATES,
+		     .formats = AIC32X4_FORMATS,},
+	.capture = {
+		    .stream_name = "Capture",
+		    .channels_min = 1,
+		    .channels_max = 2,
+		    .rates = AIC32X4_RATES,
+		    .formats = AIC32X4_FORMATS,},
+	.ops = &aic32x4_ops,
+	.symmetric_rates = 1,
+};
+
+static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int aic32x4_resume(struct snd_soc_codec *codec)
+{
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int aic32x4_probe(struct snd_soc_codec *codec)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u32 tmp_reg;
+
+	codec->hw_write = (hw_write_t) i2c_master_send;
+	codec->control_data = aic32x4->control_data;
+
+	snd_soc_write(codec, AIC32X4_RESET, 0x01);
+
+	/* Power platform configuration */
+	if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) {
+		snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN |
+						      AIC32X4_MICBIAS_2075V);
+	}
+	if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
+		snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
+	}
+	if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) {
+		snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN);
+	}
+	tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
+	if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
+		tmp_reg |= AIC32X4_LDOIN_18_36;
+	}
+	if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) {
+		tmp_reg |= AIC32X4_LDOIN2HP;
+	}
+	snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg);
+
+	/* Do DACs need to be swapped? */
+	if (aic32x4->swapdacs) {
+		snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN);
+	} else {
+		snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN);
+	}
+
+	/* Mic PGA routing */
+	if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) {
+		snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K);
+	}
+	if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) {
+		snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K);
+	}
+
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_add_controls(codec, aic32x4_snd_controls,
+			     ARRAY_SIZE(aic32x4_snd_controls));
+	aic32x4_add_widgets(codec);
+
+	return 0;
+}
+
+static int aic32x4_remove(struct snd_soc_codec *codec)
+{
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
+	.read = aic32x4_read,
+	.write = aic32x4_write,
+	.probe = aic32x4_probe,
+	.remove = aic32x4_remove,
+	.suspend = aic32x4_suspend,
+	.resume = aic32x4_resume,
+	.set_bias_level = aic32x4_set_bias_level,
+};
+
+static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct aic32x4_pdata *pdata = i2c->dev.platform_data;
+	struct aic32x4_priv *aic32x4;
+	int ret;
+
+	aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL);
+	if (aic32x4 == NULL)
+		return -ENOMEM;
+
+	aic32x4->control_data = i2c;
+	i2c_set_clientdata(i2c, aic32x4);
+
+	if (pdata) {
+		aic32x4->power_cfg = pdata->power_cfg;
+		aic32x4->swapdacs = pdata->swapdacs;
+		aic32x4->micpga_routing = pdata->micpga_routing;
+	} else {
+		aic32x4->power_cfg = 0;
+		aic32x4->swapdacs = false;
+		aic32x4->micpga_routing = 0;
+	}
+
+	ret = snd_soc_register_codec(&i2c->dev,
+			&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
+	if (ret < 0)
+		kfree(aic32x4);
+	return ret;
+}
+
+static __devexit int aic32x4_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(i2c_get_clientdata(client));
+	return 0;
+}
+
+static const struct i2c_device_id aic32x4_i2c_id[] = {
+	{ "tlv320aic32x4", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+
+static struct i2c_driver aic32x4_i2c_driver = {
+	.driver = {
+		.name = "tlv320aic32x4",
+		.owner = THIS_MODULE,
+	},
+	.probe =    aic32x4_i2c_probe,
+	.remove =   __devexit_p(aic32x4_i2c_remove),
+	.id_table = aic32x4_i2c_id,
+};
+
+static int __init aic32x4_modinit(void)
+{
+	int ret = 0;
+
+	ret = i2c_add_driver(&aic32x4_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n",
+		       ret);
+	}
+	return ret;
+}
+module_init(aic32x4_modinit);
+
+static void __exit aic32x4_exit(void)
+{
+	i2c_del_driver(&aic32x4_i2c_driver);
+}
+module_exit(aic32x4_exit);
+
+MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver");
+MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
new file mode 100644
index 00000000000..aae2b244039
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -0,0 +1,143 @@
+/*
+ * tlv320aic32x4.h
+ *
+ * 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 _TLV320AIC32X4_H
+#define _TLV320AIC32X4_H
+
+/* tlv320aic32x4 register space (in decimal to match datasheet) */
+
+#define AIC32X4_PAGE1		128
+
+#define	AIC32X4_PSEL		0
+#define	AIC32X4_RESET		1
+#define	AIC32X4_CLKMUX		4
+#define	AIC32X4_PLLPR		5
+#define	AIC32X4_PLLJ		6
+#define	AIC32X4_PLLDMSB		7
+#define	AIC32X4_PLLDLSB		8
+#define	AIC32X4_NDAC		11
+#define	AIC32X4_MDAC		12
+#define AIC32X4_DOSRMSB		13
+#define AIC32X4_DOSRLSB		14
+#define	AIC32X4_NADC		18
+#define	AIC32X4_MADC		19
+#define AIC32X4_AOSR		20
+#define AIC32X4_CLKMUX2		25
+#define AIC32X4_CLKOUTM		26
+#define AIC32X4_IFACE1		27
+#define AIC32X4_IFACE2		28
+#define AIC32X4_IFACE3		29
+#define AIC32X4_BCLKN		30
+#define AIC32X4_IFACE4		31
+#define AIC32X4_IFACE5		32
+#define AIC32X4_IFACE6		33
+#define AIC32X4_DOUTCTL		53
+#define AIC32X4_DINCTL		54
+#define AIC32X4_DACSPB		60
+#define AIC32X4_ADCSPB		61
+#define AIC32X4_DACSETUP	63
+#define AIC32X4_DACMUTE		64
+#define AIC32X4_LDACVOL		65
+#define AIC32X4_RDACVOL		66
+#define AIC32X4_ADCSETUP	81
+#define	AIC32X4_ADCFGA		82
+#define AIC32X4_LADCVOL		83
+#define AIC32X4_RADCVOL		84
+#define AIC32X4_LAGC1		86
+#define AIC32X4_LAGC2		87
+#define AIC32X4_LAGC3		88
+#define AIC32X4_LAGC4		89
+#define AIC32X4_LAGC5		90
+#define AIC32X4_LAGC6		91
+#define AIC32X4_LAGC7		92
+#define AIC32X4_RAGC1		94
+#define AIC32X4_RAGC2		95
+#define AIC32X4_RAGC3		96
+#define AIC32X4_RAGC4		97
+#define AIC32X4_RAGC5		98
+#define AIC32X4_RAGC6		99
+#define AIC32X4_RAGC7		100
+#define AIC32X4_PWRCFG		(AIC32X4_PAGE1 + 1)
+#define AIC32X4_LDOCTL		(AIC32X4_PAGE1 + 2)
+#define AIC32X4_OUTPWRCTL	(AIC32X4_PAGE1 + 9)
+#define AIC32X4_CMMODE		(AIC32X4_PAGE1 + 10)
+#define AIC32X4_HPLROUTE	(AIC32X4_PAGE1 + 12)
+#define AIC32X4_HPRROUTE	(AIC32X4_PAGE1 + 13)
+#define AIC32X4_LOLROUTE	(AIC32X4_PAGE1 + 14)
+#define AIC32X4_LORROUTE	(AIC32X4_PAGE1 + 15)
+#define	AIC32X4_HPLGAIN		(AIC32X4_PAGE1 + 16)
+#define	AIC32X4_HPRGAIN		(AIC32X4_PAGE1 + 17)
+#define	AIC32X4_LOLGAIN		(AIC32X4_PAGE1 + 18)
+#define	AIC32X4_LORGAIN		(AIC32X4_PAGE1 + 19)
+#define AIC32X4_HEADSTART	(AIC32X4_PAGE1 + 20)
+#define AIC32X4_MICBIAS		(AIC32X4_PAGE1 + 51)
+#define AIC32X4_LMICPGAPIN	(AIC32X4_PAGE1 + 52)
+#define AIC32X4_LMICPGANIN	(AIC32X4_PAGE1 + 54)
+#define AIC32X4_RMICPGAPIN	(AIC32X4_PAGE1 + 55)
+#define AIC32X4_RMICPGANIN	(AIC32X4_PAGE1 + 57)
+#define AIC32X4_FLOATINGINPUT	(AIC32X4_PAGE1 + 58)
+#define AIC32X4_LMICPGAVOL	(AIC32X4_PAGE1 + 59)
+#define AIC32X4_RMICPGAVOL	(AIC32X4_PAGE1 + 60)
+
+#define AIC32X4_FREQ_12000000 12000000
+#define AIC32X4_FREQ_24000000 24000000
+#define AIC32X4_FREQ_25000000 25000000
+
+#define AIC32X4_WORD_LEN_16BITS		0x00
+#define AIC32X4_WORD_LEN_20BITS		0x01
+#define AIC32X4_WORD_LEN_24BITS		0x02
+#define AIC32X4_WORD_LEN_32BITS		0x03
+
+#define AIC32X4_I2S_MODE		0x00
+#define AIC32X4_DSP_MODE		0x01
+#define AIC32X4_RIGHT_JUSTIFIED_MODE	0x02
+#define AIC32X4_LEFT_JUSTIFIED_MODE	0x03
+
+#define AIC32X4_AVDDWEAKDISABLE		0x08
+#define AIC32X4_LDOCTLEN		0x01
+
+#define AIC32X4_LDOIN_18_36		0x01
+#define AIC32X4_LDOIN2HP		0x02
+
+#define AIC32X4_DACSPBLOCK_MASK		0x1f
+#define AIC32X4_ADCSPBLOCK_MASK		0x1f
+
+#define AIC32X4_PLLJ_SHIFT		6
+#define AIC32X4_DOSRMSB_SHIFT		4
+
+#define AIC32X4_PLLCLKIN		0x03
+
+#define AIC32X4_MICBIAS_LDOIN		0x08
+#define AIC32X4_MICBIAS_2075V		0x60
+
+#define AIC32X4_LMICPGANIN_IN2R_10K	0x10
+#define AIC32X4_RMICPGANIN_IN1L_10K	0x10
+
+#define AIC32X4_LMICPGAVOL_NOGAIN	0x80
+#define AIC32X4_RMICPGAVOL_NOGAIN	0x80
+
+#define AIC32X4_BCLKMASTER		0x08
+#define AIC32X4_WCLKMASTER		0x04
+#define AIC32X4_PLLEN			(0x01 << 7)
+#define AIC32X4_NDACEN			(0x01 << 7)
+#define AIC32X4_MDACEN			(0x01 << 7)
+#define AIC32X4_NADCEN			(0x01 << 7)
+#define AIC32X4_MADCEN			(0x01 << 7)
+#define AIC32X4_BCLKEN			(0x01 << 7)
+#define AIC32X4_DACEN			(0x03 << 6)
+#define AIC32X4_RDAC2LCHN		(0x02 << 2)
+#define AIC32X4_LDAC2RCHN		(0x02 << 4)
+#define AIC32X4_LDAC2LCHN		(0x01 << 4)
+#define AIC32X4_RDAC2RCHN		(0x01 << 2)
+
+#define AIC32X4_SSTEP2WCLK		0x01
+#define AIC32X4_MUTEON			0x0C
+#define	AIC32X4_DACMOD2BCLK		0x01
+
+#endif				/* _TLV320AIC32X4_H */
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 71d7be8ac48..00b6d87e7bd 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1615,6 +1615,7 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = {
 	},
 	{ },
 };
+MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id);
 
 static struct i2c_driver tlv320dac33_i2c_driver = {
 	.driver = {
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index e4d464b937d..8512800f632 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -26,6 +26,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/mfd/core.h>
 #include <linux/i2c/twl.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -732,7 +733,8 @@ static int aif_event(struct snd_soc_dapm_widget *w,
 
 static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 {
-	struct twl4030_codec_audio_data *pdata = codec->dev->platform_data;
+	struct twl4030_codec_audio_data *pdata =
+			mfd_get_data(to_platform_device(codec->dev));
 	unsigned char hs_gain, hs_pop;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	/* Base values for ramp delay calculation: 2^19 - 2^26 */
@@ -2297,7 +2299,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
 
 static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 {
-	struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data;
+	struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev);
 
 	if (!pdata) {
 		dev_err(&pdev->dev, "platform_data is missing\n");
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 4bbf1b15a49..482fcdb59bf 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -724,8 +724,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-void twl6040_hs_jack_report(struct snd_soc_codec *codec,
-				struct snd_soc_jack *jack, int report)
+static void twl6040_hs_jack_report(struct snd_soc_codec *codec,
+				   struct snd_soc_jack *jack, int report)
 {
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	int status;
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index e76847a9438..48ffd406a71 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -486,7 +486,8 @@ static struct snd_soc_dai_driver uda134x_dai = {
 static int uda134x_soc_probe(struct snd_soc_codec *codec)
 {
 	struct uda134x_priv *uda134x;
-	struct uda134x_platform_data *pd = dev_get_drvdata(codec->card->dev);
+	struct uda134x_platform_data *pd = codec->card->dev->platform_data;
+
 	int ret;
 
 	printk(KERN_INFO "UDA134X SoC Audio Codec\n");
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 861b28f543d..c8a874d0d4c 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -3,7 +3,7 @@
  *
  * Author:      Matti Aaltonen, <matti.j.aaltonen@nokia.com>
  *
- * Copyright:   (C) 2010 Nokia Corporation
+ * Copyright:   (C) 2010, 2011 Nokia Corporation
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -179,7 +179,12 @@ static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
+/*
+ * TODO: Implement the audio routing in the driver. Now this control
+ * only indicates the setting that has been done elsewhere (in the user
+ * space).
+ */
+static const char * const wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
 
 static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
@@ -239,7 +244,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const char *wl1273_audio_strings[] = { "Digital", "Analog" };
+static const char * const wl1273_audio_strings[] = { "Digital", "Analog" };
 
 static const struct soc_enum wl1273_audio_enum =
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
@@ -436,7 +441,8 @@ EXPORT_SYMBOL_GPL(wl1273_get_format);
 
 static int wl1273_probe(struct snd_soc_codec *codec)
 {
-	struct wl1273_core **core = codec->dev->platform_data;
+	struct wl1273_core **core =
+			mfd_get_data(to_platform_device(codec->dev));
 	struct wl1273_priv *wl1273;
 	int r;
 
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 80ddf4fd23d..a3b9cbb20ee 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -836,24 +836,25 @@ static void wm2000_i2c_shutdown(struct i2c_client *i2c)
 }
 
 #ifdef CONFIG_PM
-static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg)
+static int wm2000_i2c_suspend(struct device *dev)
 {
+	struct i2c_client *i2c = to_i2c_client(dev);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
 
 	return wm2000_anc_transition(wm2000, ANC_OFF);
 }
 
-static int wm2000_i2c_resume(struct i2c_client *i2c)
+static int wm2000_i2c_resume(struct device *dev)
 {
+	struct i2c_client *i2c = to_i2c_client(dev);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
 
 	return wm2000_anc_set_mode(wm2000);
 }
-#else
-#define wm2000_i2c_suspend NULL
-#define wm2000_i2c_resume NULL
 #endif
 
+static SIMPLE_DEV_PM_OPS(wm2000_pm, wm2000_i2c_suspend, wm2000_i2c_resume);
+
 static const struct i2c_device_id wm2000_i2c_id[] = {
 	{ "wm2000", 0 },
 	{ }
@@ -864,11 +865,10 @@ static struct i2c_driver wm2000_i2c_driver = {
 	.driver = {
 		.name = "wm2000",
 		.owner = THIS_MODULE,
+		.pm = &wm2000_pm,
 	},
 	.probe = wm2000_i2c_probe,
 	.remove = __devexit_p(wm2000_i2c_remove),
-	.suspend = wm2000_i2c_suspend,
-	.resume = wm2000_i2c_resume,
 	.shutdown = wm2000_i2c_shutdown,
 	.id_table = wm2000_i2c_id,
 };
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 3c3bc079167..736b785e375 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -22,6 +22,7 @@
 #include <linux/regulator/consumer.h>
 #include <linux/mfd/wm8400-audio.h>
 #include <linux/mfd/wm8400-private.h>
+#include <linux/mfd/core.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -1377,7 +1378,7 @@ static void wm8400_probe_deferred(struct work_struct *work)
 
 static int wm8400_codec_probe(struct snd_soc_codec *codec)
 {
-	struct wm8400 *wm8400 = dev_get_platdata(codec->dev);
+	struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev));
 	struct wm8400_priv *priv;
 	int ret;
 	u16 reg;
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 5eb2f501ce3..4fd4d8dca0f 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -58,7 +58,7 @@ static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = {
 	0x0000,     /* R8 - ZERO_DETECT */
 };
 
-static int wm8523_volatile_register(unsigned int reg)
+static int wm8523_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM8523_DEVICE_ID:
@@ -414,7 +414,6 @@ static int wm8523_resume(struct snd_soc_codec *codec)
 static int wm8523_probe(struct snd_soc_codec *codec)
 {
 	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
-	u16 *reg_cache = codec->reg_cache;
 	int ret, i;
 
 	codec->hw_write = (hw_write_t)i2c_master_send;
@@ -471,8 +470,9 @@ static int wm8523_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Change some default settings - latch VU and enable ZC */
-	reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU;
-	reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC;
+	snd_soc_update_bits(codec, WM8523_DAC_GAINR,
+			    WM8523_DACR_VU, WM8523_DACR_VU);
+	snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC);
 
 	wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 494f2d31d75..25af901fe81 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -421,7 +421,6 @@ static int wm8741_resume(struct snd_soc_codec *codec)
 static int wm8741_probe(struct snd_soc_codec *codec)
 {
 	struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
-	u16 *reg_cache = codec->reg_cache;
 	int ret = 0;
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type);
@@ -437,10 +436,14 @@ static int wm8741_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Change some default settings - latch VU */
-	reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL;
-	reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM;
-	reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL;
-	reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM;
+	snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION,
+			    WM8741_UPDATELL, WM8741_UPDATELL);
+	snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION,
+			    WM8741_UPDATELM, WM8741_UPDATELM);
+	snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION,
+			    WM8741_UPDATERL, WM8741_UPDATERL);
+	snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION,
+			    WM8741_UPDATERM, WM8741_UPDATERM);
 
 	snd_soc_add_controls(codec, wm8741_snd_controls,
 			     ARRAY_SIZE(wm8741_snd_controls));
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 79b02ae125c..3f09deea8d9 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -55,8 +55,10 @@ static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
 
-static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, unsigned int hifi);
+static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt);
+static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt);
 
 /*
  * wm8753 register cache
@@ -87,6 +89,10 @@ struct wm8753_priv {
 	enum snd_soc_control_type control_type;
 	unsigned int sysclk;
 	unsigned int pcmclk;
+
+	unsigned int voice_fmt;
+	unsigned int hifi_fmt;
+
 	int dai_func;
 };
 
@@ -170,9 +176,9 @@ static int wm8753_get_dai(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-	int mode = snd_soc_read(codec, WM8753_IOCTL);
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.integer.value[0] = (mode & 0xc) >> 2;
+	ucontrol->value.integer.value[0] = wm8753->dai_func;
 	return 0;
 }
 
@@ -180,16 +186,26 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-	int mode = snd_soc_read(codec, WM8753_IOCTL);
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+	u16 ioctl;
+
+	if (codec->active)
+		return -EBUSY;
+
+	ioctl = snd_soc_read(codec, WM8753_IOCTL);
+
+	wm8753->dai_func = ucontrol->value.integer.value[0];
+
+	if (((ioctl >> 2) & 0x3) == wm8753->dai_func)
+		return 1;
+
+	ioctl = (ioctl & 0x1f3) | (wm8753->dai_func << 2);
+	snd_soc_write(codec, WM8753_IOCTL, ioctl);
 
-	if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0])
-		return 0;
 
-	mode &= 0xfff3;
-	mode |= (ucontrol->value.integer.value[0] << 2);
+	wm8753_hifi_write_dai_fmt(codec, wm8753->hifi_fmt);
+	wm8753_voice_write_dai_fmt(codec, wm8753->voice_fmt);
 
-	wm8753->dai_func =  ucontrol->value.integer.value[0];
 	return 1;
 }
 
@@ -828,10 +844,9 @@ static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 /*
  * Set's ADC and Voice DAC format.
  */
-static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01ec;
 
 	/* interface format */
@@ -858,13 +873,6 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int wm8753_pcm_startup(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
-{
-	wm8753_set_dai_mode(dai->codec, dai, 0);
-	return 0;
-}
-
 /*
  * Set PCM DAI bit size and sample rate.
  */
@@ -905,10 +913,9 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
 /*
  * Set's PCM dai fmt and BCLK.
  */
-static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 voice, ioctl;
 
 	voice = snd_soc_read(codec, WM8753_PCM) & 0x011f;
@@ -999,10 +1006,9 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 /*
  * Set's HiFi DAC format.
  */
-static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01e0;
 
 	/* interface format */
@@ -1032,10 +1038,9 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
 /*
  * Set's I2S DAI format.
  */
-static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 ioctl, hifi;
 
 	hifi = snd_soc_read(codec, WM8753_HIFI) & 0x011f;
@@ -1098,13 +1103,6 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int wm8753_i2s_startup(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
-{
-	wm8753_set_dai_mode(dai->codec, dai, 1);
-	return 0;
-}
-
 /*
  * Set PCM DAI bit size and sample rate.
  */
@@ -1147,61 +1145,117 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 clock;
 
 	/* set clk source as pcmclk */
 	clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
 	snd_soc_write(codec, WM8753_CLOCK, clock);
 
-	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
+	return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
 }
 
-static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+	return wm8753_hdac_set_dai_fmt(codec, fmt);
 }
 
-static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 clock;
 
 	/* set clk source as pcmclk */
 	clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
 	snd_soc_write(codec, WM8753_CLOCK, clock);
 
-	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+	return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
 }
 
-static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 clock;
 
 	/* set clk source as mclk */
 	clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
 	snd_soc_write(codec, WM8753_CLOCK, clock | 0x4);
 
-	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
+	if (wm8753_hdac_set_dai_fmt(codec, fmt) < 0)
 		return -EINVAL;
-	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+	return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
 }
 
+static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt)
+{
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	switch (wm8753->dai_func) {
+	case 0:
+		ret = wm8753_mode1h_set_dai_fmt(codec, fmt);
+		break;
+	case 1:
+		ret = wm8753_mode2_set_dai_fmt(codec, fmt);
+		break;
+	case 2:
+	case 3:
+		ret = wm8753_mode3_4_set_dai_fmt(codec, fmt);
+		break;
+	default:
+		 break;
+	}
+	if (ret)
+		return ret;
+
+	return wm8753_i2s_set_dai_fmt(codec, fmt);
+}
+
+static int wm8753_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+
+	wm8753->hifi_fmt = fmt;
+
+	return wm8753_hifi_write_dai_fmt(codec, fmt);
+};
+
+static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt)
+{
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	if (wm8753->dai_func != 0)
+		return 0;
+
+	ret = wm8753_mode1v_set_dai_fmt(codec, fmt);
+	if (ret)
+		return ret;
+	ret = wm8753_pcm_set_dai_fmt(codec, fmt);
+	if (ret)
+		return ret;
+
+	return 0;
+};
+
+static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+
+	wm8753->voice_fmt = fmt;
+
+	return wm8753_voice_write_dai_fmt(codec, fmt);
+};
+
 static int wm8753_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
@@ -1268,57 +1322,25 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
  * 3. Voice disabled - HIFI over HIFI
  * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
  */
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = {
-	.startup = wm8753_i2s_startup,
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
 	.hw_params	= wm8753_i2s_hw_params,
 	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode1h_set_dai_fmt,
-	.set_clkdiv	= wm8753_set_dai_clkdiv,
-	.set_pll	= wm8753_set_dai_pll,
-	.set_sysclk	= wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = {
-	.startup = wm8753_pcm_startup,
-	.hw_params	= wm8753_pcm_hw_params,
-	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode1v_set_dai_fmt,
+	.set_fmt	= wm8753_hifi_set_dai_fmt,
 	.set_clkdiv	= wm8753_set_dai_clkdiv,
 	.set_pll	= wm8753_set_dai_pll,
 	.set_sysclk	= wm8753_set_dai_sysclk,
 };
 
-static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = {
-	.startup = wm8753_pcm_startup,
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = {
 	.hw_params	= wm8753_pcm_hw_params,
 	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode2_set_dai_fmt,
-	.set_clkdiv	= wm8753_set_dai_clkdiv,
-	.set_pll	= wm8753_set_dai_pll,
-	.set_sysclk	= wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3	= {
-	.startup = wm8753_i2s_startup,
-	.hw_params	= wm8753_i2s_hw_params,
-	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
-	.set_clkdiv	= wm8753_set_dai_clkdiv,
-	.set_pll	= wm8753_set_dai_pll,
-	.set_sysclk	= wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4	= {
-	.startup = wm8753_i2s_startup,
-	.hw_params	= wm8753_i2s_hw_params,
-	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_fmt	= wm8753_voice_set_dai_fmt,
 	.set_clkdiv	= wm8753_set_dai_clkdiv,
 	.set_pll	= wm8753_set_dai_pll,
 	.set_sysclk	= wm8753_set_dai_sysclk,
 };
 
-static struct snd_soc_dai_driver wm8753_all_dai[] = {
+static struct snd_soc_dai_driver wm8753_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "wm8753-hifi",
 	.playback = {
@@ -1326,14 +1348,16 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS},
+		.formats = WM8753_FORMATS
+	},
 	.capture = { /* dummy for fast DAI switching */
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS},
-	.ops = &wm8753_dai_ops_hifi_mode1,
+		.formats = WM8753_FORMATS
+	},
+	.ops = &wm8753_dai_ops_hifi_mode,
 },
 /* DAI Voice mode 1 */
 {	.name = "wm8753-voice",
@@ -1342,97 +1366,19 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = {
 		.channels_min = 1,
 		.channels_max = 1,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_voice_mode1,
-},
-/* DAI HiFi mode 2 - dummy */
-{	.name = "wm8753-hifi",
-},
-/* DAI Voice mode 2 */
-{	.name = "wm8753-voice",
-	.playback = {
-		.stream_name = "Voice Playback",
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_voice_mode2,
-},
-/* DAI HiFi mode 3 */
-{	.name = "wm8753-hifi",
-	.playback = {
-		.stream_name = "HiFi Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_hifi_mode3,
-},
-/* DAI Voice mode 3 - dummy */
-{	.name = "wm8753-voice",
-},
-/* DAI HiFi mode 4 */
-{	.name = "wm8753-hifi",
-	.playback = {
-		.stream_name = "HiFi Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS,
+	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_hifi_mode4,
-},
-/* DAI Voice mode 4 - dummy */
-{	.name = "wm8753-voice",
-},
-};
-
-static struct snd_soc_dai_driver wm8753_dai[] = {
-	{
-		.name = "wm8753-aif0",
-	},
-	{
-		.name = "wm8753-aif1",
+		.formats = WM8753_FORMATS,
 	},
+	.ops = &wm8753_dai_ops_voice_mode,
+},
 };
 
-static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, unsigned int hifi)
-{
-	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
-
-	if (wm8753->dai_func < 4) {
-		if (hifi)
-			dai->driver = &wm8753_all_dai[wm8753->dai_func << 1];
-		else
-			dai->driver = &wm8753_all_dai[(wm8753->dai_func << 1) + 1];
-	}
-	snd_soc_write(codec, WM8753_IOCTL, wm8753->dai_func);
-}
-
 static void wm8753_work(struct work_struct *work)
 {
 	struct snd_soc_dapm_context *dapm =
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6dae1b40c9f..6785688f880 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -175,7 +175,7 @@ static int txsrc_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int wm8804_volatile(unsigned int reg)
+static int wm8804_volatile(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM8804_RST_DEVID1:
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index cd0959926d1..449ea09a193 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -180,7 +180,7 @@ static const u16 wm8900_reg_defaults[WM8900_MAXREG] = {
 	/* Remaining registers all zero */
 };
 
-static int wm8900_volatile_register(unsigned int reg)
+static int wm8900_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM8900_REG_ID:
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 987476a5895..ae1cadfae84 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2,6 +2,7 @@
  * wm8903.c  --  WM8903 ALSA SoC Audio driver
  *
  * Copyright 2008 Wolfson Microelectronics
+ * Copyright 2011 NVIDIA, Inc.
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -19,6 +20,7 @@
 #include <linux/init.h>
 #include <linux/completion.h>
 #include <linux/delay.h>
+#include <linux/gpio.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
@@ -213,6 +215,7 @@ static u16 wm8903_reg_defaults[] = {
 };
 
 struct wm8903_priv {
+	struct snd_soc_codec *codec;
 
 	int sysclk;
 	int irq;
@@ -220,25 +223,36 @@ struct wm8903_priv {
 	int fs;
 	int deemph;
 
+	int dcs_pending;
+	int dcs_cache[4];
+
 	/* Reference count */
 	int class_w_users;
 
-	struct completion wseq;
-
 	struct snd_soc_jack *mic_jack;
 	int mic_det;
 	int mic_short;
 	int mic_last_report;
 	int mic_delay;
+
+#ifdef CONFIG_GPIOLIB
+	struct gpio_chip gpio_chip;
+#endif
 };
 
-static int wm8903_volatile_register(unsigned int reg)
+static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM8903_SW_RESET_AND_ID:
 	case WM8903_REVISION_NUMBER:
 	case WM8903_INTERRUPT_STATUS_1:
 	case WM8903_WRITE_SEQUENCER_4:
+	case WM8903_POWER_MANAGEMENT_3:
+	case WM8903_POWER_MANAGEMENT_2:
+	case WM8903_DC_SERVO_READBACK_1:
+	case WM8903_DC_SERVO_READBACK_2:
+	case WM8903_DC_SERVO_READBACK_3:
+	case WM8903_DC_SERVO_READBACK_4:
 		return 1;
 
 	default:
@@ -246,50 +260,6 @@ static int wm8903_volatile_register(unsigned int reg)
 	}
 }
 
-static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
-{
-	u16 reg[5];
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
-	BUG_ON(start > 48);
-
-	/* Enable the sequencer if it's not already on */
-	reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0);
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0,
-		      reg[0] | WM8903_WSEQ_ENA);
-
-	dev_dbg(codec->dev, "Starting sequence at %d\n", start);
-
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_3,
-		     start | WM8903_WSEQ_START);
-
-	/* Wait for it to complete.  If we have the interrupt wired up then
-	 * that will break us out of the poll early.
-	 */
-	do {
-		wait_for_completion_timeout(&wm8903->wseq,
-					    msecs_to_jiffies(10));
-
-		reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4);
-	} while (reg[4] & WM8903_WSEQ_BUSY);
-
-	dev_dbg(codec->dev, "Sequence complete\n");
-
-	/* Disable the sequencer again if we enabled it */
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
-
-	return 0;
-}
-
-static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
-{
-	int i;
-
-	/* There really ought to be something better we can do here :/ */
-	for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
-		cache[i] = codec->hw_read(codec, i);
-}
-
 static void wm8903_reset(struct snd_soc_codec *codec)
 {
 	snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0);
@@ -297,11 +267,6 @@ static void wm8903_reset(struct snd_soc_codec *codec)
 	       sizeof(wm8903_reg_defaults));
 }
 
-#define WM8903_OUTPUT_SHORT 0x8
-#define WM8903_OUTPUT_OUT   0x4
-#define WM8903_OUTPUT_INT   0x2
-#define WM8903_OUTPUT_IN    0x1
-
 static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
 			   struct snd_kcontrol *kcontrol, int event)
 {
@@ -311,97 +276,101 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-/*
- * Event for headphone and line out amplifier power changes.  Special
- * power up/down sequences are required in order to maximise pop/click
- * performance.
- */
-static int wm8903_output_event(struct snd_soc_dapm_widget *w,
-			       struct snd_kcontrol *kcontrol, int event)
+static int wm8903_dcs_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	u16 val;
-	u16 reg;
-	u16 dcs_reg;
-	u16 dcs_bit;
-	int shift;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
-	switch (w->reg) {
-	case WM8903_POWER_MANAGEMENT_2:
-		reg = WM8903_ANALOGUE_HP_0;
-		dcs_bit = 0 + w->shift;
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		wm8903->dcs_pending |= 1 << w->shift;
 		break;
-	case WM8903_POWER_MANAGEMENT_3:
-		reg = WM8903_ANALOGUE_LINEOUT_0;
-		dcs_bit = 2 + w->shift;
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(codec, WM8903_DC_SERVO_0,
+				    1 << w->shift, 0);
 		break;
-	default:
-		BUG();
-		return -EINVAL;  /* Spurious warning from some compilers */
 	}
 
-	switch (w->shift) {
-	case 0:
-		shift = 0;
-		break;
-	case 1:
-		shift = 4;
-		break;
-	default:
-		BUG();
-		return -EINVAL;  /* Spurious warning from some compilers */
-	}
+	return 0;
+}
 
-	if (event & SND_SOC_DAPM_PRE_PMU) {
-		val = snd_soc_read(codec, reg);
+#define WM8903_DCS_MODE_WRITE_STOP 0
+#define WM8903_DCS_MODE_START_STOP 2
 
-		/* Short the output */
-		val &= ~(WM8903_OUTPUT_SHORT << shift);
-		snd_soc_write(codec, reg, val);
-	}
+static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
+				enum snd_soc_dapm_type event, int subseq)
+{
+	struct snd_soc_codec *codec = container_of(dapm,
+						   struct snd_soc_codec, dapm);
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+	int dcs_mode = WM8903_DCS_MODE_WRITE_STOP;
+	int i, val;
 
-	if (event & SND_SOC_DAPM_POST_PMU) {
-		val = snd_soc_read(codec, reg);
+	/* Complete any pending DC servo starts */
+	if (wm8903->dcs_pending) {
+		dev_dbg(codec->dev, "Starting DC servo for %x\n",
+			wm8903->dcs_pending);
 
-		val |= (WM8903_OUTPUT_IN << shift);
-		snd_soc_write(codec, reg, val);
+		/* If we've no cached values then we need to do startup */
+		for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) {
+			if (!(wm8903->dcs_pending & (1 << i)))
+				continue;
 
-		val |= (WM8903_OUTPUT_INT << shift);
-		snd_soc_write(codec, reg, val);
+			if (wm8903->dcs_cache[i]) {
+				dev_dbg(codec->dev,
+					"Restore DC servo %d value %x\n",
+					3 - i, wm8903->dcs_cache[i]);
+
+				snd_soc_write(codec, WM8903_DC_SERVO_4 + i,
+					      wm8903->dcs_cache[i] & 0xff);
+			} else {
+				dev_dbg(codec->dev,
+					"Calibrate DC servo %d\n", 3 - i);
+				dcs_mode = WM8903_DCS_MODE_START_STOP;
+			}
+		}
 
-		/* Turn on the output ENA_OUTP */
-		val |= (WM8903_OUTPUT_OUT << shift);
-		snd_soc_write(codec, reg, val);
+		/* Don't trust the cache for analogue */
+		if (wm8903->class_w_users)
+			dcs_mode = WM8903_DCS_MODE_START_STOP;
 
-		/* Enable the DC servo */
-		dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
-		dcs_reg |= dcs_bit;
-		snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+		snd_soc_update_bits(codec, WM8903_DC_SERVO_2,
+				    WM8903_DCS_MODE_MASK, dcs_mode);
 
-		/* Remove the short */
-		val |= (WM8903_OUTPUT_SHORT << shift);
-		snd_soc_write(codec, reg, val);
-	}
+		snd_soc_update_bits(codec, WM8903_DC_SERVO_0,
+				    WM8903_DCS_ENA_MASK, wm8903->dcs_pending);
 
-	if (event & SND_SOC_DAPM_PRE_PMD) {
-		val = snd_soc_read(codec, reg);
+		switch (dcs_mode) {
+		case WM8903_DCS_MODE_WRITE_STOP:
+			break;
 
-		/* Short the output */
-		val &= ~(WM8903_OUTPUT_SHORT << shift);
-		snd_soc_write(codec, reg, val);
+		case WM8903_DCS_MODE_START_STOP:
+			msleep(270);
 
-		/* Disable the DC servo */
-		dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
-		dcs_reg &= ~dcs_bit;
-		snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+			/* Cache the measured offsets for digital */
+			if (wm8903->class_w_users)
+				break;
 
-		/* Then disable the intermediate and output stages */
-		val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
-			  WM8903_OUTPUT_IN) << shift);
-		snd_soc_write(codec, reg, val);
-	}
+			for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) {
+				if (!(wm8903->dcs_pending & (1 << i)))
+					continue;
 
-	return 0;
+				val = snd_soc_read(codec,
+						   WM8903_DC_SERVO_READBACK_1 + i);
+				dev_dbg(codec->dev, "DC servo %d: %x\n",
+					3 - i, val);
+				wm8903->dcs_cache[i] = val;
+			}
+			break;
+
+		default:
+			pr_warn("DCS mode %d delay not set\n", dcs_mode);
+			break;
+		}
+
+		wm8903->dcs_pending = 0;
+	}
 }
 
 /*
@@ -667,6 +636,22 @@ static const struct soc_enum lsidetone_enum =
 static const struct soc_enum rsidetone_enum =
 	SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
 
+static const char *aif_text[] = {
+	"Left", "Right"
+};
+
+static const struct soc_enum lcapture_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text);
+
+static const struct soc_enum rcapture_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text);
+
+static const struct soc_enum lplay_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text);
+
+static const struct soc_enum rplay_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text);
+
 static const struct snd_kcontrol_new wm8903_snd_controls[] = {
 
 /* Input PGAs - No TLV since the scale depends on PGA mode */
@@ -784,6 +769,18 @@ static const struct snd_kcontrol_new lsidetone_mux =
 static const struct snd_kcontrol_new rsidetone_mux =
 	SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
 
+static const struct snd_kcontrol_new lcapture_mux =
+	SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum);
+
+static const struct snd_kcontrol_new rcapture_mux =
+	SOC_DAPM_ENUM("Right Capture Mux", rcapture_enum);
+
+static const struct snd_kcontrol_new lplay_mux =
+	SOC_DAPM_ENUM("Left Playback Mux", lplay_enum);
+
+static const struct snd_kcontrol_new rplay_mux =
+	SOC_DAPM_ENUM("Right Playback Mux", rplay_enum);
+
 static const struct snd_kcontrol_new left_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
@@ -847,14 +844,26 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux),
 SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0),
 SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0),
 
-SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
-SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
+SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0),
+SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lcapture_mux),
+SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rcapture_mux),
+
+SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
 SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
 
-SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
-SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXL", "Left Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXR", "Right Playback", 0, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Playback Mux", SND_SOC_NOPM, 0, 0, &lplay_mux),
+SND_SOC_DAPM_MUX("Right Playback Mux", SND_SOC_NOPM, 0, 0, &rplay_mux),
+
+SND_SOC_DAPM_DAC("DACL", NULL, WM8903_POWER_MANAGEMENT_6, 3, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, WM8903_POWER_MANAGEMENT_6, 2, 0),
 
 SND_SOC_DAPM_MIXER("Left Output Mixer", WM8903_POWER_MANAGEMENT_1, 1, 0,
 		   left_output_mixer, ARRAY_SIZE(left_output_mixer)),
@@ -866,23 +875,45 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
 SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
 		   right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
 
-SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
-		   1, 0, NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
-SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
-		   0, 0, NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
-
-SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
-		   NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
-SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
-		   NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+		   4, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+		   0, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+		   NULL, 0),
+
+SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+		   NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_DCS", 3, SND_SOC_NOPM, 3, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("HPR_DCS", 3, SND_SOC_NOPM, 2, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, SND_SOC_NOPM, 1, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, SND_SOC_NOPM, 0, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
 		 NULL, 0),
@@ -892,10 +923,18 @@ SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0,
 SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
 		    wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
 
+	{ "CLK_DSP", NULL, "CLK_SYS" },
+	{ "Mic Bias", NULL, "CLK_SYS" },
+	{ "HPL_DCS", NULL, "CLK_SYS" },
+	{ "HPR_DCS", NULL, "CLK_SYS" },
+	{ "LINEOUTL_DCS", NULL, "CLK_SYS" },
+	{ "LINEOUTR_DCS", NULL, "CLK_SYS" },
+
 	{ "Left Input Mux", "IN1L", "IN1L" },
 	{ "Left Input Mux", "IN2L", "IN2L" },
 	{ "Left Input Mux", "IN3L", "IN3L" },
@@ -936,18 +975,36 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Left Input PGA", NULL, "Left Input Mode Mux" },
 	{ "Right Input PGA", NULL, "Right Input Mode Mux" },
 
+	{ "Left Capture Mux", "Left", "ADCL" },
+	{ "Left Capture Mux", "Right", "ADCR" },
+
+	{ "Right Capture Mux", "Left", "ADCL" },
+	{ "Right Capture Mux", "Right", "ADCR" },
+
+	{ "AIFTXL", NULL, "Left Capture Mux" },
+	{ "AIFTXR", NULL, "Right Capture Mux" },
+
 	{ "ADCL", NULL, "Left Input PGA" },
 	{ "ADCL", NULL, "CLK_DSP" },
 	{ "ADCR", NULL, "Right Input PGA" },
 	{ "ADCR", NULL, "CLK_DSP" },
 
+	{ "Left Playback Mux", "Left", "AIFRXL" },
+	{ "Left Playback Mux", "Right", "AIFRXR" },
+
+	{ "Right Playback Mux", "Left", "AIFRXL" },
+	{ "Right Playback Mux", "Right", "AIFRXR" },
+
 	{ "DACL Sidetone", "Left", "ADCL" },
 	{ "DACL Sidetone", "Right", "ADCR" },
 	{ "DACR Sidetone", "Left", "ADCL" },
 	{ "DACR Sidetone", "Right", "ADCR" },
 
+	{ "DACL", NULL, "Left Playback Mux" },
 	{ "DACL", NULL, "DACL Sidetone" },
 	{ "DACL", NULL, "CLK_DSP" },
+
+	{ "DACR", NULL, "Right Playback Mux" },
 	{ "DACR", NULL, "DACR Sidetone" },
 	{ "DACR", NULL, "CLK_DSP" },
 
@@ -980,11 +1037,35 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Left Speaker PGA", NULL, "Left Speaker Mixer" },
 	{ "Right Speaker PGA", NULL, "Right Speaker Mixer" },
 
-	{ "HPOUTL", NULL, "Left Headphone Output PGA" },
-	{ "HPOUTR", NULL, "Right Headphone Output PGA" },
+	{ "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
+	{ "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
+	{ "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
+	{ "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
+
+	{ "HPL_DCS", NULL, "DCS Master" },
+	{ "HPR_DCS", NULL, "DCS Master" },
+	{ "LINEOUTL_DCS", NULL, "DCS Master" },
+	{ "LINEOUTR_DCS", NULL, "DCS Master" },
+
+	{ "HPL_DCS", NULL, "HPL_ENA_DLY" },
+	{ "HPR_DCS", NULL, "HPR_ENA_DLY" },
+	{ "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" },
+	{ "LINEOUTR_DCS", NULL, "LINEOUTR_ENA_DLY" },
 
-	{ "LINEOUTL", NULL, "Left Line Output PGA" },
-	{ "LINEOUTR", NULL, "Right Line Output PGA" },
+	{ "HPL_ENA_OUTP", NULL, "HPL_DCS" },
+	{ "HPR_ENA_OUTP", NULL, "HPR_DCS" },
+	{ "LINEOUTL_ENA_OUTP", NULL, "LINEOUTL_DCS" },
+	{ "LINEOUTR_ENA_OUTP", NULL, "LINEOUTR_DCS" },
+
+	{ "HPL_RMV_SHORT", NULL, "HPL_ENA_OUTP" },
+	{ "HPR_RMV_SHORT", NULL, "HPR_ENA_OUTP" },
+	{ "LINEOUTL_RMV_SHORT", NULL, "LINEOUTL_ENA_OUTP" },
+	{ "LINEOUTR_RMV_SHORT", NULL, "LINEOUTR_ENA_OUTP" },
+
+	{ "HPOUTL", NULL, "HPL_RMV_SHORT" },
+	{ "HPOUTR", NULL, "HPR_RMV_SHORT" },
+	{ "LINEOUTL", NULL, "LINEOUTL_RMV_SHORT" },
+	{ "LINEOUTR", NULL, "LINEOUTR_RMV_SHORT" },
 
 	{ "LOP", NULL, "Left Speaker PGA" },
 	{ "LON", NULL, "Left Speaker PGA" },
@@ -1012,29 +1093,71 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec)
 static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	u16 reg;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
+		break;
+
 	case SND_SOC_BIAS_PREPARE:
-		reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
-		reg &= ~(WM8903_VMID_RES_MASK);
-		reg |= WM8903_VMID_RES_50K;
-		snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_RES_MASK,
+				    WM8903_VMID_RES_50K);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			snd_soc_write(codec, WM8903_CLOCK_RATES_2,
-				     WM8903_CLK_SYS_ENA);
-
-			/* Change DC servo dither level in startup sequence */
-			snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
-			snd_soc_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
-			snd_soc_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
-
-			wm8903_run_sequence(codec, 0);
-			wm8903_sync_reg_cache(codec, codec->reg_cache);
+			snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+					    WM8903_POBCTRL | WM8903_ISEL_MASK |
+					    WM8903_STARTUP_BIAS_ENA |
+					    WM8903_BIAS_ENA,
+					    WM8903_POBCTRL |
+					    (2 << WM8903_ISEL_SHIFT) |
+					    WM8903_STARTUP_BIAS_ENA);
+
+			snd_soc_update_bits(codec,
+					    WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0,
+					    WM8903_SPK_DISCHARGE,
+					    WM8903_SPK_DISCHARGE);
+
+			msleep(33);
+
+			snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5,
+					    WM8903_SPKL_ENA | WM8903_SPKR_ENA,
+					    WM8903_SPKL_ENA | WM8903_SPKR_ENA);
+
+			snd_soc_update_bits(codec,
+					    WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0,
+					    WM8903_SPK_DISCHARGE, 0);
+
+			snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+					    WM8903_VMID_TIE_ENA |
+					    WM8903_BUFIO_ENA |
+					    WM8903_VMID_IO_ENA |
+					    WM8903_VMID_SOFT_MASK |
+					    WM8903_VMID_RES_MASK |
+					    WM8903_VMID_BUF_ENA,
+					    WM8903_VMID_TIE_ENA |
+					    WM8903_BUFIO_ENA |
+					    WM8903_VMID_IO_ENA |
+					    (2 << WM8903_VMID_SOFT_SHIFT) |
+					    WM8903_VMID_RES_250K |
+					    WM8903_VMID_BUF_ENA);
+
+			msleep(129);
+
+			snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5,
+					    WM8903_SPKL_ENA | WM8903_SPKR_ENA,
+					    0);
+
+			snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+					    WM8903_VMID_SOFT_MASK, 0);
+
+			snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+					    WM8903_VMID_RES_MASK,
+					    WM8903_VMID_RES_50K);
+
+			snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+					    WM8903_BIAS_ENA | WM8903_POBCTRL,
+					    WM8903_BIAS_ENA);
 
 			/* By default no bypass paths are enabled so
 			 * enable Class W support.
@@ -1047,17 +1170,32 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 					    WM8903_CP_DYN_V);
 		}
 
-		reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
-		reg &= ~(WM8903_VMID_RES_MASK);
-		reg |= WM8903_VMID_RES_250K;
-		snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_RES_MASK,
+				    WM8903_VMID_RES_250K);
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		wm8903_run_sequence(codec, 32);
-		reg = snd_soc_read(codec, WM8903_CLOCK_RATES_2);
-		reg &= ~WM8903_CLK_SYS_ENA;
-		snd_soc_write(codec, WM8903_CLOCK_RATES_2, reg);
+		snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+				    WM8903_BIAS_ENA, 0);
+
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_SOFT_MASK,
+				    2 << WM8903_VMID_SOFT_SHIFT);
+
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_BUF_ENA, 0);
+
+		msleep(290);
+
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_TIE_ENA | WM8903_BUFIO_ENA |
+				    WM8903_VMID_IO_ENA | WM8903_VMID_RES_MASK |
+				    WM8903_VMID_SOFT_MASK |
+				    WM8903_VMID_BUF_ENA, 0);
+
+		snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+				    WM8903_STARTUP_BIAS_ENA, 0);
 		break;
 	}
 
@@ -1482,7 +1620,7 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 			    WM8903_MICDET_EINT | WM8903_MICSHRT_EINT,
 			    irq_mask);
 
-	if (det && shrt) {
+	if (det || shrt) {
 		/* Enable mic detection, this may not have been set through
 		 * platform data (eg, if the defaults are OK). */
 		snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
@@ -1510,8 +1648,7 @@ static irqreturn_t wm8903_irq(int irq, void *data)
 	int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask;
 
 	if (int_val & WM8903_WSEQ_BUSY_EINT) {
-		dev_dbg(codec->dev, "Write sequencer done\n");
-		complete(&wm8903->wseq);
+		dev_warn(codec->dev, "Write sequencer done\n");
 	}
 
 	/*
@@ -1635,6 +1772,120 @@ static int wm8903_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
+#ifdef CONFIG_GPIOLIB
+static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip)
+{
+	return container_of(chip, struct wm8903_priv, gpio_chip);
+}
+
+static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset)
+{
+	if (offset >= WM8903_NUM_GPIO)
+		return -EINVAL;
+
+	return 0;
+}
+
+static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
+{
+	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct snd_soc_codec *codec = wm8903->codec;
+	unsigned int mask, val;
+
+	mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK;
+	val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) |
+		WM8903_GP1_DIR;
+
+	return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
+				   mask, val);
+}
+
+static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset)
+{
+	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct snd_soc_codec *codec = wm8903->codec;
+	int reg;
+
+	reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset);
+
+	return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT;
+}
+
+static int wm8903_gpio_direction_out(struct gpio_chip *chip,
+				     unsigned offset, int value)
+{
+	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct snd_soc_codec *codec = wm8903->codec;
+	unsigned int mask, val;
+
+	mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK;
+	val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) |
+		(value << WM8903_GP2_LVL_SHIFT);
+
+	return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
+				   mask, val);
+}
+
+static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
+{
+	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct snd_soc_codec *codec = wm8903->codec;
+
+	snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
+			    WM8903_GP1_LVL_MASK,
+			    !!value << WM8903_GP1_LVL_SHIFT);
+}
+
+static struct gpio_chip wm8903_template_chip = {
+	.label			= "wm8903",
+	.owner			= THIS_MODULE,
+	.request		= wm8903_gpio_request,
+	.direction_input	= wm8903_gpio_direction_in,
+	.get			= wm8903_gpio_get,
+	.direction_output	= wm8903_gpio_direction_out,
+	.set			= wm8903_gpio_set,
+	.can_sleep		= 1,
+};
+
+static void wm8903_init_gpio(struct snd_soc_codec *codec)
+{
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+	struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev);
+	int ret;
+
+	wm8903->gpio_chip = wm8903_template_chip;
+	wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO;
+	wm8903->gpio_chip.dev = codec->dev;
+
+	if (pdata && pdata->gpio_base)
+		wm8903->gpio_chip.base = pdata->gpio_base;
+	else
+		wm8903->gpio_chip.base = -1;
+
+	ret = gpiochip_add(&wm8903->gpio_chip);
+	if (ret != 0)
+		dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret);
+}
+
+static void wm8903_free_gpio(struct snd_soc_codec *codec)
+{
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	ret = gpiochip_remove(&wm8903->gpio_chip);
+	if (ret != 0)
+		dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret);
+}
+#else
+static void wm8903_init_gpio(struct snd_soc_codec *codec)
+{
+}
+
+static void wm8903_free_gpio(struct snd_soc_codec *codec)
+{
+}
+#endif
+
 static int wm8903_probe(struct snd_soc_codec *codec)
 {
 	struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev);
@@ -1643,7 +1894,7 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 	int trigger, irq_pol;
 	u16 val;
 
-	init_completion(&wm8903->wseq);
+	wm8903->codec = codec;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
 	if (ret != 0) {
@@ -1659,19 +1910,33 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 	}
 
 	val = snd_soc_read(codec, WM8903_REVISION_NUMBER);
-	dev_info(codec->dev, "WM8903 revision %d\n",
-		 val & WM8903_CHIP_REV_MASK);
+	dev_info(codec->dev, "WM8903 revision %c\n",
+		 (val & WM8903_CHIP_REV_MASK) + 'A');
 
 	wm8903_reset(codec);
 
 	/* Set up GPIOs and microphone detection */
 	if (pdata) {
+		bool mic_gpio = false;
+
 		for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
-			if (!pdata->gpio_cfg[i])
+			if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG)
 				continue;
 
 			snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
 				      pdata->gpio_cfg[i] & 0xffff);
+
+			val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
+				>> WM8903_GP1_FN_SHIFT;
+
+			switch (val) {
+			case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
+			case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
+				mic_gpio = true;
+				break;
+			default:
+				break;
+			}
 		}
 
 		snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0,
@@ -1682,6 +1947,14 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 			snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
 					    WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
 
+		/* If microphone detection is enabled by pdata but
+		 * detected via IRQ then interrupts can be lost before
+		 * the machine driver has set up microphone detection
+		 * IRQs as the IRQs are clear on read.  The detection
+		 * will be enabled when the machine driver configures.
+		 */
+		WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
+
 		wm8903->mic_delay = pdata->micdet_delay;
 	}
 	
@@ -1741,20 +2014,23 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
 
 	/* Enable DAC soft mute by default */
-	val = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
-	val |= WM8903_DAC_MUTEMODE;
-	snd_soc_write(codec, WM8903_DAC_DIGITAL_1, val);
+	snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1,
+			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
+			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
 
 	snd_soc_add_controls(codec, wm8903_snd_controls,
 				ARRAY_SIZE(wm8903_snd_controls));
 	wm8903_add_widgets(codec);
 
+	wm8903_init_gpio(codec);
+
 	return ret;
 }
 
 /* power down chip */
 static int wm8903_remove(struct snd_soc_codec *codec)
 {
+	wm8903_free_gpio(codec);
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
@@ -1769,6 +2045,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = {
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8903_reg_defaults,
 	.volatile_register = wm8903_volatile_register,
+	.seq_notifier = wm8903_seq_notifier,
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -1807,7 +2084,7 @@ MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id);
 
 static struct i2c_driver wm8903_i2c_driver = {
 	.driver = {
-		.name = "wm8903-codec",
+		.name = "wm8903",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8903_i2c_probe,
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index e8490f3edd0..db949311c0f 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -75,6 +75,14 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec,
 #define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0    0x41
 #define WM8903_DC_SERVO_0                       0x43
 #define WM8903_DC_SERVO_2                       0x45
+#define WM8903_DC_SERVO_4			0x47
+#define WM8903_DC_SERVO_5			0x48
+#define WM8903_DC_SERVO_6			0x49
+#define WM8903_DC_SERVO_7			0x4A
+#define WM8903_DC_SERVO_READBACK_1		0x51
+#define WM8903_DC_SERVO_READBACK_2		0x52
+#define WM8903_DC_SERVO_READBACK_3		0x53
+#define WM8903_DC_SERVO_READBACK_4		0x54
 #define WM8903_ANALOGUE_HP_0                    0x5A
 #define WM8903_ANALOGUE_LINEOUT_0               0x5E
 #define WM8903_CHARGE_PUMP_0                    0x62
@@ -165,7 +173,7 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec,
 
 #define WM8903_VMID_RES_50K                          2
 #define WM8903_VMID_RES_250K                         3
-#define WM8903_VMID_RES_5K                           4
+#define WM8903_VMID_RES_5K                           6
 
 /*
  * R8 (0x08) - Analogue DAC 0
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9de44a4c05c..443ae580445 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -596,7 +596,7 @@ static struct {
 	{ 0x003F, 0x003F, 0 }, /* R248 - FLL NCO Test 1 */
 };
 
-static int wm8904_volatile_register(unsigned int reg)
+static int wm8904_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	return wm8904_access[reg].vol;
 }
@@ -2436,19 +2436,28 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Change some default settings - latch VU and enable ZC */
-	reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU;
-	reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU;
-	reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU;
-	reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU;
-	reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU |
-		WM8904_HPOUTLZC;
-	reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU |
-		WM8904_HPOUTRZC;
-	reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU |
-		WM8904_LINEOUTLZC;
-	reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU |
-		WM8904_LINEOUTRZC;
-	reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE;
+	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
+			    WM8904_ADC_VU, WM8904_ADC_VU);
+	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
+			    WM8904_ADC_VU, WM8904_ADC_VU);
+	snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT,
+			    WM8904_DAC_VU, WM8904_DAC_VU);
+	snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
+			    WM8904_DAC_VU, WM8904_DAC_VU);
+	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT,
+			    WM8904_HPOUT_VU | WM8904_HPOUTLZC,
+			    WM8904_HPOUT_VU | WM8904_HPOUTLZC);
+	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT,
+			    WM8904_HPOUT_VU | WM8904_HPOUTRZC,
+			    WM8904_HPOUT_VU | WM8904_HPOUTRZC);
+	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT,
+			    WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
+			    WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
+	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT,
+			    WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
+			    WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
+	snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0,
+			    WM8904_SR_MODE, 0);
 
 	/* Apply configuration from the platform data. */
 	if (wm8904->pdata) {
@@ -2469,10 +2478,12 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 	/* Set Class W by default - this will be managed by the Class
 	 * G widget at runtime where bypass paths are available.
 	 */
-	reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR;
+	snd_soc_update_bits(codec, WM8904_CLASS_W_0,
+			    WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
 
 	/* Use normal bias source */
-	reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL;
+	snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
+			    WM8904_POBCTRL, 0);
 
 	wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 7167dfc96aa..5e0214d6293 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -934,16 +934,27 @@ static int wm8955_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Change some default settings - latch VU and enable ZC */
-	reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU;
-	reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU;
-	reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC;
-	reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC;
-	reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC;
-	reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC;
-	reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC;
+	snd_soc_update_bits(codec, WM8955_LEFT_DAC_VOLUME,
+			    WM8955_LDVU, WM8955_LDVU);
+	snd_soc_update_bits(codec, WM8955_RIGHT_DAC_VOLUME,
+			    WM8955_RDVU, WM8955_RDVU);
+	snd_soc_update_bits(codec, WM8955_LOUT1_VOLUME,
+			    WM8955_LO1VU | WM8955_LO1ZC,
+			    WM8955_LO1VU | WM8955_LO1ZC);
+	snd_soc_update_bits(codec, WM8955_ROUT1_VOLUME,
+			    WM8955_RO1VU | WM8955_RO1ZC,
+			    WM8955_RO1VU | WM8955_RO1ZC);
+	snd_soc_update_bits(codec, WM8955_LOUT2_VOLUME,
+			    WM8955_LO2VU | WM8955_LO2ZC,
+			    WM8955_LO2VU | WM8955_LO2ZC);
+	snd_soc_update_bits(codec, WM8955_ROUT2_VOLUME,
+			    WM8955_RO2VU | WM8955_RO2ZC,
+			    WM8955_RO2VU | WM8955_RO2ZC);
+	snd_soc_update_bits(codec, WM8955_MONOOUT_VOLUME,
+			    WM8955_MOZC, WM8955_MOZC);
 
 	/* Also enable adaptive bass boost by default */
-	reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB;
+	snd_soc_update_bits(codec, WM8955_BASS_CONTROL, WM8955_BB, WM8955_BB);
 
 	/* Set platform data values */
 	if (pdata) {
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 55252e7d02c..cdee8103d09 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -291,7 +291,7 @@ struct wm8961_priv {
 	int sysclk;
 };
 
-static int wm8961_volatile_register(unsigned int reg)
+static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM8961_SOFTWARE_RESET:
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index b9cb1fcf8c9..3b71dd65c96 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1938,7 +1938,7 @@ static const struct wm8962_reg_access {
 	[21139] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21139 - VSS_XTS32_0 */
 };
 
-static int wm8962_volatile_register(unsigned int reg)
+static int wm8962_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	if (wm8962_reg_access[reg].vol)
 		return 1;
@@ -1946,7 +1946,7 @@ static int wm8962_volatile_register(unsigned int reg)
 		return 0;
 }
 
-static int wm8962_readable_register(unsigned int reg)
+static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	if (wm8962_reg_access[reg].read)
 		return 1;
@@ -3635,7 +3635,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 	struct snd_soc_codec *codec = wm8962->codec;
 
 	snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset,
-			    WM8962_GP2_LVL, value << WM8962_GP2_LVL_SHIFT);
+			    WM8962_GP2_LVL, !!value << WM8962_GP2_LVL_SHIFT);
 }
 
 static int wm8962_gpio_direction_out(struct gpio_chip *chip,
@@ -3822,16 +3822,26 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Latch volume update bits */
-	reg_cache[WM8962_LEFT_INPUT_VOLUME] |= WM8962_IN_VU;
-	reg_cache[WM8962_RIGHT_INPUT_VOLUME] |= WM8962_IN_VU;
-	reg_cache[WM8962_LEFT_ADC_VOLUME] |= WM8962_ADC_VU;
-	reg_cache[WM8962_RIGHT_ADC_VOLUME] |= WM8962_ADC_VU;
-	reg_cache[WM8962_LEFT_DAC_VOLUME] |= WM8962_DAC_VU;
-	reg_cache[WM8962_RIGHT_DAC_VOLUME] |= WM8962_DAC_VU;
-	reg_cache[WM8962_SPKOUTL_VOLUME] |= WM8962_SPKOUT_VU;
-	reg_cache[WM8962_SPKOUTR_VOLUME] |= WM8962_SPKOUT_VU;
-	reg_cache[WM8962_HPOUTL_VOLUME] |= WM8962_HPOUT_VU;
-	reg_cache[WM8962_HPOUTR_VOLUME] |= WM8962_HPOUT_VU;
+	snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME,
+			    WM8962_IN_VU, WM8962_IN_VU);
+	snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME,
+			    WM8962_IN_VU, WM8962_IN_VU);
+	snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME,
+			    WM8962_ADC_VU, WM8962_ADC_VU);
+	snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME,
+			    WM8962_ADC_VU, WM8962_ADC_VU);
+	snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME,
+			    WM8962_DAC_VU, WM8962_DAC_VU);
+	snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME,
+			    WM8962_DAC_VU, WM8962_DAC_VU);
+	snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME,
+			    WM8962_SPKOUT_VU, WM8962_SPKOUT_VU);
+	snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME,
+			    WM8962_SPKOUT_VU, WM8962_SPKOUT_VU);
+	snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME,
+			    WM8962_HPOUT_VU, WM8962_HPOUT_VU);
+	snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME,
+			    WM8962_HPOUT_VU, WM8962_HPOUT_VU);
 
 	wm8962_add_widgets(codec);
 
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 4bbc3442703..85e3e630e76 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -93,6 +93,7 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
 static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
 static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
 
 static const struct snd_kcontrol_new wm8978_snd_controls[] = {
 
@@ -144,19 +145,19 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = {
 
 	SOC_SINGLE("DAC Playback Limiter Threshold",
 		WM8978_DAC_LIMITER_2, 4, 7, 0),
-	SOC_SINGLE("DAC Playback Limiter Boost",
-		WM8978_DAC_LIMITER_2, 0, 15, 0),
+	SOC_SINGLE_TLV("DAC Playback Limiter Volume",
+		WM8978_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
 
 	SOC_ENUM("ALC Enable Switch", alc1),
 	SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0),
 	SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0),
 
-	SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0),
+	SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0),
 	SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0),
 
 	SOC_ENUM("ALC Capture Mode", alc3),
-	SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0),
-	SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0),
+	SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0),
+	SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0),
 
 	SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0),
 	SOC_SINGLE("ALC Capture Noise Gate Threshold",
@@ -211,8 +212,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = {
 		WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1),
 
 	/* DAC / ADC oversampling */
-	SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0),
-	SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0),
+	SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL,
+		   5, 1, 0),
+	SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL,
+		   5, 1, 0),
 };
 
 /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */
@@ -965,7 +968,7 @@ static int wm8978_probe(struct snd_soc_codec *codec)
 	 * written.
 	 */
 	for (i = 0; i < ARRAY_SIZE(update_reg); i++)
-		((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100;
+		snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100);
 
 	/* Reset the codec */
 	ret = snd_soc_write(codec, WM8978_RESET, 0);
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
new file mode 100644
index 00000000000..28fdfd66661
--- /dev/null
+++ b/sound/soc/codecs/wm8991.c
@@ -0,0 +1,1427 @@
+/*
+ * wm8991.c  --  WM8991 ALSA Soc Audio driver
+ *
+ * Copyright 2007-2010 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ *         linux@wolfsonmicro.com
+ *
+ *  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/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include "wm8991.h"
+
+struct wm8991_priv {
+	enum snd_soc_control_type control_type;
+	unsigned int pcmclk;
+};
+
+static const u16 wm8991_reg_defs[] = {
+	0x8991,     /* R0  - Reset */
+	0x0000,     /* R1  - Power Management (1) */
+	0x6000,     /* R2  - Power Management (2) */
+	0x0000,     /* R3  - Power Management (3) */
+	0x4050,     /* R4  - Audio Interface (1) */
+	0x4000,     /* R5  - Audio Interface (2) */
+	0x01C8,     /* R6  - Clocking (1) */
+	0x0000,     /* R7  - Clocking (2) */
+	0x0040,     /* R8  - Audio Interface (3) */
+	0x0040,     /* R9  - Audio Interface (4) */
+	0x0004,     /* R10 - DAC CTRL */
+	0x00C0,     /* R11 - Left DAC Digital Volume */
+	0x00C0,     /* R12 - Right DAC Digital Volume */
+	0x0000,     /* R13 - Digital Side Tone */
+	0x0100,     /* R14 - ADC CTRL */
+	0x00C0,     /* R15 - Left ADC Digital Volume */
+	0x00C0,     /* R16 - Right ADC Digital Volume */
+	0x0000,     /* R17 */
+	0x0000,     /* R18 - GPIO CTRL 1 */
+	0x1000,     /* R19 - GPIO1 & GPIO2 */
+	0x1010,     /* R20 - GPIO3 & GPIO4 */
+	0x1010,     /* R21 - GPIO5 & GPIO6 */
+	0x8000,     /* R22 - GPIOCTRL 2 */
+	0x0800,     /* R23 - GPIO_POL */
+	0x008B,     /* R24 - Left Line Input 1&2 Volume */
+	0x008B,     /* R25 - Left Line Input 3&4 Volume */
+	0x008B,     /* R26 - Right Line Input 1&2 Volume */
+	0x008B,     /* R27 - Right Line Input 3&4 Volume */
+	0x0000,     /* R28 - Left Output Volume */
+	0x0000,     /* R29 - Right Output Volume */
+	0x0066,     /* R30 - Line Outputs Volume */
+	0x0022,     /* R31 - Out3/4 Volume */
+	0x0079,     /* R32 - Left OPGA Volume */
+	0x0079,     /* R33 - Right OPGA Volume */
+	0x0003,     /* R34 - Speaker Volume */
+	0x0003,     /* R35 - ClassD1 */
+	0x0000,     /* R36 */
+	0x0100,     /* R37 - ClassD3 */
+	0x0000,     /* R38 */
+	0x0000,     /* R39 - Input Mixer1 */
+	0x0000,     /* R40 - Input Mixer2 */
+	0x0000,     /* R41 - Input Mixer3 */
+	0x0000,     /* R42 - Input Mixer4 */
+	0x0000,     /* R43 - Input Mixer5 */
+	0x0000,     /* R44 - Input Mixer6 */
+	0x0000,     /* R45 - Output Mixer1 */
+	0x0000,     /* R46 - Output Mixer2 */
+	0x0000,     /* R47 - Output Mixer3 */
+	0x0000,     /* R48 - Output Mixer4 */
+	0x0000,     /* R49 - Output Mixer5 */
+	0x0000,     /* R50 - Output Mixer6 */
+	0x0180,     /* R51 - Out3/4 Mixer */
+	0x0000,     /* R52 - Line Mixer1 */
+	0x0000,     /* R53 - Line Mixer2 */
+	0x0000,     /* R54 - Speaker Mixer */
+	0x0000,     /* R55 - Additional Control */
+	0x0000,     /* R56 - AntiPOP1 */
+	0x0000,     /* R57 - AntiPOP2 */
+	0x0000,     /* R58 - MICBIAS */
+	0x0000,     /* R59 */
+	0x0008,     /* R60 - PLL1 */
+	0x0031,     /* R61 - PLL2 */
+	0x0026,     /* R62 - PLL3 */
+};
+
+#define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0)
+
+static const unsigned int rec_mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 7, TLV_DB_LINEAR_ITEM(-1500, 600),
+};
+
+static const unsigned int in_pga_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 0x1F, TLV_DB_LINEAR_ITEM(-1650, 3000),
+};
+
+static const unsigned int out_mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 7, TLV_DB_LINEAR_ITEM(0, -2100),
+};
+
+static const unsigned int out_pga_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 127, TLV_DB_LINEAR_ITEM(-7300, 600),
+};
+
+static const unsigned int out_omix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 7, TLV_DB_LINEAR_ITEM(-600, 0),
+};
+
+static const unsigned int out_dac_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 255, TLV_DB_LINEAR_ITEM(-7163, 0),
+};
+
+static const unsigned int in_adc_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 255, TLV_DB_LINEAR_ITEM(-7163, 1763),
+};
+
+static const unsigned int out_sidetone_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 31, TLV_DB_LINEAR_ITEM(-3600, 0),
+};
+
+static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int ret;
+	u16 val;
+
+	ret = snd_soc_put_volsw(kcontrol, ucontrol);
+	if (ret < 0)
+		return ret;
+
+	/* now hit the volume update bits (always bit 8) */
+	val = snd_soc_read(codec, reg);
+	return snd_soc_write(codec, reg, val | 0x0100);
+}
+
+static const char *wm8991_digital_sidetone[] =
+{"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8991_left_digital_sidetone_enum =
+	SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE,
+			WM8991_ADC_TO_DACL_SHIFT,
+			WM8991_ADC_TO_DACL_MASK,
+			wm8991_digital_sidetone);
+
+static const struct soc_enum wm8991_right_digital_sidetone_enum =
+	SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE,
+			WM8991_ADC_TO_DACR_SHIFT,
+			WM8991_ADC_TO_DACR_MASK,
+			wm8991_digital_sidetone);
+
+static const char *wm8991_adcmode[] =
+{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8991_right_adcmode_enum =
+	SOC_ENUM_SINGLE(WM8991_ADC_CTRL,
+			WM8991_ADC_HPF_CUT_SHIFT,
+			WM8991_ADC_HPF_CUT_MASK,
+			wm8991_adcmode);
+
+static const struct snd_kcontrol_new wm8991_snd_controls[] = {
+	/* INMIXL */
+	SOC_SINGLE("LIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L12MNBST_BIT, 1, 0),
+	SOC_SINGLE("LIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L34MNBST_BIT, 1, 0),
+	/* INMIXR */
+	SOC_SINGLE("RIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R12MNBST_BIT, 1, 0),
+	SOC_SINGLE("RIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R34MNBST_BIT, 1, 0),
+
+	/* LOMIX */
+	SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER3,
+		WM8991_LLI3LOVOL_SHIFT, WM8991_LLI3LOVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3,
+		WM8991_LR12LOVOL_SHIFT, WM8991_LR12LOVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3,
+		WM8991_LL12LOVOL_SHIFT, WM8991_LL12LOVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER5,
+		WM8991_LRI3LOVOL_SHIFT, WM8991_LRI3LOVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER5,
+		WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER5,
+		WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv),
+
+	/* ROMIX */
+	SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER4,
+		WM8991_RRI3ROVOL_SHIFT, WM8991_RRI3ROVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4,
+		WM8991_RL12ROVOL_SHIFT, WM8991_RL12ROVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4,
+		WM8991_RR12ROVOL_SHIFT, WM8991_RR12ROVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER6,
+		WM8991_RLI3ROVOL_SHIFT, WM8991_RLI3ROVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER6,
+		WM8991_RLBROVOL_SHIFT, WM8991_RLBROVOL_MASK, 1, out_mix_tlv),
+	SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER6,
+		WM8991_RRBROVOL_SHIFT, WM8991_RRBROVOL_MASK, 1, out_mix_tlv),
+
+	/* LOUT */
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8991_LEFT_OUTPUT_VOLUME,
+		WM8991_LOUTVOL_SHIFT, WM8991_LOUTVOL_MASK, 0, out_pga_tlv),
+	SOC_SINGLE("LOUT ZC", WM8991_LEFT_OUTPUT_VOLUME, WM8991_LOZC_BIT, 1, 0),
+
+	/* ROUT */
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8991_RIGHT_OUTPUT_VOLUME,
+		WM8991_ROUTVOL_SHIFT, WM8991_ROUTVOL_MASK, 0, out_pga_tlv),
+	SOC_SINGLE("ROUT ZC", WM8991_RIGHT_OUTPUT_VOLUME, WM8991_ROZC_BIT, 1, 0),
+
+	/* LOPGA */
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8991_LEFT_OPGA_VOLUME,
+		WM8991_LOPGAVOL_SHIFT, WM8991_LOPGAVOL_MASK, 0, out_pga_tlv),
+	SOC_SINGLE("LOPGA ZC Switch", WM8991_LEFT_OPGA_VOLUME,
+		WM8991_LOPGAZC_BIT, 1, 0),
+
+	/* ROPGA */
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8991_RIGHT_OPGA_VOLUME,
+		WM8991_ROPGAVOL_SHIFT, WM8991_ROPGAVOL_MASK, 0, out_pga_tlv),
+	SOC_SINGLE("ROPGA ZC Switch", WM8991_RIGHT_OPGA_VOLUME,
+		WM8991_ROPGAZC_BIT, 1, 0),
+
+	SOC_SINGLE("LON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME,
+		WM8991_LONMUTE_BIT, 1, 0),
+	SOC_SINGLE("LOP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME,
+		WM8991_LOPMUTE_BIT, 1, 0),
+	SOC_SINGLE("LOP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME,
+		WM8991_LOATTN_BIT, 1, 0),
+	SOC_SINGLE("RON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME,
+		WM8991_RONMUTE_BIT, 1, 0),
+	SOC_SINGLE("ROP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME,
+		WM8991_ROPMUTE_BIT, 1, 0),
+	SOC_SINGLE("ROP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME,
+		WM8991_ROATTN_BIT, 1, 0),
+
+	SOC_SINGLE("OUT3 Mute Switch", WM8991_OUT3_4_VOLUME,
+		WM8991_OUT3MUTE_BIT, 1, 0),
+	SOC_SINGLE("OUT3 Attenuation Switch", WM8991_OUT3_4_VOLUME,
+		WM8991_OUT3ATTN_BIT, 1, 0),
+
+	SOC_SINGLE("OUT4 Mute Switch", WM8991_OUT3_4_VOLUME,
+		WM8991_OUT4MUTE_BIT, 1, 0),
+	SOC_SINGLE("OUT4 Attenuation Switch", WM8991_OUT3_4_VOLUME,
+		WM8991_OUT4ATTN_BIT, 1, 0),
+
+	SOC_SINGLE("Speaker Mode Switch", WM8991_CLASSD1,
+		WM8991_CDMODE_BIT, 1, 0),
+
+	SOC_SINGLE("Speaker Output Attenuation Volume", WM8991_SPEAKER_VOLUME,
+		WM8991_SPKVOL_SHIFT, WM8991_SPKVOL_MASK, 0),
+	SOC_SINGLE("Speaker DC Boost Volume", WM8991_CLASSD3,
+		WM8991_DCGAIN_SHIFT, WM8991_DCGAIN_MASK, 0),
+	SOC_SINGLE("Speaker AC Boost Volume", WM8991_CLASSD3,
+		WM8991_ACGAIN_SHIFT, WM8991_ACGAIN_MASK, 0),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+		WM8991_LEFT_DAC_DIGITAL_VOLUME,
+		WM8991_DACL_VOL_SHIFT,
+		WM8991_DACL_VOL_MASK,
+		0,
+		out_dac_tlv),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+		WM8991_RIGHT_DAC_DIGITAL_VOLUME,
+		WM8991_DACR_VOL_SHIFT,
+		WM8991_DACR_VOL_MASK,
+		0,
+		out_dac_tlv),
+
+	SOC_ENUM("Left Digital Sidetone", wm8991_left_digital_sidetone_enum),
+	SOC_ENUM("Right Digital Sidetone", wm8991_right_digital_sidetone_enum),
+
+	SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE,
+		WM8991_ADCL_DAC_SVOL_SHIFT, WM8991_ADCL_DAC_SVOL_MASK, 0,
+		out_sidetone_tlv),
+	SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE,
+		WM8991_ADCR_DAC_SVOL_SHIFT, WM8991_ADCR_DAC_SVOL_MASK, 0,
+		out_sidetone_tlv),
+
+	SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8991_ADC_CTRL,
+		WM8991_ADC_HPF_ENA_BIT, 1, 0),
+
+	SOC_ENUM("ADC HPF Mode", wm8991_right_adcmode_enum),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+		WM8991_LEFT_ADC_DIGITAL_VOLUME,
+		WM8991_ADCL_VOL_SHIFT,
+		WM8991_ADCL_VOL_MASK,
+		0,
+		in_adc_tlv),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+		WM8991_RIGHT_ADC_DIGITAL_VOLUME,
+		WM8991_ADCR_VOL_SHIFT,
+		WM8991_ADCR_VOL_MASK,
+		0,
+		in_adc_tlv),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+		WM8991_LEFT_LINE_INPUT_1_2_VOLUME,
+		WM8991_LIN12VOL_SHIFT,
+		WM8991_LIN12VOL_MASK,
+		0,
+		in_pga_tlv),
+
+	SOC_SINGLE("LIN12 ZC Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME,
+		WM8991_LI12ZC_BIT, 1, 0),
+
+	SOC_SINGLE("LIN12 Mute Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME,
+		WM8991_LI12MUTE_BIT, 1, 0),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+		WM8991_LEFT_LINE_INPUT_3_4_VOLUME,
+		WM8991_LIN34VOL_SHIFT,
+		WM8991_LIN34VOL_MASK,
+		0,
+		in_pga_tlv),
+
+	SOC_SINGLE("LIN34 ZC Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME,
+		WM8991_LI34ZC_BIT, 1, 0),
+
+	SOC_SINGLE("LIN34 Mute Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME,
+		WM8991_LI34MUTE_BIT, 1, 0),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+		WM8991_RIGHT_LINE_INPUT_1_2_VOLUME,
+		WM8991_RIN12VOL_SHIFT,
+		WM8991_RIN12VOL_MASK,
+		0,
+		in_pga_tlv),
+
+	SOC_SINGLE("RIN12 ZC Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME,
+		WM8991_RI12ZC_BIT, 1, 0),
+
+	SOC_SINGLE("RIN12 Mute Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME,
+		WM8991_RI12MUTE_BIT, 1, 0),
+
+	SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+		WM8991_RIGHT_LINE_INPUT_3_4_VOLUME,
+		WM8991_RIN34VOL_SHIFT,
+		WM8991_RIN34VOL_MASK,
+		0,
+		in_pga_tlv),
+
+	SOC_SINGLE("RIN34 ZC Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME,
+		WM8991_RI34ZC_BIT, 1, 0),
+
+	SOC_SINGLE("RIN34 Mute Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME,
+		WM8991_RI34MUTE_BIT, 1, 0),
+};
+
+/*
+ * _DAPM_ Controls
+ */
+static int inmixer_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol, int event)
+{
+	u16 reg, fakepower;
+
+	reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2);
+	fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS);
+
+	if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) |
+			 (1 << WM8991_AINLMUX_PWR_BIT)))
+		reg |= WM8991_AINL_ENA;
+	else
+		reg &= ~WM8991_AINL_ENA;
+
+	if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) |
+			 (1 << WM8991_AINRMUX_PWR_BIT)))
+		reg |= WM8991_AINR_ENA;
+	else
+		reg &= ~WM8991_AINL_ENA;
+
+	snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg);
+	return 0;
+}
+
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol, int event)
+{
+	u32 reg_shift = kcontrol->private_value & 0xfff;
+	int ret = 0;
+	u16 reg;
+
+	switch (reg_shift) {
+	case WM8991_SPEAKER_MIXER | (WM8991_LDSPK_BIT << 8):
+		reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER1);
+		if (reg & WM8991_LDLO) {
+			printk(KERN_WARNING
+			       "Cannot set as Output Mixer 1 LDLO Set\n");
+			ret = -1;
+		}
+		break;
+
+	case WM8991_SPEAKER_MIXER | (WM8991_RDSPK_BIT << 8):
+		reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER2);
+		if (reg & WM8991_RDRO) {
+			printk(KERN_WARNING
+			       "Cannot set as Output Mixer 2 RDRO Set\n");
+			ret = -1;
+		}
+		break;
+
+	case WM8991_OUTPUT_MIXER1 | (WM8991_LDLO_BIT << 8):
+		reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER);
+		if (reg & WM8991_LDSPK) {
+			printk(KERN_WARNING
+			       "Cannot set as Speaker Mixer LDSPK Set\n");
+			ret = -1;
+		}
+		break;
+
+	case WM8991_OUTPUT_MIXER2 | (WM8991_RDRO_BIT << 8):
+		reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER);
+		if (reg & WM8991_RDSPK) {
+			printk(KERN_WARNING
+			       "Cannot set as Speaker Mixer RDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8991_dapm_lin12_pga_controls[] = {
+	SOC_DAPM_SINGLE("LIN1 Switch", WM8991_INPUT_MIXER2, WM8991_LMN1_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LIN2 Switch", WM8991_INPUT_MIXER2, WM8991_LMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8991_dapm_lin34_pga_controls[] = {
+	SOC_DAPM_SINGLE("LIN3 Switch", WM8991_INPUT_MIXER2, WM8991_LMN3_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LIN4 Switch", WM8991_INPUT_MIXER2, WM8991_LMP4_BIT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8991_dapm_rin12_pga_controls[] = {
+	SOC_DAPM_SINGLE("RIN1 Switch", WM8991_INPUT_MIXER2, WM8991_RMN1_BIT, 1, 0),
+	SOC_DAPM_SINGLE("RIN2 Switch", WM8991_INPUT_MIXER2, WM8991_RMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8991_dapm_rin34_pga_controls[] = {
+	SOC_DAPM_SINGLE("RIN3 Switch", WM8991_INPUT_MIXER2, WM8991_RMN3_BIT, 1, 0),
+	SOC_DAPM_SINGLE("RIN4 Switch", WM8991_INPUT_MIXER2, WM8991_RMP4_BIT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8991_dapm_inmixl_controls[] = {
+	SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8991_INPUT_MIXER3,
+		WM8991_LDBVOL_SHIFT, WM8991_LDBVOL_MASK, 0, in_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8991_INPUT_MIXER5, WM8991_LI2BVOL_SHIFT,
+		7, 0, in_mix_tlv),
+	SOC_DAPM_SINGLE("LINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT,
+		1, 0),
+	SOC_DAPM_SINGLE("LINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT,
+		1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8991_dapm_inmixr_controls[] = {
+	SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8991_INPUT_MIXER4,
+		WM8991_RDBVOL_SHIFT, WM8991_RDBVOL_MASK, 0, in_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8991_INPUT_MIXER6, WM8991_RI2BVOL_SHIFT,
+		7, 0, in_mix_tlv),
+	SOC_DAPM_SINGLE("RINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT,
+		1, 0),
+	SOC_DAPM_SINGLE("RINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT,
+		1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8991_ainlmux[] =
+{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8991_ainlmux_enum =
+	SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT,
+			ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux);
+
+static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls =
+	SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8991_ainrmux[] =
+{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8991_ainrmux_enum =
+	SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT,
+			ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux);
+
+static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls =
+	SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8991_dapm_rxvoice_controls[] = {
+	SOC_DAPM_SINGLE_TLV("LIN4RXN", WM8991_INPUT_MIXER5, WM8991_LR4BVOL_SHIFT,
+		WM8991_LR4BVOL_MASK, 0, in_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("RIN4RXP", WM8991_INPUT_MIXER6, WM8991_RL4BVOL_SHIFT,
+		WM8991_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8991_dapm_lomix_controls[] = {
+	SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER1,
+		WM8991_LRBLO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER1,
+		WM8991_LLBLO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER1,
+		WM8991_LRI3LO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER1,
+		WM8991_LLI3LO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1,
+		WM8991_LR12LO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1,
+		WM8991_LL12LO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8991_OUTPUT_MIXER1,
+		WM8991_LDLO_BIT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8991_dapm_romix_controls[] = {
+	SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER2,
+		WM8991_RLBRO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER2,
+		WM8991_RRBRO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER2,
+		WM8991_RLI3RO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER2,
+		WM8991_RRI3RO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2,
+		WM8991_RL12RO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2,
+		WM8991_RR12RO_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8991_OUTPUT_MIXER2,
+		WM8991_RDRO_BIT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8991_dapm_lonmix_controls[] = {
+	SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1,
+		WM8991_LLOPGALON_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER1,
+		WM8991_LROPGALON_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8991_LINE_MIXER1,
+		WM8991_LOPLON_BIT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8991_dapm_lopmix_controls[] = {
+	SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER1,
+		WM8991_LR12LOP_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER1,
+		WM8991_LL12LOP_BIT, 1, 0),
+	SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1,
+		WM8991_LLOPGALOP_BIT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8991_dapm_ronmix_controls[] = {
+	SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2,
+		WM8991_RROPGARON_BIT, 1, 0),
+	SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER2,
+		WM8991_RLOPGARON_BIT, 1, 0),
+	SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8991_LINE_MIXER2,
+		WM8991_ROPRON_BIT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8991_dapm_ropmix_controls[] = {
+	SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER2,
+		WM8991_RL12ROP_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER2,
+		WM8991_RR12ROP_BIT, 1, 0),
+	SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2,
+		WM8991_RROPGAROP_BIT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8991_dapm_out3mix_controls[] = {
+	SOC_DAPM_SINGLE("OUT3MIX LIN4RXN Bypass Switch", WM8991_OUT3_4_MIXER,
+		WM8991_LI4O3_BIT, 1, 0),
+	SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8991_OUT3_4_MIXER,
+		WM8991_LPGAO3_BIT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8991_dapm_out4mix_controls[] = {
+	SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8991_OUT3_4_MIXER,
+		WM8991_RPGAO4_BIT, 1, 0),
+	SOC_DAPM_SINGLE("OUT4MIX RIN4RXP Bypass Switch", WM8991_OUT3_4_MIXER,
+		WM8991_RI4O4_BIT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8991_dapm_spkmix_controls[] = {
+	SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8991_SPEAKER_MIXER,
+		WM8991_LI2SPK_BIT, 1, 0),
+	SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8991_SPEAKER_MIXER,
+		WM8991_LB2SPK_BIT, 1, 0),
+	SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8991_SPEAKER_MIXER,
+		WM8991_LOPGASPK_BIT, 1, 0),
+	SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8991_SPEAKER_MIXER,
+		WM8991_LDSPK_BIT, 1, 0),
+	SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8991_SPEAKER_MIXER,
+		WM8991_RDSPK_BIT, 1, 0),
+	SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8991_SPEAKER_MIXER,
+		WM8991_ROPGASPK_BIT, 1, 0),
+	SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8991_SPEAKER_MIXER,
+		WM8991_RL12ROP_BIT, 1, 0),
+	SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8991_SPEAKER_MIXER,
+		WM8991_RI2SPK_BIT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = {
+	/* Input Side */
+	/* Input Lines */
+	SND_SOC_DAPM_INPUT("LIN1"),
+	SND_SOC_DAPM_INPUT("LIN2"),
+	SND_SOC_DAPM_INPUT("LIN3"),
+	SND_SOC_DAPM_INPUT("LIN4RXN"),
+	SND_SOC_DAPM_INPUT("RIN3"),
+	SND_SOC_DAPM_INPUT("RIN4RXP"),
+	SND_SOC_DAPM_INPUT("RIN1"),
+	SND_SOC_DAPM_INPUT("RIN2"),
+	SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+	/* DACs */
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2,
+		WM8991_ADCL_ENA_BIT, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8991_POWER_MANAGEMENT_2,
+		WM8991_ADCR_ENA_BIT, 0),
+
+	/* Input PGAs */
+	SND_SOC_DAPM_MIXER("LIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN12_ENA_BIT,
+		0, &wm8991_dapm_lin12_pga_controls[0],
+		ARRAY_SIZE(wm8991_dapm_lin12_pga_controls)),
+	SND_SOC_DAPM_MIXER("LIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN34_ENA_BIT,
+		0, &wm8991_dapm_lin34_pga_controls[0],
+		ARRAY_SIZE(wm8991_dapm_lin34_pga_controls)),
+	SND_SOC_DAPM_MIXER("RIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN12_ENA_BIT,
+		0, &wm8991_dapm_rin12_pga_controls[0],
+		ARRAY_SIZE(wm8991_dapm_rin12_pga_controls)),
+	SND_SOC_DAPM_MIXER("RIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN34_ENA_BIT,
+		0, &wm8991_dapm_rin34_pga_controls[0],
+		ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)),
+
+	/* INMIXL */
+	SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0,
+		&wm8991_dapm_inmixl_controls[0],
+		ARRAY_SIZE(wm8991_dapm_inmixl_controls),
+		inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* AINLMUX */
+	SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0,
+		&wm8991_dapm_ainlmux_controls, inmixer_event,
+		SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* INMIXR */
+	SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0,
+		&wm8991_dapm_inmixr_controls[0],
+		ARRAY_SIZE(wm8991_dapm_inmixr_controls),
+		inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* AINRMUX */
+	SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0,
+		&wm8991_dapm_ainrmux_controls, inmixer_event,
+		SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* Output Side */
+	/* DACs */
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8991_POWER_MANAGEMENT_3,
+		WM8991_DACL_ENA_BIT, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8991_POWER_MANAGEMENT_3,
+		WM8991_DACR_ENA_BIT, 0),
+
+	/* LOMIX */
+	SND_SOC_DAPM_MIXER_E("LOMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOMIX_ENA_BIT,
+		0, &wm8991_dapm_lomix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_lomix_controls),
+		outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+	/* LONMIX */
+	SND_SOC_DAPM_MIXER("LONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LON_ENA_BIT, 0,
+		&wm8991_dapm_lonmix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_lonmix_controls)),
+
+	/* LOPMIX */
+	SND_SOC_DAPM_MIXER("LOPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOP_ENA_BIT, 0,
+		&wm8991_dapm_lopmix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_lopmix_controls)),
+
+	/* OUT3MIX */
+	SND_SOC_DAPM_MIXER("OUT3MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT3_ENA_BIT, 0,
+		&wm8991_dapm_out3mix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_out3mix_controls)),
+
+	/* SPKMIX */
+	SND_SOC_DAPM_MIXER_E("SPKMIX", WM8991_POWER_MANAGEMENT_1, WM8991_SPK_ENA_BIT, 0,
+		&wm8991_dapm_spkmix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_spkmix_controls), outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+
+	/* OUT4MIX */
+	SND_SOC_DAPM_MIXER("OUT4MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT4_ENA_BIT, 0,
+		&wm8991_dapm_out4mix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_out4mix_controls)),
+
+	/* ROPMIX */
+	SND_SOC_DAPM_MIXER("ROPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROP_ENA_BIT, 0,
+		&wm8991_dapm_ropmix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_ropmix_controls)),
+
+	/* RONMIX */
+	SND_SOC_DAPM_MIXER("RONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_RON_ENA_BIT, 0,
+		&wm8991_dapm_ronmix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_ronmix_controls)),
+
+	/* ROMIX */
+	SND_SOC_DAPM_MIXER_E("ROMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROMIX_ENA_BIT,
+		0, &wm8991_dapm_romix_controls[0],
+		ARRAY_SIZE(wm8991_dapm_romix_controls),
+		outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+	/* LOUT PGA */
+	SND_SOC_DAPM_PGA("LOUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_LOUT_ENA_BIT, 0,
+		NULL, 0),
+
+	/* ROUT PGA */
+	SND_SOC_DAPM_PGA("ROUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_ROUT_ENA_BIT, 0,
+		NULL, 0),
+
+	/* LOPGA */
+	SND_SOC_DAPM_PGA("LOPGA", WM8991_POWER_MANAGEMENT_3, WM8991_LOPGA_ENA_BIT, 0,
+		NULL, 0),
+
+	/* ROPGA */
+	SND_SOC_DAPM_PGA("ROPGA", WM8991_POWER_MANAGEMENT_3, WM8991_ROPGA_ENA_BIT, 0,
+		NULL, 0),
+
+	/* MICBIAS */
+	SND_SOC_DAPM_MICBIAS("MICBIAS", WM8991_POWER_MANAGEMENT_1,
+		WM8991_MICBIAS_ENA_BIT, 0),
+
+	SND_SOC_DAPM_OUTPUT("LON"),
+	SND_SOC_DAPM_OUTPUT("LOP"),
+	SND_SOC_DAPM_OUTPUT("OUT3"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("SPKN"),
+	SND_SOC_DAPM_OUTPUT("SPKP"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+	SND_SOC_DAPM_OUTPUT("OUT4"),
+	SND_SOC_DAPM_OUTPUT("ROP"),
+	SND_SOC_DAPM_OUTPUT("RON"),
+	SND_SOC_DAPM_OUTPUT("OUT"),
+
+	SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Make DACs turn on when playing even if not mixed into any outputs */
+	{"Internal DAC Sink", NULL, "Left DAC"},
+	{"Internal DAC Sink", NULL, "Right DAC"},
+
+	/* Make ADCs turn on when recording even if not mixed from any inputs */
+	{"Left ADC", NULL, "Internal ADC Source"},
+	{"Right ADC", NULL, "Internal ADC Source"},
+
+	/* Input Side */
+	/* LIN12 PGA */
+	{"LIN12 PGA", "LIN1 Switch", "LIN1"},
+	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
+	/* LIN34 PGA */
+	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
+	{"LIN34 PGA", "LIN4 Switch", "LIN4RXN"},
+	/* INMIXL */
+	{"INMIXL", "Record Left Volume", "LOMIX"},
+	{"INMIXL", "LIN2 Volume", "LIN2"},
+	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+	/* AINLMUX */
+	{"AINLMUX", "INMIXL Mix", "INMIXL"},
+	{"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+	{"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+	{"AINLMUX", "RXVOICE Mix", "LIN4RXN"},
+	{"AINLMUX", "RXVOICE Mix", "RIN4RXP"},
+	/* ADC */
+	{"Left ADC", NULL, "AINLMUX"},
+
+	/* RIN12 PGA */
+	{"RIN12 PGA", "RIN1 Switch", "RIN1"},
+	{"RIN12 PGA", "RIN2 Switch", "RIN2"},
+	/* RIN34 PGA */
+	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
+	{"RIN34 PGA", "RIN4 Switch", "RIN4RXP"},
+	/* INMIXL */
+	{"INMIXR", "Record Right Volume", "ROMIX"},
+	{"INMIXR", "RIN2 Volume", "RIN2"},
+	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+	/* AINRMUX */
+	{"AINRMUX", "INMIXR Mix", "INMIXR"},
+	{"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+	{"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+	{"AINRMUX", "RXVOICE Mix", "LIN4RXN"},
+	{"AINRMUX", "RXVOICE Mix", "RIN4RXP"},
+	/* ADC */
+	{"Right ADC", NULL, "AINRMUX"},
+
+	/* LOMIX */
+	{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+	{"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+	{"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"},
+	{"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"},
+	{"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+	/* ROMIX */
+	{"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+	{"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+	{"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"},
+	{"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"},
+	{"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+	/* SPKMIX */
+	{"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+	{"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+	{"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"},
+	{"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"},
+	{"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+	{"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+	{"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+	{"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+
+	/* LONMIX */
+	{"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+	/* LOPMIX */
+	{"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+	/* OUT3MIX */
+	{"OUT3MIX", "OUT3MIX LIN4RXN Bypass Switch", "LIN4RXN"},
+	{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+	/* OUT4MIX */
+	{"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+	{"OUT4MIX", "OUT4MIX RIN4RXP Bypass Switch", "RIN4RXP"},
+
+	/* RONMIX */
+	{"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+	/* ROPMIX */
+	{"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+	/* Out Mixer PGAs */
+	{"LOPGA", NULL, "LOMIX"},
+	{"ROPGA", NULL, "ROMIX"},
+
+	{"LOUT PGA", NULL, "LOMIX"},
+	{"ROUT PGA", NULL, "ROMIX"},
+
+	/* Output Pins */
+	{"LON", NULL, "LONMIX"},
+	{"LOP", NULL, "LOPMIX"},
+	{"OUT", NULL, "OUT3MIX"},
+	{"LOUT", NULL, "LOUT PGA"},
+	{"SPKN", NULL, "SPKMIX"},
+	{"ROUT", NULL, "ROUT PGA"},
+	{"OUT4", NULL, "OUT4MIX"},
+	{"ROP", NULL, "ROPMIX"},
+	{"RON", NULL, "RONMIX"},
+};
+
+/* PLL divisors */
+struct _pll_div {
+	u32 div2;
+	u32 n;
+	u32 k;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 16) * 10)
+
+static void pll_factors(struct _pll_div *pll_div, unsigned int target,
+			unsigned int source)
+{
+	u64 Kpart;
+	unsigned int K, Ndiv, Nmod;
+
+
+	Ndiv = target / source;
+	if (Ndiv < 6) {
+		source >>= 1;
+		pll_div->div2 = 1;
+		Ndiv = target / source;
+	} else
+		pll_div->div2 = 0;
+
+	if ((Ndiv < 6) || (Ndiv > 12))
+		printk(KERN_WARNING
+		       "WM8991 N value outwith recommended range! N = %d\n", Ndiv);
+
+	pll_div->n = Ndiv;
+	Nmod = target % source;
+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, source);
+
+	K = Kpart & 0xFFFFFFFF;
+
+	/* Check if we need to round */
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	K /= 10;
+
+	pll_div->k = K;
+}
+
+static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
+			      int pll_id, int src, unsigned int freq_in, unsigned int freq_out)
+{
+	u16 reg;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct _pll_div pll_div;
+
+	if (freq_in && freq_out) {
+		pll_factors(&pll_div, freq_out * 4, freq_in);
+
+		/* Turn on PLL */
+		reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2);
+		reg |= WM8991_PLL_ENA;
+		snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg);
+
+		/* sysclk comes from PLL */
+		reg = snd_soc_read(codec, WM8991_CLOCKING_2);
+		snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC);
+
+		/* set up N , fractional mode and pre-divisor if neccessary */
+		snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM |
+			      (pll_div.div2 ? WM8991_PRESCALE : 0));
+		snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8));
+		snd_soc_write(codec, WM8991_PLL3, (u8)(pll_div.k & 0xFF));
+	} else {
+		/* Turn on PLL */
+		reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2);
+		reg &= ~WM8991_PLL_ENA;
+		snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg);
+	}
+	return 0;
+}
+
+/*
+ * Set's ADC and Voice DAC format.
+ */
+static int wm8991_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 audio1, audio3;
+
+	audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1);
+	audio3 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_3);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		audio3 &= ~WM8991_AIF_MSTR1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		audio3 |= WM8991_AIF_MSTR1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	audio1 &= ~WM8991_AIF_FMT_MASK;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		audio1 |= WM8991_AIF_TMF_I2S;
+		audio1 &= ~WM8991_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		audio1 |= WM8991_AIF_TMF_RIGHTJ;
+		audio1 &= ~WM8991_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		audio1 |= WM8991_AIF_TMF_LEFTJ;
+		audio1 &= ~WM8991_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		audio1 |= WM8991_AIF_TMF_DSP;
+		audio1 &= ~WM8991_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		audio1 |= WM8991_AIF_TMF_DSP | WM8991_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1);
+	snd_soc_write(codec, WM8991_AUDIO_INTERFACE_3, audio3);
+	return 0;
+}
+
+static int wm8991_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+				 int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	switch (div_id) {
+	case WM8991_MCLK_DIV:
+		reg = snd_soc_read(codec, WM8991_CLOCKING_2) &
+		      ~WM8991_MCLK_DIV_MASK;
+		snd_soc_write(codec, WM8991_CLOCKING_2, reg | div);
+		break;
+	case WM8991_DACCLK_DIV:
+		reg = snd_soc_read(codec, WM8991_CLOCKING_2) &
+		      ~WM8991_DAC_CLKDIV_MASK;
+		snd_soc_write(codec, WM8991_CLOCKING_2, reg | div);
+		break;
+	case WM8991_ADCCLK_DIV:
+		reg = snd_soc_read(codec, WM8991_CLOCKING_2) &
+		      ~WM8991_ADC_CLKDIV_MASK;
+		snd_soc_write(codec, WM8991_CLOCKING_2, reg | div);
+		break;
+	case WM8991_BCLK_DIV:
+		reg = snd_soc_read(codec, WM8991_CLOCKING_1) &
+		      ~WM8991_BCLK_DIV_MASK;
+		snd_soc_write(codec, WM8991_CLOCKING_1, reg | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8991_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1);
+
+	audio1 &= ~WM8991_AIF_WL_MASK;
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		audio1 |= WM8991_AIF_WL_20BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		audio1 |= WM8991_AIF_WL_24BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		audio1 |= WM8991_AIF_WL_32BITS;
+		break;
+	}
+
+	snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1);
+	return 0;
+}
+
+static int wm8991_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 val;
+
+	val  = snd_soc_read(codec, WM8991_DAC_CTRL) & ~WM8991_DAC_MUTE;
+	if (mute)
+		snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE);
+	else
+		snd_soc_write(codec, WM8991_DAC_CTRL, val);
+	return 0;
+}
+
+static int wm8991_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 val;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VMID=2*50k */
+		val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) &
+		      ~WM8991_VMID_MODE_MASK;
+		snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x2);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_cache_sync(codec);
+			/* Enable all output discharge bits */
+			snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE |
+				      WM8991_DIS_RLINE | WM8991_DIS_OUT3 |
+				      WM8991_DIS_OUT4 | WM8991_DIS_LOUT |
+				      WM8991_DIS_ROUT);
+
+			/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+			snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST |
+				      WM8991_BUFDCOPEN | WM8991_POBCTRL |
+				      WM8991_VMIDTOG);
+
+			/* Delay to allow output caps to discharge */
+			msleep(300);
+
+			/* Disable VMIDTOG */
+			snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST |
+				      WM8991_BUFDCOPEN | WM8991_POBCTRL);
+
+			/* disable all output discharge bits */
+			snd_soc_write(codec, WM8991_ANTIPOP1, 0);
+
+			/* Enable outputs */
+			snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1b00);
+
+			msleep(50);
+
+			/* Enable VMID at 2x50k */
+			snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f02);
+
+			msleep(100);
+
+			/* Enable VREF */
+			snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03);
+
+			msleep(600);
+
+			/* Enable BUFIOEN */
+			snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST |
+				      WM8991_BUFDCOPEN | WM8991_POBCTRL |
+				      WM8991_BUFIOEN);
+
+			/* Disable outputs */
+			snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x3);
+
+			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+			snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_BUFIOEN);
+		}
+
+		/* VMID=2*250k */
+		val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) &
+		      ~WM8991_VMID_MODE_MASK;
+		snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x4);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* Enable POBCTRL and SOFT_ST */
+		snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST |
+			      WM8991_POBCTRL | WM8991_BUFIOEN);
+
+		/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+		snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST |
+			      WM8991_BUFDCOPEN | WM8991_POBCTRL |
+			      WM8991_BUFIOEN);
+
+		/* mute DAC */
+		val = snd_soc_read(codec, WM8991_DAC_CTRL);
+		snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE);
+
+		/* Enable any disabled outputs */
+		snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03);
+
+		/* Disable VMID */
+		snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f01);
+
+		msleep(300);
+
+		/* Enable all output discharge bits */
+		snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE |
+			      WM8991_DIS_RLINE | WM8991_DIS_OUT3 |
+			      WM8991_DIS_OUT4 | WM8991_DIS_LOUT |
+			      WM8991_DIS_ROUT);
+
+		/* Disable VREF */
+		snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x0);
+
+		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+		snd_soc_write(codec, WM8991_ANTIPOP2, 0x0);
+		codec->cache_sync = 1;
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static int wm8991_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8991_resume(struct snd_soc_codec *codec)
+{
+	wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+/* power down chip */
+static int wm8991_remove(struct snd_soc_codec *codec)
+{
+	wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8991_probe(struct snd_soc_codec *codec)
+{
+	struct wm8991_priv *wm8991;
+	int ret;
+	unsigned int reg;
+
+	wm8991 = snd_soc_codec_get_drvdata(codec);
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8991->control_type);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
+		return ret;
+	}
+
+	ret = wm8991_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
+	}
+
+	wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	reg = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_4);
+	snd_soc_write(codec, WM8991_AUDIO_INTERFACE_4, reg | WM8991_ALRCGPIO1);
+
+	reg = snd_soc_read(codec, WM8991_GPIO1_GPIO2) &
+	      ~WM8991_GPIO1_SEL_MASK;
+	snd_soc_write(codec, WM8991_GPIO1_GPIO2, reg | 1);
+
+	reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1);
+	snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, reg | WM8991_VREF_ENA|
+		      WM8991_VMID_MODE_MASK);
+
+	reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2);
+	snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg | WM8991_OPCLK_ENA);
+
+	snd_soc_write(codec, WM8991_DAC_CTRL, 0);
+	snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+	snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+	snd_soc_add_controls(codec, wm8991_snd_controls,
+			     ARRAY_SIZE(wm8991_snd_controls));
+
+	snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets,
+				  ARRAY_SIZE(wm8991_dapm_widgets));
+	snd_soc_dapm_add_routes(&codec->dapm, audio_map,
+				ARRAY_SIZE(audio_map));
+	return 0;
+}
+
+#define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8991_ops = {
+	.hw_params = wm8991_hw_params,
+	.digital_mute = wm8991_mute,
+	.set_fmt = wm8991_set_dai_fmt,
+	.set_clkdiv = wm8991_set_dai_clkdiv,
+	.set_pll = wm8991_set_dai_pll
+};
+
+/*
+ * The WM8991 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+static struct snd_soc_dai_driver wm8991_dai = {
+	/* ADC/DAC on primary */
+	.name = "wm8991",
+	.id = 1,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = WM8991_FORMATS
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = WM8991_FORMATS
+	},
+	.ops = &wm8991_ops
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8991 = {
+	.probe = wm8991_probe,
+	.remove = wm8991_remove,
+	.suspend = wm8991_suspend,
+	.resume = wm8991_resume,
+	.set_bias_level = wm8991_set_bias_level,
+	.reg_cache_size = WM8991_MAX_REGISTER + 1,
+	.reg_word_size = sizeof(u16),
+	.reg_cache_default = wm8991_reg_defs
+};
+
+static __devinit int wm8991_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct wm8991_priv *wm8991;
+	int ret;
+
+	wm8991 = kzalloc(sizeof *wm8991, GFP_KERNEL);
+	if (!wm8991)
+		return -ENOMEM;
+
+	wm8991->control_type = SND_SOC_I2C;
+	i2c_set_clientdata(i2c, wm8991);
+
+	ret = snd_soc_register_codec(&i2c->dev,
+				     &soc_codec_dev_wm8991, &wm8991_dai, 1);
+	if (ret < 0)
+		kfree(wm8991);
+	return ret;
+}
+
+static __devexit int wm8991_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(i2c_get_clientdata(client));
+	return 0;
+}
+
+static const struct i2c_device_id wm8991_i2c_id[] = {
+	{ "wm8991", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id);
+
+static struct i2c_driver wm8991_i2c_driver = {
+	.driver = {
+		.name = "wm8991",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8991_i2c_probe,
+	.remove = __devexit_p(wm8991_i2c_remove),
+	.id_table = wm8991_i2c_id,
+};
+
+static int __init wm8991_modinit(void)
+{
+	int ret;
+	ret = i2c_add_driver(&wm8991_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n",
+		       ret);
+	}
+	return 0;
+}
+module_init(wm8991_modinit);
+
+static void __exit wm8991_exit(void)
+{
+	i2c_del_driver(&wm8991_i2c_driver);
+}
+module_exit(wm8991_exit);
+
+MODULE_DESCRIPTION("ASoC WM8991 driver");
+MODULE_AUTHOR("Graeme Gregory");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h
new file mode 100644
index 00000000000..8a942efd18a
--- /dev/null
+++ b/sound/soc/codecs/wm8991.h
@@ -0,0 +1,833 @@
+/*
+ * wm8991.h  --  audio driver for WM8991
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ *  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.
+ */
+
+#ifndef _WM8991_H
+#define _WM8991_H
+
+/*
+ * Register values.
+ */
+#define WM8991_RESET                            0x00
+#define WM8991_POWER_MANAGEMENT_1               0x01
+#define WM8991_POWER_MANAGEMENT_2               0x02
+#define WM8991_POWER_MANAGEMENT_3               0x03
+#define WM8991_AUDIO_INTERFACE_1                0x04
+#define WM8991_AUDIO_INTERFACE_2                0x05
+#define WM8991_CLOCKING_1                       0x06
+#define WM8991_CLOCKING_2                       0x07
+#define WM8991_AUDIO_INTERFACE_3                0x08
+#define WM8991_AUDIO_INTERFACE_4                0x09
+#define WM8991_DAC_CTRL                         0x0A
+#define WM8991_LEFT_DAC_DIGITAL_VOLUME          0x0B
+#define WM8991_RIGHT_DAC_DIGITAL_VOLUME         0x0C
+#define WM8991_DIGITAL_SIDE_TONE                0x0D
+#define WM8991_ADC_CTRL                         0x0E
+#define WM8991_LEFT_ADC_DIGITAL_VOLUME          0x0F
+#define WM8991_RIGHT_ADC_DIGITAL_VOLUME         0x10
+#define WM8991_GPIO_CTRL_1                      0x12
+#define WM8991_GPIO1_GPIO2                      0x13
+#define WM8991_GPIO3_GPIO4                      0x14
+#define WM8991_GPIO5_GPIO6                      0x15
+#define WM8991_GPIOCTRL_2                       0x16
+#define WM8991_GPIO_POL                         0x17
+#define WM8991_LEFT_LINE_INPUT_1_2_VOLUME       0x18
+#define WM8991_LEFT_LINE_INPUT_3_4_VOLUME       0x19
+#define WM8991_RIGHT_LINE_INPUT_1_2_VOLUME      0x1A
+#define WM8991_RIGHT_LINE_INPUT_3_4_VOLUME      0x1B
+#define WM8991_LEFT_OUTPUT_VOLUME               0x1C
+#define WM8991_RIGHT_OUTPUT_VOLUME              0x1D
+#define WM8991_LINE_OUTPUTS_VOLUME              0x1E
+#define WM8991_OUT3_4_VOLUME                    0x1F
+#define WM8991_LEFT_OPGA_VOLUME                 0x20
+#define WM8991_RIGHT_OPGA_VOLUME                0x21
+#define WM8991_SPEAKER_VOLUME                   0x22
+#define WM8991_CLASSD1                          0x23
+#define WM8991_CLASSD3                          0x25
+#define WM8991_INPUT_MIXER1                     0x27
+#define WM8991_INPUT_MIXER2                     0x28
+#define WM8991_INPUT_MIXER3                     0x29
+#define WM8991_INPUT_MIXER4                     0x2A
+#define WM8991_INPUT_MIXER5                     0x2B
+#define WM8991_INPUT_MIXER6                     0x2C
+#define WM8991_OUTPUT_MIXER1                    0x2D
+#define WM8991_OUTPUT_MIXER2                    0x2E
+#define WM8991_OUTPUT_MIXER3                    0x2F
+#define WM8991_OUTPUT_MIXER4                    0x30
+#define WM8991_OUTPUT_MIXER5                    0x31
+#define WM8991_OUTPUT_MIXER6                    0x32
+#define WM8991_OUT3_4_MIXER                     0x33
+#define WM8991_LINE_MIXER1                      0x34
+#define WM8991_LINE_MIXER2                      0x35
+#define WM8991_SPEAKER_MIXER                    0x36
+#define WM8991_ADDITIONAL_CONTROL               0x37
+#define WM8991_ANTIPOP1                         0x38
+#define WM8991_ANTIPOP2                         0x39
+#define WM8991_MICBIAS                          0x3A
+#define WM8991_PLL1                             0x3C
+#define WM8991_PLL2                             0x3D
+#define WM8991_PLL3                             0x3E
+#define WM8991_INTDRIVBITS			0x3F
+
+#define WM8991_REGISTER_COUNT                   60
+#define WM8991_MAX_REGISTER                     0x3F
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Reset
+ */
+#define WM8991_SW_RESET_CHIP_ID_MASK            0xFFFF  /* SW_RESET_CHIP_ID - [15:0] */
+
+/*
+ * R1 (0x01) - Power Management (1)
+ */
+#define WM8991_SPK_ENA                          0x1000  /* SPK_ENA */
+#define WM8991_SPK_ENA_BIT			12
+#define WM8991_OUT3_ENA                         0x0800  /* OUT3_ENA */
+#define WM8991_OUT3_ENA_BIT			11
+#define WM8991_OUT4_ENA                         0x0400  /* OUT4_ENA */
+#define WM8991_OUT4_ENA_BIT			10
+#define WM8991_LOUT_ENA                         0x0200  /* LOUT_ENA */
+#define WM8991_LOUT_ENA_BIT			9
+#define WM8991_ROUT_ENA                         0x0100  /* ROUT_ENA */
+#define WM8991_ROUT_ENA_BIT			8
+#define WM8991_MICBIAS_ENA                      0x0010  /* MICBIAS_ENA */
+#define WM8991_MICBIAS_ENA_BIT			4
+#define WM8991_VMID_MODE_MASK                   0x0006  /* VMID_MODE - [2:1] */
+#define WM8991_VREF_ENA                         0x0001  /* VREF_ENA */
+#define WM8991_VREF_ENA_BIT			0
+
+/*
+ * R2 (0x02) - Power Management (2)
+ */
+#define WM8991_PLL_ENA                          0x8000  /* PLL_ENA */
+#define WM8991_PLL_ENA_BIT			15
+#define WM8991_TSHUT_ENA                        0x4000  /* TSHUT_ENA */
+#define WM8991_TSHUT_ENA_BIT			14
+#define WM8991_TSHUT_OPDIS                      0x2000  /* TSHUT_OPDIS */
+#define WM8991_TSHUT_OPDIS_BIT			13
+#define WM8991_OPCLK_ENA                        0x0800  /* OPCLK_ENA */
+#define WM8991_OPCLK_ENA_BIT			11
+#define WM8991_AINL_ENA                         0x0200  /* AINL_ENA */
+#define WM8991_AINL_ENA_BIT			9
+#define WM8991_AINR_ENA                         0x0100  /* AINR_ENA */
+#define WM8991_AINR_ENA_BIT			8
+#define WM8991_LIN34_ENA                        0x0080  /* LIN34_ENA */
+#define WM8991_LIN34_ENA_BIT			7
+#define WM8991_LIN12_ENA                        0x0040  /* LIN12_ENA */
+#define WM8991_LIN12_ENA_BIT			6
+#define WM8991_RIN34_ENA                        0x0020  /* RIN34_ENA */
+#define WM8991_RIN34_ENA_BIT			5
+#define WM8991_RIN12_ENA                        0x0010  /* RIN12_ENA */
+#define WM8991_RIN12_ENA_BIT			4
+#define WM8991_ADCL_ENA                         0x0002  /* ADCL_ENA */
+#define WM8991_ADCL_ENA_BIT			1
+#define WM8991_ADCR_ENA                         0x0001  /* ADCR_ENA */
+#define WM8991_ADCR_ENA_BIT			0
+
+/*
+ * R3 (0x03) - Power Management (3)
+ */
+#define WM8991_LON_ENA                          0x2000  /* LON_ENA */
+#define WM8991_LON_ENA_BIT			13
+#define WM8991_LOP_ENA                          0x1000  /* LOP_ENA */
+#define WM8991_LOP_ENA_BIT			12
+#define WM8991_RON_ENA                          0x0800  /* RON_ENA */
+#define WM8991_RON_ENA_BIT			11
+#define WM8991_ROP_ENA                          0x0400  /* ROP_ENA */
+#define WM8991_ROP_ENA_BIT			10
+#define WM8991_LOPGA_ENA                        0x0080  /* LOPGA_ENA */
+#define WM8991_LOPGA_ENA_BIT			7
+#define WM8991_ROPGA_ENA                        0x0040  /* ROPGA_ENA */
+#define WM8991_ROPGA_ENA_BIT			6
+#define WM8991_LOMIX_ENA                        0x0020  /* LOMIX_ENA */
+#define WM8991_LOMIX_ENA_BIT			5
+#define WM8991_ROMIX_ENA                        0x0010  /* ROMIX_ENA */
+#define WM8991_ROMIX_ENA_BIT			4
+#define WM8991_DACL_ENA                         0x0002  /* DACL_ENA */
+#define WM8991_DACL_ENA_BIT			1
+#define WM8991_DACR_ENA                         0x0001  /* DACR_ENA */
+#define WM8991_DACR_ENA_BIT			0
+
+/*
+ * R4 (0x04) - Audio Interface (1)
+ */
+#define WM8991_AIFADCL_SRC                      0x8000  /* AIFADCL_SRC */
+#define WM8991_AIFADCR_SRC                      0x4000  /* AIFADCR_SRC */
+#define WM8991_AIFADC_TDM                       0x2000  /* AIFADC_TDM */
+#define WM8991_AIFADC_TDM_CHAN                  0x1000  /* AIFADC_TDM_CHAN */
+#define WM8991_AIF_BCLK_INV                     0x0100  /* AIF_BCLK_INV */
+#define WM8991_AIF_LRCLK_INV                    0x0080  /* AIF_LRCLK_INV */
+#define WM8991_AIF_WL_MASK                      0x0060  /* AIF_WL - [6:5] */
+#define WM8991_AIF_WL_16BITS			(0 << 5)
+#define WM8991_AIF_WL_20BITS			(1 << 5)
+#define WM8991_AIF_WL_24BITS			(2 << 5)
+#define WM8991_AIF_WL_32BITS			(3 << 5)
+#define WM8991_AIF_FMT_MASK                     0x0018  /* AIF_FMT - [4:3] */
+#define WM8991_AIF_TMF_RIGHTJ			(0 << 3)
+#define WM8991_AIF_TMF_LEFTJ			(1 << 3)
+#define WM8991_AIF_TMF_I2S			(2 << 3)
+#define WM8991_AIF_TMF_DSP			(3 << 3)
+
+/*
+ * R5 (0x05) - Audio Interface (2)
+ */
+#define WM8991_DACL_SRC                         0x8000  /* DACL_SRC */
+#define WM8991_DACR_SRC                         0x4000  /* DACR_SRC */
+#define WM8991_AIFDAC_TDM                       0x2000  /* AIFDAC_TDM */
+#define WM8991_AIFDAC_TDM_CHAN                  0x1000  /* AIFDAC_TDM_CHAN */
+#define WM8991_DAC_BOOST_MASK                   0x0C00  /* DAC_BOOST - [11:10] */
+#define WM8991_DAC_COMP                         0x0010  /* DAC_COMP */
+#define WM8991_DAC_COMPMODE                     0x0008  /* DAC_COMPMODE */
+#define WM8991_ADC_COMP                         0x0004  /* ADC_COMP */
+#define WM8991_ADC_COMPMODE                     0x0002  /* ADC_COMPMODE */
+#define WM8991_LOOPBACK                         0x0001  /* LOOPBACK */
+
+/*
+ * R6 (0x06) - Clocking (1)
+ */
+#define WM8991_TOCLK_RATE                       0x8000  /* TOCLK_RATE */
+#define WM8991_TOCLK_ENA                        0x4000  /* TOCLK_ENA */
+#define WM8991_OPCLKDIV_MASK                    0x1E00  /* OPCLKDIV - [12:9] */
+#define WM8991_DCLKDIV_MASK                     0x01C0  /* DCLKDIV - [8:6] */
+#define WM8991_BCLK_DIV_MASK                    0x001E  /* BCLK_DIV - [4:1] */
+#define WM8991_BCLK_DIV_1			(0x0 << 1)
+#define WM8991_BCLK_DIV_1_5			(0x1 << 1)
+#define WM8991_BCLK_DIV_2			(0x2 << 1)
+#define WM8991_BCLK_DIV_3			(0x3 << 1)
+#define WM8991_BCLK_DIV_4			(0x4 << 1)
+#define WM8991_BCLK_DIV_5_5			(0x5 << 1)
+#define WM8991_BCLK_DIV_6			(0x6 << 1)
+#define WM8991_BCLK_DIV_8			(0x7 << 1)
+#define WM8991_BCLK_DIV_11			(0x8 << 1)
+#define WM8991_BCLK_DIV_12			(0x9 << 1)
+#define WM8991_BCLK_DIV_16			(0xA << 1)
+#define WM8991_BCLK_DIV_22			(0xB << 1)
+#define WM8991_BCLK_DIV_24			(0xC << 1)
+#define WM8991_BCLK_DIV_32			(0xD << 1)
+#define WM8991_BCLK_DIV_44			(0xE << 1)
+#define WM8991_BCLK_DIV_48			(0xF << 1)
+
+/*
+ * R7 (0x07) - Clocking (2)
+ */
+#define WM8991_MCLK_SRC                         0x8000  /* MCLK_SRC */
+#define WM8991_SYSCLK_SRC                       0x4000  /* SYSCLK_SRC */
+#define WM8991_CLK_FORCE                        0x2000  /* CLK_FORCE */
+#define WM8991_MCLK_DIV_MASK                    0x1800  /* MCLK_DIV - [12:11] */
+#define WM8991_MCLK_DIV_1			(0 << 11)
+#define WM8991_MCLK_DIV_2			( 2 << 11)
+#define WM8991_MCLK_INV                         0x0400  /* MCLK_INV */
+#define WM8991_ADC_CLKDIV_MASK                  0x00E0  /* ADC_CLKDIV - [7:5] */
+#define WM8991_ADC_CLKDIV_1			(0 << 5)
+#define WM8991_ADC_CLKDIV_1_5			(1 << 5)
+#define WM8991_ADC_CLKDIV_2			(2 << 5)
+#define WM8991_ADC_CLKDIV_3			(3 << 5)
+#define WM8991_ADC_CLKDIV_4			(4 << 5)
+#define WM8991_ADC_CLKDIV_5_5			(5 << 5)
+#define WM8991_ADC_CLKDIV_6			(6 << 5)
+#define WM8991_DAC_CLKDIV_MASK                  0x001C  /* DAC_CLKDIV - [4:2] */
+#define WM8991_DAC_CLKDIV_1			(0 << 2)
+#define WM8991_DAC_CLKDIV_1_5			(1 << 2)
+#define WM8991_DAC_CLKDIV_2			(2 << 2)
+#define WM8991_DAC_CLKDIV_3			(3 << 2)
+#define WM8991_DAC_CLKDIV_4			(4 << 2)
+#define WM8991_DAC_CLKDIV_5_5			(5 << 2)
+#define WM8991_DAC_CLKDIV_6			(6 << 2)
+
+/*
+ * R8 (0x08) - Audio Interface (3)
+ */
+#define WM8991_AIF_MSTR1                        0x8000  /* AIF_MSTR1 */
+#define WM8991_AIF_MSTR2                        0x4000  /* AIF_MSTR2 */
+#define WM8991_AIF_SEL                          0x2000  /* AIF_SEL */
+#define WM8991_ADCLRC_DIR                       0x0800  /* ADCLRC_DIR */
+#define WM8991_ADCLRC_RATE_MASK                 0x07FF  /* ADCLRC_RATE - [10:0] */
+
+/*
+ * R9 (0x09) - Audio Interface (4)
+ */
+#define WM8991_ALRCGPIO1                        0x8000  /* ALRCGPIO1 */
+#define WM8991_ALRCBGPIO6                       0x4000  /* ALRCBGPIO6 */
+#define WM8991_AIF_TRIS                         0x2000  /* AIF_TRIS */
+#define WM8991_DACLRC_DIR                       0x0800  /* DACLRC_DIR */
+#define WM8991_DACLRC_RATE_MASK                 0x07FF  /* DACLRC_RATE - [10:0] */
+
+/*
+ * R10 (0x0A) - DAC CTRL
+ */
+#define WM8991_AIF_LRCLKRATE                    0x0400  /* AIF_LRCLKRATE */
+#define WM8991_DAC_MONO                         0x0200  /* DAC_MONO */
+#define WM8991_DAC_SB_FILT                      0x0100  /* DAC_SB_FILT */
+#define WM8991_DAC_MUTERATE                     0x0080  /* DAC_MUTERATE */
+#define WM8991_DAC_MUTEMODE                     0x0040  /* DAC_MUTEMODE */
+#define WM8991_DEEMP_MASK                       0x0030  /* DEEMP - [5:4] */
+#define WM8991_DAC_MUTE                         0x0004  /* DAC_MUTE */
+#define WM8991_DACL_DATINV                      0x0002  /* DACL_DATINV */
+#define WM8991_DACR_DATINV                      0x0001  /* DACR_DATINV */
+
+/*
+ * R11 (0x0B) - Left DAC Digital Volume
+ */
+#define WM8991_DAC_VU                           0x0100  /* DAC_VU */
+#define WM8991_DACL_VOL_MASK                    0x00FF  /* DACL_VOL - [7:0] */
+#define WM8991_DACL_VOL_SHIFT			0
+/*
+ * R12 (0x0C) - Right DAC Digital Volume
+ */
+#define WM8991_DAC_VU                           0x0100  /* DAC_VU */
+#define WM8991_DACR_VOL_MASK                    0x00FF  /* DACR_VOL - [7:0] */
+#define WM8991_DACR_VOL_SHIFT			0
+/*
+ * R13 (0x0D) - Digital Side Tone
+ */
+#define WM8991_ADCL_DAC_SVOL_MASK               0x0F  /* ADCL_DAC_SVOL - [12:9] */
+#define WM8991_ADCL_DAC_SVOL_SHIFT		9
+#define WM8991_ADCR_DAC_SVOL_MASK               0x0F  /* ADCR_DAC_SVOL - [8:5] */
+#define WM8991_ADCR_DAC_SVOL_SHIFT		5
+#define WM8991_ADC_TO_DACL_MASK                 0x03  /* ADC_TO_DACL - [3:2] */
+#define WM8991_ADC_TO_DACL_SHIFT		2
+#define WM8991_ADC_TO_DACR_MASK                 0x03  /* ADC_TO_DACR - [1:0] */
+#define WM8991_ADC_TO_DACR_SHIFT		0
+
+/*
+ * R14 (0x0E) - ADC CTRL
+ */
+#define WM8991_ADC_HPF_ENA                      0x0100  /* ADC_HPF_ENA */
+#define WM8991_ADC_HPF_ENA_BIT			8
+#define WM8991_ADC_HPF_CUT_MASK                 0x03  /* ADC_HPF_CUT - [6:5] */
+#define WM8991_ADC_HPF_CUT_SHIFT		5
+#define WM8991_ADCL_DATINV                      0x0002  /* ADCL_DATINV */
+#define WM8991_ADCL_DATINV_BIT			1
+#define WM8991_ADCR_DATINV                      0x0001  /* ADCR_DATINV */
+#define WM8991_ADCR_DATINV_BIT			0
+
+/*
+ * R15 (0x0F) - Left ADC Digital Volume
+ */
+#define WM8991_ADC_VU                           0x0100  /* ADC_VU */
+#define WM8991_ADCL_VOL_MASK                    0x00FF  /* ADCL_VOL - [7:0] */
+#define WM8991_ADCL_VOL_SHIFT			0
+
+/*
+ * R16 (0x10) - Right ADC Digital Volume
+ */
+#define WM8991_ADC_VU                           0x0100  /* ADC_VU */
+#define WM8991_ADCR_VOL_MASK                    0x00FF  /* ADCR_VOL - [7:0] */
+#define WM8991_ADCR_VOL_SHIFT			0
+
+/*
+ * R18 (0x12) - GPIO CTRL 1
+ */
+#define WM8991_IRQ                              0x1000  /* IRQ */
+#define WM8991_TEMPOK                           0x0800  /* TEMPOK */
+#define WM8991_MICSHRT                          0x0400  /* MICSHRT */
+#define WM8991_MICDET                           0x0200  /* MICDET */
+#define WM8991_PLL_LCK                          0x0100  /* PLL_LCK */
+#define WM8991_GPI8_STATUS                      0x0080  /* GPI8_STATUS */
+#define WM8991_GPI7_STATUS                      0x0040  /* GPI7_STATUS */
+#define WM8991_GPIO6_STATUS                     0x0020  /* GPIO6_STATUS */
+#define WM8991_GPIO5_STATUS                     0x0010  /* GPIO5_STATUS */
+#define WM8991_GPIO4_STATUS                     0x0008  /* GPIO4_STATUS */
+#define WM8991_GPIO3_STATUS                     0x0004  /* GPIO3_STATUS */
+#define WM8991_GPIO2_STATUS                     0x0002  /* GPIO2_STATUS */
+#define WM8991_GPIO1_STATUS                     0x0001  /* GPIO1_STATUS */
+
+/*
+ * R19 (0x13) - GPIO1 & GPIO2
+ */
+#define WM8991_GPIO2_DEB_ENA                    0x8000  /* GPIO2_DEB_ENA */
+#define WM8991_GPIO2_IRQ_ENA                    0x4000  /* GPIO2_IRQ_ENA */
+#define WM8991_GPIO2_PU                         0x2000  /* GPIO2_PU */
+#define WM8991_GPIO2_PD                         0x1000  /* GPIO2_PD */
+#define WM8991_GPIO2_SEL_MASK                   0x0F00  /* GPIO2_SEL - [11:8] */
+#define WM8991_GPIO1_DEB_ENA                    0x0080  /* GPIO1_DEB_ENA */
+#define WM8991_GPIO1_IRQ_ENA                    0x0040  /* GPIO1_IRQ_ENA */
+#define WM8991_GPIO1_PU                         0x0020  /* GPIO1_PU */
+#define WM8991_GPIO1_PD                         0x0010  /* GPIO1_PD */
+#define WM8991_GPIO1_SEL_MASK                   0x000F  /* GPIO1_SEL - [3:0] */
+
+/*
+ * R20 (0x14) - GPIO3 & GPIO4
+ */
+#define WM8991_GPIO4_DEB_ENA                    0x8000  /* GPIO4_DEB_ENA */
+#define WM8991_GPIO4_IRQ_ENA                    0x4000  /* GPIO4_IRQ_ENA */
+#define WM8991_GPIO4_PU                         0x2000  /* GPIO4_PU */
+#define WM8991_GPIO4_PD                         0x1000  /* GPIO4_PD */
+#define WM8991_GPIO4_SEL_MASK                   0x0F00  /* GPIO4_SEL - [11:8] */
+#define WM8991_GPIO3_DEB_ENA                    0x0080  /* GPIO3_DEB_ENA */
+#define WM8991_GPIO3_IRQ_ENA                    0x0040  /* GPIO3_IRQ_ENA */
+#define WM8991_GPIO3_PU                         0x0020  /* GPIO3_PU */
+#define WM8991_GPIO3_PD                         0x0010  /* GPIO3_PD */
+#define WM8991_GPIO3_SEL_MASK                   0x000F  /* GPIO3_SEL - [3:0] */
+
+/*
+ * R21 (0x15) - GPIO5 & GPIO6
+ */
+#define WM8991_GPIO6_DEB_ENA                    0x8000  /* GPIO6_DEB_ENA */
+#define WM8991_GPIO6_IRQ_ENA                    0x4000  /* GPIO6_IRQ_ENA */
+#define WM8991_GPIO6_PU                         0x2000  /* GPIO6_PU */
+#define WM8991_GPIO6_PD                         0x1000  /* GPIO6_PD */
+#define WM8991_GPIO6_SEL_MASK                   0x0F00  /* GPIO6_SEL - [11:8] */
+#define WM8991_GPIO5_DEB_ENA                    0x0080  /* GPIO5_DEB_ENA */
+#define WM8991_GPIO5_IRQ_ENA                    0x0040  /* GPIO5_IRQ_ENA */
+#define WM8991_GPIO5_PU                         0x0020  /* GPIO5_PU */
+#define WM8991_GPIO5_PD                         0x0010  /* GPIO5_PD */
+#define WM8991_GPIO5_SEL_MASK                   0x000F  /* GPIO5_SEL - [3:0] */
+
+/*
+ * R22 (0x16) - GPIOCTRL 2
+ */
+#define WM8991_RD_3W_ENA                        0x8000  /* RD_3W_ENA */
+#define WM8991_MODE_3W4W                        0x4000  /* MODE_3W4W */
+#define WM8991_TEMPOK_IRQ_ENA                   0x0800  /* TEMPOK_IRQ_ENA */
+#define WM8991_MICSHRT_IRQ_ENA                  0x0400  /* MICSHRT_IRQ_ENA */
+#define WM8991_MICDET_IRQ_ENA                   0x0200  /* MICDET_IRQ_ENA */
+#define WM8991_PLL_LCK_IRQ_ENA                  0x0100  /* PLL_LCK_IRQ_ENA */
+#define WM8991_GPI8_DEB_ENA                     0x0080  /* GPI8_DEB_ENA */
+#define WM8991_GPI8_IRQ_ENA                     0x0040  /* GPI8_IRQ_ENA */
+#define WM8991_GPI8_ENA                         0x0010  /* GPI8_ENA */
+#define WM8991_GPI7_DEB_ENA                     0x0008  /* GPI7_DEB_ENA */
+#define WM8991_GPI7_IRQ_ENA                     0x0004  /* GPI7_IRQ_ENA */
+#define WM8991_GPI7_ENA                         0x0001  /* GPI7_ENA */
+
+/*
+ * R23 (0x17) - GPIO_POL
+ */
+#define WM8991_IRQ_INV                          0x1000  /* IRQ_INV */
+#define WM8991_TEMPOK_POL                       0x0800  /* TEMPOK_POL */
+#define WM8991_MICSHRT_POL                      0x0400  /* MICSHRT_POL */
+#define WM8991_MICDET_POL                       0x0200  /* MICDET_POL */
+#define WM8991_PLL_LCK_POL                      0x0100  /* PLL_LCK_POL */
+#define WM8991_GPI8_POL                         0x0080  /* GPI8_POL */
+#define WM8991_GPI7_POL                         0x0040  /* GPI7_POL */
+#define WM8991_GPIO6_POL                        0x0020  /* GPIO6_POL */
+#define WM8991_GPIO5_POL                        0x0010  /* GPIO5_POL */
+#define WM8991_GPIO4_POL                        0x0008  /* GPIO4_POL */
+#define WM8991_GPIO3_POL                        0x0004  /* GPIO3_POL */
+#define WM8991_GPIO2_POL                        0x0002  /* GPIO2_POL */
+#define WM8991_GPIO1_POL                        0x0001  /* GPIO1_POL */
+
+/*
+ * R24 (0x18) - Left Line Input 1&2 Volume
+ */
+#define WM8991_IPVU                             0x0100  /* IPVU */
+#define WM8991_LI12MUTE                         0x0080  /* LI12MUTE */
+#define WM8991_LI12MUTE_BIT			7
+#define WM8991_LI12ZC                           0x0040  /* LI12ZC */
+#define WM8991_LI12ZC_BIT			6
+#define WM8991_LIN12VOL_MASK                    0x001F  /* LIN12VOL - [4:0] */
+#define WM8991_LIN12VOL_SHIFT			0
+/*
+ * R25 (0x19) - Left Line Input 3&4 Volume
+ */
+#define WM8991_IPVU                             0x0100  /* IPVU */
+#define WM8991_LI34MUTE                         0x0080  /* LI34MUTE */
+#define WM8991_LI34MUTE_BIT			7
+#define WM8991_LI34ZC                           0x0040  /* LI34ZC */
+#define WM8991_LI34ZC_BIT			6
+#define WM8991_LIN34VOL_MASK                    0x001F  /* LIN34VOL - [4:0] */
+#define WM8991_LIN34VOL_SHIFT			0
+
+/*
+ * R26 (0x1A) - Right Line Input 1&2 Volume
+ */
+#define WM8991_IPVU                             0x0100  /* IPVU */
+#define WM8991_RI12MUTE                         0x0080  /* RI12MUTE */
+#define WM8991_RI12MUTE_BIT			7
+#define WM8991_RI12ZC                           0x0040  /* RI12ZC */
+#define WM8991_RI12ZC_BIT			6
+#define WM8991_RIN12VOL_MASK                    0x001F  /* RIN12VOL - [4:0] */
+#define WM8991_RIN12VOL_SHIFT			0
+
+/*
+ * R27 (0x1B) - Right Line Input 3&4 Volume
+ */
+#define WM8991_IPVU                             0x0100  /* IPVU */
+#define WM8991_RI34MUTE                         0x0080  /* RI34MUTE */
+#define WM8991_RI34MUTE_BIT			7
+#define WM8991_RI34ZC                           0x0040  /* RI34ZC */
+#define WM8991_RI34ZC_BIT			6
+#define WM8991_RIN34VOL_MASK                    0x001F  /* RIN34VOL - [4:0] */
+#define WM8991_RIN34VOL_SHIFT			0
+
+/*
+ * R28 (0x1C) - Left Output Volume
+ */
+#define WM8991_OPVU                             0x0100  /* OPVU */
+#define WM8991_LOZC                             0x0080  /* LOZC */
+#define WM8991_LOZC_BIT				7
+#define WM8991_LOUTVOL_MASK                     0x007F  /* LOUTVOL - [6:0] */
+#define WM8991_LOUTVOL_SHIFT			0
+/*
+ * R29 (0x1D) - Right Output Volume
+ */
+#define WM8991_OPVU                             0x0100  /* OPVU */
+#define WM8991_ROZC                             0x0080  /* ROZC */
+#define WM8991_ROZC_BIT				7
+#define WM8991_ROUTVOL_MASK                     0x007F  /* ROUTVOL - [6:0] */
+#define WM8991_ROUTVOL_SHIFT			0
+/*
+ * R30 (0x1E) - Line Outputs Volume
+ */
+#define WM8991_LONMUTE                          0x0040  /* LONMUTE */
+#define WM8991_LONMUTE_BIT			6
+#define WM8991_LOPMUTE                          0x0020  /* LOPMUTE */
+#define WM8991_LOPMUTE_BIT			5
+#define WM8991_LOATTN                           0x0010  /* LOATTN */
+#define WM8991_LOATTN_BIT			4
+#define WM8991_RONMUTE                          0x0004  /* RONMUTE */
+#define WM8991_RONMUTE_BIT			2
+#define WM8991_ROPMUTE                          0x0002  /* ROPMUTE */
+#define WM8991_ROPMUTE_BIT			1
+#define WM8991_ROATTN                           0x0001  /* ROATTN */
+#define WM8991_ROATTN_BIT			0
+
+/*
+ * R31 (0x1F) - Out3/4 Volume
+ */
+#define WM8991_OUT3MUTE                         0x0020  /* OUT3MUTE */
+#define WM8991_OUT3MUTE_BIT			5
+#define WM8991_OUT3ATTN                         0x0010  /* OUT3ATTN */
+#define WM8991_OUT3ATTN_BIT			4
+#define WM8991_OUT4MUTE                         0x0002  /* OUT4MUTE */
+#define WM8991_OUT4MUTE_BIT			1
+#define WM8991_OUT4ATTN                         0x0001  /* OUT4ATTN */
+#define WM8991_OUT4ATTN_BIT			0
+
+/*
+ * R32 (0x20) - Left OPGA Volume
+ */
+#define WM8991_OPVU                             0x0100  /* OPVU */
+#define WM8991_LOPGAZC                          0x0080  /* LOPGAZC */
+#define WM8991_LOPGAZC_BIT			7
+#define WM8991_LOPGAVOL_MASK                    0x007F  /* LOPGAVOL - [6:0] */
+#define WM8991_LOPGAVOL_SHIFT			0
+
+/*
+ * R33 (0x21) - Right OPGA Volume
+ */
+#define WM8991_OPVU                             0x0100  /* OPVU */
+#define WM8991_ROPGAZC                          0x0080  /* ROPGAZC */
+#define WM8991_ROPGAZC_BIT			7
+#define WM8991_ROPGAVOL_MASK                    0x007F  /* ROPGAVOL - [6:0] */
+#define WM8991_ROPGAVOL_SHIFT			0
+/*
+ * R34 (0x22) - Speaker Volume
+ */
+#define WM8991_SPKVOL_MASK                      0x0003  /* SPKVOL - [1:0] */
+#define WM8991_SPKVOL_SHIFT			0
+
+/*
+ * R35 (0x23) - ClassD1
+ */
+#define WM8991_CDMODE                           0x0100  /* CDMODE */
+#define WM8991_CDMODE_BIT			8
+
+/*
+ * R37 (0x25) - ClassD3
+ */
+#define WM8991_DCGAIN_MASK                      0x0007  /* DCGAIN - [5:3] */
+#define WM8991_DCGAIN_SHIFT			3
+#define WM8991_ACGAIN_MASK                      0x0007  /* ACGAIN - [2:0] */
+#define WM8991_ACGAIN_SHIFT			0
+/*
+ * R39 (0x27) - Input Mixer1
+ */
+#define WM8991_AINLMODE_MASK                    0x000C  /* AINLMODE - [3:2] */
+#define WM8991_AINLMODE_SHIFT			2
+#define WM8991_AINRMODE_MASK                    0x0003  /* AINRMODE - [1:0] */
+#define WM8991_AINRMODE_SHIFT			0
+
+/*
+ * R40 (0x28) - Input Mixer2
+ */
+#define WM8991_LMP4								0x0080	/* LMP4 */
+#define WM8991_LMP4_BIT                         7		/* LMP4 */
+#define WM8991_LMN3                             0x0040  /* LMN3 */
+#define WM8991_LMN3_BIT                         6       /* LMN3 */
+#define WM8991_LMP2                             0x0020  /* LMP2 */
+#define WM8991_LMP2_BIT                         5       /* LMP2 */
+#define WM8991_LMN1                             0x0010  /* LMN1 */
+#define WM8991_LMN1_BIT                         4       /* LMN1 */
+#define WM8991_RMP4                             0x0008  /* RMP4 */
+#define WM8991_RMP4_BIT                         3       /* RMP4 */
+#define WM8991_RMN3                             0x0004  /* RMN3 */
+#define WM8991_RMN3_BIT                         2       /* RMN3 */
+#define WM8991_RMP2                             0x0002  /* RMP2 */
+#define WM8991_RMP2_BIT                         1       /* RMP2 */
+#define WM8991_RMN1                             0x0001  /* RMN1 */
+#define WM8991_RMN1_BIT                         0       /* RMN1 */
+
+/*
+ * R41 (0x29) - Input Mixer3
+ */
+#define WM8991_L34MNB                           0x0100  /* L34MNB */
+#define WM8991_L34MNB_BIT			8
+#define WM8991_L34MNBST                         0x0080  /* L34MNBST */
+#define WM8991_L34MNBST_BIT			7
+#define WM8991_L12MNB                           0x0020  /* L12MNB */
+#define WM8991_L12MNB_BIT			5
+#define WM8991_L12MNBST                         0x0010  /* L12MNBST */
+#define WM8991_L12MNBST_BIT			4
+#define WM8991_LDBVOL_MASK                      0x0007  /* LDBVOL - [2:0] */
+#define WM8991_LDBVOL_SHIFT			0
+
+/*
+ * R42 (0x2A) - Input Mixer4
+ */
+#define WM8991_R34MNB                           0x0100  /* R34MNB */
+#define WM8991_R34MNB_BIT			8
+#define WM8991_R34MNBST                         0x0080  /* R34MNBST */
+#define WM8991_R34MNBST_BIT			7
+#define WM8991_R12MNB                           0x0020  /* R12MNB */
+#define WM8991_R12MNB_BIT			5
+#define WM8991_R12MNBST                         0x0010  /* R12MNBST */
+#define WM8991_R12MNBST_BIT			4
+#define WM8991_RDBVOL_MASK                      0x0007  /* RDBVOL - [2:0] */
+#define WM8991_RDBVOL_SHIFT			0
+
+/*
+ * R43 (0x2B) - Input Mixer5
+ */
+#define WM8991_LI2BVOL_MASK                     0x07  /* LI2BVOL - [8:6] */
+#define WM8991_LI2BVOL_SHIFT			6
+#define WM8991_LR4BVOL_MASK                     0x07  /* LR4BVOL - [5:3] */
+#define WM8991_LR4BVOL_SHIFT			3
+#define WM8991_LL4BVOL_MASK                     0x07  /* LL4BVOL - [2:0] */
+#define WM8991_LL4BVOL_SHIFT			0
+
+/*
+ * R44 (0x2C) - Input Mixer6
+ */
+#define WM8991_RI2BVOL_MASK                     0x07  /* RI2BVOL - [8:6] */
+#define WM8991_RI2BVOL_SHIFT			6
+#define WM8991_RL4BVOL_MASK                     0x07  /* RL4BVOL - [5:3] */
+#define WM8991_RL4BVOL_SHIFT			3
+#define WM8991_RR4BVOL_MASK                     0x07  /* RR4BVOL - [2:0] */
+#define WM8991_RR4BVOL_SHIFT			0
+
+/*
+ * R45 (0x2D) - Output Mixer1
+ */
+#define WM8991_LRBLO                            0x0080  /* LRBLO */
+#define WM8991_LRBLO_BIT			7
+#define WM8991_LLBLO                            0x0040  /* LLBLO */
+#define WM8991_LLBLO_BIT			6
+#define WM8991_LRI3LO                           0x0020  /* LRI3LO */
+#define WM8991_LRI3LO_BIT			5
+#define WM8991_LLI3LO                           0x0010  /* LLI3LO */
+#define WM8991_LLI3LO_BIT			4
+#define WM8991_LR12LO                           0x0008  /* LR12LO */
+#define WM8991_LR12LO_BIT			3
+#define WM8991_LL12LO                           0x0004  /* LL12LO */
+#define WM8991_LL12LO_BIT			2
+#define WM8991_LDLO                             0x0001  /* LDLO */
+#define WM8991_LDLO_BIT				0
+
+/*
+ * R46 (0x2E) - Output Mixer2
+ */
+#define WM8991_RLBRO                            0x0080  /* RLBRO */
+#define WM8991_RLBRO_BIT			7
+#define WM8991_RRBRO                            0x0040  /* RRBRO */
+#define WM8991_RRBRO_BIT			6
+#define WM8991_RLI3RO                           0x0020  /* RLI3RO */
+#define WM8991_RLI3RO_BIT			5
+#define WM8991_RRI3RO                           0x0010  /* RRI3RO */
+#define WM8991_RRI3RO_BIT			4
+#define WM8991_RL12RO                           0x0008  /* RL12RO */
+#define WM8991_RL12RO_BIT			3
+#define WM8991_RR12RO                           0x0004  /* RR12RO */
+#define WM8991_RR12RO_BIT			2
+#define WM8991_RDRO                             0x0001  /* RDRO */
+#define WM8991_RDRO_BIT				0
+
+/*
+ * R47 (0x2F) - Output Mixer3
+ */
+#define WM8991_LLI3LOVOL_MASK                   0x07  /* LLI3LOVOL - [8:6] */
+#define WM8991_LLI3LOVOL_SHIFT			6
+#define WM8991_LR12LOVOL_MASK                   0x07  /* LR12LOVOL - [5:3] */
+#define WM8991_LR12LOVOL_SHIFT			3
+#define WM8991_LL12LOVOL_MASK                   0x07  /* LL12LOVOL - [2:0] */
+#define WM8991_LL12LOVOL_SHIFT			0
+
+/*
+ * R48 (0x30) - Output Mixer4
+ */
+#define WM8991_RRI3ROVOL_MASK                   0x07  /* RRI3ROVOL - [8:6] */
+#define WM8991_RRI3ROVOL_SHIFT			6
+#define WM8991_RL12ROVOL_MASK                   0x07  /* RL12ROVOL - [5:3] */
+#define WM8991_RL12ROVOL_SHIFT			3
+#define WM8991_RR12ROVOL_MASK                   0x07  /* RR12ROVOL - [2:0] */
+#define WM8991_RR12ROVOL_SHIFT			0
+
+/*
+ * R49 (0x31) - Output Mixer5
+ */
+#define WM8991_LRI3LOVOL_MASK                   0x07  /* LRI3LOVOL - [8:6] */
+#define WM8991_LRI3LOVOL_SHIFT			6
+#define WM8991_LRBLOVOL_MASK                    0x07  /* LRBLOVOL - [5:3] */
+#define WM8991_LRBLOVOL_SHIFT			3
+#define WM8991_LLBLOVOL_MASK                    0x07  /* LLBLOVOL - [2:0] */
+#define WM8991_LLBLOVOL_SHIFT			0
+
+/*
+ * R50 (0x32) - Output Mixer6
+ */
+#define WM8991_RLI3ROVOL_MASK                   0x07  /* RLI3ROVOL - [8:6] */
+#define WM8991_RLI3ROVOL_SHIFT			6
+#define WM8991_RLBROVOL_MASK                    0x07  /* RLBROVOL - [5:3] */
+#define WM8991_RLBROVOL_SHIFT			3
+#define WM8991_RRBROVOL_MASK                    0x07  /* RRBROVOL - [2:0] */
+#define WM8991_RRBROVOL_SHIFT			0
+
+/*
+ * R51 (0x33) - Out3/4 Mixer
+ */
+#define WM8991_VSEL_MASK                        0x0180  /* VSEL - [8:7] */
+#define WM8991_LI4O3                            0x0020  /* LI4O3 */
+#define WM8991_LI4O3_BIT			5
+#define WM8991_LPGAO3                           0x0010  /* LPGAO3 */
+#define WM8991_LPGAO3_BIT			4
+#define WM8991_RI4O4                            0x0002  /* RI4O4 */
+#define WM8991_RI4O4_BIT			1
+#define WM8991_RPGAO4                           0x0001  /* RPGAO4 */
+#define WM8991_RPGAO4_BIT			0
+/*
+ * R52 (0x34) - Line Mixer1
+ */
+#define WM8991_LLOPGALON                        0x0040  /* LLOPGALON */
+#define WM8991_LLOPGALON_BIT			6
+#define WM8991_LROPGALON                        0x0020  /* LROPGALON */
+#define WM8991_LROPGALON_BIT			5
+#define WM8991_LOPLON                           0x0010  /* LOPLON */
+#define WM8991_LOPLON_BIT			4
+#define WM8991_LR12LOP                          0x0004  /* LR12LOP */
+#define WM8991_LR12LOP_BIT			2
+#define WM8991_LL12LOP                          0x0002  /* LL12LOP */
+#define WM8991_LL12LOP_BIT			1
+#define WM8991_LLOPGALOP                        0x0001  /* LLOPGALOP */
+#define WM8991_LLOPGALOP_BIT			0
+/*
+ * R53 (0x35) - Line Mixer2
+ */
+#define WM8991_RROPGARON                        0x0040  /* RROPGARON */
+#define WM8991_RROPGARON_BIT			6
+#define WM8991_RLOPGARON                        0x0020  /* RLOPGARON */
+#define WM8991_RLOPGARON_BIT			5
+#define WM8991_ROPRON                           0x0010  /* ROPRON */
+#define WM8991_ROPRON_BIT			4
+#define WM8991_RL12ROP                          0x0004  /* RL12ROP */
+#define WM8991_RL12ROP_BIT			2
+#define WM8991_RR12ROP                          0x0002  /* RR12ROP */
+#define WM8991_RR12ROP_BIT			1
+#define WM8991_RROPGAROP                        0x0001  /* RROPGAROP */
+#define WM8991_RROPGAROP_BIT			0
+
+/*
+ * R54 (0x36) - Speaker Mixer
+ */
+#define WM8991_LB2SPK                           0x0080  /* LB2SPK */
+#define WM8991_LB2SPK_BIT			7
+#define WM8991_RB2SPK                           0x0040  /* RB2SPK */
+#define WM8991_RB2SPK_BIT			6
+#define WM8991_LI2SPK                           0x0020  /* LI2SPK */
+#define WM8991_LI2SPK_BIT			5
+#define WM8991_RI2SPK                           0x0010  /* RI2SPK */
+#define WM8991_RI2SPK_BIT			4
+#define WM8991_LOPGASPK                         0x0008  /* LOPGASPK */
+#define WM8991_LOPGASPK_BIT			3
+#define WM8991_ROPGASPK                         0x0004  /* ROPGASPK */
+#define WM8991_ROPGASPK_BIT			2
+#define WM8991_LDSPK                            0x0002  /* LDSPK */
+#define WM8991_LDSPK_BIT			1
+#define WM8991_RDSPK                            0x0001  /* RDSPK */
+#define WM8991_RDSPK_BIT			0
+
+/*
+ * R55 (0x37) - Additional Control
+ */
+#define WM8991_VROI                             0x0001  /* VROI */
+
+/*
+ * R56 (0x38) - AntiPOP1
+ */
+#define WM8991_DIS_LLINE                        0x0020  /* DIS_LLINE */
+#define WM8991_DIS_RLINE                        0x0010  /* DIS_RLINE */
+#define WM8991_DIS_OUT3                         0x0008  /* DIS_OUT3 */
+#define WM8991_DIS_OUT4                         0x0004  /* DIS_OUT4 */
+#define WM8991_DIS_LOUT                         0x0002  /* DIS_LOUT */
+#define WM8991_DIS_ROUT                         0x0001  /* DIS_ROUT */
+
+/*
+ * R57 (0x39) - AntiPOP2
+ */
+#define WM8991_SOFTST                           0x0040  /* SOFTST */
+#define WM8991_BUFIOEN                          0x0008  /* BUFIOEN */
+#define WM8991_BUFDCOPEN                        0x0004  /* BUFDCOPEN */
+#define WM8991_POBCTRL                          0x0002  /* POBCTRL */
+#define WM8991_VMIDTOG                          0x0001  /* VMIDTOG */
+
+/*
+ * R58 (0x3A) - MICBIAS
+ */
+#define WM8991_MCDSCTH_MASK                     0x00C0  /* MCDSCTH - [7:6] */
+#define WM8991_MCDTHR_MASK                      0x0038  /* MCDTHR - [5:3] */
+#define WM8991_MCD                              0x0004  /* MCD */
+#define WM8991_MBSEL                            0x0001  /* MBSEL */
+
+/*
+ * R60 (0x3C) - PLL1
+ */
+#define WM8991_SDM                              0x0080  /* SDM */
+#define WM8991_PRESCALE                         0x0040  /* PRESCALE */
+#define WM8991_PLLN_MASK                        0x000F  /* PLLN - [3:0] */
+
+/*
+ * R61 (0x3D) - PLL2
+ */
+#define WM8991_PLLK1_MASK                       0x00FF  /* PLLK1 - [7:0] */
+
+/*
+ * R62 (0x3E) - PLL3
+ */
+#define WM8991_PLLK2_MASK                       0x00FF  /* PLLK2 - [7:0] */
+
+/*
+ * R63 (0x3F) - Internal Driver Bits
+ */
+#define WM8991_INMIXL_PWR_BIT			0
+#define WM8991_AINLMUX_PWR_BIT			1
+#define WM8991_INMIXR_PWR_BIT			2
+#define WM8991_AINRMUX_PWR_BIT			3
+
+#define WM8991_MCLK_DIV 0
+#define WM8991_DACCLK_DIV 1
+#define WM8991_ADCCLK_DIV 2
+#define WM8991_BCLK_DIV 3
+
+#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\
+					 tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+#endif /* _WM8991_H */
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 18c0d9ce7c3..379fa22c5b6 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -242,7 +242,7 @@ struct wm8993_priv {
 	int fll_src;
 };
 
-static int wm8993_volatile(unsigned int reg)
+static int wm8993_volatile(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM8993_SOFTWARE_RESET:
diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c
index 68e9b024dd4..a87adbd05ee 100644
--- a/sound/soc/codecs/wm8994-tables.c
+++ b/sound/soc/codecs/wm8994-tables.c
@@ -62,8 +62,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = {
 	{ 0x00FF, 0x00FF }, /* R58    - MICBIAS */
 	{ 0x000F, 0x000F }, /* R59    - LDO 1 */
 	{ 0x0007, 0x0007 }, /* R60    - LDO 2 */
-	{ 0x0000, 0x0000 }, /* R61 */
-	{ 0x0000, 0x0000 }, /* R62 */
+	{ 0xFFFF, 0xFFFF }, /* R61 */
+	{ 0xFFFF, 0xFFFF }, /* R62 */
 	{ 0x0000, 0x0000 }, /* R63 */
 	{ 0x0000, 0x0000 }, /* R64 */
 	{ 0x0000, 0x0000 }, /* R65 */
@@ -209,9 +209,9 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = {
 	{ 0x0000, 0x0000 }, /* R205 */
 	{ 0x0000, 0x0000 }, /* R206 */
 	{ 0x0000, 0x0000 }, /* R207 */
-	{ 0x0000, 0x0000 }, /* R208 */
-	{ 0x0000, 0x0000 }, /* R209 */
-	{ 0x0000, 0x0000 }, /* R210 */
+	{ 0xFFFF, 0xFFFF }, /* R208 */
+	{ 0xFFFF, 0xFFFF }, /* R209 */
+	{ 0xFFFF, 0xFFFF }, /* R210 */
 	{ 0x0000, 0x0000 }, /* R211 */
 	{ 0x0000, 0x0000 }, /* R212 */
 	{ 0x0000, 0x0000 }, /* R213 */
@@ -1573,7 +1573,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = {
 	{ 0x03C3, 0x03C3 }, /* R1569  - Sidetone */
 };
 
-const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
+const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
 	0x8994,     /* R0     - Software Reset */
 	0x0000,     /* R1     - Power Management (1) */
 	0x6000,     /* R2     - Power Management (2) */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 247a6a99feb..3dc64c8b6a5 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -102,14 +102,19 @@ struct wm8994_priv {
 
 	wm8958_micdet_cb jack_cb;
 	void *jack_cb_data;
-	bool jack_is_mic;
-	bool jack_is_video;
+	int micdet_irq;
 
 	int revision;
 	struct wm8994_pdata *pdata;
+
+	unsigned int aif1clk_enable:1;
+	unsigned int aif2clk_enable:1;
+
+	unsigned int aif1clk_disable:1;
+	unsigned int aif2clk_disable:1;
 };
 
-static int wm8994_readable(unsigned int reg)
+static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM8994_GPIO_1:
@@ -136,7 +141,7 @@ static int wm8994_readable(unsigned int reg)
 	return wm8994_access_masks[reg].readable != 0;
 }
 
-static int wm8994_volatile(unsigned int reg)
+static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg)
 {
 	if (reg >= WM8994_CACHE_SIZE)
 		return 1;
@@ -164,7 +169,7 @@ static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	BUG_ON(reg > WM8994_MAX_REGISTER);
 
-	if (!wm8994_volatile(reg)) {
+	if (!wm8994_volatile(codec, reg)) {
 		ret = snd_soc_cache_write(codec, reg, value);
 		if (ret != 0)
 			dev_err(codec->dev, "Cache write to %x failed: %d\n",
@@ -182,7 +187,7 @@ static unsigned int wm8994_read(struct snd_soc_codec *codec,
 
 	BUG_ON(reg > WM8994_MAX_REGISTER);
 
-	if (!wm8994_volatile(reg) && wm8994_readable(reg) &&
+	if (!wm8994_volatile(codec, reg) && wm8994_readable(codec, reg) &&
 	    reg < codec->driver->reg_cache_size) {
 		ret = snd_soc_cache_read(codec, reg, &val);
 		if (ret >= 0)
@@ -523,7 +528,7 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8994_priv *wm8994 =snd_soc_codec_get_drvdata(codec);
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 
 	ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
@@ -1004,6 +1009,117 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
 	}
 }
 
+static int late_enable_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (wm8994->aif1clk_enable) {
+			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+					    WM8994_AIF1CLK_ENA_MASK,
+					    WM8994_AIF1CLK_ENA);
+			wm8994->aif1clk_enable = 0;
+		}
+		if (wm8994->aif2clk_enable) {
+			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+					    WM8994_AIF2CLK_ENA_MASK,
+					    WM8994_AIF2CLK_ENA);
+			wm8994->aif2clk_enable = 0;
+		}
+		break;
+	}
+
+	return 0;
+}
+
+static int late_disable_ev(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMD:
+		if (wm8994->aif1clk_disable) {
+			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+					    WM8994_AIF1CLK_ENA_MASK, 0);
+			wm8994->aif1clk_disable = 0;
+		}
+		if (wm8994->aif2clk_disable) {
+			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+					    WM8994_AIF2CLK_ENA_MASK, 0);
+			wm8994->aif2clk_disable = 0;
+		}
+		break;
+	}
+
+	return 0;
+}
+
+static int aif1clk_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		wm8994->aif1clk_enable = 1;
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		wm8994->aif1clk_disable = 1;
+		break;
+	}
+
+	return 0;
+}
+
+static int aif2clk_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		wm8994->aif2clk_enable = 1;
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		wm8994->aif2clk_disable = 1;
+		break;
+	}
+
+	return 0;
+}
+
+static int adc_mux_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	late_enable_ev(w, kcontrol, event);
+	return 0;
+}
+
+static int micbias_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	late_enable_ev(w, kcontrol, event);
+	return 0;
+}
+
+static int dac_ev(struct snd_soc_dapm_widget *w,
+		  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int mask = 1 << w->shift;
+
+	snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+			    mask, mask);
+	return 0;
+}
+
 static const char *hp_mux_text[] = {
 	"Mixer",
 	"DAC",
@@ -1272,11 +1388,68 @@ static const struct soc_enum aif2dacr_src_enum =
 static const struct snd_kcontrol_new aif2dacr_src_mux =
 	SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
 
+static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev,
+	SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev,
+	SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+
+SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
+};
+
+static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0)
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
+SND_SOC_DAPM_DAC_E("DAC2L", NULL, SND_SOC_NOPM, 3, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC2R", NULL, SND_SOC_NOPM, 2, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1L", NULL, SND_SOC_NOPM, 1, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = {
+SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
+SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
+SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
+SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
+SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+		   adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+		   adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
+SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+};
+
 static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("DMIC1DAT"),
 SND_SOC_DAPM_INPUT("DMIC2DAT"),
 SND_SOC_DAPM_INPUT("Clock"),
 
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0),
+SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
+		      SND_SOC_DAPM_PRE_PMU),
+
 SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
 		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
@@ -1284,12 +1457,9 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
 
-SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
-
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1468,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1515,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1368,14 +1539,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0),
 SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0),
 SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0),
 
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
-
-SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
-SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
-SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
-SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
-
 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
 
@@ -1515,14 +1678,12 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" },
 
 	/* DAC1 inputs */
-	{ "DAC1L", NULL, "DAC1L Mixer" },
 	{ "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" },
 	{ "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
 	{ "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
 	{ "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" },
 	{ "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
-	{ "DAC1R", NULL, "DAC1R Mixer" },
 	{ "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" },
 	{ "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
 	{ "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
@@ -1531,7 +1692,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 
 	/* DAC2/AIF2 outputs  */
 	{ "AIF2ADCL", NULL, "AIF2DAC2L Mixer" },
-	{ "DAC2L", NULL, "AIF2DAC2L Mixer" },
 	{ "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" },
 	{ "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
 	{ "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
@@ -1539,13 +1699,17 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
 	{ "AIF2ADCR", NULL, "AIF2DAC2R Mixer" },
-	{ "DAC2R", NULL, "AIF2DAC2R Mixer" },
 	{ "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" },
 	{ "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
 	{ "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
 	{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
 	{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
 	{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
 
 	/* AIF3 output */
@@ -1578,6 +1742,33 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Right Headphone Mux", "DAC", "DAC1R" },
 };
 
+static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = {
+	{ "DAC1L", NULL, "Late DAC1L Enable PGA" },
+	{ "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" },
+	{ "DAC1R", NULL, "Late DAC1R Enable PGA" },
+	{ "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" },
+	{ "DAC2L", NULL, "Late DAC2L Enable PGA" },
+	{ "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" },
+	{ "DAC2R", NULL, "Late DAC2R Enable PGA" },
+	{ "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" }
+};
+
+static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = {
+	{ "DAC1L", NULL, "DAC1L Mixer" },
+	{ "DAC1R", NULL, "DAC1R Mixer" },
+	{ "DAC2L", NULL, "AIF2DAC2L Mixer" },
+	{ "DAC2R", NULL, "AIF2DAC2R Mixer" },
+};
+
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+	{ "AIF1DACDAT", NULL, "AIF2DACDAT" },
+	{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
+	{ "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+	{ "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+	{ "MICBIAS", NULL, "CLK_SYS" },
+	{ "MICBIAS", NULL, "MICBIAS Supply" },
+};
+
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
 	{ "AIF2DACL", NULL, "AIF2DAC Mux" },
 	{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -2386,7 +2577,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 	else
 		val = 0;
 
-	return snd_soc_update_bits(codec, reg, mask, reg);
+	return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 #define WM8994_RATES SNDRV_PCM_RATE_8000_96000
@@ -2501,6 +2692,22 @@ static int wm8994_resume(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
+	unsigned int val, mask;
+
+	if (wm8994->revision < 4) {
+		/* force a HW read */
+		val = wm8994_reg_read(codec->control_data,
+				      WM8994_POWER_MANAGEMENT_5);
+
+		/* modify the cache only */
+		codec->cache_only = 1;
+		mask =  WM8994_DAC1R_ENA | WM8994_DAC1L_ENA |
+			WM8994_DAC2R_ENA | WM8994_DAC2L_ENA;
+		val &= mask;
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    mask, val);
+		codec->cache_only = 0;
+	}
 
 	/* Restore the registers */
 	ret = snd_soc_cache_sync(codec);
@@ -2688,6 +2895,13 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994)
 	else
 		snd_soc_add_controls(wm8994->codec, wm8994_eq_controls,
 				     ARRAY_SIZE(wm8994_eq_controls));
+
+	for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) {
+		if (pdata->micbias[i]) {
+			snd_soc_write(codec, WM8958_MICBIAS1 + i,
+				pdata->micbias[i] & 0xffff);
+		}
+	}
 }
 
 /**
@@ -2798,47 +3012,18 @@ static void wm8958_default_micdet(u16 status, void *data)
 	int report = 0;
 
 	/* If nothing present then clear our statuses */
-	if (!(status & WM8958_MICD_STS)) {
-		wm8994->jack_is_video = false;
-		wm8994->jack_is_mic = false;
+	if (!(status & WM8958_MICD_STS))
 		goto done;
-	}
-
-	/* Assume anything over 475 ohms is a microphone and remember
-	 * that we've seen one (since buttons override it) */
-	if (status & 0x600)
-		wm8994->jack_is_mic = true;
-	if (wm8994->jack_is_mic)
-		report |= SND_JACK_MICROPHONE;
 
-	/* Video has an impedence of approximately 75 ohms; assume
-	 * this isn't used as a button and remember it since buttons
-	 * override it. */
-	if (status & 0x40)
-		wm8994->jack_is_video = true;
-	if (wm8994->jack_is_video)
-		report |= SND_JACK_VIDEOOUT;
+	report = SND_JACK_MICROPHONE;
 
 	/* Everything else is buttons; just assign slots */
-	if (status & 0x4)
+	if (status & 0x1c0)
 		report |= SND_JACK_BTN_0;
-	if (status & 0x8)
-		report |= SND_JACK_BTN_1;
-	if (status & 0x10)
-		report |= SND_JACK_BTN_2;
-	if (status & 0x20)
-		report |= SND_JACK_BTN_3;
-	if (status & 0x80)
-		report |= SND_JACK_BTN_4;
-	if (status & 0x100)
-		report |= SND_JACK_BTN_5;
 
 done:
-	snd_soc_jack_report(wm8994->micdet[0].jack,
-			    SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 |
-			    SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 |
-			    SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT,
-			    report);
+	snd_soc_jack_report(wm8994->micdet[0].jack, report,
+			    SND_JACK_BTN_0 | SND_JACK_MICROPHONE);
 }
 
 /**
@@ -2937,13 +3122,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	wm8994->pdata = dev_get_platdata(codec->dev->parent);
 	wm8994->codec = codec;
 
+	if (wm8994->pdata && wm8994->pdata->micdet_irq)
+		wm8994->micdet_irq = wm8994->pdata->micdet_irq;
+	else if (wm8994->pdata && wm8994->pdata->irq_base)
+		wm8994->micdet_irq = wm8994->pdata->irq_base +
+				     WM8994_IRQ_MIC1_DET;
+
 	pm_runtime_enable(codec->dev);
 	pm_runtime_resume(codec->dev);
 
 	/* Read our current status back from the chip - we don't want to
 	 * reset as this may interfere with the GPIO or LDO operation. */
 	for (i = 0; i < WM8994_CACHE_SIZE; i++) {
-		if (!wm8994_readable(i) || wm8994_volatile(i))
+		if (!wm8994_readable(codec, i) || wm8994_volatile(codec, i))
 			continue;
 
 		ret = wm8994_reg_read(codec->control_data, i);
@@ -2985,14 +3176,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 
 	switch (control->type) {
 	case WM8994:
-		ret = wm8994_request_irq(codec->control_data,
-					 WM8994_IRQ_MIC1_DET,
-					 wm8994_mic_irq, "Mic 1 detect",
-					 wm8994);
-		if (ret != 0)
-			dev_warn(codec->dev,
-				 "Failed to request Mic1 detect IRQ: %d\n",
-				 ret);
+		if (wm8994->micdet_irq) {
+			ret = request_threaded_irq(wm8994->micdet_irq, NULL,
+						   wm8994_mic_irq,
+						   IRQF_TRIGGER_RISING,
+						   "Mic1 detect",
+						   wm8994);
+			if (ret != 0)
+				dev_warn(codec->dev,
+					 "Failed to request Mic1 detect IRQ: %d\n",
+					 ret);
+		}
 
 		ret = wm8994_request_irq(codec->control_data,
 					 WM8994_IRQ_MIC1_SHRT,
@@ -3023,15 +3217,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		break;
 
 	case WM8958:
-		ret = wm8994_request_irq(codec->control_data,
-					 WM8994_IRQ_MIC1_DET,
-					 wm8958_mic_irq, "Mic detect",
-					 wm8994);
-		if (ret != 0)
-			dev_warn(codec->dev,
-				 "Failed to request Mic detect IRQ: %d\n",
-				 ret);
-		break;
+		if (wm8994->micdet_irq) {
+			ret = request_threaded_irq(wm8994->micdet_irq, NULL,
+						   wm8958_mic_irq,
+						   IRQF_TRIGGER_RISING,
+						   "Mic detect",
+						   wm8994);
+			if (ret != 0)
+				dev_warn(codec->dev,
+					 "Failed to request Mic detect IRQ: %d\n",
+					 ret);
+		}
 	}
 
 	/* Remember if AIFnLRCLK is configured as a GPIO.  This should be
@@ -3112,10 +3308,31 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	case WM8994:
 		snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
 					  ARRAY_SIZE(wm8994_specific_dapm_widgets));
+		if (wm8994->revision < 4) {
+			snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
+						  ARRAY_SIZE(wm8994_lateclk_revd_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
+						  ARRAY_SIZE(wm8994_adc_revd_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets,
+						  ARRAY_SIZE(wm8994_dac_revd_widgets));
+		} else {
+			snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+						  ARRAY_SIZE(wm8994_lateclk_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+						  ARRAY_SIZE(wm8994_adc_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+						  ARRAY_SIZE(wm8994_dac_widgets));
+		}
 		break;
 	case WM8958:
 		snd_soc_add_controls(codec, wm8958_snd_controls,
 				     ARRAY_SIZE(wm8958_snd_controls));
+		snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+					  ARRAY_SIZE(wm8994_lateclk_widgets));
+		snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+					  ARRAY_SIZE(wm8994_adc_widgets));
+		snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+					  ARRAY_SIZE(wm8994_dac_widgets));
 		snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
 					  ARRAY_SIZE(wm8958_dapm_widgets));
 		break;
@@ -3129,8 +3346,20 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	case WM8994:
 		snd_soc_dapm_add_routes(dapm, wm8994_intercon,
 					ARRAY_SIZE(wm8994_intercon));
+
+		if (wm8994->revision < 4) {
+			snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+						ARRAY_SIZE(wm8994_revd_intercon));
+			snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
+						ARRAY_SIZE(wm8994_lateclk_revd_intercon));
+		} else {
+			snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+						ARRAY_SIZE(wm8994_lateclk_intercon));
+		}
 		break;
 	case WM8958:
+		snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+					ARRAY_SIZE(wm8994_lateclk_intercon));
 		snd_soc_dapm_add_routes(dapm, wm8958_intercon,
 					ARRAY_SIZE(wm8958_intercon));
 		break;
@@ -3142,7 +3371,8 @@ err_irq:
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
-	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
+	if (wm8994->micdet_irq)
+		free_irq(wm8994->micdet_irq, wm8994);
 err:
 	kfree(wm8994);
 	return ret;
@@ -3159,8 +3389,8 @@ static int  wm8994_codec_remove(struct snd_soc_codec *codec)
 
 	switch (control->type) {
 	case WM8994:
-		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT,
-				wm8994);
+		if (wm8994->micdet_irq)
+			free_irq(wm8994->micdet_irq, wm8994);
 		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET,
 				wm8994);
 		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT,
@@ -3170,8 +3400,8 @@ static int  wm8994_codec_remove(struct snd_soc_codec *codec)
 		break;
 
 	case WM8958:
-		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET,
-				wm8994);
+		if (wm8994->micdet_irq)
+			free_irq(wm8994->micdet_irq, wm8994);
 		break;
 	}
 	kfree(wm8994->retune_mobile_texts);
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 0c355bfc88f..999b8851226 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -43,6 +43,6 @@ struct wm8994_access_mask {
 };
 
 extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE];
-extern const __devinitdata  u16 wm8994_reg_defaults[WM8994_CACHE_SIZE];
+extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE];
 
 #endif
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 6045cbde492..67eaaecbb42 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -19,6 +19,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
+#include <linux/regulator/consumer.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -30,6 +31,18 @@
 
 #include "wm8995.h"
 
+#define WM8995_NUM_SUPPLIES 8
+static const char *wm8995_supply_names[WM8995_NUM_SUPPLIES] = {
+	"DCVDD",
+	"DBVDD1",
+	"DBVDD2",
+	"DBVDD3",
+	"AVDD1",
+	"AVDD2",
+	"CPVDD",
+	"MICVDD"
+};
+
 static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = {
 	[0]     = 0x8995, [5]     = 0x0100, [16]    = 0x000b, [17]    = 0x000b,
 	[24]    = 0x02c0, [25]    = 0x02c0, [26]    = 0x02c0, [27]    = 0x02c0,
@@ -126,8 +139,37 @@ struct wm8995_priv {
 	int mclk[2];
 	int aifclk[2];
 	struct fll_config fll[2], fll_suspend[2];
+	struct regulator_bulk_data supplies[WM8995_NUM_SUPPLIES];
+	struct notifier_block disable_nb[WM8995_NUM_SUPPLIES];
+	struct snd_soc_codec *codec;
 };
 
+/*
+ * We can't use the same notifier block for more than one supply and
+ * there's no way I can see to get from a callback to the caller
+ * except container_of().
+ */
+#define WM8995_REGULATOR_EVENT(n) \
+static int wm8995_regulator_event_##n(struct notifier_block *nb, \
+				      unsigned long event, void *data)    \
+{ \
+	struct wm8995_priv *wm8995 = container_of(nb, struct wm8995_priv, \
+				     disable_nb[n]); \
+	if (event & REGULATOR_EVENT_DISABLE) { \
+		wm8995->codec->cache_sync = 1; \
+	} \
+	return 0; \
+}
+
+WM8995_REGULATOR_EVENT(0)
+WM8995_REGULATOR_EVENT(1)
+WM8995_REGULATOR_EVENT(2)
+WM8995_REGULATOR_EVENT(3)
+WM8995_REGULATOR_EVENT(4)
+WM8995_REGULATOR_EVENT(5)
+WM8995_REGULATOR_EVENT(6)
+WM8995_REGULATOR_EVENT(7)
+
 static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
 static const DECLARE_TLV_DB_SCALE(in1lr_pga_tlv, -1650, 150, 0);
 static const DECLARE_TLV_DB_SCALE(in1l_boost_tlv, 0, 600, 0);
@@ -909,7 +951,7 @@ static const struct snd_soc_dapm_route wm8995_intercon[] = {
 	{ "SPK2R", NULL, "SPK2R Driver" }
 };
 
-static int wm8995_volatile(unsigned int reg)
+static int wm8995_volatile(struct snd_soc_codec *codec, unsigned int reg)
 {
 	/* out of bounds registers are generally considered
 	 * volatile to support register banks that are partially
@@ -1223,7 +1265,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 	else
 		val = 0;
 
-	return snd_soc_update_bits(codec, reg, mask, reg);
+	return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 /* The size in bits of the FLL divide multiplied by 10
@@ -1483,6 +1525,11 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies),
+						    wm8995->supplies);
+			if (ret)
+				return ret;
+
 			ret = snd_soc_cache_sync(codec);
 			if (ret) {
 				dev_err(codec->dev,
@@ -1492,12 +1539,13 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec,
 
 			snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1,
 					    WM8995_BG_ENA_MASK, WM8995_BG_ENA);
-
 		}
 		break;
 	case SND_SOC_BIAS_OFF:
 		snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1,
 				    WM8995_BG_ENA_MASK, 0);
+		regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies),
+				       wm8995->supplies);
 		break;
 	}
 
@@ -1536,10 +1584,12 @@ static int wm8995_remove(struct snd_soc_codec *codec)
 static int wm8995_probe(struct snd_soc_codec *codec)
 {
 	struct wm8995_priv *wm8995;
+	int i;
 	int ret;
 
 	codec->dapm.idle_bias_off = 1;
 	wm8995 = snd_soc_codec_get_drvdata(codec);
+	wm8995->codec = codec;
 
 	ret = snd_soc_codec_set_cache_io(codec, 16, 16, wm8995->control_type);
 	if (ret < 0) {
@@ -1547,21 +1597,58 @@ static int wm8995_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
+	for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++)
+		wm8995->supplies[i].supply = wm8995_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8995->supplies),
+				 wm8995->supplies);
+	if (ret) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	wm8995->disable_nb[0].notifier_call = wm8995_regulator_event_0;
+	wm8995->disable_nb[1].notifier_call = wm8995_regulator_event_1;
+	wm8995->disable_nb[2].notifier_call = wm8995_regulator_event_2;
+	wm8995->disable_nb[3].notifier_call = wm8995_regulator_event_3;
+	wm8995->disable_nb[4].notifier_call = wm8995_regulator_event_4;
+	wm8995->disable_nb[5].notifier_call = wm8995_regulator_event_5;
+	wm8995->disable_nb[6].notifier_call = wm8995_regulator_event_6;
+	wm8995->disable_nb[7].notifier_call = wm8995_regulator_event_7;
+
+	/* This should really be moved into the regulator core */
+	for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) {
+		ret = regulator_register_notifier(wm8995->supplies[i].consumer,
+						  &wm8995->disable_nb[i]);
+		if (ret) {
+			dev_err(codec->dev,
+				"Failed to register regulator notifier: %d\n",
+				ret);
+		}
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies),
+				    wm8995->supplies);
+	if (ret) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_reg_get;
+	}
+
 	ret = snd_soc_read(codec, WM8995_SOFTWARE_RESET);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to read device ID: %d\n", ret);
-		return ret;
+		goto err_reg_enable;
 	}
 
 	if (ret != 0x8995) {
 		dev_err(codec->dev, "Invalid device ID: %#x\n", ret);
-		return -EINVAL;
+		goto err_reg_enable;
 	}
 
 	ret = snd_soc_write(codec, WM8995_SOFTWARE_RESET, 0);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
-		return ret;
+		goto err_reg_enable;
 	}
 
 	wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1596,6 +1683,12 @@ static int wm8995_probe(struct snd_soc_codec *codec)
 				ARRAY_SIZE(wm8995_intercon));
 
 	return 0;
+
+err_reg_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), wm8995->supplies);
+err_reg_get:
+	regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies);
+	return ret;
 }
 
 #define WM8995_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 43825b2102a..55cdf298202 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -15,6 +15,7 @@
 #include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
+#include <linux/device.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
@@ -166,10 +167,10 @@ struct wm9081_priv {
 	int fll_fref;
 	int fll_fout;
 	int tdm_width;
-	struct wm9081_retune_mobile_config *retune;
+	struct wm9081_pdata pdata;
 };
 
-static int wm9081_volatile_register(unsigned int reg)
+static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM9081_SOFTWARE_RESET:
@@ -388,27 +389,6 @@ SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0),
 SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0),
 };
 
-static int speaker_event(struct snd_soc_dapm_widget *w,
-			 struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = w->codec;
-	unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT);
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		reg |= WM9081_SPK_ENA;
-		break;
-
-	case SND_SOC_DAPM_PRE_PMD:
-		reg &= ~WM9081_SPK_ENA;
-		break;
-	}
-
-	snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg);
-
-	return 0;
-}
-
 struct _fll_div {
 	u16 fll_fratio;
 	u16 fll_outdiv;
@@ -746,9 +726,8 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
 
 SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0,
-		   speaker_event,
-		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
 
 SND_SOC_DAPM_OUTPUT("LINEOUT"),
 SND_SOC_DAPM_OUTPUT("SPKN"),
@@ -761,7 +740,7 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0),
 };
 
 
-static const struct snd_soc_dapm_route audio_paths[] = {
+static const struct snd_soc_dapm_route wm9081_audio_paths[] = {
 	{ "DAC", NULL, "CLK_SYS" },
 	{ "DAC", NULL, "CLK_DSP" },
 
@@ -779,8 +758,10 @@ static const struct snd_soc_dapm_route audio_paths[] = {
 	{ "Speaker PGA", NULL, "TOCLK" },
 	{ "Speaker PGA", NULL, "CLK_SYS" },
 
-	{ "SPKN", NULL, "Speaker PGA" },
-	{ "SPKP", NULL, "Speaker PGA" },
+	{ "Speaker", NULL, "Speaker PGA" },
+
+	{ "SPKN", NULL, "Speaker" },
+	{ "SPKP", NULL, "Speaker" },
 };
 
 static int wm9081_set_bias_level(struct snd_soc_codec *codec,
@@ -1081,21 +1062,22 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
 	aif4 |= wm9081->bclk / wm9081->fs;
 
 	/* Apply a ReTune Mobile configuration if it's in use */
-	if (wm9081->retune) {
-		struct wm9081_retune_mobile_config *retune = wm9081->retune;
+	if (wm9081->pdata.num_retune_configs) {
+		struct wm9081_pdata *pdata = &wm9081->pdata;
 		struct wm9081_retune_mobile_setting *s;
 		int eq1;
 
 		best = 0;
-		best_val = abs(retune->configs[0].rate - wm9081->fs);
-		for (i = 0; i < retune->num_configs; i++) {
-			cur_val = abs(retune->configs[i].rate - wm9081->fs);
+		best_val = abs(pdata->retune_configs[0].rate - wm9081->fs);
+		for (i = 0; i < pdata->num_retune_configs; i++) {
+			cur_val = abs(pdata->retune_configs[i].rate -
+				      wm9081->fs);
 			if (cur_val < best_val) {
 				best_val = cur_val;
 				best = i;
 			}
 		}
-		s = &retune->configs[best];
+		s = &pdata->retune_configs[best];
 
 		dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n",
 			s->name, s->rate);
@@ -1138,10 +1120,9 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 	return 0;
 }
 
-static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
+static int wm9081_set_sysclk(struct snd_soc_codec *codec,
 			     int clk_id, unsigned int freq, int dir)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 
 	switch (clk_id) {
@@ -1206,7 +1187,6 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
 
 static struct snd_soc_dai_ops wm9081_dai_ops = {
 	.hw_params = wm9081_hw_params,
-	.set_sysclk = wm9081_set_sysclk,
 	.set_fmt = wm9081_set_dai_fmt,
 	.digital_mute = wm9081_digital_mute,
 	.set_tdm_slot = wm9081_set_tdm_slot,
@@ -1230,7 +1210,6 @@ static struct snd_soc_dai_driver wm9081_dai = {
 static int wm9081_probe(struct snd_soc_codec *codec)
 {
 	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 	u16 reg;
 
@@ -1254,6 +1233,14 @@ static int wm9081_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
+	reg = 0;
+	if (wm9081->pdata.irq_high)
+		reg |= WM9081_IRQ_POL;
+	if (!wm9081->pdata.irq_cmos)
+		reg |= WM9081_IRQ_OP_CTRL;
+	snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL,
+			    WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg);
+
 	wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Enable zero cross by default */
@@ -1265,17 +1252,13 @@ static int wm9081_probe(struct snd_soc_codec *codec)
 
 	snd_soc_add_controls(codec, wm9081_snd_controls,
 			     ARRAY_SIZE(wm9081_snd_controls));
-	if (!wm9081->retune) {
+	if (!wm9081->pdata.num_retune_configs) {
 		dev_dbg(codec->dev,
 			"No ReTune Mobile data, using normal EQ\n");
 		snd_soc_add_controls(codec, wm9081_eq_controls,
 				     ARRAY_SIZE(wm9081_eq_controls));
 	}
 
-	snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets,
-				  ARRAY_SIZE(wm9081_dapm_widgets));
-	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
-
 	return ret;
 }
 
@@ -1319,11 +1302,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = {
 	.remove = 	wm9081_remove,
 	.suspend =	wm9081_suspend,
 	.resume =	wm9081_resume,
+
+	.set_sysclk = wm9081_set_sysclk,
 	.set_bias_level = wm9081_set_bias_level,
+
 	.reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm9081_reg_defaults,
 	.volatile_register = wm9081_volatile_register,
+
+	.dapm_widgets	  = wm9081_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets),
+	.dapm_routes     = wm9081_audio_paths,
+	.num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths),
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -1341,6 +1332,10 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c,
 	wm9081->control_type = SND_SOC_I2C;
 	wm9081->control_data = i2c;
 
+	if (dev_get_platdata(&i2c->dev))
+		memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev),
+		       sizeof(wm9081->pdata));
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm9081, &wm9081_dai, 1);
 	if (ret < 0)
@@ -1363,7 +1358,7 @@ MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id);
 
 static struct i2c_driver wm9081_i2c_driver = {
 	.driver = {
-		.name = "wm9081-codec",
+		.name = "wm9081",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm9081_i2c_probe,
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index a788c429704..4de12203e61 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -144,7 +144,7 @@ struct wm9090_priv {
 	void *control_data;
 };
 
-static int wm9090_volatile(unsigned int reg)
+static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg)
 {
 	switch (reg) {
 	case WM9090_SOFTWARE_RESET:
@@ -518,7 +518,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec,
 			for (i = 1; i < codec->driver->reg_cache_size; i++) {
 				if (reg_cache[i] == wm9090_reg_defaults[i])
 					continue;
-				if (wm9090_volatile(i))
+				if (wm9090_volatile(codec, i))
 					continue;
 
 				ret = snd_soc_write(codec, i, reg_cache[i]);
@@ -551,7 +551,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec,
 static int wm9090_probe(struct snd_soc_codec *codec)
 {
 	struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec);
-	u16 *reg_cache = codec->reg_cache;
 	int ret;
 
 	codec->control_data = wm9090->control_data;
@@ -576,22 +575,30 @@ static int wm9090_probe(struct snd_soc_codec *codec)
 	/* Configure some defaults; they will be written out when we
 	 * bring the bias up.
 	 */
-	reg_cache[WM9090_IN1_LINE_INPUT_A_VOLUME] |= WM9090_IN1_VU
-		| WM9090_IN1A_ZC;
-	reg_cache[WM9090_IN1_LINE_INPUT_B_VOLUME] |= WM9090_IN1_VU
-		| WM9090_IN1B_ZC;
-	reg_cache[WM9090_IN2_LINE_INPUT_A_VOLUME] |= WM9090_IN2_VU
-		| WM9090_IN2A_ZC;
-	reg_cache[WM9090_IN2_LINE_INPUT_B_VOLUME] |= WM9090_IN2_VU
-		| WM9090_IN2B_ZC;
-	reg_cache[WM9090_SPEAKER_VOLUME_LEFT] |=
-		WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC;
-	reg_cache[WM9090_LEFT_OUTPUT_VOLUME] |=
-		WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC;
-	reg_cache[WM9090_RIGHT_OUTPUT_VOLUME] |=
-		WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC;
-
-	reg_cache[WM9090_CLOCKING_1] |= WM9090_TOCLK_ENA;
+	snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_A_VOLUME,
+			    WM9090_IN1_VU | WM9090_IN1A_ZC,
+			    WM9090_IN1_VU | WM9090_IN1A_ZC);
+	snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_B_VOLUME,
+			    WM9090_IN1_VU | WM9090_IN1B_ZC,
+			    WM9090_IN1_VU | WM9090_IN1B_ZC);
+	snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_A_VOLUME,
+			    WM9090_IN2_VU | WM9090_IN2A_ZC,
+			    WM9090_IN2_VU | WM9090_IN2A_ZC);
+	snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_B_VOLUME,
+			    WM9090_IN2_VU | WM9090_IN2B_ZC,
+			    WM9090_IN2_VU | WM9090_IN2B_ZC);
+	snd_soc_update_bits(codec, WM9090_SPEAKER_VOLUME_LEFT,
+			    WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC,
+			    WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC);
+	snd_soc_update_bits(codec, WM9090_LEFT_OUTPUT_VOLUME,
+			    WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC,
+			    WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC);
+	snd_soc_update_bits(codec, WM9090_RIGHT_OUTPUT_VOLUME,
+			    WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC,
+			    WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC);
+
+	snd_soc_update_bits(codec, WM9090_CLOCKING_1,
+			    WM9090_TOCLK_ENA, WM9090_TOCLK_ENA);
 
 	wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index c466982eed2..7b6b3c18e29 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -82,7 +82,8 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
 	} while (reg & op && count < 400);
 
 	if (reg & op)
-		dev_err(codec->dev, "Timed out waiting for DC Servo\n");
+		dev_err(codec->dev, "Timed out waiting for DC Servo %x\n",
+			op);
 }
 
 /*
@@ -91,6 +92,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+	s8 offset;
 	u16 reg, reg_l, reg_r, dcs_cfg;
 
 	/* If we're using a digital only path and have a previously
@@ -149,16 +151,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 			hubs->dcs_codes);
 
 		/* HPOUT1L */
-		if (reg_l + hubs->dcs_codes > 0 &&
-		    reg_l + hubs->dcs_codes < 0xff)
-			reg_l += hubs->dcs_codes;
-		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		offset = reg_l;
+		offset += hubs->dcs_codes;
+		dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
 		/* HPOUT1R */
-		if (reg_r + hubs->dcs_codes > 0 &&
-		    reg_r + hubs->dcs_codes < 0xff)
-			reg_r += hubs->dcs_codes;
-		dcs_cfg |= reg_r;
+		offset = reg_r;
+		offset += hubs->dcs_codes;
+		dcs_cfg |= (u8)offset;
 
 		dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
 
@@ -675,6 +675,9 @@ SND_SOC_DAPM_OUTPUT("LINEOUT2N"),
 };
 
 static const struct snd_soc_dapm_route analogue_routes[] = {
+	{ "MICBIAS1", NULL, "CLK_SYS" },
+	{ "MICBIAS2", NULL, "CLK_SYS" },
+
 	{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
 	{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
 
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6bacc68..fe7984221eb 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,12 +218,24 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
 		.ops = &evm_spdif_ops,
 	},
 };
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai_name = "davinci-mcasp.1",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.codec_name = "tlv320aic3x-codec.1-0018",
+	.platform_name = "davinci-pcm-audio",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
 	.cpu_dai_name= "davinci-mcasp.0",
 	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.0-001a",
+	.codec_name = "tlv320aic3x-codec.1-0018",
 	.platform_name = "davinci-pcm-audio",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
 
 static struct snd_soc_card da830_snd_soc_card = {
 	.name = "DA830/OMAP-L137 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da830_evm_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
 	.name = "DA850/OMAP-L138 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da850_evm_dai,
 	.num_links = 1,
 };
 
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 9e0e565e6ed..d0d60b8a54d 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -658,7 +658,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1,
+	ioarea = request_mem_region(mem->start, resource_size(mem),
 				    pdev->name);
 	if (!ioarea) {
 		dev_err(&pdev->dev, "McBSP region already claimed\n");
@@ -694,20 +694,25 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 	}
 	clk_enable(dev->clk);
 
-	dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+	dev->base = ioremap(mem->start, resource_size(mem));
+	if (!dev->base) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_release_clk;
+	}
 
 	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
-	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
+	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
 
 	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
-	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
+	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
 
 	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENXIO;
-		goto err_free_mem;
+		goto err_iounmap;
 	}
 	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
 
@@ -715,7 +720,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENXIO;
-		goto err_free_mem;
+		goto err_iounmap;
 	}
 	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 	dev->dev = &pdev->dev;
@@ -724,14 +729,19 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 
 	ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai);
 	if (ret != 0)
-		goto err_free_mem;
+		goto err_iounmap;
 
 	return 0;
 
+err_iounmap:
+	iounmap(dev->base);
+err_release_clk:
+	clk_disable(dev->clk);
+	clk_put(dev->clk);
 err_free_mem:
 	kfree(dev);
 err_release_region:
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 
 	return ret;
 }
@@ -747,7 +757,7 @@ static int davinci_i2s_remove(struct platform_device *pdev)
 	dev->clk = NULL;
 	kfree(dev);
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 
 	return 0;
 }
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index fb55d2c5d70..a5af834c8ef 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -868,7 +868,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	}
 
 	ioarea = request_mem_region(mem->start,
-			(mem->end - mem->start) + 1, pdev->name);
+			resource_size(mem), pdev->name);
 	if (!ioarea) {
 		dev_err(&pdev->dev, "Audio region already claimed\n");
 		ret = -EBUSY;
@@ -885,7 +885,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	clk_enable(dev->clk);
 	dev->clk_active = 1;
 
-	dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+	dev->base = ioremap(mem->start, resource_size(mem));
+	if (!dev->base) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_release_clk;
+	}
+
 	dev->op_mode = pdata->op_mode;
 	dev->tdm_slots = pdata->tdm_slots;
 	dev->num_serializer = pdata->num_serializer;
@@ -899,14 +905,14 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
-							io_v2p(dev->base));
+							mem->start);
 
 	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENODEV;
-		goto err_release_region;
+		goto err_iounmap;
 	}
 
 	dma_data->channel = res->start;
@@ -915,13 +921,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
-							io_v2p(dev->base));
+							mem->start);
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENODEV;
-		goto err_release_region;
+		goto err_iounmap;
 	}
 
 	dma_data->channel = res->start;
@@ -929,11 +935,16 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]);
 
 	if (ret != 0)
-		goto err_release_region;
+		goto err_iounmap;
 	return 0;
 
+err_iounmap:
+	iounmap(dev->base);
+err_release_clk:
+	clk_disable(dev->clk);
+	clk_put(dev->clk);
 err_release_region:
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 err_release_data:
 	kfree(dev);
 
@@ -951,7 +962,7 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
 	dev->clk = NULL;
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 
 	kfree(dev);
 
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 9d2afccc3a2..13e05a302a9 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = {
 
 static int davinci_vcif_probe(struct platform_device *pdev)
 {
-	struct davinci_vc *davinci_vc = platform_get_drvdata(pdev);
+	struct davinci_vc *davinci_vc = mfd_get_data(pdev);
 	struct davinci_vcif_dev *davinci_vcif_dev;
 	int ret;
 
diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig
index 57429041189..91a28de9410 100644
--- a/sound/soc/ep93xx/Kconfig
+++ b/sound/soc/ep93xx/Kconfig
@@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE
 	help
 	  Say Y or M here if you want to add support for AC97 audio on the
 	  Simplemachines Sim.One board.
+
+config SND_EP93XX_SOC_EDB93XX
+	tristate "SoC Audio support for Cirrus Logic EDB93xx boards"
+	depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A)
+	select SND_EP93XX_SOC_I2S
+	select SND_SOC_CS4271
+	help
+	  Say Y or M here if you want to add support for I2S audio on the
+	  Cirrus Logic EDB93xx boards.
diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile
index 8e7977fb6b7..5514146cbdf 100644
--- a/sound/soc/ep93xx/Makefile
+++ b/sound/soc/ep93xx/Makefile
@@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97)		+= snd-soc-ep93xx-ac97.o
 # EP93XX Machine Support
 snd-soc-snappercl15-objs			:= snappercl15.o
 snd-soc-simone-objs				:= simone.o
+snd-soc-edb93xx-objs				:= edb93xx.o
 
 obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15)	+= snd-soc-snappercl15.o
 obj-$(CONFIG_SND_EP93XX_SOC_SIMONE)		+= snd-soc-simone.o
+obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX)		+= snd-soc-edb93xx.o
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c
new file mode 100644
index 00000000000..d3aa15119d2
--- /dev/null
+++ b/sound/soc/ep93xx/edb93xx.c
@@ -0,0 +1,142 @@
+/*
+ * SoC audio for EDB93xx
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * This driver support CS4271 codec being master or slave, working
+ * in control port mode, connected either via SPI or I2C.
+ * The data format accepted is I2S or left-justified.
+ * DAPM support not implemented.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include "ep93xx-pcm.h"
+
+#define edb93xx_has_audio() (machine_is_edb9301() ||	\
+			     machine_is_edb9302() ||	\
+			     machine_is_edb9302a() ||	\
+			     machine_is_edb9307a() ||	\
+			     machine_is_edb9315a())
+
+static int edb93xx_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 err;
+	unsigned int mclk_rate;
+	unsigned int rate = params_rate(params);
+
+	/*
+	 * According to CS4271 datasheet we use MCLK/LRCK=256 for
+	 * rates below 50kHz and 128 for higher sample rates
+	 */
+	if (rate < 50000)
+		mclk_rate = rate * 64 * 4;
+	else
+		mclk_rate = rate * 64 * 2;
+
+	err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (err)
+		return err;
+
+	err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (err)
+		return err;
+
+	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate,
+				     SND_SOC_CLOCK_IN);
+	if (err)
+		return err;
+
+	return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate,
+				      SND_SOC_CLOCK_OUT);
+}
+
+static struct snd_soc_ops edb93xx_ops = {
+	.hw_params	= edb93xx_hw_params,
+};
+
+static struct snd_soc_dai_link edb93xx_dai = {
+	.name		= "CS4271",
+	.stream_name	= "CS4271 HiFi",
+	.platform_name	= "ep93xx-pcm-audio",
+	.cpu_dai_name	= "ep93xx-i2s",
+	.codec_name	= "spi0.0",
+	.codec_dai_name	= "cs4271-hifi",
+	.ops		= &edb93xx_ops,
+};
+
+static struct snd_soc_card snd_soc_edb93xx = {
+	.name		= "EDB93XX",
+	.dai_link	= &edb93xx_dai,
+	.num_links	= 1,
+};
+
+static struct platform_device *edb93xx_snd_device;
+
+static int __init edb93xx_init(void)
+{
+	int ret;
+
+	if (!edb93xx_has_audio())
+		return -ENODEV;
+
+	ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
+				 EP93XX_SYSCON_I2SCLKDIV_ORIDE |
+				 EP93XX_SYSCON_I2SCLKDIV_SPOL);
+	if (ret)
+		return ret;
+
+	edb93xx_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!edb93xx_snd_device) {
+		ret = -ENOMEM;
+		goto free_i2s;
+	}
+
+	platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx);
+	ret = platform_device_add(edb93xx_snd_device);
+	if (ret)
+		goto device_put;
+
+	return 0;
+
+device_put:
+	platform_device_put(edb93xx_snd_device);
+free_i2s:
+	ep93xx_i2s_release();
+	return ret;
+}
+module_init(edb93xx_init);
+
+static void __exit edb93xx_exit(void)
+{
+	platform_device_unregister(edb93xx_snd_device);
+	ep93xx_i2s_release();
+}
+module_exit(edb93xx_exit);
+
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("ALSA SoC EDB93xx");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 68a0bae1208..104e95cda0a 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -253,7 +253,6 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream,
 	struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai);
 	unsigned v = 0;
 
-
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index fff579a1c13..042f4e93746 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 	unsigned word_len, div, sdiv, lrdiv;
-	int found = 0, err;
+	int err;
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
@@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	 * the codec uses.
 	 */
 	div = clk_get_rate(info->mclk) / params_rate(params);
-	for (sdiv = 2; sdiv <= 4; sdiv += 2)
-		for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1)
-			if (sdiv * lrdiv == div) {
-				found = 1;
-				goto out;
-			}
-out:
-	if (!found)
-		return -EINVAL;
+	sdiv = 4;
+	if (div > (256 + 512) / 2) {
+		lrdiv = 128;
+	} else {
+		lrdiv = 64;
+		if (div < (128 + 256) / 2)
+			sdiv = 2;
+	}
 
 	err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv);
 	if (err)
@@ -314,10 +313,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai)
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
 	if (!dai->active)
-		return;
+		return 0;
 
 	ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK);
 	ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+	return 0;
 }
 
 static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
@@ -325,10 +326,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
 	if (!dai->active)
-		return;
+		return 0;
 
 	ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK);
 	ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+	return 0;
 }
 #else
 #define ep93xx_i2s_suspend	NULL
@@ -352,13 +355,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = {
 	.playback	= {
 		.channels_min	= 2,
 		.channels_max	= 2,
-		.rates		= SNDRV_PCM_RATE_8000_96000,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
 		.formats	= EP93XX_I2S_FORMATS,
 	},
 	.capture	= {
 		 .channels_min	= 2,
 		 .channels_max	= 2,
-		 .rates		= SNDRV_PCM_RATE_8000_96000,
+		 .rates		= SNDRV_PCM_RATE_8000_192000,
 		 .formats	= EP93XX_I2S_FORMATS,
 	},
 	.ops		= &ep93xx_i2s_dai_ops,
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 06670776f64..a456e491155 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
 				   SNDRV_PCM_INFO_INTERLEAVED	|
 				   SNDRV_PCM_INFO_BLOCK_TRANSFER),
 				   
-	.rates			= SNDRV_PCM_RATE_8000_96000,
+	.rates			= SNDRV_PCM_RATE_8000_192000,
 	.rate_min		= SNDRV_PCM_RATE_8000,
-	.rate_max		= SNDRV_PCM_RATE_96000,
+	.rate_max		= SNDRV_PCM_RATE_192000,
 	
 	.formats		= (SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE |
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 4cf98c03af2..15dac0f20cd 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -896,8 +896,7 @@ static struct snd_pcm_ops fsl_dma_ops = {
 	.pointer	= fsl_dma_pointer,
 };
 
-static int __devinit fsl_soc_dma_probe(struct platform_device *pdev,
-				       const struct of_device_id *match)
+static int __devinit fsl_soc_dma_probe(struct platform_device *pdev)
  {
 	struct dma_object *dma;
 	struct device_node *np = pdev->dev.of_node;
@@ -979,7 +978,7 @@ static const struct of_device_id fsl_soc_dma_ids[] = {
 };
 MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids);
 
-static struct of_platform_driver fsl_soc_dma_driver = {
+static struct platform_driver fsl_soc_dma_driver = {
 	.driver = {
 		.name = "fsl-pcm-audio",
 		.owner = THIS_MODULE,
@@ -993,12 +992,12 @@ static int __init fsl_soc_dma_init(void)
 {
 	pr_info("Freescale Elo DMA ASoC PCM Driver\n");
 
-	return of_register_platform_driver(&fsl_soc_dma_driver);
+	return platform_driver_register(&fsl_soc_dma_driver);
 }
 
 static void __exit fsl_soc_dma_exit(void)
 {
-	of_unregister_platform_driver(&fsl_soc_dma_driver);
+	platform_driver_unregister(&fsl_soc_dma_driver);
 }
 
 module_init(fsl_soc_dma_init);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 4cc167a7aeb..313e0ccedd5 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -624,8 +624,7 @@ static void make_lowercase(char *s)
 	}
 }
 
-static int __devinit fsl_ssi_probe(struct platform_device *pdev,
-				   const struct of_device_id *match)
+static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 {
 	struct fsl_ssi_private *ssi_private;
 	int ret = 0;
@@ -774,7 +773,7 @@ static const struct of_device_id fsl_ssi_ids[] = {
 };
 MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
 
-static struct of_platform_driver fsl_ssi_driver = {
+static struct platform_driver fsl_ssi_driver = {
 	.driver = {
 		.name = "fsl-ssi-dai",
 		.owner = THIS_MODULE,
@@ -788,12 +787,12 @@ static int __init fsl_ssi_init(void)
 {
 	printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n");
 
-	return of_register_platform_driver(&fsl_ssi_driver);
+	return platform_driver_register(&fsl_ssi_driver);
 }
 
 static void __exit fsl_ssi_exit(void)
 {
-	of_unregister_platform_driver(&fsl_ssi_driver);
+	platform_driver_unregister(&fsl_ssi_driver);
 }
 
 module_init(fsl_ssi_init);
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index f92dca07cd3..fff695ccdd3 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -368,8 +368,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
 	.pcm_free	= &psc_dma_free,
 };
 
-static int mpc5200_hpcd_probe(struct of_device *op,
-		const struct of_device_id *match)
+static int mpc5200_hpcd_probe(struct of_device *op)
 {
 	phys_addr_t fifo;
 	struct psc_dma *psc_dma;
@@ -511,32 +510,31 @@ static int mpc5200_hpcd_remove(struct of_device *op)
 }
 
 static struct of_device_id mpc5200_hpcd_match[] = {
-	{
-		.compatible = "fsl,mpc5200-pcm",
-	},
+	{ .compatible = "fsl,mpc5200-pcm", },
 	{}
 };
 MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match);
 
-static struct of_platform_driver mpc5200_hpcd_of_driver = {
-	.owner		= THIS_MODULE,
-	.name		= "mpc5200-pcm-audio",
-	.match_table    = mpc5200_hpcd_match,
+static struct platform_driver mpc5200_hpcd_of_driver = {
 	.probe		= mpc5200_hpcd_probe,
 	.remove		= mpc5200_hpcd_remove,
+	.dev = {
+		.owner		= THIS_MODULE,
+		.name		= "mpc5200-pcm-audio",
+		.of_match_table    = mpc5200_hpcd_match,
+	}
 };
 
 static int __init mpc5200_hpcd_init(void)
 {
-	return of_register_platform_driver(&mpc5200_hpcd_of_driver);
+	return platform_driver_register(&mpc5200_hpcd_of_driver);
 }
+module_init(mpc5200_hpcd_init);
 
 static void __exit mpc5200_hpcd_exit(void)
 {
-	of_unregister_platform_driver(&mpc5200_hpcd_of_driver);
+	platform_driver_unregister(&mpc5200_hpcd_of_driver);
 }
-
-module_init(mpc5200_hpcd_init);
 module_exit(mpc5200_hpcd_exit);
 
 MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 40acc8e2b1c..ad36b095bb7 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -272,8 +272,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = {
  * - Probe/remove operations
  * - OF device match table
  */
-static int __devinit psc_ac97_of_probe(struct platform_device *op,
-				      const struct of_device_id *match)
+static int __devinit psc_ac97_of_probe(struct platform_device *op)
 {
 	int rc;
 	struct snd_ac97 ac97;
@@ -316,7 +315,7 @@ static struct of_device_id psc_ac97_match[] __devinitdata = {
 };
 MODULE_DEVICE_TABLE(of, psc_ac97_match);
 
-static struct of_platform_driver psc_ac97_driver = {
+static struct platform_driver psc_ac97_driver = {
 	.probe = psc_ac97_of_probe,
 	.remove = __devexit_p(psc_ac97_of_remove),
 	.driver = {
@@ -332,13 +331,13 @@ static struct of_platform_driver psc_ac97_driver = {
  */
 static int __init psc_ac97_init(void)
 {
-	return of_register_platform_driver(&psc_ac97_driver);
+	return platform_driver_register(&psc_ac97_driver);
 }
 module_init(psc_ac97_init);
 
 static void __exit psc_ac97_exit(void)
 {
-	of_unregister_platform_driver(&psc_ac97_driver);
+	platform_driver_unregister(&psc_ac97_driver);
 }
 module_exit(psc_ac97_exit);
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9018fa5bf0d..87cf2a5c2b2 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -150,8 +150,7 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{
  * - Probe/remove operations
  * - OF device match table
  */
-static int __devinit psc_i2s_of_probe(struct platform_device *op,
-				      const struct of_device_id *match)
+static int __devinit psc_i2s_of_probe(struct platform_device *op)
 {
 	int rc;
 	struct psc_dma *psc_dma;
@@ -213,7 +212,7 @@ static struct of_device_id psc_i2s_match[] __devinitdata = {
 };
 MODULE_DEVICE_TABLE(of, psc_i2s_match);
 
-static struct of_platform_driver psc_i2s_driver = {
+static struct platform_driver psc_i2s_driver = {
 	.probe = psc_i2s_of_probe,
 	.remove = __devexit_p(psc_i2s_of_remove),
 	.driver = {
@@ -229,13 +228,13 @@ static struct of_platform_driver psc_i2s_driver = {
  */
 static int __init psc_i2s_init(void)
 {
-	return of_register_platform_driver(&psc_i2s_driver);
+	return platform_driver_register(&psc_i2s_driver);
 }
 module_init(psc_i2s_init);
 
 static void __exit psc_i2s_exit(void)
 {
-	of_unregister_platform_driver(&psc_i2s_driver);
+	platform_driver_unregister(&psc_i2s_driver);
 }
 module_exit(psc_i2s_exit);
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 7d7847a1e66..c16c6b2eff9 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -53,9 +53,8 @@ struct mpc8610_hpcd_data {
  *
  * Here we program the DMACR and PMUXCR registers.
  */
-static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device)
+static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = platform_get_drvdata(sound_device);
 	struct mpc8610_hpcd_data *machine_data =
 		container_of(card, struct mpc8610_hpcd_data, card);
 	struct ccsr_guts_86xx __iomem *guts;
@@ -138,9 +137,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
  * This function is called to remove the sound device for one SSI.  We
  * de-program the DMACR and PMUXCR register.
  */
-static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device)
+static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = platform_get_drvdata(sound_device);
 	struct mpc8610_hpcd_data *machine_data =
 		container_of(card, struct mpc8610_hpcd_data, card);
 	struct ccsr_guts_86xx __iomem *guts;
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 026b756961e..66e0b68af14 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -85,9 +85,8 @@ struct machine_data {
  *
  * Here we program the DMACR and PMUXCR registers.
  */
-static int p1022_ds_machine_probe(struct platform_device *sound_device)
+static int p1022_ds_machine_probe(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = platform_get_drvdata(sound_device);
 	struct machine_data *mdata =
 		container_of(card, struct machine_data, card);
 	struct ccsr_guts_85xx __iomem *guts;
@@ -160,9 +159,8 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream)
  * This function is called to remove the sound device for one SSI.  We
  * de-program the DMACR and PMUXCR register.
  */
-static int p1022_ds_machine_remove(struct platform_device *sound_device)
+static int p1022_ds_machine_remove(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = platform_get_drvdata(sound_device);
 	struct machine_data *mdata =
 		container_of(card, struct machine_data, card);
 	struct ccsr_guts_85xx __iomem *guts;
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index 642270a635e..d8f130d39dd 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -30,6 +30,16 @@ config SND_MXC_SOC_WM1133_EV1
 	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
 	  PMIC board with WM8835x fitted.
 
+config SND_SOC_MX27VIS_AIC32X4
+	tristate "SoC audio support for Visstrim M10 boards"
+	depends on MACH_IMX27_VISSTRIM_M10
+	select SND_SOC_TVL320AIC32X4
+	select SND_MXC_SOC_SSI
+	select SND_MXC_SOC_MX2
+	help
+	  Say Y if you want to add support for SoC audio on Visstrim SM10
+	  board with TLV320AIC32X4 codec.
+
 config SND_SOC_PHYCORE_AC97
 	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
 	depends on MACH_PCM043 || MACH_PCA100
@@ -44,7 +54,8 @@ config SND_SOC_EUKREA_TLV320
 	tristate "Eukrea TLV320"
 	depends on MACH_EUKREA_MBIMX27_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
-		|| MACH_EUKREA_MBIMXSD35_BASEBOARD
+		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
+		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
 	select SND_SOC_TLV320AIC23
 	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
index b67fc02a4ec..d6d609ba7e2 100644
--- a/sound/soc/imx/Makefile
+++ b/sound/soc/imx/Makefile
@@ -10,8 +10,10 @@ obj-$(CONFIG_SND_MXC_SOC_MX2) += snd-soc-imx-mx2.o
 # i.MX Machine Support
 snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
 snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index e20c9e1457c..75fb4b83548 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -79,7 +79,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
 	.codec_dai_name	= "tlv320aic23-hifi",
-	.platform_name	= "imx-pcm-audio.0",
+	.platform_name	= "imx-fiq-pcm-audio.0",
 	.codec_name	= "tlv320aic23-codec.0-001a",
 	.cpu_dai_name	= "imx-ssi.0",
 	.ops		= &eukrea_tlv320_snd_ops,
@@ -98,7 +98,8 @@ static int __init eukrea_tlv320_init(void)
 	int ret;
 
 	if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()
-		&& !machine_is_eukrea_cpuimx35sd())
+		&& !machine_is_eukrea_cpuimx35sd()
+		&& !machine_is_eukrea_cpuimx51sd())
 		/* return happy. We might run on a totally different machine */
 		return 0;
 
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 30894ea7f33..bc92ec62000 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -108,7 +108,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		/* data on rising edge of bclk, frame high with data */
-		strcr |= SSI_STCR_TFSL;
+		strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* data on rising edge of bclk, frame high 1clk before data */
@@ -656,6 +656,9 @@ static int imx_ssi_probe(struct platform_device *pdev)
 	ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0;
 	ssi->dma_params_tx.dma_addr = res->start + SSI_STX0;
 
+	ssi->dma_params_tx.burstsize = 4;
+	ssi->dma_params_rx.burstsize = 4;
+
 	res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0");
 	if (res)
 		ssi->dma_params_tx.dma = res->start;
diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c
new file mode 100644
index 00000000000..054110b91d4
--- /dev/null
+++ b/sound/soc/imx/mx27vis-aic32x4.c
@@ -0,0 +1,137 @@
+/*
+ * mx27vis-aic32x4.c
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+#include <mach/audmux.h>
+
+#include "../codecs/tlv320aic32x4.h"
+#include "imx-ssi.h"
+
+static int mx27vis_aic32x4_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;
+	u32 dai_format;
+
+	dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM;
+
+	/* set codec DAI configuration */
+	snd_soc_dai_set_fmt(codec_dai, dai_format);
+
+	/* set cpu DAI configuration */
+	snd_soc_dai_set_fmt(cpu_dai, dai_format);
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+				     25000000, SND_SOC_CLOCK_OUT);
+	if (ret) {
+		pr_err("%s: failed setting codec sysclk\n", __func__);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret) {
+		pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops mx27vis_aic32x4_snd_ops = {
+	.hw_params	= mx27vis_aic32x4_hw_params,
+};
+
+static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
+	.name		= "tlv320aic32x4",
+	.stream_name	= "TLV320AIC32X4",
+	.codec_dai_name	= "tlv320aic32x4-hifi",
+	.platform_name	= "imx-pcm-audio.0",
+	.codec_name	= "tlv320aic32x4.0-0018",
+	.cpu_dai_name	= "imx-ssi.0",
+	.ops		= &mx27vis_aic32x4_snd_ops,
+};
+
+static struct snd_soc_card mx27vis_aic32x4 = {
+	.name		= "visstrim_m10-audio",
+	.dai_link	= &mx27vis_aic32x4_dai,
+	.num_links	= 1,
+};
+
+static struct platform_device *mx27vis_aic32x4_snd_device;
+
+static int __init mx27vis_aic32x4_init(void)
+{
+	int ret;
+
+	mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!mx27vis_aic32x4_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4);
+	ret = platform_device_add(mx27vis_aic32x4_snd_device);
+
+	if (ret) {
+		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+		platform_device_put(mx27vis_aic32x4_snd_device);
+	}
+
+	/* Connect SSI0 as clock slave to SSI1 external pins */
+	mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+			MXC_AUDMUX_V1_PCR_SYN |
+			MXC_AUDMUX_V1_PCR_TFSDIR |
+			MXC_AUDMUX_V1_PCR_TCLKDIR |
+			MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) |
+			MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1)
+	);
+	mxc_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1,
+			MXC_AUDMUX_V1_PCR_SYN |
+			MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+	);
+
+	return ret;
+}
+
+static void __exit mx27vis_aic32x4_exit(void)
+{
+	platform_device_unregister(mx27vis_aic32x4_snd_device);
+}
+
+module_init(mx27vis_aic32x4_init);
+module_exit(mx27vis_aic32x4_exit);
+
+MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
+MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig
new file mode 100644
index 00000000000..29350428f1c
--- /dev/null
+++ b/sound/soc/mid-x86/Kconfig
@@ -0,0 +1,14 @@
+config SND_MFLD_MACHINE
+	tristate "SOC Machine Audio driver for Intel Medfield MID platform"
+	depends on INTEL_SCU_IPC
+	depends on SND_INTEL_SST
+	select SND_SOC_SN95031
+	select SND_SST_PLATFORM
+	help
+          This adds support for ASoC machine driver for Intel(R) MID Medfield platform
+          used as alsa device in audio substem in Intel(R) MID devices
+          Say Y if you have such a device
+          If unsure select "N".
+
+config SND_SST_PLATFORM
+	tristate
diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/mid-x86/Makefile
new file mode 100644
index 00000000000..63988333946
--- /dev/null
+++ b/sound/soc/mid-x86/Makefile
@@ -0,0 +1,5 @@
+snd-soc-sst-platform-objs := sst_platform.o
+snd-soc-mfld-machine-objs := mfld_machine.o
+
+obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o
+obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
new file mode 100644
index 00000000000..429aa1be2cf
--- /dev/null
+++ b/sound/soc/mid-x86/mfld_machine.c
@@ -0,0 +1,452 @@
+/*
+ *  mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/sn95031.h"
+
+#define MID_MONO 1
+#define MID_STEREO 2
+#define MID_MAX_CAP 5
+#define MFLD_JACK_INSERT 0x04
+
+enum soc_mic_bias_zones {
+	MFLD_MV_START = 0,
+	/* mic bias volutage range for Headphones*/
+	MFLD_MV_HP = 400,
+	/* mic bias volutage range for American Headset*/
+	MFLD_MV_AM_HS = 650,
+	/* mic bias volutage range for Headset*/
+	MFLD_MV_HS = 2000,
+	MFLD_MV_UNDEFINED,
+};
+
+static unsigned int	hs_switch;
+static unsigned int	lo_dac;
+
+struct mfld_mc_private {
+	struct platform_device *socdev;
+	void __iomem *int_base;
+	struct snd_soc_codec *codec;
+	u8 interrupt_status;
+};
+
+struct snd_soc_jack mfld_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mfld_jack_pins[] = {
+	{
+		.pin = "Headphones",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin = "AMIC1",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+/* jack detection voltage zones */
+static struct snd_soc_jack_zone mfld_zones[] = {
+	{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
+	{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
+};
+
+/* sound card controls */
+static const char *headset_switch_text[] = {"Earpiece", "Headset"};
+
+static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"};
+
+static const struct soc_enum headset_enum =
+	SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
+
+static const struct soc_enum lo_enum =
+	SOC_ENUM_SINGLE_EXT(4, lo_text);
+
+static int headset_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = hs_switch;
+	return 0;
+}
+
+static int headset_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] == hs_switch)
+		return 0;
+
+	if (ucontrol->value.integer.value[0]) {
+		pr_debug("hs_set HS path\n");
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+	} else {
+		pr_debug("hs_set EP path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+	}
+	snd_soc_dapm_sync(&codec->dapm);
+	hs_switch = ucontrol->value.integer.value[0];
+
+	return 0;
+}
+
+static void lo_enable_out_pins(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL");
+	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR");
+	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL");
+	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR");
+	snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
+	snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
+	if (hs_switch) {
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+	} else {
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+	}
+}
+
+static int lo_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = lo_dac;
+	return 0;
+}
+
+static int lo_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] == lo_dac)
+		return 0;
+
+	/* we dont want to work with last state of lineout so just enable all
+	 * pins and then disable pins not required
+	 */
+	lo_enable_out_pins(codec);
+	switch (ucontrol->value.integer.value[0]) {
+	case 0:
+		pr_debug("set vibra path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT");
+		snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0);
+		break;
+
+	case 1:
+		pr_debug("set hs  path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
+		break;
+
+	case 2:
+		pr_debug("set spkr path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL");
+		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44);
+		break;
+
+	case 3:
+		pr_debug("set null path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL");
+		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66);
+		break;
+	}
+	snd_soc_dapm_sync(&codec->dapm);
+	lo_dac = ucontrol->value.integer.value[0];
+	return 0;
+}
+
+static const struct snd_kcontrol_new mfld_snd_controls[] = {
+	SOC_ENUM_EXT("Playback Switch", headset_enum,
+			headset_get_switch, headset_set_switch),
+	SOC_ENUM_EXT("Lineout Mux", lo_enum,
+			lo_get_switch, lo_set_switch),
+};
+
+static const struct snd_soc_dapm_widget mfld_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mfld_map[] = {
+	{"Headphones", NULL, "HPOUTR"},
+	{"Headphones", NULL, "HPOUTL"},
+	{"Mic", NULL, "AMIC1"},
+};
+
+static void mfld_jack_check(unsigned int intr_status)
+{
+	struct mfld_jack_data jack_data;
+
+	jack_data.mfld_jack = &mfld_jack;
+	jack_data.intr_id = intr_status;
+
+	sn95031_jack_detection(&jack_data);
+	/* TODO: add american headset detection post gpiolib support */
+}
+
+static int mfld_init(struct snd_soc_pcm_runtime *runtime)
+{
+	struct snd_soc_codec *codec = runtime->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret_val;
+
+	/* Add jack sense widgets */
+	snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets));
+
+	/* Set up the map */
+	snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map));
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Mic");
+	snd_soc_dapm_sync(dapm);
+
+	ret_val = snd_soc_add_controls(codec, mfld_snd_controls,
+				ARRAY_SIZE(mfld_snd_controls));
+	if (ret_val) {
+		pr_err("soc_add_controls failed %d", ret_val);
+		return ret_val;
+	}
+	/* default is earpiece pin, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "Headphones");
+	/* default is lineout NC, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
+	lo_dac = 3;
+	hs_switch = 0;
+	/* we dont use linein in this so set to NC */
+	snd_soc_dapm_disable_pin(dapm, "LINEINL");
+	snd_soc_dapm_disable_pin(dapm, "LINEINR");
+	snd_soc_dapm_sync(dapm);
+
+	/* Headset and button jack detection */
+	ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 |
+			SND_JACK_BTN_1, &mfld_jack);
+	if (ret_val) {
+		pr_err("jack creation failed\n");
+		return ret_val;
+	}
+
+	ret_val = snd_soc_jack_add_pins(&mfld_jack,
+			ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins);
+	if (ret_val) {
+		pr_err("adding jack pins failed\n");
+		return ret_val;
+	}
+	ret_val = snd_soc_jack_add_zones(&mfld_jack,
+			ARRAY_SIZE(mfld_zones), mfld_zones);
+	if (ret_val) {
+		pr_err("adding jack zones failed\n");
+		return ret_val;
+	}
+
+	/* we want to check if anything is inserted at boot,
+	 * so send a fake event to codec and it will read adc
+	 * to find if anything is there or not */
+	mfld_jack_check(MFLD_JACK_INSERT);
+	return ret_val;
+}
+
+struct snd_soc_dai_link mfld_msic_dailink[] = {
+	{
+		.name = "Medfield Headset",
+		.stream_name = "Headset",
+		.cpu_dai_name = "Headset-cpu-dai",
+		.codec_dai_name = "SN95031 Headset",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = mfld_init,
+	},
+	{
+		.name = "Medfield Speaker",
+		.stream_name = "Speaker",
+		.cpu_dai_name = "Speaker-cpu-dai",
+		.codec_dai_name = "SN95031 Speaker",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Vibra",
+		.stream_name = "Vibra1",
+		.cpu_dai_name = "Vibra1-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra1",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Haptics",
+		.stream_name = "Vibra2",
+		.cpu_dai_name = "Vibra2-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra2",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_mfld = {
+	.name = "medfield_audio",
+	.dai_link = mfld_msic_dailink,
+	.num_links = ARRAY_SIZE(mfld_msic_dailink),
+};
+
+static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
+{
+	struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
+
+	memcpy_fromio(&mc_private->interrupt_status,
+			((void *)(mc_private->int_base)),
+			sizeof(u8));
+	return IRQ_WAKE_THREAD;
+}
+
+static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
+{
+	struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
+
+	if (mfld_jack.codec == NULL)
+		return IRQ_HANDLED;
+	mfld_jack_check(mc_drv_ctx->interrupt_status);
+
+	return IRQ_HANDLED;
+}
+
+static int __devinit snd_mfld_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0, irq;
+	struct mfld_mc_private *mc_drv_ctx;
+	struct resource *irq_mem;
+
+	pr_debug("snd_mfld_mc_probe called\n");
+
+	/* retrive the irq number */
+	irq = platform_get_irq(pdev, 0);
+
+	/* audio interrupt base of SRAM location where
+	 * interrupts are stored by System FW */
+	mc_drv_ctx = kzalloc(sizeof(*mc_drv_ctx), GFP_ATOMIC);
+	if (!mc_drv_ctx) {
+		pr_err("allocation failed\n");
+		return -ENOMEM;
+	}
+
+	irq_mem = platform_get_resource_byname(
+				pdev, IORESOURCE_MEM, "IRQ_BASE");
+	if (!irq_mem) {
+		pr_err("no mem resource given\n");
+		ret_val = -ENODEV;
+		goto unalloc;
+	}
+	mc_drv_ctx->int_base = ioremap_nocache(irq_mem->start,
+					resource_size(irq_mem));
+	if (!mc_drv_ctx->int_base) {
+		pr_err("Mapping of cache failed\n");
+		ret_val = -ENOMEM;
+		goto unalloc;
+	}
+	/* register for interrupt */
+	ret_val = request_threaded_irq(irq, snd_mfld_jack_intr_handler,
+			snd_mfld_jack_detection,
+			IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
+	if (ret_val) {
+		pr_err("cannot register IRQ\n");
+		goto unalloc;
+	}
+	/* register the soc card */
+	snd_soc_card_mfld.dev = &pdev->dev;
+	ret_val = snd_soc_register_card(&snd_soc_card_mfld);
+	if (ret_val) {
+		pr_debug("snd_soc_register_card failed %d\n", ret_val);
+		goto freeirq;
+	}
+	platform_set_drvdata(pdev, mc_drv_ctx);
+	pr_debug("successfully exited probe\n");
+	return ret_val;
+
+freeirq:
+	free_irq(irq, mc_drv_ctx);
+unalloc:
+	kfree(mc_drv_ctx);
+	return ret_val;
+}
+
+static int __devexit snd_mfld_mc_remove(struct platform_device *pdev)
+{
+	struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev);
+
+	pr_debug("snd_mfld_mc_remove called\n");
+	free_irq(platform_get_irq(pdev, 0), mc_drv_ctx);
+	snd_soc_unregister_card(&snd_soc_card_mfld);
+	kfree(mc_drv_ctx);
+	platform_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+static struct platform_driver snd_mfld_mc_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "msic_audio",
+	},
+	.probe = snd_mfld_mc_probe,
+	.remove = __devexit_p(snd_mfld_mc_remove),
+};
+
+static int __init snd_mfld_driver_init(void)
+{
+	pr_debug("snd_mfld_driver_init called\n");
+	return platform_driver_register(&snd_mfld_mc_driver);
+}
+module_init(snd_mfld_driver_init);
+
+static void __exit snd_mfld_driver_exit(void)
+{
+	pr_debug("snd_mfld_driver_exit called\n");
+	platform_driver_unregister(&snd_mfld_mc_driver);
+}
+module_exit(snd_mfld_driver_exit);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:msic-audio");
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
new file mode 100644
index 00000000000..ee2c22475a7
--- /dev/null
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -0,0 +1,474 @@
+/*
+ *  sst_platform.c - Intel MID Platform driver
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *
+ */
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../../drivers/staging/intel_sst/intel_sst_ioctl.h"
+#include "../../../drivers/staging/intel_sst/intel_sst.h"
+#include "sst_platform.h"
+
+static struct snd_pcm_hardware sst_platform_pcm_hw = {
+	.info =	(SNDRV_PCM_INFO_INTERLEAVED |
+			SNDRV_PCM_INFO_DOUBLE |
+			SNDRV_PCM_INFO_PAUSE |
+			SNDRV_PCM_INFO_RESUME |
+			SNDRV_PCM_INFO_MMAP|
+			SNDRV_PCM_INFO_MMAP_VALID |
+			SNDRV_PCM_INFO_BLOCK_TRANSFER |
+			SNDRV_PCM_INFO_SYNC_START),
+	.formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 |
+			SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 |
+			SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32),
+	.rates = (SNDRV_PCM_RATE_8000|
+			SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000),
+	.rate_min = SST_MIN_RATE,
+	.rate_max = SST_MAX_RATE,
+	.channels_min =	SST_MIN_CHANNEL,
+	.channels_max =	SST_MAX_CHANNEL,
+	.buffer_bytes_max = SST_MAX_BUFFER,
+	.period_bytes_min = SST_MIN_PERIOD_BYTES,
+	.period_bytes_max = SST_MAX_PERIOD_BYTES,
+	.periods_min = SST_MIN_PERIODS,
+	.periods_max = SST_MAX_PERIODS,
+	.fifo_size = SST_FIFO_SIZE,
+};
+
+/* MFLD - MSIC */
+struct snd_soc_dai_driver sst_platform_dai[] = {
+{
+	.name = "Headset-cpu-dai",
+	.id = 0,
+	.playback = {
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 5,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Speaker-cpu-dai",
+	.id = 1,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Vibra1-cpu-dai",
+	.id = 2,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_MONO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Vibra2-cpu-dai",
+	.id = 3,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+};
+
+/* helper functions */
+static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
+					int state)
+{
+	spin_lock(&stream->status_lock);
+	stream->stream_status = state;
+	spin_unlock(&stream->status_lock);
+}
+
+static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
+{
+	int state;
+
+	spin_lock(&stream->status_lock);
+	state = stream->stream_status;
+	spin_unlock(&stream->status_lock);
+	return state;
+}
+
+static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
+				struct snd_sst_stream_params *param)
+{
+
+	param->uc.pcm_params.codec = SST_CODEC_TYPE_PCM;
+	param->uc.pcm_params.num_chan = (u8) substream->runtime->channels;
+	param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits;
+	param->uc.pcm_params.reserved = 0;
+	param->uc.pcm_params.sfreq = substream->runtime->rate;
+	param->uc.pcm_params.ring_buffer_size =
+					snd_pcm_lib_buffer_bytes(substream);
+	param->uc.pcm_params.period_count = substream->runtime->period_size;
+	param->uc.pcm_params.ring_buffer_addr =
+				virt_to_phys(substream->dma_buffer.area);
+	pr_debug("period_cnt = %d\n", param->uc.pcm_params.period_count);
+	pr_debug("sfreq= %d, wd_sz = %d\n",
+		 param->uc.pcm_params.sfreq, param->uc.pcm_params.pcm_wd_sz);
+}
+
+static int sst_platform_alloc_stream(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream =
+			substream->runtime->private_data;
+	struct snd_sst_stream_params param = {{{0,},},};
+	struct snd_sst_params str_params = {0};
+	int ret_val;
+
+	/* set codec params and inform SST driver the same */
+	sst_fill_pcm_params(substream, &param);
+	substream->runtime->dma_area = substream->dma_buffer.area;
+	str_params.sparams = param;
+	str_params.codec =  param.uc.pcm_params.codec;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		str_params.ops = STREAM_OPS_PLAYBACK;
+		str_params.device_type = substream->pcm->device + 1;
+		pr_debug("Playbck stream,Device %d\n",
+					substream->pcm->device);
+	} else {
+		str_params.ops = STREAM_OPS_CAPTURE;
+		str_params.device_type = SND_SST_DEVICE_CAPTURE;
+		pr_debug("Capture stream,Device %d\n",
+					substream->pcm->device);
+	}
+	ret_val = stream->sstdrv_ops->pcm_control->open(&str_params);
+	pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val);
+	if (ret_val < 0)
+		return ret_val;
+
+	stream->stream_info.str_id = ret_val;
+	pr_debug("str id :  %d\n", stream->stream_info.str_id);
+	return ret_val;
+}
+
+static void sst_period_elapsed(void *mad_substream)
+{
+	struct snd_pcm_substream *substream = mad_substream;
+	struct sst_runtime_stream *stream;
+	int status;
+
+	if (!substream || !substream->runtime)
+		return;
+	stream = substream->runtime->private_data;
+	if (!stream)
+		return;
+	status = sst_get_stream_status(stream);
+	if (status != SST_PLATFORM_RUNNING)
+		return;
+	snd_pcm_period_elapsed(substream);
+}
+
+static int sst_platform_init_stream(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream =
+			substream->runtime->private_data;
+	int ret_val;
+
+	pr_debug("setting buffer ptr param\n");
+	sst_set_stream_status(stream, SST_PLATFORM_INIT);
+	stream->stream_info.period_elapsed = sst_period_elapsed;
+	stream->stream_info.mad_substream = substream;
+	stream->stream_info.buffer_ptr = 0;
+	stream->stream_info.sfreq = substream->runtime->rate;
+	ret_val = stream->sstdrv_ops->pcm_control->device_control(
+			SST_SND_STREAM_INIT, &stream->stream_info);
+	if (ret_val)
+		pr_err("control_set ret error %d\n", ret_val);
+	return ret_val;
+
+}
+/* end -- helper functions */
+
+static int sst_platform_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime;
+	struct sst_runtime_stream *stream;
+	int ret_val = 0;
+
+	pr_debug("sst_platform_open called\n");
+	runtime = substream->runtime;
+	runtime->hw = sst_platform_pcm_hw;
+	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+	if (!stream)
+		return -ENOMEM;
+	spin_lock_init(&stream->status_lock);
+	stream->stream_info.str_id = 0;
+	sst_set_stream_status(stream, SST_PLATFORM_INIT);
+	stream->stream_info.mad_substream = substream;
+	/* allocate memory for SST API set */
+	stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops),
+							GFP_KERNEL);
+	if (!stream->sstdrv_ops) {
+		pr_err("sst: mem allocation for ops fail\n");
+		kfree(stream);
+		return -ENOMEM;
+	}
+	stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID;
+	/* registering with SST driver to get access to SST APIs to use */
+	ret_val = register_sst_card(stream->sstdrv_ops);
+	if (ret_val) {
+		pr_err("sst: sst card registration failed\n");
+		return ret_val;
+	}
+	runtime->private_data = stream;
+	return snd_pcm_hw_constraint_integer(runtime,
+			 SNDRV_PCM_HW_PARAM_PERIODS);
+}
+
+static int sst_platform_close(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val = 0, str_id;
+
+	pr_debug("sst_platform_close called\n");
+	stream = substream->runtime->private_data;
+	str_id = stream->stream_info.str_id;
+	if (str_id)
+		ret_val = stream->sstdrv_ops->pcm_control->close(str_id);
+	kfree(stream->sstdrv_ops);
+	kfree(stream);
+	return ret_val;
+}
+
+static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val = 0, str_id;
+
+	pr_debug("sst_platform_pcm_prepare called\n");
+	stream = substream->runtime->private_data;
+	str_id = stream->stream_info.str_id;
+	if (stream->stream_info.str_id) {
+		ret_val = stream->sstdrv_ops->pcm_control->device_control(
+					SST_SND_DROP, &str_id);
+		return ret_val;
+	}
+
+	ret_val = sst_platform_alloc_stream(substream);
+	if (ret_val < 0)
+		return ret_val;
+	snprintf(substream->pcm->id, sizeof(substream->pcm->id),
+			"%d", stream->stream_info.str_id);
+
+	ret_val = sst_platform_init_stream(substream);
+	if (ret_val)
+		return ret_val;
+	substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER;
+	return ret_val;
+}
+
+static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
+					int cmd)
+{
+	int ret_val = 0, str_id;
+	struct sst_runtime_stream *stream;
+	int str_cmd, status;
+
+	pr_debug("sst_platform_pcm_trigger called\n");
+	stream = substream->runtime->private_data;
+	str_id = stream->stream_info.str_id;
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		pr_debug("sst: Trigger Start\n");
+		str_cmd = SST_SND_START;
+		status = SST_PLATFORM_RUNNING;
+		stream->stream_info.mad_substream = substream;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		pr_debug("sst: in stop\n");
+		str_cmd = SST_SND_DROP;
+		status = SST_PLATFORM_DROPPED;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		pr_debug("sst: in pause\n");
+		str_cmd = SST_SND_PAUSE;
+		status = SST_PLATFORM_PAUSED;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		pr_debug("sst: in pause release\n");
+		str_cmd = SST_SND_RESUME;
+		status = SST_PLATFORM_RUNNING;
+		break;
+	default:
+		return -EINVAL;
+	}
+	ret_val = stream->sstdrv_ops->pcm_control->device_control(str_cmd,
+								&str_id);
+	if (!ret_val)
+		sst_set_stream_status(stream, status);
+
+	return ret_val;
+}
+
+
+static snd_pcm_uframes_t sst_platform_pcm_pointer
+			(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val, status;
+	struct pcm_stream_info *str_info;
+
+	stream = substream->runtime->private_data;
+	status = sst_get_stream_status(stream);
+	if (status == SST_PLATFORM_INIT)
+		return 0;
+	str_info = &stream->stream_info;
+	ret_val = stream->sstdrv_ops->pcm_control->device_control(
+				SST_SND_BUFFER_POINTER, str_info);
+	if (ret_val) {
+		pr_err("sst: error code = %d\n", ret_val);
+		return ret_val;
+	}
+	return stream->stream_info.buffer_ptr;
+}
+
+static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params)
+{
+	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+
+	return 0;
+}
+
+static struct snd_pcm_ops sst_platform_ops = {
+	.open = sst_platform_open,
+	.close = sst_platform_close,
+	.ioctl = snd_pcm_lib_ioctl,
+	.prepare = sst_platform_pcm_prepare,
+	.trigger = sst_platform_pcm_trigger,
+	.pointer = sst_platform_pcm_pointer,
+	.hw_params = sst_platform_pcm_hw_params,
+};
+
+static void sst_pcm_free(struct snd_pcm *pcm)
+{
+	pr_debug("sst_pcm_free called\n");
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+			struct snd_pcm *pcm)
+{
+	int retval = 0;
+
+	pr_debug("sst_pcm_new called\n");
+	if (dai->driver->playback.channels_min ||
+			dai->driver->capture.channels_min) {
+		retval =  snd_pcm_lib_preallocate_pages_for_all(pcm,
+			SNDRV_DMA_TYPE_CONTINUOUS,
+			snd_dma_continuous_data(GFP_KERNEL),
+			SST_MIN_BUFFER, SST_MAX_BUFFER);
+		if (retval) {
+			pr_err("dma buffer allocationf fail\n");
+			return retval;
+		}
+	}
+	return retval;
+}
+struct snd_soc_platform_driver sst_soc_platform_drv = {
+	.ops		= &sst_platform_ops,
+	.pcm_new	= sst_pcm_new,
+	.pcm_free	= sst_pcm_free,
+};
+
+static int sst_platform_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	pr_debug("sst_platform_probe called\n");
+	ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
+	if (ret) {
+		pr_err("registering soc platform failed\n");
+		return ret;
+	}
+
+	ret = snd_soc_register_dais(&pdev->dev,
+				sst_platform_dai, ARRAY_SIZE(sst_platform_dai));
+	if (ret) {
+		pr_err("registering cpu dais failed\n");
+		snd_soc_unregister_platform(&pdev->dev);
+	}
+	return ret;
+}
+
+static int sst_platform_remove(struct platform_device *pdev)
+{
+
+	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai));
+	snd_soc_unregister_platform(&pdev->dev);
+	pr_debug("sst_platform_remove sucess\n");
+	return 0;
+}
+
+static struct platform_driver sst_platform_driver = {
+	.driver		= {
+		.name		= "sst-platform",
+		.owner		= THIS_MODULE,
+	},
+	.probe		= sst_platform_probe,
+	.remove		= sst_platform_remove,
+};
+
+static int __init sst_soc_platform_init(void)
+{
+	pr_debug("sst_soc_platform_init called\n");
+	return  platform_driver_register(&sst_platform_driver);
+}
+module_init(sst_soc_platform_init);
+
+static void __exit sst_soc_platform_exit(void)
+{
+	platform_driver_unregister(&sst_platform_driver);
+	pr_debug("sst_soc_platform_exit sucess\n");
+}
+module_exit(sst_soc_platform_exit);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sst-platform");
diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h
new file mode 100644
index 00000000000..df370286694
--- /dev/null
+++ b/sound/soc/mid-x86/sst_platform.h
@@ -0,0 +1,63 @@
+/*
+ *  sst_platform.h - Intel MID Platform driver header file
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *
+ */
+
+#ifndef __SST_PLATFORMDRV_H__
+#define __SST_PLATFORMDRV_H__
+
+#define SST_MONO		1
+#define SST_STEREO		2
+#define SST_MAX_CAP		5
+
+#define SST_MIN_RATE		8000
+#define SST_MAX_RATE		48000
+#define SST_MIN_CHANNEL		1
+#define SST_MAX_CHANNEL		5
+#define SST_MAX_BUFFER		(800*1024)
+#define SST_MIN_BUFFER		(800*1024)
+#define SST_MIN_PERIOD_BYTES	32
+#define SST_MAX_PERIOD_BYTES	SST_MAX_BUFFER
+#define SST_MIN_PERIODS		2
+#define SST_MAX_PERIODS		(1024*2)
+#define SST_FIFO_SIZE		0
+#define SST_CARD_NAMES		"intel_mid_card"
+#define MSIC_VENDOR_ID		3
+
+struct sst_runtime_stream {
+	int     stream_status;
+	struct pcm_stream_info stream_info;
+	struct intel_sst_card_ops *sstdrv_ops;
+	spinlock_t	status_lock;
+};
+
+enum sst_drv_status {
+	SST_PLATFORM_INIT = 1,
+	SST_PLATFORM_STARTED,
+	SST_PLATFORM_RUNNING,
+	SST_PLATFORM_PAUSED,
+	SST_PLATFORM_DROPPED,
+};
+
+#endif
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index a088db6d509..b5922984eac 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -24,6 +24,7 @@ config SND_OMAP_SOC_RX51
 	select OMAP_MCBSP
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TLV320AIC3X
+	select SND_SOC_TPA6130A2
 	help
 	  Say Y if you want to add support for SoC audio on Nokia RX-51
 	  hardware. This is also known as Nokia N900 product.
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 161750443eb..73dde4a1adc 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = {
 	.cpu_dai_name ="omap-mcbsp-dai.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.platform_name = "omap-pcm-audio",
-	.codec_name = "tlv320aic23-codec",
+	.codec_name = "tlv320aic23-codec.2-001a",
 	.init = am3517evm_aic23_init,
 	.ops = &am3517evm_ops,
 };
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdcee21..3167be68962 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 	/* Set up digital mute if not provided by the codec */
 	if (!codec_dai->driver->ops) {
 		codec_dai->driver->ops = &ams_delta_dai_ops;
-	} else if (!codec_dai->driver->ops->digital_mute) {
-		codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
 	} else {
 		ams_delta_ops.startup = ams_delta_startup;
 		ams_delta_ops.shutdown = ams_delta_shutdown;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d203f4da18a..2175f09e57b 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -69,110 +69,6 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
  */
 static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2];
 
-#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
-static const int omap1_dma_reqs[][2] = {
-	{ OMAP_DMA_MCBSP1_TX, OMAP_DMA_MCBSP1_RX },
-	{ OMAP_DMA_MCBSP2_TX, OMAP_DMA_MCBSP2_RX },
-	{ OMAP_DMA_MCBSP3_TX, OMAP_DMA_MCBSP3_RX },
-};
-static const unsigned long omap1_mcbsp_port[][2] = {
-	{ OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
-	  OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
-	{ OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
-	  OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
-	{ OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DXR1,
-	  OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DRR1 },
-};
-#else
-static const int omap1_dma_reqs[][2] = {};
-static const unsigned long omap1_mcbsp_port[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3)
-static const int omap24xx_dma_reqs[][2] = {
-	{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
-	{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
-	{ OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
-	{ OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
-	{ OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
-#endif
-};
-#else
-static const int omap24xx_dma_reqs[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP4)
-static const int omap44xx_dma_reqs[][2] = {
-	{ OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX },
-	{ OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX },
-	{ OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX },
-	{ OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX },
-};
-#else
-static const int omap44xx_dma_reqs[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP2420)
-static const unsigned long omap2420_mcbsp_port[][2] = {
-	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
-	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
-	{ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
-	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
-};
-#else
-static const unsigned long omap2420_mcbsp_port[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP2430)
-static const unsigned long omap2430_mcbsp_port[][2] = {
-	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
-};
-#else
-static const unsigned long omap2430_mcbsp_port[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP3)
-static const unsigned long omap34xx_mcbsp_port[][2] = {
-	{ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
-};
-#else
-static const unsigned long omap34xx_mcbsp_port[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP4)
-static const unsigned long omap44xx_mcbsp_port[][2] = {
-	{ OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
-};
-#else
-static const unsigned long omap44xx_mcbsp_port[][2] = {};
-#endif
-
 static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -346,24 +242,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	unsigned int format, div, framesize, master;
 
 	dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream];
-	if (cpu_class_is_omap1()) {
-		dma = omap1_dma_reqs[bus_id][substream->stream];
-		port = omap1_mcbsp_port[bus_id][substream->stream];
-	} else if (cpu_is_omap2420()) {
-		dma = omap24xx_dma_reqs[bus_id][substream->stream];
-		port = omap2420_mcbsp_port[bus_id][substream->stream];
-	} else if (cpu_is_omap2430()) {
-		dma = omap24xx_dma_reqs[bus_id][substream->stream];
-		port = omap2430_mcbsp_port[bus_id][substream->stream];
-	} else if (cpu_is_omap343x()) {
-		dma = omap24xx_dma_reqs[bus_id][substream->stream];
-		port = omap34xx_mcbsp_port[bus_id][substream->stream];
-	 } else if (cpu_is_omap44xx()) {
-		dma = omap44xx_dma_reqs[bus_id][substream->stream];
-		port = omap44xx_mcbsp_port[bus_id][substream->stream];
-	} else {
-		return -ENODEV;
-	}
+
+	dma = omap_mcbsp_dma_ch_params(bus_id, substream->stream);
+	port = omap_mcbsp_dma_reg_params(bus_id, substream->stream);
+
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		dma_data->data_type = OMAP_DMA_DATA_TYPE_S16;
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 110c106611d..37dc7211ed3 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -43,7 +43,7 @@ enum omap_mcbsp_div {
 	OMAP_MCBSP_CLKGDV,		/* Sample rate generator divider */
 };
 
-#if defined(CONFIG_ARCH_OMAP2420)
+#if defined(CONFIG_SOC_OMAP2420)
 #define NUM_LINKS	2
 #endif
 #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
@@ -54,7 +54,7 @@ enum omap_mcbsp_div {
 #undef  NUM_LINKS
 #define NUM_LINKS	4
 #endif
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
+#if defined(CONFIG_ARCH_OMAP3) || defined(CONFIG_SOC_OMAP2430)
 #undef  NUM_LINKS
 #define NUM_LINKS	5
 #endif
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 09fb0df8d41..d0986220eff 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -31,6 +31,7 @@
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <plat/mcbsp.h>
+#include "../codecs/tpa6130a2.h"
 
 #include <asm/mach-types.h>
 
@@ -39,6 +40,7 @@
 
 #define RX51_TVOUT_SEL_GPIO		40
 #define RX51_JACK_DETECT_GPIO		177
+#define RX51_ECI_SW_GPIO		182
 /*
  * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
  * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
@@ -47,7 +49,9 @@
 
 enum {
 	RX51_JACK_DISABLED,
-	RX51_JACK_TVOUT,		/* tv-out */
+	RX51_JACK_TVOUT,		/* tv-out with stereo output */
+	RX51_JACK_HP,			/* headphone: stereo output, no mic */
+	RX51_JACK_HS,			/* headset: stereo output with mic */
 };
 
 static int rx51_spk_func;
@@ -57,6 +61,19 @@ static int rx51_jack_func;
 static void rx51_ext_control(struct snd_soc_codec *codec)
 {
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int hp = 0, hs = 0, tvout = 0;
+
+	switch (rx51_jack_func) {
+	case RX51_JACK_TVOUT:
+		tvout = 1;
+		hp = 1;
+		break;
+	case RX51_JACK_HS:
+		hs = 1;
+	case RX51_JACK_HP:
+		hp = 1;
+		break;
+	}
 
 	if (rx51_spk_func)
 		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
@@ -66,9 +83,16 @@ static void rx51_ext_control(struct snd_soc_codec *codec)
 		snd_soc_dapm_enable_pin(dapm, "DMic");
 	else
 		snd_soc_dapm_disable_pin(dapm, "DMic");
+	if (hp)
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+	if (hs)
+		snd_soc_dapm_enable_pin(dapm, "HS Mic");
+	else
+		snd_soc_dapm_disable_pin(dapm, "HS Mic");
 
-	gpio_set_value(RX51_TVOUT_SEL_GPIO,
-		       rx51_jack_func == RX51_JACK_TVOUT);
+	gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
 
 	snd_soc_dapm_sync(dapm);
 }
@@ -153,6 +177,19 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rx51_hp_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_codec *codec = w->dapm->codec;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		tpa6130a2_stereo_enable(codec, 1);
+	else
+		tpa6130a2_stereo_enable(codec, 0);
+
+	return 0;
+}
+
 static int rx51_get_input(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
@@ -203,7 +240,7 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
 	{
 		.gpio = RX51_JACK_DETECT_GPIO,
 		.name = "avdet-gpio",
-		.report = SND_JACK_VIDEOOUT,
+		.report = SND_JACK_HEADSET,
 		.invert = 1,
 		.debounce_time = 200,
 	},
@@ -212,19 +249,38 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
 static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
 	SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
 	SND_SOC_DAPM_MIC("DMic", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event),
+	SND_SOC_DAPM_MIC("HS Mic", NULL),
+	SND_SOC_DAPM_LINE("FM Transmitter", NULL),
+};
+
+static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = {
+	SND_SOC_DAPM_SPK("Earphone", NULL),
 };
 
 static const struct snd_soc_dapm_route audio_map[] = {
 	{"Ext Spk", NULL, "HPLOUT"},
 	{"Ext Spk", NULL, "HPROUT"},
+	{"Headphone Jack", NULL, "LLOUT"},
+	{"Headphone Jack", NULL, "RLOUT"},
+	{"FM Transmitter", NULL, "LLOUT"},
+	{"FM Transmitter", NULL, "RLOUT"},
 
 	{"DMic Rate 64", NULL, "Mic Bias 2V"},
 	{"Mic Bias 2V", NULL, "DMic"},
 };
 
+static const struct snd_soc_dapm_route audio_mapb[] = {
+	{"b LINE2R", NULL, "MONO_LOUT"},
+	{"Earphone", NULL, "b HPLOUT"},
+
+	{"LINE1L", NULL, "b Mic Bias 2.5V"},
+	{"b Mic Bias 2.5V", NULL, "HS Mic"}
+};
+
 static const char *spk_function[] = {"Off", "On"};
 static const char *input_function[] = {"ADC", "Digital Mic"};
-static const char *jack_function[] = {"Off", "TV-OUT"};
+static const char *jack_function[] = {"Off", "TV-OUT", "Headphone", "Headset"};
 
 static const struct soc_enum rx51_enum[] = {
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
@@ -239,6 +295,11 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = {
 		     rx51_get_input, rx51_set_input),
 	SOC_ENUM_EXT("Jack Function", rx51_enum[2],
 		     rx51_get_jack, rx51_set_jack),
+	SOC_DAPM_PIN_SWITCH("FM Transmitter"),
+};
+
+static const struct snd_kcontrol_new aic34_rx51_controlsb[] = {
+	SOC_DAPM_PIN_SWITCH("Earphone"),
 };
 
 static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
@@ -265,11 +326,21 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 	/* Set up RX-51 specific audio path audio_map */
 	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
+	err = tpa6130a2_add_controls(codec);
+	if (err < 0)
+		return err;
+	snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
+
+	err = omap_mcbsp_st_add_controls(codec, 1);
+	if (err < 0)
+		return err;
+
 	snd_soc_dapm_sync(dapm);
 
 	/* AV jack detection */
 	err = snd_soc_jack_new(codec, "AV Jack",
-			       SND_JACK_VIDEOOUT, &rx51_av_jack);
+			       SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
+			       &rx51_av_jack);
 	if (err)
 		return err;
 	err = snd_soc_jack_add_gpios(&rx51_av_jack,
@@ -279,6 +350,24 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 	return err;
 }
 
+static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
+{
+	int err;
+
+	err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb,
+				   ARRAY_SIZE(aic34_rx51_controlsb));
+	if (err < 0)
+		return err;
+
+	err = snd_soc_dapm_new_controls(dapm, aic34_dapm_widgetsb,
+					ARRAY_SIZE(aic34_dapm_widgetsb));
+	if (err < 0)
+		return 0;
+
+	return snd_soc_dapm_add_routes(dapm, audio_mapb,
+				       ARRAY_SIZE(audio_mapb));
+}
+
 /* Digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link rx51_dai[] = {
 	{
@@ -293,11 +382,30 @@ static struct snd_soc_dai_link rx51_dai[] = {
 	},
 };
 
+struct snd_soc_aux_dev rx51_aux_dev[] = {
+	{
+		.name = "TLV320AIC34b",
+		.codec_name = "tlv320aic3x-codec.2-0019",
+		.init = rx51_aic34b_init,
+	},
+};
+
+static struct snd_soc_codec_conf rx51_codec_conf[] = {
+	{
+		.dev_name = "tlv320aic3x-codec.2-0019",
+		.name_prefix = "b",
+	},
+};
+
 /* Audio card */
 static struct snd_soc_card rx51_sound_card = {
 	.name = "RX-51",
 	.dai_link = rx51_dai,
 	.num_links = ARRAY_SIZE(rx51_dai),
+	.aux_dev = rx51_aux_dev,
+	.num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
+	.codec_conf = rx51_codec_conf,
+	.num_configs = ARRAY_SIZE(rx51_codec_conf),
 };
 
 static struct platform_device *rx51_snd_device;
@@ -309,10 +417,14 @@ static int __init rx51_soc_init(void)
 	if (!machine_is_nokia_rx51())
 		return -ENODEV;
 
-	err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel");
+	err = gpio_request_one(RX51_TVOUT_SEL_GPIO,
+			       GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel");
 	if (err)
 		goto err_gpio_tvout_sel;
-	gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0);
+	err = gpio_request_one(RX51_ECI_SW_GPIO,
+			       GPIOF_DIR_OUT | GPIOF_INIT_HIGH, "eci_sw");
+	if (err)
+		goto err_gpio_eci_sw;
 
 	rx51_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!rx51_snd_device) {
@@ -330,6 +442,8 @@ static int __init rx51_soc_init(void)
 err2:
 	platform_device_put(rx51_snd_device);
 err1:
+	gpio_free(RX51_ECI_SW_GPIO);
+err_gpio_eci_sw:
 	gpio_free(RX51_TVOUT_SEL_GPIO);
 err_gpio_tvout_sel:
 
@@ -342,6 +456,7 @@ static void __exit rx51_soc_exit(void)
 				rx51_av_jack_gpios);
 
 	platform_device_unregister(rx51_snd_device);
+	gpio_free(RX51_ECI_SW_GPIO);
 	gpio_free(RX51_TVOUT_SEL_GPIO);
 }
 
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0d5fc..784cff5f67e 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link corgi_dai = {
 	.name = "WM8731",
 	.stream_name = "WM8731",
-	.cpu_dai_name = "pxa-is2-dai",
+	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8731-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8731-codec-0.001a",
+	.codec_name = "wm8731-codec-0.001b",
 	.init = corgi_wm8731_init,
 	.ops = &corgi_ops,
 };
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 28333e7d9c5..dc65650a6fa 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -117,7 +117,7 @@ static struct snd_soc_dai_link e740_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9705-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
@@ -126,7 +126,7 @@ static struct snd_soc_dai_link e740_dai[] = {
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name = "wm9705-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 01bf31675c5..51897fcd911 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -99,7 +99,7 @@ static struct snd_soc_dai_link e750_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9705-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
@@ -109,7 +109,7 @@ static struct snd_soc_dai_link e750_dai[] = {
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9705-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index c6a37c6ef23..053ed208e59 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -89,7 +89,7 @@ static struct snd_soc_dai_link e800_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9712-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
@@ -98,7 +98,7 @@ static struct snd_soc_dai_link e800_dai[] = {
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9712-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index fc22e6eefc9..b13a4252812 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9712-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
@@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9712-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 0d70fc8c12b..38ca6759907 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -162,7 +162,7 @@ static struct snd_soc_dai_link mioa701_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9713-hifi",
 		.codec_name = "wm9713-codec",
 		.init = mioa701_wm9713_init,
@@ -172,7 +172,7 @@ static struct snd_soc_dai_link mioa701_dai[] = {
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9713-aux",
 		.codec_name = "wm9713-codec",
 		.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 857db96d4a4..504e4004f00 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -132,7 +132,7 @@ static struct snd_soc_dai_link palm27x_dai[] = {
 {
 	.name = "AC97 HiFi",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai_name = "pxa-ac97.0",
+	.cpu_dai_name = "pxa2xx-ac97",
 	.codec_dai_name =  "wm9712-hifi",
 	.codec_name = "wm9712-codec",
 	.platform_name = "pxa-pcm-audio",
@@ -141,7 +141,7 @@ static struct snd_soc_dai_link palm27x_dai[] = {
 {
 	.name = "AC97 Aux",
 	.stream_name = "AC97 Aux",
-	.cpu_dai_name = "pxa-ac97.1",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
 	.codec_dai_name = "wm9712-aux",
 	.codec_name = "wm9712-codec",
 	.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee115e2..a7d4999f9b2 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
 	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8731-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8731-codec.0-001a",
+	.codec_name = "wm8731-codec.0-001b",
 	.init = poodle_wm8731_init,
 	.ops = &poodle_ops,
 };
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index 0fd60f42303..2afabaf5949 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = {
 	.hw_params = raumfeld_cs4270_hw_params,
 };
 
-static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state)
+static int raumfeld_line_suspend(struct snd_soc_card *card)
 {
 	raumfeld_enable_audio(false);
 	return 0;
 }
 
-static int raumfeld_line_resume(struct platform_device *pdev)
+static int raumfeld_line_resume(struct snd_soc_card *card)
 {
 	raumfeld_enable_audio(true);
 	return 0;
@@ -229,19 +229,19 @@ static struct snd_soc_dai_link raumfeld_dai[] = {
 {
 	.name		= "ak4104",
 	.stream_name	= "Playback",
-	.cpu_dai_name = "pxa-ssp-dai.1",
-	.codec_dai_name = "ak4104-hifi",
-	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name	= "pxa-ssp-dai.1",
+	.codec_dai_name	= "ak4104-hifi",
+	.platform_name	= "pxa-pcm-audio",
 	.ops		= &raumfeld_ak4104_ops,
-	.codec_name = "ak4104-codec.0",
+	.codec_name	= "ak4104-codec.0",
 },
 {
 	.name		= "CS4270",
 	.stream_name	= "CS4270",
-	.cpu_dai_name = "pxa-ssp-dai.0",
-	.platform_name = "pxa-pcm-audio",
-	.codec_dai_name = "cs4270-hifi",
-	.codec_name = "cs4270-codec.0-0048",
+	.cpu_dai_name	= "pxa-ssp-dai.0",
+	.platform_name	= "pxa-pcm-audio",
+	.codec_dai_name	= "cs4270-hifi",
+	.codec_name	= "cs4270-codec.0-0048",
 	.ops		= &raumfeld_cs4270_ops,
 },};
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69b957..8e157135063 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link spitz_dai = {
 	.name = "wm8750",
 	.stream_name = "WM8750",
-	.cpu_dai_name = "pxa-is2",
+	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8750-codec.0-001a",
+	.codec_name = "wm8750-codec.0-001b",
 	.init = spitz_wm8750_init,
 	.ops = &spitz_ops,
 };
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f75804ef089..9a235136695 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -219,7 +219,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
 {
 	.name = "AC97",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai_name = "pxa-ac97.0",
+	.cpu_dai_name = "pxa2xx-ac97",
 	.codec_dai_name = "wm9712-hifi",
 	.platform_name = "pxa-pcm-audio",
 	.codec_name = "wm9712-codec",
@@ -229,7 +229,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
 {
 	.name = "AC97 Aux",
 	.stream_name = "AC97 Aux",
-	.cpu_dai_name = "pxa-ac97.1",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
 	.codec_dai_name = "wm9712-aux",
 	.platform_name = "pxa-pcm-audio",
 	.codec_name = "wm9712-codec",
@@ -237,7 +237,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
 },
 };
 
-static int tosa_probe(struct platform_device *dev)
+static int tosa_probe(struct snd_soc_card *card)
 {
 	int ret;
 
@@ -251,7 +251,7 @@ static int tosa_probe(struct platform_device *dev)
 	return ret;
 }
 
-static int tosa_remove(struct platform_device *dev)
+static int tosa_remove(struct snd_soc_card *card)
 {
 	gpio_free(TOSA_GPIO_L_MUTE);
 	return 0;
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 3ceaef68e01..d69d9fc3223 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -95,6 +95,11 @@ static struct snd_soc_jack_pin hs_jack_pins[] = {
 		.pin	= "Headphone Jack",
 		.mask	= SND_JACK_HEADPHONE,
 	},
+	{
+		.pin    = "Ext Spk",
+		.mask   = SND_JACK_HEADPHONE,
+		.invert = 1
+	},
 };
 
 /* Headset jack detection gpios */
@@ -147,7 +152,7 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_disable_pin(dapm, "LINPUT3");
 	snd_soc_dapm_disable_pin(dapm, "RINPUT3");
 	snd_soc_dapm_disable_pin(dapm, "OUT3");
-	snd_soc_dapm_disable_pin(dapm, "MONO");
+	snd_soc_dapm_disable_pin(dapm, "MONO1");
 
 	/* Add z2 specific widgets */
 	snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index b222a7d7202..ac577263b3e 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -166,7 +166,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 	.stream_name = "AC97 HiFi",
 	.codec_name = "wm9713-codec",
 	.platform_name = "pxa-pcm-audio",
-	.cpu_dai_name = "pxa-ac97.0",
+	.cpu_dai_name = "pxa2xx-ac97",
 	.codec_name = "wm9713-hifi",
 	.init = zylonite_wm9713_init,
 },
@@ -175,7 +175,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 	.stream_name = "AC97 Aux",
 	.codec_name = "wm9713-codec",
 	.platform_name = "pxa-pcm-audio",
-	.cpu_dai_name = "pxa-ac97.1",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
 	.codec_name = "wm9713-aux",
 },
 {
@@ -189,7 +189,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 },
 };
 
-static int zylonite_probe(struct platform_device *pdev)
+static int zylonite_probe(struct snd_soc_card *card)
 {
 	int ret;
 
@@ -216,7 +216,7 @@ static int zylonite_probe(struct platform_device *pdev)
 	return 0;
 }
 
-static int zylonite_remove(struct platform_device *pdev)
+static int zylonite_remove(struct snd_soc_card *card)
 {
 	if (clk_pout) {
 		clk_disable(pout);
@@ -226,8 +226,7 @@ static int zylonite_remove(struct platform_device *pdev)
 	return 0;
 }
 
-static int zylonite_suspend_post(struct platform_device *pdev,
-				 pm_message_t state)
+static int zylonite_suspend_post(struct snd_soc_card *card)
 {
 	if (clk_pout)
 		clk_disable(pout);
@@ -235,7 +234,7 @@ static int zylonite_suspend_post(struct platform_device *pdev,
 	return 0;
 }
 
-static int zylonite_resume_pre(struct platform_device *pdev)
+static int zylonite_resume_pre(struct snd_soc_card *card)
 {
 	int ret = 0;
 
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index a6a6b5fa2f2..a3fdfb63146 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,6 +1,6 @@
 config SND_SOC_SAMSUNG
 	tristate "ASoC support for Samsung"
-	depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_S5P6442 || ARCH_S5PV310
+	depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_S5P6442 || ARCH_EXYNOS4
 	select S3C64XX_DMA if ARCH_S3C64XX
 	select S3C2410_DMA if ARCH_S3C2410
 	help
@@ -35,23 +35,16 @@ config SND_SAMSUNG_I2S
 	tristate
 
 config SND_SOC_SAMSUNG_NEO1973_WM8753
-	tristate "SoC I2S Audio support for NEO1973 - WM8753"
-	depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA01
+	tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)"
+	depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02)
 	select SND_S3C24XX_I2S
 	select SND_SOC_WM8753
+	select SND_SOC_LM4857 if MACH_NEO1973_GTA01
+	select SND_SOC_DFBMCS320
 	help
-	  Say Y if you want to add support for SoC audio on smdk2440
-	  with the WM8753.
+	  Say Y here to enable audio support for the Openmoko Neo1973
+	  Smartphones.
 
-config SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753
-	tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
-	depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02
-	select SND_S3C24XX_I2S
-	select SND_SOC_WM8753
-	help
-	  This driver provides audio support for the Openmoko Neo FreeRunner
-	  smartphone.
-	  
 config SND_SOC_SAMSUNG_JIVE_WM8750
 	tristate "SoC I2S Audio support for Jive"
 	depends on SND_SOC_SAMSUNG && MACH_JIVE
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 705d4e8a672..294dec05c26 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -20,7 +20,6 @@ obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o
 # S3C24XX Machine Support
 snd-soc-jive-wm8750-objs := jive_wm8750.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
-snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
 snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
@@ -38,7 +37,6 @@ snd-soc-smdk-spdif-objs := smdk_spdif.o
 
 obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
-obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 4770a955034..f97110e72e8 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -12,24 +12,24 @@
  * published by the Free Software Foundation.
  */
 
-#include <linux/init.h>
-#include <linux/module.h>
 #include <linux/io.h>
 #include <linux/delay.h>
 #include <linux/clk.h>
 
 #include <sound/soc.h>
 
-#include <plat/regs-ac97.h>
 #include <mach/dma.h>
+#include <plat/regs-ac97.h>
 #include <plat/audio.h>
 
 #include "dma.h"
-#include "ac97.h"
 
 #define AC_CMD_ADDR(x) (x << 16)
 #define AC_CMD_DATA(x) (x & 0xffff)
 
+#define S3C_AC97_DAI_PCM 0
+#define S3C_AC97_DAI_MIC 1
+
 struct s3c_ac97_info {
 	struct clk         *ac97_clk;
 	void __iomem	   *regs;
diff --git a/sound/soc/samsung/ac97.h b/sound/soc/samsung/ac97.h
deleted file mode 100644
index 0d0e1b51145..00000000000
--- a/sound/soc/samsung/ac97.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/* sound/soc/samsung/ac97.h
- *
- * ALSA SoC Audio Layer - S3C AC97 Controller driver
- *	Evolved from s3c2443-ac97.h
- *
- * Copyright (c) 2010 Samsung Electronics Co. Ltd
- *	Author: Jaswinder Singh <jassi.brar@samsung.com>
- *	Credits: Graeme Gregory, Sean Choi
- *
- * 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 __S3C_AC97_H_
-#define __S3C_AC97_H_
-
-#define S3C_AC97_DAI_PCM 0
-#define S3C_AC97_DAI_MIC 1
-
-#endif /* __S3C_AC97_H_ */
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 21240198c5d..5cb3b880f0d 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -14,17 +14,11 @@
  *  option) any later version.
  */
 
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/io.h>
-#include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
 
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/pcm_params.h>
 
 #include <asm/dma.h>
 #include <mach/hardware.h>
@@ -32,6 +26,9 @@
 
 #include "dma.h"
 
+#define ST_RUNNING		(1<<0)
+#define ST_OPENED		(1<<1)
+
 static const struct snd_pcm_hardware dma_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
 				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -313,7 +310,7 @@ dma_pointer(struct snd_pcm_substream *substream)
 	/* we seem to be getting the odd error from the pcm library due
 	 * to out-of-bounds pointers. this is maybe due to the dma engine
 	 * not having loaded the new values for the channel before being
-	 * callled... (todo - fix )
+	 * called... (todo - fix )
 	 */
 
 	if (res >= snd_pcm_lib_buffer_bytes(substream)) {
diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h
index f8cd2b4223a..c50659269a4 100644
--- a/sound/soc/samsung/dma.h
+++ b/sound/soc/samsung/dma.h
@@ -12,9 +12,6 @@
 #ifndef _S3C_AUDIO_H
 #define _S3C_AUDIO_H
 
-#define ST_RUNNING		(1<<0)
-#define ST_OPENED		(1<<1)
-
 struct s3c_dma_params {
 	struct s3c2410_dma_client *client;	/* stream identifier */
 	int channel;				/* Channel ID */
@@ -22,9 +19,4 @@ struct s3c_dma_params {
 	int dma_size;			/* Size of the DMA transfer */
 };
 
-#define S3C24XX_DAI_I2S			0
-
-/* platform data */
-extern struct snd_ac97_bus_ops s3c24xx_ac97_ops;
-
 #endif
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index 34dd9ef1b9c..f6b3a3ce591 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -11,21 +11,13 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/io.h>
-#include <linux/platform_device.h>
 #include <sound/soc.h>
 #include <sound/jack.h>
+
 #include <asm/mach-types.h>
 #include <mach/gpio.h>
-#include <mach/regs-clock.h>
 
-#include <linux/mfd/wm8994/core.h>
-#include <linux/mfd/wm8994/registers.h>
 #include "../codecs/wm8994.h"
-#include "dma.h"
-#include "i2s.h"
 
 #define MACHINE_NAME	0
 #define CPU_VOICE_DAI	1
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index c45f7ce14d6..241f55d0066 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,25 +13,16 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/platform_device.h>
-#include <linux/i2c.h>
 #include <linux/gpio.h>
 
 #include <sound/soc.h>
-#include <sound/uda1380.h>
 #include <sound/jack.h>
 
 #include <plat/regs-iis.h>
-
 #include <mach/h1940-latch.h>
-
 #include <asm/mach-types.h>
 
-#include "dma.h"
 #include "s3c24xx-i2s.h"
-#include "../codecs/uda1380.h"
 
 static unsigned int rates[] = {
 	11025,
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index d00ac3a7102..ffa09b3b2ca 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -15,9 +15,8 @@
 #include <linux/clk.h>
 #include <linux/io.h>
 
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/pcm_params.h>
 
 #include <plat/audio.h>
 
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 08802520e01..3b53ad54bc3 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -11,22 +11,11 @@
  * published by the Free Software Foundation.
 */
 
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/clk.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
 #include <sound/soc.h>
 
 #include <asm/mach-types.h>
 
-#include "dma.h"
 #include "s3c2412-i2s.h"
-
 #include "../codecs/wm8750.h"
 
 static const struct snd_soc_dapm_route audio_map[] = {
diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h
deleted file mode 100644
index 0cf5b7011d6..00000000000
--- a/sound/soc/samsung/lm4857.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * lm4857.h  --  ALSA Soc Audio Layer
- *
- * Copyright 2007 Wolfson Microelectronics PLC.
- * Author: Graeme Gregory
- *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
- *
- *  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.
- *
- *  Revision history
- *    18th Jun 2007   Initial version.
- */
-
-#ifndef LM4857_H_
-#define LM4857_H_
-
-/* The register offsets in the cache array */
-#define LM4857_MVOL 0
-#define LM4857_LVOL 1
-#define LM4857_RVOL 2
-#define LM4857_CTRL 3
-
-/* the shifts required to set these bits */
-#define LM4857_3D 5
-#define LM4857_WAKEUP 5
-#define LM4857_EPGAIN 4
-
-#endif /*LM4857_H_*/
-
diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c
index a2bb34def74..bd91c19a6c0 100644
--- a/sound/soc/samsung/ln2440sbc_alc650.c
+++ b/sound/soc/samsung/ln2440sbc_alc650.c
@@ -16,15 +16,8 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
 #include <sound/soc.h>
 
-#include "dma.h"
-#include "ac97.h"
-
 static struct snd_soc_card ln2440sbc;
 
 static struct snd_soc_dai_link ln2440sbc_dai[] = {
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
deleted file mode 100644
index 3eec610c10f..00000000000
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ /dev/null
@@ -1,504 +0,0 @@
-/*
- * neo1973_gta02_wm8753.c  --  SoC audio for Openmoko Freerunner(GTA02)
- *
- * Copyright 2007 Openmoko Inc
- * Author: Graeme Gregory <graeme@openmoko.org>
- * Copyright 2007 Wolfson Microelectronics PLC.
- * Author: Graeme Gregory <linux@wolfsonmicro.com>
- * Copyright 2009 Wolfson Microelectronics
- *
- *  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/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/gpio.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-
-#include <plat/regs-iis.h>
-
-#include <mach/regs-clock.h>
-#include <asm/io.h>
-#include <mach/gta02.h>
-#include "../codecs/wm8753.h"
-#include "dma.h"
-#include "s3c24xx-i2s.h"
-
-static struct snd_soc_card neo1973_gta02;
-
-static int neo1973_gta02_hifi_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;
-	unsigned int pll_out = 0, bclk = 0;
-	int ret = 0;
-	unsigned long iis_clkrate;
-
-	iis_clkrate = s3c24xx_i2s_get_clockrate();
-
-	switch (params_rate(params)) {
-	case 8000:
-	case 16000:
-		pll_out = 12288000;
-		break;
-	case 48000:
-		bclk = WM8753_BCLK_DIV_4;
-		pll_out = 12288000;
-		break;
-	case 96000:
-		bclk = WM8753_BCLK_DIV_2;
-		pll_out = 12288000;
-		break;
-	case 11025:
-		bclk = WM8753_BCLK_DIV_16;
-		pll_out = 11289600;
-		break;
-	case 22050:
-		bclk = WM8753_BCLK_DIV_8;
-		pll_out = 11289600;
-		break;
-	case 44100:
-		bclk = WM8753_BCLK_DIV_4;
-		pll_out = 11289600;
-		break;
-	case 88200:
-		bclk = WM8753_BCLK_DIV_2;
-		pll_out = 11289600;
-		break;
-	}
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai,
-		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-		SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-		SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
-		SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* set MCLK division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
-		S3C2410_IISMOD_32FS);
-	if (ret < 0)
-		return ret;
-
-	/* set codec BCLK division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(codec_dai,
-					WM8753_BCLKDIV, bclk);
-	if (ret < 0)
-		return ret;
-
-	/* set prescaler division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
-		S3C24XX_PRESCALE(4, 4));
-	if (ret < 0)
-		return ret;
-
-	/* codec PLL input is PCLK/4 */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
-		iis_clkrate / 4, pll_out);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
-	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
-}
-
-/*
- * Neo1973 WM8753 HiFi DAI opserations.
- */
-static struct snd_soc_ops neo1973_gta02_hifi_ops = {
-	.hw_params = neo1973_gta02_hifi_hw_params,
-	.hw_free = neo1973_gta02_hifi_hw_free,
-};
-
-static int neo1973_gta02_voice_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;
-	unsigned int pcmdiv = 0;
-	int ret = 0;
-	unsigned long iis_clkrate;
-
-	iis_clkrate = s3c24xx_i2s_get_clockrate();
-
-	if (params_rate(params) != 8000)
-		return -EINVAL;
-	if (params_channels(params) != 1)
-		return -EINVAL;
-
-	pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
-
-	/* todo: gg check mode (DSP_B) against CSR datasheet */
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
-		12288000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* set codec PCM division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV,
-					pcmdiv);
-	if (ret < 0)
-		return ret;
-
-	/* configure and enable PLL for 12.288MHz output */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
-		iis_clkrate / 4, 12288000);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
-	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
-}
-
-static struct snd_soc_ops neo1973_gta02_voice_ops = {
-	.hw_params = neo1973_gta02_voice_hw_params,
-	.hw_free = neo1973_gta02_voice_hw_free,
-};
-
-#define LM4853_AMP 1
-#define LM4853_SPK 2
-
-static u8 lm4853_state;
-
-/* This has no effect, it exists only to maintain compatibility with
- * existing ALSA state files.
- */
-static int lm4853_set_state(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	int val = ucontrol->value.integer.value[0];
-
-	if (val)
-		lm4853_state |= LM4853_AMP;
-	else
-		lm4853_state &= ~LM4853_AMP;
-
-	return 0;
-}
-
-static int lm4853_get_state(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
-
-	return 0;
-}
-
-static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	int val = ucontrol->value.integer.value[0];
-
-	if (val) {
-		lm4853_state |= LM4853_SPK;
-		gpio_set_value(GTA02_GPIO_HP_IN, 0);
-	} else {
-		lm4853_state &= ~LM4853_SPK;
-		gpio_set_value(GTA02_GPIO_HP_IN, 1);
-	}
-
-	return 0;
-}
-
-static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
-
-	return 0;
-}
-
-static int lm4853_event(struct snd_soc_dapm_widget *w,
-			struct snd_kcontrol *k,
-			int event)
-{
-	gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event));
-
-	return 0;
-}
-
-static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
-	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
-	SND_SOC_DAPM_LINE("GSM Line In", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Handset Mic", NULL),
-	SND_SOC_DAPM_SPK("Handset Spk", NULL),
-};
-
-
-/* example machine audio_mapnections */
-static const struct snd_soc_dapm_route audio_map[] = {
-
-	/* Connections to the lm4853 amp */
-	{"Stereo Out", NULL, "LOUT1"},
-	{"Stereo Out", NULL, "ROUT1"},
-
-	/* Connections to the GSM Module */
-	{"GSM Line Out", NULL, "MONO1"},
-	{"GSM Line Out", NULL, "MONO2"},
-	{"RXP", NULL, "GSM Line In"},
-	{"RXN", NULL, "GSM Line In"},
-
-	/* Connections to Headset */
-	{"MIC1", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Headset Mic"},
-
-	/* Call Mic */
-	{"MIC2", NULL, "Mic Bias"},
-	{"MIC2N", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Handset Mic"},
-
-	/* Call Speaker */
-	{"Handset Spk", NULL, "LOUT2"},
-	{"Handset Spk", NULL, "ROUT2"},
-
-	/* Connect the ALC pins */
-	{"ACIN", NULL, "ACOP"},
-};
-
-static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
-	SOC_DAPM_PIN_SWITCH("Stereo Out"),
-	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
-	SOC_DAPM_PIN_SWITCH("GSM Line In"),
-	SOC_DAPM_PIN_SWITCH("Headset Mic"),
-	SOC_DAPM_PIN_SWITCH("Handset Mic"),
-	SOC_DAPM_PIN_SWITCH("Handset Spk"),
-
-	/* This has no effect, it exists only to maintain compatibility with
-	 * existing ALSA state files.
-	 */
-	SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
-		lm4853_get_state,
-		lm4853_set_state),
-	SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
-		lm4853_get_spk,
-		lm4853_set_spk),
-};
-
-/*
- * This is an example machine initialisation for a wm8753 connected to a
- * neo1973 GTA02.
- */
-static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
-
-	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(dapm, "OUT3");
-	snd_soc_dapm_nc_pin(dapm, "OUT4");
-	snd_soc_dapm_nc_pin(dapm, "LINE1");
-	snd_soc_dapm_nc_pin(dapm, "LINE2");
-
-	/* Add neo1973 gta02 specific widgets */
-	snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
-				  ARRAY_SIZE(wm8753_dapm_widgets));
-
-	/* add neo1973 gta02 specific controls */
-	err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls,
-		ARRAY_SIZE(wm8753_neo1973_gta02_controls));
-
-	if (err < 0)
-		return err;
-
-	/* set up neo1973 gta02 specific audio path audio_map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
-	/* set endpoints to default off mode */
-	snd_soc_dapm_disable_pin(dapm, "Stereo Out");
-	snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-	snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-	snd_soc_dapm_disable_pin(dapm, "Handset Mic");
-	snd_soc_dapm_disable_pin(dapm, "Handset Spk");
-
-	/* allow audio paths from the GSM modem to run during suspend */
-	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
-	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
-
-	snd_soc_dapm_sync(dapm);
-
-	return 0;
-}
-
-/*
- * BT Codec DAI
- */
-static struct snd_soc_dai_driver bt_dai = {
-	.name = "bluetooth-dai",
-	.playback = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-};
-
-static struct snd_soc_dai_link neo1973_gta02_dai[] = {
-{ /* Hifi Playback - for similatious use with voice below */
-	.name = "WM8753",
-	.stream_name = "WM8753 HiFi",
-	.cpu_dai_name = "s3c24xx-i2s",
-	.codec_dai_name = "wm8753-hifi",
-	.init = neo1973_gta02_wm8753_init,
-	.platform_name = "samsung-audio",
-	.codec_name = "wm8753-codec.0-0x1a",
-	.ops = &neo1973_gta02_hifi_ops,
-},
-{ /* Voice via BT */
-	.name = "Bluetooth",
-	.stream_name = "Voice",
-	.cpu_dai_name = "bluetooth-dai",
-	.codec_dai_name = "wm8753-voice",
-	.ops = &neo1973_gta02_voice_ops,
-	.codec_name = "wm8753-codec.0-0x1a",
-	.platform_name = "samsung-audio",
-},
-};
-
-static struct snd_soc_card neo1973_gta02 = {
-	.name = "neo1973-gta02",
-	.dai_link = neo1973_gta02_dai,
-	.num_links = ARRAY_SIZE(neo1973_gta02_dai),
-};
-
-static struct platform_device *neo1973_gta02_snd_device;
-
-static int __init neo1973_gta02_init(void)
-{
-	int ret;
-
-	if (!machine_is_neo1973_gta02()) {
-		printk(KERN_INFO
-		       "Only GTA02 is supported by this ASoC driver\n");
-		return -ENODEV;
-	}
-
-	neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!neo1973_gta02_snd_device)
-		return -ENOMEM;
-
-	/* register bluetooth DAI here */
-	ret = snd_soc_register_dai(&neo1973_gta02_snd_device->dev, &bt_dai);
-	if (ret)
-		goto err_put_device;
-
-	platform_set_drvdata(neo1973_gta02_snd_device, &neo1973_gta02);
-	ret = platform_device_add(neo1973_gta02_snd_device);
-
-	if (ret)
-		goto err_unregister_dai;
-
-	/* Initialise GPIOs used by amp */
-	ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN");
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN);
-		goto err_del_device;
-	}
-
-	ret = gpio_direction_output(GTA02_GPIO_HP_IN, 1);
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN);
-		goto err_free_gpio_hp_in;
-	}
-
-	ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT");
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT);
-		goto err_free_gpio_hp_in;
-	}
-
-	ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1);
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT);
-		goto err_free_gpio_amp_shut;
-	}
-
-	return 0;
-
-err_free_gpio_amp_shut:
-	gpio_free(GTA02_GPIO_AMP_SHUT);
-err_free_gpio_hp_in:
-	gpio_free(GTA02_GPIO_HP_IN);
-err_del_device:
-	platform_device_del(neo1973_gta02_snd_device);
-err_unregister_dai:
-	snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev);
-err_put_device:
-	platform_device_put(neo1973_gta02_snd_device);
-	return ret;
-}
-module_init(neo1973_gta02_init);
-
-static void __exit neo1973_gta02_exit(void)
-{
-	snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev);
-	platform_device_unregister(neo1973_gta02_snd_device);
-	gpio_free(GTA02_GPIO_HP_IN);
-	gpio_free(GTA02_GPIO_AMP_SHUT);
-}
-module_exit(neo1973_gta02_exit);
-
-/* Module information */
-MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
-MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a24514beb..78bfdb3f5d7 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -1,57 +1,32 @@
 /*
- * neo1973_wm8753.c  --  SoC audio for Neo1973
+ * neo1973_wm8753.c  --  SoC audio for Openmoko Neo1973 and Freerunner devices
  *
+ * Copyright 2007 Openmoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
  * Copyright 2007 Wolfson Microelectronics PLC.
  * Author: Graeme Gregory
  *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Copyright 2009 Wolfson Microelectronics
  *
  *  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/timer.h>
-#include <linux/interrupt.h>
 #include <linux/platform_device.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
+#include <linux/gpio.h>
+
 #include <sound/soc.h>
-#include <sound/tlv.h>
 
 #include <asm/mach-types.h>
-#include <asm/hardware/scoop.h>
-#include <mach/regs-clock.h>
-#include <mach/regs-gpio.h>
-#include <mach/hardware.h>
-#include <linux/io.h>
-#include <mach/spi-gpio.h>
-
 #include <plat/regs-iis.h>
+#include <mach/gta02.h>
 
 #include "../codecs/wm8753.h"
-#include "lm4857.h"
-#include "dma.h"
 #include "s3c24xx-i2s.h"
 
-/* define the scenarios */
-#define NEO_AUDIO_OFF			0
-#define NEO_GSM_CALL_AUDIO_HANDSET	1
-#define NEO_GSM_CALL_AUDIO_HEADSET	2
-#define NEO_GSM_CALL_AUDIO_BLUETOOTH	3
-#define NEO_STEREO_TO_SPEAKERS		4
-#define NEO_STEREO_TO_HEADPHONES	5
-#define NEO_CAPTURE_HANDSET		6
-#define NEO_CAPTURE_HEADSET		7
-#define NEO_CAPTURE_BLUETOOTH		8
-
-static struct snd_soc_card neo1973;
-static struct i2c_client *i2c;
-
 static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
@@ -62,8 +37,6 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	pr_debug("Entered %s\n", __func__);
-
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	switch (params_rate(params)) {
@@ -148,8 +121,6 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 
-	pr_debug("Entered %s\n", __func__);
-
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
 }
@@ -171,8 +142,6 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	pr_debug("Entered %s\n", __func__);
-
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	if (params_rate(params) != 8000)
@@ -214,8 +183,6 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 
-	pr_debug("Entered %s\n", __func__);
-
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
 }
@@ -225,343 +192,240 @@ static struct snd_soc_ops neo1973_voice_ops = {
 	.hw_free = neo1973_voice_hw_free,
 };
 
-static int neo1973_scenario;
-
-static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = neo1973_scenario;
-	return 0;
-}
-
-static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	pr_debug("Entered %s\n", __func__);
-
-	switch (neo1973_scenario) {
-	case NEO_AUDIO_OFF:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_GSM_CALL_AUDIO_HANDSET:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_enable_pin(dapm, "Call Mic");
-		break;
-	case NEO_GSM_CALL_AUDIO_HEADSET:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_GSM_CALL_AUDIO_BLUETOOTH:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_STEREO_TO_SPEAKERS:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_STEREO_TO_HEADPHONES:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_CAPTURE_HANDSET:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_enable_pin(dapm, "Call Mic");
-		break;
-	case NEO_CAPTURE_HEADSET:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_CAPTURE_BLUETOOTH:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	default:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-	}
+/* Shared routes and controls */
 
-	snd_soc_dapm_sync(dapm);
+static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line In", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Handset Mic", NULL),
+};
 
-	return 0;
-}
+static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
+	/* Connections to the GSM Module */
+	{"GSM Line Out", NULL, "MONO1"},
+	{"GSM Line Out", NULL, "MONO2"},
+	{"RXP", NULL, "GSM Line In"},
+	{"RXN", NULL, "GSM Line In"},
 
-static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	/* Connections to Headset */
+	{"MIC1", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Headset Mic"},
 
-	pr_debug("Entered %s\n", __func__);
+	/* Call Mic */
+	{"MIC2", NULL, "Mic Bias"},
+	{"MIC2N", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Handset Mic"},
 
-	if (neo1973_scenario == ucontrol->value.integer.value[0])
-		return 0;
+	/* Connect the ALC pins */
+	{"ACIN", NULL, "ACOP"},
+};
 
-	neo1973_scenario = ucontrol->value.integer.value[0];
-	set_scenario_endpoints(codec, neo1973_scenario);
-	return 1;
-}
+static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
+	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+	SOC_DAPM_PIN_SWITCH("GSM Line In"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Handset Mic"),
+};
 
-static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
+/* GTA02 specific routes and controlls */
 
-static void lm4857_write_regs(void)
-{
-	pr_debug("Entered %s\n", __func__);
+#ifdef CONFIG_MACH_NEO1973_GTA02
 
-	if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
-		printk(KERN_ERR "lm4857: i2c write failed\n");
-}
+static int gta02_speaker_enabled;
 
-static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
+static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	int reg = mc->reg;
-	int shift = mc->shift;
-	int mask = mc->max;
+	gta02_speaker_enabled = ucontrol->value.integer.value[0];
 
-	pr_debug("Entered %s\n", __func__);
+	gpio_set_value(GTA02_GPIO_HP_IN, !gta02_speaker_enabled);
 
-	ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
 	return 0;
 }
 
-static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
+static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	int reg = mc->reg;
-	int shift = mc->shift;
-	int mask = mc->max;
-
-	if (((lm4857_regs[reg] >> shift) & mask) ==
-		ucontrol->value.integer.value[0])
-		return 0;
-
-	lm4857_regs[reg] &= ~(mask << shift);
-	lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
-	lm4857_write_regs();
-	return 1;
+	ucontrol->value.integer.value[0] = gta02_speaker_enabled;
+	return 0;
 }
 
-static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
+static int lm4853_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
 {
-	u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
-
-	pr_debug("Entered %s\n", __func__);
-
-	if (value)
-		value -= 5;
+	gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event));
 
-	ucontrol->value.integer.value[0] = value;
 	return 0;
 }
 
-static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	u8 value = ucontrol->value.integer.value[0];
-
-	pr_debug("Entered %s\n", __func__);
-
-	if (value)
-		value += 5;
-
-	if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
-		return 0;
-
-	lm4857_regs[LM4857_CTRL] &= 0xF0;
-	lm4857_regs[LM4857_CTRL] |= value;
-	lm4857_write_regs();
-	return 1;
-}
+static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
+	/* Connections to the amp */
+	{"Stereo Out", NULL, "LOUT1"},
+	{"Stereo Out", NULL, "ROUT1"},
 
-static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_LINE("Audio Out", NULL),
-	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
-	SND_SOC_DAPM_LINE("GSM Line In", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Call Mic", NULL),
+	/* Call Speaker */
+	{"Handset Spk", NULL, "LOUT2"},
+	{"Handset Spk", NULL, "ROUT2"},
 };
 
+static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Handset Spk"),
+	SOC_DAPM_PIN_SWITCH("Stereo Out"),
 
-static const struct snd_soc_dapm_route dapm_routes[] = {
-
-	/* Connections to the lm4857 amp */
-	{"Audio Out", NULL, "LOUT1"},
-	{"Audio Out", NULL, "ROUT1"},
-
-	/* Connections to the GSM Module */
-	{"GSM Line Out", NULL, "MONO1"},
-	{"GSM Line Out", NULL, "MONO2"},
-	{"RXP", NULL, "GSM Line In"},
-	{"RXN", NULL, "GSM Line In"},
+	SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0,
+		lm4853_get_spk,
+		lm4853_set_spk),
+};
 
-	/* Connections to Headset */
-	{"MIC1", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Headset Mic"},
+static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+};
 
-	/* Call Mic */
-	{"MIC2", NULL, "Mic Bias"},
-	{"MIC2N", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Call Mic"},
+static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
 
-	/* Connect the ALC pins */
-	{"ACIN", NULL, "ACOP"},
-};
+	ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets,
+			ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets));
+	if (ret)
+		return ret;
 
-static const char *lm4857_mode[] = {
-	"Off",
-	"Call Speaker",
-	"Stereo Speakers",
-	"Stereo Speakers + Headphones",
-	"Headphones"
-};
+	ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes,
+			ARRAY_SIZE(neo1973_gta02_routes));
+	if (ret)
+		return ret;
 
-static const struct soc_enum lm4857_mode_enum[] = {
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
-};
+	ret = snd_soc_add_controls(codec, neo1973_gta02_wm8753_controls,
+			ARRAY_SIZE(neo1973_gta02_wm8753_controls));
+	if (ret)
+		return ret;
 
-static const char *neo_scenarios[] = {
-	"Off",
-	"GSM Handset",
-	"GSM Headset",
-	"GSM Bluetooth",
-	"Speakers",
-	"Headphones",
-	"Capture Handset",
-	"Capture Headset",
-	"Capture Bluetooth"
-};
+	snd_soc_dapm_disable_pin(dapm, "Stereo Out");
+	snd_soc_dapm_disable_pin(dapm, "Handset Spk");
+	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
 
-static const struct soc_enum neo_scenario_enum[] = {
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios),
-};
+	return 0;
+}
 
-static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
-static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
-
-static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
-	SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg, stereo_tlv),
-	SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg, stereo_tlv),
-	SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg, mono_tlv),
-	SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
-		lm4857_get_mode, lm4857_set_mode),
-	SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
-		neo1973_get_scenario, neo1973_set_scenario),
-	SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-};
+#else
+static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; }
+#endif
 
-/*
- * This is an example machine initialisation for a wm8753 connected to a
- * neo1973 II. It is missing logic to detect hp/mic insertions and logic
- * to re-route the audio in such an event.
- */
 static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
-
-	pr_debug("Entered %s\n", __func__);
+	int ret;
 
 	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(dapm, "LOUT2");
-	snd_soc_dapm_nc_pin(dapm, "ROUT2");
+	if (machine_is_neo1973_gta01()) {
+		snd_soc_dapm_nc_pin(dapm, "LOUT2");
+		snd_soc_dapm_nc_pin(dapm, "ROUT2");
+	}
 	snd_soc_dapm_nc_pin(dapm, "OUT3");
 	snd_soc_dapm_nc_pin(dapm, "OUT4");
 	snd_soc_dapm_nc_pin(dapm, "LINE1");
 	snd_soc_dapm_nc_pin(dapm, "LINE2");
 
 	/* Add neo1973 specific widgets */
-	snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
-				  ARRAY_SIZE(wm8753_dapm_widgets));
-
-	/* set endpoints to default mode */
-	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+	ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets,
+			ARRAY_SIZE(neo1973_wm8753_dapm_widgets));
+	if (ret)
+		return ret;
 
 	/* add neo1973 specific controls */
-	err = snd_soc_add_controls(codec, wm8753_neo1973_controls,
-				ARRAY_SIZE(8753_neo1973_controls));
-	if (err < 0)
-		return err;
+	ret = snd_soc_add_controls(codec, neo1973_wm8753_controls,
+			ARRAY_SIZE(neo1973_wm8753_controls));
+	if (ret)
+		return ret;
 
 	/* set up neo1973 specific audio routes */
-	err = snd_soc_dapm_add_routes(dapm, dapm_routes,
-				      ARRAY_SIZE(dapm_routes));
+	ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes,
+			ARRAY_SIZE(neo1973_wm8753_routes));
+	if (ret)
+		return ret;
+
+	/* set endpoints to default off mode */
+	snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+	snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+
+	/* allow audio paths from the GSM modem to run during suspend */
+	snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
+	snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
+	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
+
+	if (machine_is_neo1973_gta02()) {
+		ret = neo1973_gta02_wm8753_init(codec);
+		if (ret)
+			return ret;
+	}
 
 	snd_soc_dapm_sync(dapm);
+
 	return 0;
 }
 
-/*
- * BT Codec DAI
- */
-static struct snd_soc_dai bt_dai = {
-	.name = "bluetooth-dai",
-	.playback = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+/* GTA01 specific controlls */
+
+#ifdef CONFIG_MACH_NEO1973_GTA01
+
+static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = {
+	{"Amp IN", NULL, "ROUT1"},
+	{"Amp IN", NULL, "LOUT1"},
+
+	{"Handset Spk", NULL, "Amp EP"},
+	{"Stereo Out", NULL, "Amp LS"},
+	{"Headphone", NULL, "Amp HP"},
+};
+
+static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+	SND_SOC_DAPM_SPK("Stereo Out", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
 };
 
+static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets,
+			ARRAY_SIZE(neo1973_lm4857_dapm_widgets));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes,
+			ARRAY_SIZE(neo1973_lm4857_routes));
+	if (ret)
+		return ret;
+
+	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
+	snd_soc_dapm_ignore_suspend(dapm, "Headphone");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+#else
+static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; };
+#endif
+
 static struct snd_soc_dai_link neo1973_dai[] = {
 { /* Hifi Playback - for similatious use with voice below */
 	.name = "WM8753",
 	.stream_name = "WM8753 HiFi",
 	.platform_name = "samsung-audio",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.codec_dai_name = "wm8753-hifi",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.init = neo1973_wm8753_init,
 	.ops = &neo1973_hifi_ops,
 },
@@ -569,90 +433,49 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.name = "Bluetooth",
 	.stream_name = "Voice",
 	.platform_name = "samsung-audio",
-	.cpu_dai_name = "bluetooth-dai",
+	.cpu_dai_name = "dfbmcs320-pcm",
 	.codec_dai_name = "wm8753-voice",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.ops = &neo1973_voice_ops,
 },
 };
 
-static struct snd_soc_card neo1973 = {
-	.name = "neo1973",
-	.dai_link = neo1973_dai,
-	.num_links = ARRAY_SIZE(neo1973_dai),
+static struct snd_soc_aux_dev neo1973_aux_devs[] = {
+	{
+		.name = "dfbmcs320",
+		.codec_name = "dfbmcs320.0",
+	},
+	{
+		.name = "lm4857",
+		.codec_name = "lm4857.0-007c",
+		.init = neo1973_lm4857_init,
+	},
 };
 
-static int lm4857_i2c_probe(struct i2c_client *client,
-			    const struct i2c_device_id *id)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	i2c = client;
-
-	lm4857_write_regs();
-	return 0;
-}
-
-static int lm4857_i2c_remove(struct i2c_client *client)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	i2c = NULL;
-
-	return 0;
-}
-
-static u8 lm4857_state;
-
-static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	dev_dbg(&dev->dev, "lm4857_suspend\n");
-	lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
-	if (lm4857_state) {
-		lm4857_regs[LM4857_CTRL] &= 0xf0;
-		lm4857_write_regs();
-	}
-	return 0;
-}
-
-static int lm4857_resume(struct i2c_client *dev)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	if (lm4857_state) {
-		lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
-		lm4857_write_regs();
-	}
-	return 0;
-}
-
-static void lm4857_shutdown(struct i2c_client *dev)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	dev_dbg(&dev->dev, "lm4857_shutdown\n");
-	lm4857_regs[LM4857_CTRL] &= 0xf0;
-	lm4857_write_regs();
-}
+static struct snd_soc_codec_conf neo1973_codec_conf[] = {
+	{
+		.dev_name = "lm4857.0-007c",
+		.name_prefix = "Amp",
+	},
+};
 
-static const struct i2c_device_id lm4857_i2c_id[] = {
-	{ "neo1973_lm4857", 0 },
-	{ }
+#ifdef CONFIG_MACH_NEO1973_GTA02
+static const struct gpio neo1973_gta02_gpios[] = {
+	{ GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" },
+	{ GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" },
 };
+#else
+static const struct gpio neo1973_gta02_gpios[] = {};
+#endif
 
-static struct i2c_driver lm4857_i2c_driver = {
-	.driver = {
-		.name = "LM4857 I2C Amp",
-		.owner = THIS_MODULE,
-	},
-	.suspend =        lm4857_suspend,
-	.resume	=         lm4857_resume,
-	.shutdown =       lm4857_shutdown,
-	.probe =          lm4857_i2c_probe,
-	.remove =         lm4857_i2c_remove,
-	.id_table =       lm4857_i2c_id,
+static struct snd_soc_card neo1973 = {
+	.name = "neo1973",
+	.dai_link = neo1973_dai,
+	.num_links = ARRAY_SIZE(neo1973_dai),
+	.aux_dev = neo1973_aux_devs,
+	.num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
+	.codec_conf = neo1973_codec_conf,
+	.num_configs = ARRAY_SIZE(neo1973_codec_conf),
 };
 
 static struct platform_device *neo1973_snd_device;
@@ -661,46 +484,56 @@ static int __init neo1973_init(void)
 {
 	int ret;
 
-	pr_debug("Entered %s\n", __func__);
-
-	if (!machine_is_neo1973_gta01()) {
-		printk(KERN_INFO
-			"Only GTA01 hardware supported by ASoC driver\n");
+	if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02())
 		return -ENODEV;
+
+	if (machine_is_neo1973_gta02()) {
+		neo1973.name = "neo1973gta02";
+		neo1973.num_aux_devs = 1;
+
+		ret = gpio_request_array(neo1973_gta02_gpios,
+				ARRAY_SIZE(neo1973_gta02_gpios));
+		if (ret)
+			return ret;
 	}
 
 	neo1973_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!neo1973_snd_device)
-		return -ENOMEM;
+	if (!neo1973_snd_device) {
+		ret = -ENOMEM;
+		goto err_gpio_free;
+	}
 
 	platform_set_drvdata(neo1973_snd_device, &neo1973);
 	ret = platform_device_add(neo1973_snd_device);
 
-	if (ret) {
-		platform_device_put(neo1973_snd_device);
-		return ret;
-	}
-
-	ret = i2c_add_driver(&lm4857_i2c_driver);
+	if (ret)
+		goto err_put_device;
 
-	if (ret != 0)
-		platform_device_unregister(neo1973_snd_device);
+	return 0;
 
+err_put_device:
+	platform_device_put(neo1973_snd_device);
+err_gpio_free:
+	if (machine_is_neo1973_gta02()) {
+		gpio_free_array(neo1973_gta02_gpios,
+				ARRAY_SIZE(neo1973_gta02_gpios));
+	}
 	return ret;
 }
+module_init(neo1973_init);
 
 static void __exit neo1973_exit(void)
 {
-	pr_debug("Entered %s\n", __func__);
-
-	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
-}
 
-module_init(neo1973_init);
+	if (machine_is_neo1973_gta02()) {
+		gpio_free_array(neo1973_gta02_gpios,
+				ARRAY_SIZE(neo1973_gta02_gpios));
+	}
+}
 module_exit(neo1973_exit);
 
 /* Module information */
 MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org");
-MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 48d0b750406..38aac7d57a5 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -11,20 +11,11 @@
  * published by the Free Software Foundation.
  */
 
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/device.h>
-#include <linux/delay.h>
 #include <linux/clk.h>
-#include <linux/kernel.h>
-#include <linux/gpio.h>
 #include <linux/io.h>
 
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
 #include <sound/soc.h>
+#include <sound/pcm_params.h>
 
 #include <plat/audio.h>
 #include <plat/dma.h>
@@ -32,6 +23,113 @@
 #include "dma.h"
 #include "pcm.h"
 
+/*Register Offsets */
+#define S3C_PCM_CTL		0x00
+#define S3C_PCM_CLKCTL		0x04
+#define S3C_PCM_TXFIFO		0x08
+#define S3C_PCM_RXFIFO		0x0C
+#define S3C_PCM_IRQCTL		0x10
+#define S3C_PCM_IRQSTAT		0x14
+#define S3C_PCM_FIFOSTAT	0x18
+#define S3C_PCM_CLRINT		0x20
+
+/* PCM_CTL Bit-Fields */
+#define S3C_PCM_CTL_TXDIPSTICK_MASK	0x3f
+#define S3C_PCM_CTL_TXDIPSTICK_SHIFT	13
+#define S3C_PCM_CTL_RXDIPSTICK_MASK	0x3f
+#define S3C_PCM_CTL_RXDIPSTICK_SHIFT	7
+#define S3C_PCM_CTL_TXDMA_EN		(0x1 << 6)
+#define S3C_PCM_CTL_RXDMA_EN		(0x1 << 5)
+#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC	(0x1 << 4)
+#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC	(0x1 << 3)
+#define S3C_PCM_CTL_TXFIFO_EN		(0x1 << 2)
+#define S3C_PCM_CTL_RXFIFO_EN		(0x1 << 1)
+#define S3C_PCM_CTL_ENABLE		(0x1 << 0)
+
+/* PCM_CLKCTL Bit-Fields */
+#define S3C_PCM_CLKCTL_SERCLK_EN	(0x1 << 19)
+#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK	(0x1 << 18)
+#define S3C_PCM_CLKCTL_SCLKDIV_MASK	0x1ff
+#define S3C_PCM_CLKCTL_SYNCDIV_MASK	0x1ff
+#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT	9
+#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT	0
+
+/* PCM_TXFIFO Bit-Fields */
+#define S3C_PCM_TXFIFO_DVALID	(0x1 << 16)
+#define S3C_PCM_TXFIFO_DATA_MSK	(0xffff << 0)
+
+/* PCM_RXFIFO Bit-Fields */
+#define S3C_PCM_RXFIFO_DVALID	(0x1 << 16)
+#define S3C_PCM_RXFIFO_DATA_MSK	(0xffff << 0)
+
+/* PCM_IRQCTL Bit-Fields */
+#define S3C_PCM_IRQCTL_IRQEN		(0x1 << 14)
+#define S3C_PCM_IRQCTL_WRDEN		(0x1 << 12)
+#define S3C_PCM_IRQCTL_TXEMPTYEN	(0x1 << 11)
+#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN	(0x1 << 10)
+#define S3C_PCM_IRQCTL_TXFULLEN		(0x1 << 9)
+#define S3C_PCM_IRQCTL_TXALMSTFULLEN	(0x1 << 8)
+#define S3C_PCM_IRQCTL_TXSTARVEN	(0x1 << 7)
+#define S3C_PCM_IRQCTL_TXERROVRFLEN	(0x1 << 6)
+#define S3C_PCM_IRQCTL_RXEMPTEN		(0x1 << 5)
+#define S3C_PCM_IRQCTL_RXALMSTEMPTEN	(0x1 << 4)
+#define S3C_PCM_IRQCTL_RXFULLEN		(0x1 << 3)
+#define S3C_PCM_IRQCTL_RXALMSTFULLEN	(0x1 << 2)
+#define S3C_PCM_IRQCTL_RXSTARVEN	(0x1 << 1)
+#define S3C_PCM_IRQCTL_RXERROVRFLEN	(0x1 << 0)
+
+/* PCM_IRQSTAT Bit-Fields */
+#define S3C_PCM_IRQSTAT_IRQPND		(0x1 << 13)
+#define S3C_PCM_IRQSTAT_WRD_XFER	(0x1 << 12)
+#define S3C_PCM_IRQSTAT_TXEMPTY		(0x1 << 11)
+#define S3C_PCM_IRQSTAT_TXALMSTEMPTY	(0x1 << 10)
+#define S3C_PCM_IRQSTAT_TXFULL		(0x1 << 9)
+#define S3C_PCM_IRQSTAT_TXALMSTFULL	(0x1 << 8)
+#define S3C_PCM_IRQSTAT_TXSTARV		(0x1 << 7)
+#define S3C_PCM_IRQSTAT_TXERROVRFL	(0x1 << 6)
+#define S3C_PCM_IRQSTAT_RXEMPT		(0x1 << 5)
+#define S3C_PCM_IRQSTAT_RXALMSTEMPT	(0x1 << 4)
+#define S3C_PCM_IRQSTAT_RXFULL		(0x1 << 3)
+#define S3C_PCM_IRQSTAT_RXALMSTFULL	(0x1 << 2)
+#define S3C_PCM_IRQSTAT_RXSTARV		(0x1 << 1)
+#define S3C_PCM_IRQSTAT_RXERROVRFL	(0x1 << 0)
+
+/* PCM_FIFOSTAT Bit-Fields */
+#define S3C_PCM_FIFOSTAT_TXCNT_MSK		(0x3f << 14)
+#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY		(0x1 << 13)
+#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY	(0x1 << 12)
+#define S3C_PCM_FIFOSTAT_TXFIFOFULL		(0x1 << 11)
+#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL	(0x1 << 10)
+#define S3C_PCM_FIFOSTAT_RXCNT_MSK		(0x3f << 4)
+#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY		(0x1 << 3)
+#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY	(0x1 << 2)
+#define S3C_PCM_FIFOSTAT_RXFIFOFULL		(0x1 << 1)
+#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL	(0x1 << 0)
+
+/**
+ * struct s3c_pcm_info - S3C PCM Controller information
+ * @dev: The parent device passed to use from the probe.
+ * @regs: The pointer to the device register block.
+ * @dma_playback: DMA information for playback channel.
+ * @dma_capture: DMA information for capture channel.
+ */
+struct s3c_pcm_info {
+	spinlock_t lock;
+	struct device	*dev;
+	void __iomem	*regs;
+
+	unsigned int sclk_per_fs;
+
+	/* Whether to keep PCMSCLK enabled even when idle(no active xfer) */
+	unsigned int idleclk;
+
+	struct clk	*pclk;
+	struct clk	*cclk;
+
+	struct s3c_dma_params	*dma_playback;
+	struct s3c_dma_params	*dma_capture;
+};
+
 static struct s3c2410_dma_client s3c_pcm_dma_client_out = {
 	.name		= "PCM Stereo out"
 };
diff --git a/sound/soc/samsung/pcm.h b/sound/soc/samsung/pcm.h
index 03393dcf852..726baf81461 100644
--- a/sound/soc/samsung/pcm.h
+++ b/sound/soc/samsung/pcm.h
@@ -9,116 +9,9 @@
 #ifndef __S3C_PCM_H
 #define __S3C_PCM_H __FILE__
 
-/*Register Offsets */
-#define S3C_PCM_CTL	(0x00)
-#define S3C_PCM_CLKCTL	(0x04)
-#define S3C_PCM_TXFIFO	(0x08)
-#define S3C_PCM_RXFIFO	(0x0C)
-#define S3C_PCM_IRQCTL	(0x10)
-#define S3C_PCM_IRQSTAT	(0x14)
-#define S3C_PCM_FIFOSTAT	(0x18)
-#define S3C_PCM_CLRINT	(0x20)
-
-/* PCM_CTL Bit-Fields */
-#define S3C_PCM_CTL_TXDIPSTICK_MASK		(0x3f)
-#define S3C_PCM_CTL_TXDIPSTICK_SHIFT	(13)
-#define S3C_PCM_CTL_RXDIPSTICK_MASK		(0x3f)
-#define S3C_PCM_CTL_RXDIPSTICK_SHIFT	(7)
-#define S3C_PCM_CTL_TXDMA_EN		(0x1<<6)
-#define S3C_PCM_CTL_RXDMA_EN		(0x1<<5)
-#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC	(0x1<<4)
-#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC	(0x1<<3)
-#define S3C_PCM_CTL_TXFIFO_EN		(0x1<<2)
-#define S3C_PCM_CTL_RXFIFO_EN		(0x1<<1)
-#define S3C_PCM_CTL_ENABLE			(0x1<<0)
-
-/* PCM_CLKCTL Bit-Fields */
-#define S3C_PCM_CLKCTL_SERCLK_EN		(0x1<<19)
-#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK	(0x1<<18)
-#define S3C_PCM_CLKCTL_SCLKDIV_MASK		(0x1ff)
-#define S3C_PCM_CLKCTL_SYNCDIV_MASK		(0x1ff)
-#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT	(9)
-#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT	(0)
-
-/* PCM_TXFIFO Bit-Fields */
-#define S3C_PCM_TXFIFO_DVALID	(0x1<<16)
-#define S3C_PCM_TXFIFO_DATA_MSK	(0xffff<<0)
-
-/* PCM_RXFIFO Bit-Fields */
-#define S3C_PCM_RXFIFO_DVALID	(0x1<<16)
-#define S3C_PCM_RXFIFO_DATA_MSK	(0xffff<<0)
-
-/* PCM_IRQCTL Bit-Fields */
-#define S3C_PCM_IRQCTL_IRQEN		(0x1<<14)
-#define S3C_PCM_IRQCTL_WRDEN		(0x1<<12)
-#define S3C_PCM_IRQCTL_TXEMPTYEN		(0x1<<11)
-#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN	(0x1<<10)
-#define S3C_PCM_IRQCTL_TXFULLEN		(0x1<<9)
-#define S3C_PCM_IRQCTL_TXALMSTFULLEN	(0x1<<8)
-#define S3C_PCM_IRQCTL_TXSTARVEN		(0x1<<7)
-#define S3C_PCM_IRQCTL_TXERROVRFLEN		(0x1<<6)
-#define S3C_PCM_IRQCTL_RXEMPTEN		(0x1<<5)
-#define S3C_PCM_IRQCTL_RXALMSTEMPTEN	(0x1<<4)
-#define S3C_PCM_IRQCTL_RXFULLEN		(0x1<<3)
-#define S3C_PCM_IRQCTL_RXALMSTFULLEN	(0x1<<2)
-#define S3C_PCM_IRQCTL_RXSTARVEN		(0x1<<1)
-#define S3C_PCM_IRQCTL_RXERROVRFLEN		(0x1<<0)
-
-/* PCM_IRQSTAT Bit-Fields */
-#define S3C_PCM_IRQSTAT_IRQPND		(0x1<<13)
-#define S3C_PCM_IRQSTAT_WRD_XFER		(0x1<<12)
-#define S3C_PCM_IRQSTAT_TXEMPTY		(0x1<<11)
-#define S3C_PCM_IRQSTAT_TXALMSTEMPTY	(0x1<<10)
-#define S3C_PCM_IRQSTAT_TXFULL		(0x1<<9)
-#define S3C_PCM_IRQSTAT_TXALMSTFULL		(0x1<<8)
-#define S3C_PCM_IRQSTAT_TXSTARV		(0x1<<7)
-#define S3C_PCM_IRQSTAT_TXERROVRFL		(0x1<<6)
-#define S3C_PCM_IRQSTAT_RXEMPT		(0x1<<5)
-#define S3C_PCM_IRQSTAT_RXALMSTEMPT		(0x1<<4)
-#define S3C_PCM_IRQSTAT_RXFULL		(0x1<<3)
-#define S3C_PCM_IRQSTAT_RXALMSTFULL		(0x1<<2)
-#define S3C_PCM_IRQSTAT_RXSTARV		(0x1<<1)
-#define S3C_PCM_IRQSTAT_RXERROVRFL		(0x1<<0)
-
-/* PCM_FIFOSTAT Bit-Fields */
-#define S3C_PCM_FIFOSTAT_TXCNT_MSK		(0x3f<<14)
-#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY	(0x1<<13)
-#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY	(0x1<<12)
-#define S3C_PCM_FIFOSTAT_TXFIFOFULL		(0x1<<11)
-#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL	(0x1<<10)
-#define S3C_PCM_FIFOSTAT_RXCNT_MSK		(0x3f<<4)
-#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY	(0x1<<3)
-#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY	(0x1<<2)
-#define S3C_PCM_FIFOSTAT_RXFIFOFULL		(0x1<<1)
-#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL	(0x1<<0)
-
 #define S3C_PCM_CLKSRC_PCLK	0
 #define S3C_PCM_CLKSRC_MUX	1
 
 #define S3C_PCM_SCLK_PER_FS	0
 
-/**
- * struct s3c_pcm_info - S3C PCM Controller information
- * @dev: The parent device passed to use from the probe.
- * @regs: The pointer to the device register block.
- * @dma_playback: DMA information for playback channel.
- * @dma_capture: DMA information for capture channel.
- */
-struct s3c_pcm_info {
-	spinlock_t lock;
-	struct device	*dev;
-	void __iomem	*regs;
-
-	unsigned int sclk_per_fs;
-
-	/* Whether to keep PCMSCLK enabled even when idle(no active xfer) */
-	unsigned int idleclk;
-
-	struct clk	*pclk;
-	struct clk	*cclk;
-
-	struct s3c_dma_params	*dma_playback;
-	struct s3c_dma_params	*dma_capture;
-};
-
 #endif /* __S3C_PCM_H */
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index f40027445dd..1e574a5d440 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,26 +17,15 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/platform_device.h>
-#include <linux/i2c.h>
 #include <linux/gpio.h>
-#include <linux/clk.h>
 
 #include <sound/soc.h>
-#include <sound/uda1380.h>
 #include <sound/jack.h>
 
 #include <plat/regs-iis.h>
-
-#include <mach/regs-clock.h>
-
 #include <asm/mach-types.h>
 
-#include "dma.h"
 #include "s3c24xx-i2s.h"
-#include "../codecs/uda1380.h"
 
 static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd);
 static int rx1950_startup(struct snd_pcm_substream *substream);
diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 094f36e41e8..52074a2b069 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -20,9 +20,8 @@
 #include <linux/clk.h>
 #include <linux/io.h>
 
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/pcm_params.h>
 
 #include <mach/dma.h>
 
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 7ea83786712..841ab14c110 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -16,21 +16,13 @@
  * option) any later version.
  */
 
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/gpio.h>
 #include <linux/clk.h>
-#include <linux/kernel.h>
 #include <linux/io.h>
 
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
 #include <sound/soc.h>
-#include <mach/hardware.h>
+#include <sound/pcm_params.h>
 
 #include <mach/regs-gpio.h>
 #include <mach/dma.h>
@@ -39,8 +31,6 @@
 #include "regs-i2s-v2.h"
 #include "s3c2412-i2s.h"
 
-#define S3C2412_I2S_DEBUG 0
-
 static struct s3c2410_dma_client s3c2412_dma_client_out = {
 	.name		= "I2S PCM Stereo out"
 };
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 13e41ed8e22..63d8849d80b 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -14,28 +14,16 @@
  *  option) any later version.
  */
 
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/clk.h>
-#include <linux/jiffies.h>
 #include <linux/io.h>
 #include <linux/gpio.h>
 
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
 #include <sound/soc.h>
+#include <sound/pcm_params.h>
 
-#include <mach/hardware.h>
 #include <mach/regs-gpio.h>
-#include <mach/regs-clock.h>
-
-#include <asm/dma.h>
 #include <mach/dma.h>
-
 #include <plat/regs-iis.h>
 
 #include "dma.h"
diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c
index a434032d183..349566f0686 100644
--- a/sound/soc/samsung/s3c24xx_simtec.c
+++ b/sound/soc/samsung/s3c24xx_simtec.c
@@ -7,20 +7,13 @@
  * published by the Free Software Foundation.
 */
 
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/platform_device.h>
 #include <linux/gpio.h>
 #include <linux/clk.h>
-#include <linux/i2c.h>
 
-#include <sound/core.h>
-#include <sound/pcm.h>
 #include <sound/soc.h>
 
 #include <plat/audio-simtec.h>
 
-#include "dma.h"
 #include "s3c24xx-i2s.h"
 #include "s3c24xx_simtec.h"
 
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index bb4292e3596..ce6aef60417 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -7,18 +7,8 @@
  * published by the Free Software Foundation.
 */
 
-#include <linux/module.h>
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
 #include <sound/soc.h>
 
-#include <plat/audio-simtec.h>
-
-#include "dma.h"
-#include "s3c24xx-i2s.h"
 #include "s3c24xx_simtec.h"
 
 static const struct snd_soc_dapm_widget dapm_widgets[] = {
@@ -94,8 +84,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic33 = {
 	.name		= "tlv320aic33",
 	.stream_name	= "TLV320AIC33",
-	.codec_name	= "tlv320aic3x-codec.0-0x1a",
-	.cpu_dai_name	= "s3c24xx-i2s",
+	.codec_name	= "tlv320aic3x-codec.0-001a",
+	.cpu_dai_name	= "s3c24xx-iis",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
 	.init		= simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index fbba4e36772..a7ef7db5468 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -7,22 +7,10 @@
  * published by the Free Software Foundation.
 */
 
-#include <linux/module.h>
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
 #include <sound/soc.h>
 
-#include <plat/audio-simtec.h>
-
-#include "dma.h"
-#include "s3c24xx-i2s.h"
 #include "s3c24xx_simtec.h"
 
-#include "../codecs/tlv320aic23.h"
-
 /* supported machines:
  *
  * Machine	Connections		AMP
@@ -85,8 +73,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic23 = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
-	.codec_name	= "tlv320aic3x-codec.0-0x1a",
-	.cpu_dai_name	= "s3c24xx-i2s",
+	.codec_name	= "tlv320aic3x-codec.0-001a",
+	.cpu_dai_name	= "s3c24xx-iis",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
 	.init		= simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index cdc8ecbcb8e..dc9d551f678 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -11,22 +11,15 @@
  * published by the Free Software Foundation.
  */
 
-#include <linux/module.h>
 #include <linux/clk.h>
-#include <linux/mutex.h>
 #include <linux/gpio.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
+
 #include <sound/soc.h>
 #include <sound/s3c24xx_uda134x.h>
-#include <sound/uda134x.h>
 
 #include <plat/regs-iis.h>
 
-#include "dma.h"
 #include "s3c24xx-i2s.h"
-#include "../codecs/uda134x.h"
-
 
 /* #define ENFORCE_RATES 1 */
 /*
@@ -226,9 +219,9 @@ static struct snd_soc_ops s3c24xx_uda134x_ops = {
 static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
 	.name = "UDA134X",
 	.stream_name = "UDA134X",
-	.codec_name = "uda134x-hifi",
+	.codec_name = "uda134x-codec",
 	.codec_dai_name = "uda134x-hifi",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.ops = &s3c24xx_uda134x_ops,
 	.platform_name	= "samsung-audio",
 };
@@ -321,6 +314,7 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(s3c24xx_uda134x_snd_device,
 			     &snd_soc_s3c24xx_uda134x);
+	platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x));
 	ret = platform_device_add(s3c24xx_uda134x_snd_device);
 	if (ret) {
 		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index 61e2b5219d4..0a2c4f22303 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -13,20 +13,14 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/platform_device.h>
 #include <linux/gpio.h>
 
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/jack.h>
 
 #include <asm/mach-types.h>
 
-#include "dma.h"
 #include "i2s.h"
-
 #include "../codecs/wm8750.h"
 
 /*
diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c
index 3be7e7e92d6..3a0dbfc793f 100644
--- a/sound/soc/samsung/smdk2443_wm9710.c
+++ b/sound/soc/samsung/smdk2443_wm9710.c
@@ -12,15 +12,8 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
 #include <sound/soc.h>
 
-#include "dma.h"
-#include "ac97.h"
-
 static struct snd_soc_card smdk2443;
 
 static struct snd_soc_dai_link smdk2443_dai[] = {
diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c
index b5c3fad01bb..e8ac961c6ba 100644
--- a/sound/soc/samsung/smdk_spdif.c
+++ b/sound/soc/samsung/smdk_spdif.c
@@ -10,15 +10,10 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/device.h>
 #include <linux/clk.h>
 
-#include <plat/devs.h>
-
 #include <sound/soc.h>
 
-#include "dma.h"
 #include "spdif.h"
 
 /* Audio clock settings are belonged to board specific part. Every
diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c
index b2cff1a44ae..8aacf23d6f3 100644
--- a/sound/soc/samsung/smdk_wm8580.c
+++ b/sound/soc/samsung/smdk_wm8580.c
@@ -10,17 +10,12 @@
  *  option) any later version.
  */
 
-#include <linux/platform_device.h>
-#include <linux/clk.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/pcm_params.h>
 
 #include <asm/mach-types.h>
 
 #include "../codecs/wm8580.h"
-#include "dma.h"
 #include "i2s.h"
 
 /*
diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c
index ae5fed6f772..fffe3c1dd1b 100644
--- a/sound/soc/samsung/smdk_wm9713.c
+++ b/sound/soc/samsung/smdk_wm9713.c
@@ -11,13 +11,8 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/device.h>
 #include <sound/soc.h>
 
-#include "dma.h"
-#include "ac97.h"
-
 static struct snd_soc_card smdk;
 
 /*
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index f0816404ea3..28c491dacf7 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -13,9 +13,8 @@
 #include <linux/clk.h>
 #include <linux/io.h>
 
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/pcm_params.h>
 
 #include <plat/audio.h>
 #include <mach/dma.h>
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index a14820ac966..d6f4703b3c0 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -18,18 +18,26 @@ struct fsi_ak4642_data {
 	const char *cpu_dai;
 	const char *codec;
 	const char *platform;
+	int id;
 };
 
 static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *codec = rtd->codec_dai;
+	struct snd_soc_dai *cpu = rtd->cpu_dai;
 	int ret;
 
-	ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_CBM_CFM);
+	ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J |
+					 SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_sysclk(dai, 0, 11289600, 0);
+	ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J |
+				       SND_SOC_DAIFMT_CBS_CFS);
 
 	return ret;
 }
@@ -60,7 +68,7 @@ static int fsi_ak4642_probe(struct platform_device *pdev)
 
 	pdata = (struct fsi_ak4642_data *)id_entry->driver_data;
 
-	fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A);
+	fsi_snd_device = platform_device_alloc("soc-audio", pdata->id);
 	if (!fsi_snd_device)
 		goto out;
 
@@ -93,6 +101,7 @@ static struct fsi_ak4642_data fsi_a_ak4642 = {
 	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0012",
 	.platform	= "sh_fsi.0",
+	.id		= FSI_PORT_A,
 };
 
 static struct fsi_ak4642_data fsi_b_ak4642 = {
@@ -101,6 +110,7 @@ static struct fsi_ak4642_data fsi_b_ak4642 = {
 	.cpu_dai	= "fsib-dai",
 	.codec		= "ak4642-codec.0-0012",
 	.platform	= "sh_fsi.0",
+	.id		= FSI_PORT_B,
 };
 
 static struct fsi_ak4642_data fsi_a_ak4643 = {
@@ -109,6 +119,7 @@ static struct fsi_ak4642_data fsi_a_ak4643 = {
 	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi.0",
+	.id		= FSI_PORT_A,
 };
 
 static struct fsi_ak4642_data fsi_b_ak4643 = {
@@ -117,6 +128,7 @@ static struct fsi_ak4642_data fsi_b_ak4643 = {
 	.cpu_dai	= "fsib-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi.0",
+	.id		= FSI_PORT_B,
 };
 
 static struct fsi_ak4642_data fsi2_a_ak4642 = {
@@ -125,6 +137,7 @@ static struct fsi_ak4642_data fsi2_a_ak4642 = {
 	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0012",
 	.platform	= "sh_fsi2",
+	.id		= FSI_PORT_A,
 };
 
 static struct fsi_ak4642_data fsi2_b_ak4642 = {
@@ -133,6 +146,7 @@ static struct fsi_ak4642_data fsi2_b_ak4642 = {
 	.cpu_dai	= "fsib-dai",
 	.codec		= "ak4642-codec.0-0012",
 	.platform	= "sh_fsi2",
+	.id		= FSI_PORT_B,
 };
 
 static struct fsi_ak4642_data fsi2_a_ak4643 = {
@@ -141,6 +155,7 @@ static struct fsi_ak4642_data fsi2_a_ak4643 = {
 	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi2",
+	.id		= FSI_PORT_A,
 };
 
 static struct fsi_ak4642_data fsi2_b_ak4643 = {
@@ -149,6 +164,7 @@ static struct fsi_ak4642_data fsi2_b_ak4643 = {
 	.cpu_dai	= "fsib-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi2",
+	.id		= FSI_PORT_B,
 };
 
 static struct platform_device_id fsi_id_table[] = {
diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c
index e8df9da92f7..dbafd7ac559 100644
--- a/sound/soc/sh/fsi-da7210.c
+++ b/sound/soc/sh/fsi-da7210.c
@@ -15,11 +15,20 @@
 
 static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *codec = rtd->codec_dai;
+	struct snd_soc_dai *cpu = rtd->cpu_dai;
+	int ret;
 
-	return snd_soc_dai_set_fmt(dai,
+	ret = snd_soc_dai_set_fmt(codec,
 				   SND_SOC_DAIFMT_I2S |
 				   SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S |
+				       SND_SOC_DAIFMT_CBS_CFS);
+
+	return ret;
 }
 
 static struct snd_soc_dai_link fsi_da7210_dai = {
diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c
index a52dd8ec71d..9719985eb82 100644
--- a/sound/soc/sh/fsi-hdmi.c
+++ b/sound/soc/sh/fsi-hdmi.c
@@ -12,31 +12,59 @@
 #include <linux/platform_device.h>
 #include <sound/sh_fsi.h>
 
+struct fsi_hdmi_data {
+	const char *cpu_dai;
+	const char *card;
+	int id;
+};
+
+static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *cpu = rtd->cpu_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM);
+
+	return ret;
+}
+
 static struct snd_soc_dai_link fsi_dai_link = {
 	.name		= "HDMI",
 	.stream_name	= "HDMI",
-	.cpu_dai_name	= "fsib-dai", /* fsi B */
 	.codec_dai_name	= "sh_mobile_hdmi-hifi",
 	.platform_name	= "sh_fsi2",
 	.codec_name	= "sh-mobile-hdmi",
+	.init		= fsi_hdmi_dai_init,
 };
 
 static struct snd_soc_card fsi_soc_card  = {
-	.name		= "FSI (SH MOBILE HDMI)",
 	.dai_link	= &fsi_dai_link,
 	.num_links	= 1,
 };
 
 static struct platform_device *fsi_snd_device;
 
-static int __init fsi_hdmi_init(void)
+static int fsi_hdmi_probe(struct platform_device *pdev)
 {
 	int ret = -ENOMEM;
+	const struct platform_device_id	*id_entry;
+	struct fsi_hdmi_data *pdata;
+
+	id_entry = pdev->id_entry;
+	if (!id_entry) {
+		dev_err(&pdev->dev, "unknown fsi hdmi\n");
+		return -ENODEV;
+	}
 
-	fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B);
+	pdata = (struct fsi_hdmi_data *)id_entry->driver_data;
+
+	fsi_snd_device = platform_device_alloc("soc-audio", pdata->id);
 	if (!fsi_snd_device)
 		goto out;
 
+	fsi_dai_link.cpu_dai_name	= pdata->cpu_dai;
+	fsi_soc_card.name		= pdata->card;
+
 	platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
 	ret = platform_device_add(fsi_snd_device);
 
@@ -47,9 +75,48 @@ out:
 	return ret;
 }
 
-static void __exit fsi_hdmi_exit(void)
+static int fsi_hdmi_remove(struct platform_device *pdev)
 {
 	platform_device_unregister(fsi_snd_device);
+	return 0;
+}
+
+static struct fsi_hdmi_data fsi2_a_hdmi = {
+	.cpu_dai	= "fsia-dai",
+	.card		= "FSI2A (SH MOBILE HDMI)",
+	.id		= FSI_PORT_A,
+};
+
+static struct fsi_hdmi_data fsi2_b_hdmi = {
+	.cpu_dai	= "fsib-dai",
+	.card		= "FSI2B (SH MOBILE HDMI)",
+	.id		= FSI_PORT_B,
+};
+
+static struct platform_device_id fsi_id_table[] = {
+	/* FSI 2 */
+	{ "sh_fsi2_a_hdmi",	(kernel_ulong_t)&fsi2_a_hdmi },
+	{ "sh_fsi2_b_hdmi",	(kernel_ulong_t)&fsi2_b_hdmi },
+	{},
+};
+
+static struct platform_driver fsi_hdmi = {
+	.driver = {
+		.name	= "fsi-hdmi-audio",
+	},
+	.probe		= fsi_hdmi_probe,
+	.remove		= fsi_hdmi_remove,
+	.id_table	= fsi_id_table,
+};
+
+static int __init fsi_hdmi_init(void)
+{
+	return platform_driver_register(&fsi_hdmi);
+}
+
+static void __exit fsi_hdmi_exit(void)
+{
+	platform_driver_unregister(&fsi_hdmi);
 }
 
 module_init(fsi_hdmi_init);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 2b06402801e..0c9997e2d8c 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -78,6 +78,8 @@
 /* CKG1 */
 #define ACKMD_MASK	0x00007000
 #define BPFMD_MASK	0x00000700
+#define DIMD		(1 << 4)
+#define DOMD		(1 << 0)
 
 /* A/B MST_CTLR */
 #define BP	(1 << 4)	/* Fix the signal of Biphase output */
@@ -111,6 +113,8 @@
 
 #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
 
+typedef int (*set_rate_func)(struct device *dev, int is_porta, int rate, int enable);
+
 /*
  * FSI driver use below type name for variable
  *
@@ -128,7 +132,6 @@ struct fsi_stream {
 	struct snd_pcm_substream *substream;
 
 	int fifo_max_num;
-	int chan_num;
 
 	int buff_offset;
 	int buff_len;
@@ -143,6 +146,7 @@ struct fsi_priv {
 	void __iomem *base;
 	struct fsi_master *master;
 
+	int chan_num;
 	struct fsi_stream playback;
 	struct fsi_stream capture;
 
@@ -252,9 +256,8 @@ static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream)
 	return  rtd->cpu_dai;
 }
 
-static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream)
+static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *dai)
 {
-	struct snd_soc_dai *dai = fsi_get_dai(substream);
 	struct fsi_master *master = snd_soc_dai_get_drvdata(dai);
 
 	if (dai->id == 0)
@@ -263,11 +266,27 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream)
 		return &master->fsib;
 }
 
+static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream)
+{
+	return fsi_get_priv_frm_dai(fsi_get_dai(substream));
+}
+
+static set_rate_func fsi_get_info_set_rate(struct fsi_master *master)
+{
+	if (!master->info)
+		return NULL;
+
+	return master->info->set_rate;
+}
+
 static u32 fsi_get_info_flags(struct fsi_priv *fsi)
 {
 	int is_porta = fsi_is_port_a(fsi);
 	struct fsi_master *master = fsi_get_master(fsi);
 
+	if (!master->info)
+		return 0;
+
 	return is_porta ? master->info->porta_flags :
 		master->info->portb_flags;
 }
@@ -288,21 +307,6 @@ static inline struct fsi_stream *fsi_get_stream(struct fsi_priv *fsi,
 	return is_play ? &fsi->playback : &fsi->capture;
 }
 
-static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play)
-{
-	u32 mode;
-	u32 flags = fsi_get_info_flags(fsi);
-
-	mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE;
-
-	/* return
-	 * 1 : master mode
-	 * 0 : slave mode
-	 */
-
-	return (mode & flags) != mode;
-}
-
 static u32 fsi_get_port_shift(struct fsi_priv *fsi, int is_play)
 {
 	int is_porta = fsi_is_port_a(fsi);
@@ -357,7 +361,6 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play)
 static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
 {
 	u32 status;
-	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
 	int data_num;
 
 	status = is_play ?
@@ -365,7 +368,7 @@ static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
 		fsi_reg_read(fsi, DIFF_ST);
 
 	data_num = 0x1ff & (status >> 8);
-	data_num *= io->chan_num;
+	data_num *= fsi->chan_num;
 
 	return data_num;
 }
@@ -387,7 +390,7 @@ static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play)
 	struct snd_pcm_substream *substream = io->substream;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	return frames_to_bytes(runtime, 1) / io->chan_num;
+	return frames_to_bytes(runtime, 1) / fsi->chan_num;
 }
 
 static void fsi_count_fifo_err(struct fsi_priv *fsi)
@@ -580,10 +583,10 @@ static void fsi_fifo_init(struct fsi_priv *fsi,
 	 * 7 channels:  32 ( 32 x 7 = 224)
 	 * 8 channels:  32 ( 32 x 8 = 256)
 	 */
-	for (i = 1; i < io->chan_num; i <<= 1)
+	for (i = 1; i < fsi->chan_num; i <<= 1)
 		io->fifo_max_num >>= 1;
 	dev_dbg(dai->dev, "%d channel %d store\n",
-		io->chan_num, io->fifo_max_num);
+		fsi->chan_num, io->fifo_max_num);
 
 	/*
 	 * set interrupt generation factor
@@ -659,7 +662,7 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream)
 		 * data_num_max	: number of FSI fifo free space
 		 * data_num	: number of ALSA residue data
 		 */
-		data_num_max  = io->fifo_max_num * io->chan_num;
+		data_num_max  = io->fifo_max_num * fsi->chan_num;
 		data_num_max -= fsi_get_fifo_data_num(fsi, is_play);
 
 		data_num = data_residue_num;
@@ -754,25 +757,12 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
-	struct fsi_master *master = fsi_get_master(fsi);
-	struct fsi_stream *io;
 	u32 flags = fsi_get_info_flags(fsi);
-	u32 fmt;
 	u32 data;
 	int is_play = fsi_is_play(substream);
-	int is_master;
-
-	io = fsi_get_stream(fsi, is_play);
 
 	pm_runtime_get_sync(dai->dev);
 
-	/* CKG1 */
-	data = is_play ? (1 << 0) : (1 << 4);
-	is_master = fsi_is_master_mode(fsi, is_play);
-	if (is_master)
-		fsi_reg_mask_set(fsi, CKG1, data, data);
-	else
-		fsi_reg_mask_set(fsi, CKG1, data, 0);
 
 	/* clock inversion (CKG2) */
 	data = 0;
@@ -787,54 +777,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 
 	fsi_reg_write(fsi, CKG2, data);
 
-	/* do fmt, di fmt */
-	data = 0;
-	fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags);
-	switch (fmt) {
-	case SH_FSI_FMT_MONO:
-		data = CR_MONO;
-		io->chan_num = 1;
-		break;
-	case SH_FSI_FMT_MONO_DELAY:
-		data = CR_MONO_D;
-		io->chan_num = 1;
-		break;
-	case SH_FSI_FMT_PCM:
-		data = CR_PCM;
-		io->chan_num = 2;
-		break;
-	case SH_FSI_FMT_I2S:
-		data = CR_I2S;
-		io->chan_num = 2;
-		break;
-	case SH_FSI_FMT_TDM:
-		io->chan_num = is_play ?
-			SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
-		data = CR_TDM | (io->chan_num - 1);
-		break;
-	case SH_FSI_FMT_TDM_DELAY:
-		io->chan_num = is_play ?
-			SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
-		data = CR_TDM_D | (io->chan_num - 1);
-		break;
-	case SH_FSI_FMT_SPDIF:
-		if (master->core->ver < 2) {
-			dev_err(dai->dev, "This FSI can not use SPDIF\n");
-			return -EINVAL;
-		}
-		data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
-		io->chan_num = 2;
-		fsi_spdif_clk_ctrl(fsi, 1);
-		fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
-		break;
-	default:
-		dev_err(dai->dev, "unknown format.\n");
-		return -EINVAL;
-	}
-	is_play ?
-		fsi_reg_write(fsi, DO_FMT, data) :
-		fsi_reg_write(fsi, DI_FMT, data);
-
 	/* irq clear */
 	fsi_irq_disable(fsi, is_play);
 	fsi_irq_clear_status(fsi);
@@ -851,12 +793,12 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	int is_play = fsi_is_play(substream);
 	struct fsi_master *master = fsi_get_master(fsi);
-	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
+	set_rate_func set_rate;
 
 	fsi_irq_disable(fsi, is_play);
 	fsi_clk_ctrl(fsi, 0);
 
-	set_rate = master->info->set_rate;
+	set_rate = fsi_get_info_set_rate(master);
 	if (set_rate && fsi->rate)
 		set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
 	fsi->rate = 0;
@@ -889,18 +831,100 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt)
+{
+	u32 data = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		data = CR_I2S;
+		fsi->chan_num = 2;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		data = CR_PCM;
+		fsi->chan_num = 2;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	fsi_reg_write(fsi, DO_FMT, data);
+	fsi_reg_write(fsi, DI_FMT, data);
+
+	return 0;
+}
+
+static int fsi_set_fmt_spdif(struct fsi_priv *fsi)
+{
+	struct fsi_master *master = fsi_get_master(fsi);
+	u32 data = 0;
+
+	if (master->core->ver < 2)
+		return -EINVAL;
+
+	data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
+	fsi->chan_num = 2;
+	fsi_spdif_clk_ctrl(fsi, 1);
+	fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+
+	fsi_reg_write(fsi, DO_FMT, data);
+	fsi_reg_write(fsi, DI_FMT, data);
+
+	return 0;
+}
+
+static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai);
+	u32 flags = fsi_get_info_flags(fsi);
+	u32 data = 0;
+	int ret;
+
+	pm_runtime_get_sync(dai->dev);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		data = DIMD | DOMD;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		ret = -EINVAL;
+		goto set_fmt_exit;
+	}
+	fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
+
+	/* set format */
+	switch (flags & SH_FSI_FMT_MASK) {
+	case SH_FSI_FMT_DAI:
+		ret = fsi_set_fmt_dai(fsi, fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		break;
+	case SH_FSI_FMT_SPDIF:
+		ret = fsi_set_fmt_spdif(fsi);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+set_fmt_exit:
+	pm_runtime_put_sync(dai->dev);
+
+	return ret;
+}
+
 static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params,
 			     struct snd_soc_dai *dai)
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	struct fsi_master *master = fsi_get_master(fsi);
-	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
+	set_rate_func set_rate;
 	int fsi_ver = master->core->ver;
 	long rate = params_rate(params);
 	int ret;
 
-	set_rate = master->info->set_rate;
+	set_rate = fsi_get_info_set_rate(master);
 	if (!set_rate)
 		return 0;
 
@@ -975,6 +999,7 @@ static struct snd_soc_dai_ops fsi_dai_ops = {
 	.startup	= fsi_dai_startup,
 	.shutdown	= fsi_dai_shutdown,
 	.trigger	= fsi_dai_trigger,
+	.set_fmt	= fsi_dai_set_fmt,
 	.hw_params	= fsi_dai_hw_params,
 };
 
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 8c2a21a978a..5d76da43b14 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -18,6 +18,8 @@
 #include <linux/bitmap.h>
 #include <linux/rbtree.h>
 
+#include <trace/events/asoc.h>
+
 static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
 				     unsigned int reg)
 {
@@ -25,7 +27,8 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
 	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
-		snd_soc_codec_volatile_register(codec, reg)) {
+		snd_soc_codec_volatile_register(codec, reg) ||
+		codec->cache_bypass) {
 			if (codec->cache_only)
 				return -1;
 
@@ -49,7 +52,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
 	data[1] = value & 0x00ff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size) {
+		reg < codec->driver->reg_cache_size &&
+		!codec->cache_bypass) {
 		ret = snd_soc_cache_write(codec, reg, value);
 		if (ret < 0)
 			return -1;
@@ -106,7 +110,8 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
 	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
-		snd_soc_codec_volatile_register(codec, reg)) {
+		snd_soc_codec_volatile_register(codec, reg) ||
+		codec->cache_bypass) {
 			if (codec->cache_only)
 				return -1;
 
@@ -130,7 +135,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
 	data[1] = value & 0x00ff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size) {
+		reg < codec->driver->reg_cache_size &&
+		!codec->cache_bypass) {
 		ret = snd_soc_cache_write(codec, reg, value);
 		if (ret < 0)
 			return -1;
@@ -191,7 +197,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
 	data[1] = value & 0xff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size) {
+		reg < codec->driver->reg_cache_size &&
+		!codec->cache_bypass) {
 		ret = snd_soc_cache_write(codec, reg, value);
 		if (ret < 0)
 			return -1;
@@ -216,7 +223,8 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
 
 	reg &= 0xff;
 	if (reg >= codec->driver->reg_cache_size ||
-		snd_soc_codec_volatile_register(codec, reg)) {
+		snd_soc_codec_volatile_register(codec, reg) ||
+		codec->cache_bypass) {
 			if (codec->cache_only)
 				return -1;
 
@@ -271,7 +279,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
 	data[2] = value & 0xff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size) {
+		reg < codec->driver->reg_cache_size &&
+		!codec->cache_bypass) {
 		ret = snd_soc_cache_write(codec, reg, value);
 		if (ret < 0)
 			return -1;
@@ -295,7 +304,8 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
 	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
-	    snd_soc_codec_volatile_register(codec, reg)) {
+	    snd_soc_codec_volatile_register(codec, reg) ||
+	    codec->cache_bypass) {
 		if (codec->cache_only)
 			return -1;
 
@@ -450,7 +460,8 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec,
 
 	reg &= 0xff;
 	if (reg >= codec->driver->reg_cache_size ||
-		snd_soc_codec_volatile_register(codec, reg)) {
+		snd_soc_codec_volatile_register(codec, reg) ||
+		codec->cache_bypass) {
 			if (codec->cache_only)
 				return -1;
 
@@ -476,7 +487,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	reg &= 0xff;
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size) {
+		reg < codec->driver->reg_cache_size &&
+		!codec->cache_bypass) {
 		ret = snd_soc_cache_write(codec, reg, value);
 		if (ret < 0)
 			return -1;
@@ -568,7 +580,8 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec,
 	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
-	    snd_soc_codec_volatile_register(codec, reg)) {
+	    snd_soc_codec_volatile_register(codec, reg) ||
+	    codec->cache_bypass) {
 		if (codec->cache_only)
 			return -1;
 
@@ -595,7 +608,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
 	data[3] = value & 0xff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size) {
+		reg < codec->driver->reg_cache_size &&
+		!codec->cache_bypass) {
 		ret = snd_soc_cache_write(codec, reg, value);
 		if (ret < 0)
 			return -1;
@@ -761,6 +775,49 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
 
+static bool snd_soc_set_cache_val(void *base, unsigned int idx,
+				  unsigned int val, unsigned int word_size)
+{
+	switch (word_size) {
+	case 1: {
+		u8 *cache = base;
+		if (cache[idx] == val)
+			return true;
+		cache[idx] = val;
+		break;
+	}
+	case 2: {
+		u16 *cache = base;
+		if (cache[idx] == val)
+			return true;
+		cache[idx] = val;
+		break;
+	}
+	default:
+		BUG();
+	}
+	return false;
+}
+
+static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx,
+		unsigned int word_size)
+{
+	switch (word_size) {
+	case 1: {
+		const u8 *cache = base;
+		return cache[idx];
+	}
+	case 2: {
+		const u16 *cache = base;
+		return cache[idx];
+	}
+	default:
+		BUG();
+	}
+	/* unreachable */
+	return -1;
+}
+
 struct snd_soc_rbtree_node {
 	struct rb_node node;
 	unsigned int reg;
@@ -835,7 +892,9 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec)
 		ret = snd_soc_cache_read(codec, rbnode->reg, &val);
 		if (ret)
 			return ret;
+		codec->cache_bypass = 1;
 		ret = snd_soc_write(codec, rbnode->reg, val);
+		codec->cache_bypass = 0;
 		if (ret)
 			return ret;
 		dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
@@ -924,7 +983,12 @@ static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec)
 
 static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
 {
+	struct snd_soc_rbtree_node *rbtree_node;
 	struct snd_soc_rbtree_ctx *rbtree_ctx;
+	unsigned int val;
+	unsigned int word_size;
+	int i;
+	int ret;
 
 	codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL);
 	if (!codec->reg_cache)
@@ -936,53 +1000,25 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
 	if (!codec->reg_def_copy)
 		return 0;
 
-/*
- * populate the rbtree with the initialized registers.  All other
- * registers will be inserted into the tree when they are first written.
- *
- * The reasoning behind this, is that we need to step through and
- * dereference the cache in u8/u16 increments without sacrificing
- * portability.  This could also be done using memcpy() but that would
- * be slightly more cryptic.
- */
-#define snd_soc_rbtree_populate(cache)					\
-({									\
-	int ret, i;							\
-	struct snd_soc_rbtree_node *rbtree_node;			\
-									\
-	ret = 0;							\
-	cache = codec->reg_def_copy;					\
-	for (i = 0; i < codec->driver->reg_cache_size; ++i) {		\
-		if (!cache[i])						\
-			continue;					\
-		rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL);	\
-		if (!rbtree_node) {					\
-			ret = -ENOMEM;					\
-			snd_soc_cache_exit(codec);			\
-			break;						\
-		}							\
-		rbtree_node->reg = i;					\
-		rbtree_node->value = cache[i];				\
-		rbtree_node->defval = cache[i];				\
-		snd_soc_rbtree_insert(&rbtree_ctx->root,		\
-				      rbtree_node);			\
-	}								\
-	ret;								\
-})
-
-	switch (codec->driver->reg_word_size) {
-	case 1: {
-		const u8 *cache;
-
-		return snd_soc_rbtree_populate(cache);
-	}
-	case 2: {
-		const u16 *cache;
-
-		return snd_soc_rbtree_populate(cache);
-	}
-	default:
-		BUG();
+	/*
+	 * populate the rbtree with the initialized registers.  All other
+	 * registers will be inserted when they are first modified.
+	 */
+	word_size = codec->driver->reg_word_size;
+	for (i = 0; i < codec->driver->reg_cache_size; ++i) {
+		val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size);
+		if (!val)
+			continue;
+		rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL);
+		if (!rbtree_node) {
+			ret = -ENOMEM;
+			snd_soc_cache_exit(codec);
+			break;
+		}
+		rbtree_node->reg = i;
+		rbtree_node->value = val;
+		rbtree_node->defval = val;
+		snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node);
 	}
 
 	return 0;
@@ -1080,34 +1116,28 @@ static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec,
 		unsigned int reg)
 {
 	const struct snd_soc_codec_driver *codec_drv;
-	size_t reg_size;
 
 	codec_drv = codec->driver;
-	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
 	return (reg * codec_drv->reg_word_size) /
-	       DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count());
+	       DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count());
 }
 
 static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec,
 		unsigned int reg)
 {
 	const struct snd_soc_codec_driver *codec_drv;
-	size_t reg_size;
 
 	codec_drv = codec->driver;
-	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
-	return reg % (DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()) /
+	return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) /
 		      codec_drv->reg_word_size);
 }
 
 static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec)
 {
 	const struct snd_soc_codec_driver *codec_drv;
-	size_t reg_size;
 
 	codec_drv = codec->driver;
-	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
-	return DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count());
+	return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count());
 }
 
 static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
@@ -1122,7 +1152,9 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
 		ret = snd_soc_cache_read(codec, i, &val);
 		if (ret)
 			return ret;
+		codec->cache_bypass = 1;
 		ret = snd_soc_write(codec, i, val);
+		codec->cache_bypass = 0;
 		if (ret)
 			return ret;
 		dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
@@ -1165,29 +1197,10 @@ static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec,
 	}
 
 	/* write the new value to the cache */
-	switch (codec->driver->reg_word_size) {
-	case 1: {
-		u8 *cache;
-		cache = lzo_block->dst;
-		if (cache[blkpos] == value) {
-			kfree(lzo_block->dst);
-			goto out;
-		}
-		cache[blkpos] = value;
-	}
-	break;
-	case 2: {
-		u16 *cache;
-		cache = lzo_block->dst;
-		if (cache[blkpos] == value) {
-			kfree(lzo_block->dst);
-			goto out;
-		}
-		cache[blkpos] = value;
-	}
-	break;
-	default:
-		BUG();
+	if (snd_soc_set_cache_val(lzo_block->dst, blkpos, value,
+				  codec->driver->reg_word_size)) {
+		kfree(lzo_block->dst);
+		goto out;
 	}
 
 	/* prepare the source to be the decompressed block */
@@ -1241,25 +1254,10 @@ static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec,
 
 	/* decompress the block */
 	ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block);
-	if (ret >= 0) {
+	if (ret >= 0)
 		/* fetch the value from the cache */
-		switch (codec->driver->reg_word_size) {
-		case 1: {
-			u8 *cache;
-			cache = lzo_block->dst;
-			*value = cache[blkpos];
-		}
-		break;
-		case 2: {
-			u16 *cache;
-			cache = lzo_block->dst;
-			*value = cache[blkpos];
-		}
-		break;
-		default:
-			BUG();
-		}
-	}
+		*value = snd_soc_get_cache_val(lzo_block->dst, blkpos,
+					       codec->driver->reg_word_size);
 
 	kfree(lzo_block->dst);
 	/* restore the pointer and length of the compressed block */
@@ -1301,7 +1299,7 @@ static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec)
 static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
 {
 	struct snd_soc_lzo_ctx **lzo_blocks;
-	size_t reg_size, bmp_size;
+	size_t bmp_size;
 	const struct snd_soc_codec_driver *codec_drv;
 	int ret, tofree, i, blksize, blkcount;
 	const char *p, *end;
@@ -1309,7 +1307,6 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
 
 	ret = 0;
 	codec_drv = codec->driver;
-	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
 
 	/*
 	 * If we have not been given a default register cache
@@ -1321,8 +1318,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
 		tofree = 1;
 
 	if (!codec->reg_def_copy) {
-		codec->reg_def_copy = kzalloc(reg_size,
-						       GFP_KERNEL);
+		codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL);
 		if (!codec->reg_def_copy)
 			return -ENOMEM;
 	}
@@ -1370,7 +1366,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
 
 	blksize = snd_soc_lzo_get_blksize(codec);
 	p = codec->reg_def_copy;
-	end = codec->reg_def_copy + reg_size;
+	end = codec->reg_def_copy + codec->reg_size;
 	/* compress the register map and fill the lzo blocks */
 	for (i = 0; i < blkcount; ++i, p += blksize) {
 		lzo_blocks[i]->src = p;
@@ -1414,28 +1410,10 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
 		ret = snd_soc_cache_read(codec, i, &val);
 		if (ret)
 			return ret;
-		if (codec_drv->reg_cache_default) {
-			switch (codec_drv->reg_word_size) {
-			case 1: {
-				const u8 *cache;
-
-				cache = codec_drv->reg_cache_default;
-				if (cache[i] == val)
-					continue;
-			}
-			break;
-			case 2: {
-				const u16 *cache;
-
-				cache = codec_drv->reg_cache_default;
-				if (cache[i] == val)
-					continue;
-			}
-			break;
-			default:
-				BUG();
-			}
-		}
+		if (codec->reg_def_copy)
+			if (snd_soc_get_cache_val(codec->reg_def_copy,
+						  i, codec_drv->reg_word_size) == val)
+				continue;
 		ret = snd_soc_write(codec, i, val);
 		if (ret)
 			return ret;
@@ -1448,50 +1426,16 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
 static int snd_soc_flat_cache_write(struct snd_soc_codec *codec,
 				    unsigned int reg, unsigned int value)
 {
-	switch (codec->driver->reg_word_size) {
-	case 1: {
-		u8 *cache;
-
-		cache = codec->reg_cache;
-		cache[reg] = value;
-	}
-	break;
-	case 2: {
-		u16 *cache;
-
-		cache = codec->reg_cache;
-		cache[reg] = value;
-	}
-	break;
-	default:
-		BUG();
-	}
-
+	snd_soc_set_cache_val(codec->reg_cache, reg, value,
+			      codec->driver->reg_word_size);
 	return 0;
 }
 
 static int snd_soc_flat_cache_read(struct snd_soc_codec *codec,
 				   unsigned int reg, unsigned int *value)
 {
-	switch (codec->driver->reg_word_size) {
-	case 1: {
-		u8 *cache;
-
-		cache = codec->reg_cache;
-		*value = cache[reg];
-	}
-	break;
-	case 2: {
-		u16 *cache;
-
-		cache = codec->reg_cache;
-		*value = cache[reg];
-	}
-	break;
-	default:
-		BUG();
-	}
-
+	*value = snd_soc_get_cache_val(codec->reg_cache, reg,
+				       codec->driver->reg_word_size);
 	return 0;
 }
 
@@ -1507,24 +1451,14 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec)
 static int snd_soc_flat_cache_init(struct snd_soc_codec *codec)
 {
 	const struct snd_soc_codec_driver *codec_drv;
-	size_t reg_size;
 
 	codec_drv = codec->driver;
-	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
 
-	/*
-	 * for flat compression, we don't need to keep a copy of the
-	 * original defaults register cache as it will definitely not
-	 * be marked as __devinitconst
-	 */
-	kfree(codec->reg_def_copy);
-	codec->reg_def_copy = NULL;
-
-	if (codec_drv->reg_cache_default)
-		codec->reg_cache = kmemdup(codec_drv->reg_cache_default,
-					   reg_size, GFP_KERNEL);
+	if (codec->reg_def_copy)
+		codec->reg_cache = kmemdup(codec->reg_def_copy,
+					   codec->reg_size, GFP_KERNEL);
 	else
-		codec->reg_cache = kzalloc(reg_size, GFP_KERNEL);
+		codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL);
 	if (!codec->reg_cache)
 		return -ENOMEM;
 
@@ -1669,21 +1603,77 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write);
 int snd_soc_cache_sync(struct snd_soc_codec *codec)
 {
 	int ret;
+	const char *name;
 
 	if (!codec->cache_sync) {
 		return 0;
 	}
 
-	if (codec->cache_ops && codec->cache_ops->sync) {
-		if (codec->cache_ops->name)
-			dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
-				codec->cache_ops->name, codec->name);
-		ret = codec->cache_ops->sync(codec);
-		if (!ret)
-			codec->cache_sync = 0;
-		return ret;
-	}
+	if (!codec->cache_ops || !codec->cache_ops->sync)
+		return -EINVAL;
 
-	return -EINVAL;
+	if (codec->cache_ops->name)
+		name = codec->cache_ops->name;
+	else
+		name = "unknown";
+
+	if (codec->cache_ops->name)
+		dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
+			codec->cache_ops->name, codec->name);
+	trace_snd_soc_cache_sync(codec, name, "start");
+	ret = codec->cache_ops->sync(codec);
+	if (!ret)
+		codec->cache_sync = 0;
+	trace_snd_soc_cache_sync(codec, name, "end");
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_cache_sync);
+
+static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec,
+					unsigned int reg)
+{
+	const struct snd_soc_codec_driver *codec_drv;
+	unsigned int min, max, index;
+
+	codec_drv = codec->driver;
+	min = 0;
+	max = codec_drv->reg_access_size - 1;
+	do {
+		index = (min + max) / 2;
+		if (codec_drv->reg_access_default[index].reg == reg)
+			return index;
+		if (codec_drv->reg_access_default[index].reg < reg)
+			min = index + 1;
+		else
+			max = index;
+	} while (min <= max);
+	return -1;
+}
+
+int snd_soc_default_volatile_register(struct snd_soc_codec *codec,
+				      unsigned int reg)
+{
+	int index;
+
+	if (reg >= codec->driver->reg_cache_size)
+		return 1;
+	index = snd_soc_get_reg_access_index(codec, reg);
+	if (index < 0)
+		return 0;
+	return codec->driver->reg_access_default[index].vol;
+}
+EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register);
+
+int snd_soc_default_readable_register(struct snd_soc_codec *codec,
+				      unsigned int reg)
+{
+	int index;
+
+	if (reg >= codec->driver->reg_cache_size)
+		return 1;
+	index = snd_soc_get_reg_access_index(codec, reg);
+	if (index < 0)
+		return 0;
+	return codec->driver->reg_access_default[index].read;
+}
+EXPORT_SYMBOL_GPL(snd_soc_default_readable_register);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bac7291b6ff..4dda58926bc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -48,7 +48,8 @@ static DEFINE_MUTEX(pcm_mutex);
 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
 
 #ifdef CONFIG_DEBUG_FS
-static struct dentry *debugfs_root;
+struct dentry *snd_soc_debugfs_root;
+EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
 #endif
 
 static DEFINE_MUTEX(client_mutex);
@@ -57,8 +58,6 @@ static LIST_HEAD(dai_list);
 static LIST_HEAD(platform_list);
 static LIST_HEAD(codec_list);
 
-static int snd_soc_register_card(struct snd_soc_card *card);
-static int snd_soc_unregister_card(struct snd_soc_card *card);
 static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
 
 /*
@@ -70,10 +69,73 @@ static int pmdown_time = 5000;
 module_param(pmdown_time, int, 0);
 MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
 
+/* returns the minimum number of bytes needed to represent
+ * a particular given value */
+static int min_bytes_needed(unsigned long val)
+{
+	int c = 0;
+	int i;
+
+	for (i = (sizeof val * 8) - 1; i >= 0; --i, ++c)
+		if (val & (1UL << i))
+			break;
+	c = (sizeof val * 8) - c;
+	if (!c || (c % 8))
+		c = (c + 8) / 8;
+	else
+		c /= 8;
+	return c;
+}
+
+/* fill buf which is 'len' bytes with a formatted
+ * string of the form 'reg: value\n' */
+static int format_register_str(struct snd_soc_codec *codec,
+			       unsigned int reg, char *buf, size_t len)
+{
+	int wordsize = codec->driver->reg_word_size * 2;
+	int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+	int ret;
+	char tmpbuf[len + 1];
+	char regbuf[regsize + 1];
+
+	/* since tmpbuf is allocated on the stack, warn the callers if they
+	 * try to abuse this function */
+	WARN_ON(len > 63);
+
+	/* +2 for ': ' and + 1 for '\n' */
+	if (wordsize + regsize + 2 + 1 != len)
+		return -EINVAL;
+
+	ret = snd_soc_read(codec , reg);
+	if (ret < 0) {
+		memset(regbuf, 'X', regsize);
+		regbuf[regsize] = '\0';
+	} else {
+		snprintf(regbuf, regsize + 1, "%.*x", regsize, ret);
+	}
+
+	/* prepare the buffer */
+	snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf);
+	/* copy it back to the caller without the '\0' */
+	memcpy(buf, tmpbuf, len);
+
+	return 0;
+}
+
 /* codec register dump */
-static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
+static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
+				  size_t count, loff_t pos)
 {
-	int ret, i, step = 1, count = 0;
+	int i, step = 1;
+	int wordsize, regsize;
+	int len;
+	size_t total = 0;
+	loff_t p = 0;
+
+	wordsize = codec->driver->reg_word_size * 2;
+	regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+
+	len = wordsize + regsize + 2 + 1;
 
 	if (!codec->driver->reg_cache_size)
 		return 0;
@@ -81,55 +143,37 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
 	if (codec->driver->reg_cache_step)
 		step = codec->driver->reg_cache_step;
 
-	count += sprintf(buf, "%s registers\n", codec->name);
 	for (i = 0; i < codec->driver->reg_cache_size; i += step) {
-		if (codec->driver->readable_register && !codec->driver->readable_register(i))
+		if (codec->readable_register && !codec->readable_register(codec, i))
 			continue;
-
-		count += sprintf(buf + count, "%2x: ", i);
-		if (count >= PAGE_SIZE - 1)
-			break;
-
 		if (codec->driver->display_register) {
 			count += codec->driver->display_register(codec, buf + count,
 							 PAGE_SIZE - count, i);
 		} else {
-			/* If the read fails it's almost certainly due to
-			 * the register being volatile and the device being
-			 * powered off.
-			 */
-			ret = snd_soc_read(codec, i);
-			if (ret >= 0)
-				count += snprintf(buf + count,
-						  PAGE_SIZE - count,
-						  "%4x", ret);
-			else
-				count += snprintf(buf + count,
-						  PAGE_SIZE - count,
-						  "<no data: %d>", ret);
+			/* only support larger than PAGE_SIZE bytes debugfs
+			 * entries for the default case */
+			if (p >= pos) {
+				if (total + len >= count - 1)
+					break;
+				format_register_str(codec, i, buf + total, len);
+				total += len;
+			}
+			p += len;
 		}
-
-		if (count >= PAGE_SIZE - 1)
-			break;
-
-		count += snprintf(buf + count, PAGE_SIZE - count, "\n");
-		if (count >= PAGE_SIZE - 1)
-			break;
 	}
 
-	/* Truncate count; min() would cause a warning */
-	if (count >= PAGE_SIZE)
-		count = PAGE_SIZE - 1;
+	total = min(total, count - 1);
 
-	return count;
+	return total;
 }
+
 static ssize_t codec_reg_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_pcm_runtime *rtd =
 			container_of(dev, struct snd_soc_pcm_runtime, dev);
 
-	return soc_codec_reg_show(rtd->codec, buf);
+	return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0);
 }
 
 static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
@@ -168,16 +212,28 @@ static int codec_reg_open_file(struct inode *inode, struct file *file)
 }
 
 static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
-			       size_t count, loff_t *ppos)
+				   size_t count, loff_t *ppos)
 {
 	ssize_t ret;
 	struct snd_soc_codec *codec = file->private_data;
-	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	char *buf;
+
+	if (*ppos < 0 || !count)
+		return -EINVAL;
+
+	buf = kmalloc(count, GFP_KERNEL);
 	if (!buf)
 		return -ENOMEM;
-	ret = soc_codec_reg_show(codec, buf);
-	if (ret >= 0)
-		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+
+	ret = soc_codec_reg_show(codec, buf, count, *ppos);
+	if (ret >= 0) {
+		if (copy_to_user(user_buf, buf, ret)) {
+			kfree(buf);
+			return -EFAULT;
+		}
+		*ppos += ret;
+	}
+
 	kfree(buf);
 	return ret;
 }
@@ -203,12 +259,14 @@ static ssize_t codec_reg_write_file(struct file *file,
 	while (*start == ' ')
 		start++;
 	reg = simple_strtoul(start, &start, 16);
-	if ((reg >= codec->driver->reg_cache_size) || (reg % step))
-		return -EINVAL;
 	while (*start == ' ')
 		start++;
 	if (strict_strtoul(start, 16, &value))
 		return -EINVAL;
+
+	/* Userspace has been fiddling around behind the kernel's back */
+	add_taint(TAINT_USER);
+
 	snd_soc_write(codec, reg, value);
 	return buf_size;
 }
@@ -232,6 +290,11 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
 		return;
 	}
 
+	debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root,
+			    &codec->cache_sync);
+	debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root,
+			    &codec->cache_only);
+
 	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
 						 codec->debugfs_codec_root,
 						 codec, &codec_reg_fops);
@@ -356,7 +419,7 @@ static const struct file_operations platform_list_fops = {
 static void soc_init_card_debugfs(struct snd_soc_card *card)
 {
 	card->debugfs_card_root = debugfs_create_dir(card->name,
-						     debugfs_root);
+						     snd_soc_debugfs_root);
 	if (!card->debugfs_card_root) {
 		dev_warn(card->dev,
 			 "ASoC: Failed to create codec debugfs directory\n");
@@ -435,20 +498,30 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	int ret;
 
-	if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates ||
-			rtd->dai_link->symmetric_rates) {
-		dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n",
-				rtd->rate);
+	if (!codec_dai->driver->symmetric_rates &&
+	    !cpu_dai->driver->symmetric_rates &&
+	    !rtd->dai_link->symmetric_rates)
+		return 0;
 
-		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
-						   SNDRV_PCM_HW_PARAM_RATE,
-						   rtd->rate,
-						   rtd->rate);
-		if (ret < 0) {
-			dev_err(&rtd->dev,
-				"Unable to apply rate symmetry constraint: %d\n", ret);
-			return ret;
-		}
+	/* This can happen if multiple streams are starting simultaneously -
+	 * the second can need to get its constraints before the first has
+	 * picked a rate.  Complain and allow the application to carry on.
+	 */
+	if (!rtd->rate) {
+		dev_warn(&rtd->dev,
+			 "Not enforcing symmetric_rates due to race\n");
+		return 0;
+	}
+
+	dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
+
+	ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+					   SNDRV_PCM_HW_PARAM_RATE,
+					   rtd->rate, rtd->rate);
+	if (ret < 0) {
+		dev_err(&rtd->dev,
+			"Unable to apply rate symmetry constraint: %d\n", ret);
+		return ret;
 	}
 
 	return 0;
@@ -962,12 +1035,11 @@ static struct snd_pcm_ops soc_pcm_ops = {
 	.pointer	= soc_pcm_pointer,
 };
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 /* powers down audio subsystem for suspend */
-static int soc_suspend(struct device *dev)
+int snd_soc_suspend(struct device *dev)
 {
-	struct platform_device *pdev = to_platform_device(dev);
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = dev_get_drvdata(dev);
 	struct snd_soc_codec *codec;
 	int i;
 
@@ -1008,7 +1080,7 @@ static int soc_suspend(struct device *dev)
 	}
 
 	if (card->suspend_pre)
-		card->suspend_pre(pdev, PMSG_SUSPEND);
+		card->suspend_pre(card);
 
 	for (i = 0; i < card->num_rtd; i++) {
 		struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
@@ -1075,10 +1147,11 @@ static int soc_suspend(struct device *dev)
 	}
 
 	if (card->suspend_post)
-		card->suspend_post(pdev, PMSG_SUSPEND);
+		card->suspend_post(card);
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_suspend);
 
 /* deferred resume work, so resume can complete before we finished
  * setting our codec back up, which can be very slow on I2C
@@ -1087,7 +1160,6 @@ static void soc_resume_deferred(struct work_struct *work)
 {
 	struct snd_soc_card *card =
 			container_of(work, struct snd_soc_card, deferred_resume_work);
-	struct platform_device *pdev = to_platform_device(card->dev);
 	struct snd_soc_codec *codec;
 	int i;
 
@@ -1101,7 +1173,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2);
 
 	if (card->resume_pre)
-		card->resume_pre(pdev);
+		card->resume_pre(card);
 
 	/* resume AC97 DAIs */
 	for (i = 0; i < card->num_rtd; i++) {
@@ -1176,7 +1248,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	if (card->resume_post)
-		card->resume_post(pdev);
+		card->resume_post(card);
 
 	dev_dbg(card->dev, "resume work completed\n");
 
@@ -1185,10 +1257,9 @@ static void soc_resume_deferred(struct work_struct *work)
 }
 
 /* powers up audio subsystem after a suspend */
-static int soc_resume(struct device *dev)
+int snd_soc_resume(struct device *dev)
 {
-	struct platform_device *pdev = to_platform_device(dev);
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = dev_get_drvdata(dev);
 	int i;
 
 	/* AC97 devices might have other drivers hanging off them so
@@ -1210,9 +1281,10 @@ static int soc_resume(struct device *dev)
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_resume);
 #else
-#define soc_suspend	NULL
-#define soc_resume	NULL
+#define snd_soc_suspend NULL
+#define snd_soc_resume NULL
 #endif
 
 static struct snd_soc_dai_ops null_dai_ops = {
@@ -1400,31 +1472,44 @@ static int soc_probe_codec(struct snd_soc_card *card,
 			   struct snd_soc_codec *codec)
 {
 	int ret = 0;
+	const struct snd_soc_codec_driver *driver = codec->driver;
 
 	codec->card = card;
 	codec->dapm.card = card;
 	soc_set_name_prefix(card, codec);
 
-	if (codec->driver->probe) {
-		ret = codec->driver->probe(codec);
+	if (!try_module_get(codec->dev->driver->owner))
+		return -ENODEV;
+
+	if (driver->probe) {
+		ret = driver->probe(codec);
 		if (ret < 0) {
 			dev_err(codec->dev,
 				"asoc: failed to probe CODEC %s: %d\n",
 				codec->name, ret);
-			return ret;
+			goto err_probe;
 		}
 	}
 
+	if (driver->dapm_widgets)
+		snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets,
+					  driver->num_dapm_widgets);
+	if (driver->dapm_routes)
+		snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes,
+					driver->num_dapm_routes);
+
 	soc_init_codec_debugfs(codec);
 
 	/* mark codec as probed and add to card codec list */
-	if (!try_module_get(codec->dev->driver->owner))
-		return -ENODEV;
-
 	codec->probed = 1;
 	list_add(&codec->card_list, &card->codec_dev_list);
 	list_add(&codec->dapm.list, &card->dapm_list);
 
+	return 0;
+
+err_probe:
+	module_put(codec->dev->driver->owner);
+
 	return ret;
 }
 
@@ -1449,6 +1534,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
 		rtd = &card->rtd_aux[num];
 		name = aux_dev->name;
 	}
+	rtd->card = card;
 
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
@@ -1467,11 +1553,9 @@ static int soc_post_component_init(struct snd_soc_card *card,
 
 	/* Make sure all DAPM widgets are instantiated */
 	snd_soc_dapm_new_widgets(&codec->dapm);
-	snd_soc_dapm_sync(&codec->dapm);
 
 	/* register the rtd device */
 	rtd->codec = codec;
-	rtd->card = card;
 	rtd->dev.parent = card->dev;
 	rtd->dev.release = rtd_release;
 	rtd->dev.init_name = name;
@@ -1543,19 +1627,19 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
 
 	/* probe the platform */
 	if (!platform->probed) {
+		if (!try_module_get(platform->dev->driver->owner))
+			return -ENODEV;
+
 		if (platform->driver->probe) {
 			ret = platform->driver->probe(platform);
 			if (ret < 0) {
 				printk(KERN_ERR "asoc: failed to probe platform %s\n",
 						platform->name);
+				module_put(platform->dev->driver->owner);
 				return ret;
 			}
 		}
 		/* mark platform as probed and add to card platform list */
-
-		if (!try_module_get(platform->dev->driver->owner))
-			return -ENODEV;
-
 		platform->probed = 1;
 		list_add(&platform->card_list, &card->platform_dev_list);
 	}
@@ -1664,9 +1748,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
 	goto out;
 
 found:
-	if (!try_module_get(codec->dev->driver->owner))
-		return -ENODEV;
-
 	ret = soc_probe_codec(card, codec);
 	if (ret < 0)
 		return ret;
@@ -1716,7 +1797,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec,
 
 static void snd_soc_instantiate_card(struct snd_soc_card *card)
 {
-	struct platform_device *pdev = to_platform_device(card->dev);
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_conf *codec_conf;
 	enum snd_soc_compress_type compress_type;
@@ -1743,6 +1823,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 	list_for_each_entry(codec, &codec_list, list) {
 		if (codec->cache_init)
 			continue;
+		/* by default we don't override the compress_type */
+		compress_type = 0;
 		/* check to see if we need to override the compress_type */
 		for (i = 0; i < card->num_configs; ++i) {
 			codec_conf = &card->codec_conf[i];
@@ -1753,18 +1835,6 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 					break;
 			}
 		}
-		if (i == card->num_configs) {
-			/* no need to override the compress_type so
-			 * go ahead and do the standard thing */
-			ret = snd_soc_init_codec_cache(codec, 0);
-			if (ret < 0) {
-				mutex_unlock(&card->mutex);
-				return;
-			}
-			continue;
-		}
-		/* override the compress_type with the one supplied in
-		 * the machine driver */
 		ret = snd_soc_init_codec_cache(codec, compress_type);
 		if (ret < 0) {
 			mutex_unlock(&card->mutex);
@@ -1783,14 +1853,19 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 	card->snd_card->dev = card->dev;
 
-#ifdef CONFIG_PM
+	card->dapm.bias_level = SND_SOC_BIAS_OFF;
+	card->dapm.dev = card->dev;
+	card->dapm.card = card;
+	list_add(&card->dapm.list, &card->dapm_list);
+
+#ifdef CONFIG_PM_SLEEP
 	/* deferred resume work */
 	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
 #endif
 
 	/* initialise the sound card only once */
 	if (card->probe) {
-		ret = card->probe(pdev);
+		ret = card->probe(card);
 		if (ret < 0)
 			goto card_probe_error;
 	}
@@ -1813,11 +1888,37 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 		}
 	}
 
+	if (card->dapm_widgets)
+		snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
+					  card->num_dapm_widgets);
+	if (card->dapm_routes)
+		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
+					card->num_dapm_routes);
+
+#ifdef CONFIG_DEBUG_FS
+	card->dapm.debugfs_dapm = debugfs_create_dir("dapm",
+						     card->debugfs_card_root);
+	if (!card->dapm.debugfs_dapm)
+		printk(KERN_WARNING
+		       "Failed to create card DAPM debugfs directory\n");
+
+	snd_soc_dapm_debugfs_init(&card->dapm);
+#endif
+
 	snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname),
 		 "%s",  card->name);
 	snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
 		 "%s", card->name);
 
+	if (card->late_probe) {
+		ret = card->late_probe(card);
+		if (ret < 0) {
+			dev_err(card->dev, "%s late_probe() failed: %d\n",
+				card->name, ret);
+			goto probe_aux_dev_err;
+		}
+	}
+
 	ret = snd_card_register(card->snd_card);
 	if (ret < 0) {
 		printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name);
@@ -1851,7 +1952,7 @@ probe_dai_err:
 
 card_probe_error:
 	if (card->remove)
-		card->remove(pdev);
+		card->remove(card);
 
 	snd_card_free(card->snd_card);
 
@@ -1875,16 +1976,15 @@ static int soc_probe(struct platform_device *pdev)
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	int ret = 0;
 
+	/*
+	 * no card, so machine driver should be registering card
+	 * we should not be here in that case so ret error
+	 */
+	if (!card)
+		return -EINVAL;
+
 	/* Bodge while we unpick instantiation */
 	card->dev = &pdev->dev;
-	INIT_LIST_HEAD(&card->dai_dev_list);
-	INIT_LIST_HEAD(&card->codec_dev_list);
-	INIT_LIST_HEAD(&card->platform_dev_list);
-	INIT_LIST_HEAD(&card->widgets);
-	INIT_LIST_HEAD(&card->paths);
-	INIT_LIST_HEAD(&card->dapm_list);
-
-	soc_init_card_debugfs(card);
 
 	ret = snd_soc_register_card(card);
 	if (ret != 0) {
@@ -1895,45 +1995,48 @@ static int soc_probe(struct platform_device *pdev)
 	return 0;
 }
 
-/* removes a socdev */
-static int soc_remove(struct platform_device *pdev)
+static int soc_cleanup_card_resources(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	int i;
 
-		if (card->instantiated) {
+	/* make sure any delayed work runs */
+	for (i = 0; i < card->num_rtd; i++) {
+		struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
+		flush_delayed_work_sync(&rtd->delayed_work);
+	}
 
-		/* make sure any delayed work runs */
-		for (i = 0; i < card->num_rtd; i++) {
-			struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
-			flush_delayed_work_sync(&rtd->delayed_work);
-		}
+	/* remove auxiliary devices */
+	for (i = 0; i < card->num_aux_devs; i++)
+		soc_remove_aux_dev(card, i);
+
+	/* remove and free each DAI */
+	for (i = 0; i < card->num_rtd; i++)
+		soc_remove_dai_link(card, i);
 
-		/* remove auxiliary devices */
-		for (i = 0; i < card->num_aux_devs; i++)
-			soc_remove_aux_dev(card, i);
+	soc_cleanup_card_debugfs(card);
 
-		/* remove and free each DAI */
-		for (i = 0; i < card->num_rtd; i++)
-			soc_remove_dai_link(card, i);
+	/* remove the card */
+	if (card->remove)
+		card->remove(card);
 
-		soc_cleanup_card_debugfs(card);
+	kfree(card->rtd);
+	snd_card_free(card->snd_card);
+	return 0;
 
-		/* remove the card */
-		if (card->remove)
-			card->remove(pdev);
+}
+
+/* removes a socdev */
+static int soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
 
-		kfree(card->rtd);
-		snd_card_free(card->snd_card);
-	}
 	snd_soc_unregister_card(card);
 	return 0;
 }
 
-static int soc_poweroff(struct device *dev)
+int snd_soc_poweroff(struct device *dev)
 {
-	struct platform_device *pdev = to_platform_device(dev);
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = dev_get_drvdata(dev);
 	int i;
 
 	if (!card->instantiated)
@@ -1950,11 +2053,12 @@ static int soc_poweroff(struct device *dev)
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_poweroff);
 
-static const struct dev_pm_ops soc_pm_ops = {
-	.suspend = soc_suspend,
-	.resume = soc_resume,
-	.poweroff = soc_poweroff,
+const struct dev_pm_ops snd_soc_pm_ops = {
+	.suspend = snd_soc_suspend,
+	.resume = snd_soc_resume,
+	.poweroff = snd_soc_poweroff,
 };
 
 /* ASoC platform driver */
@@ -1962,7 +2066,7 @@ static struct platform_driver soc_driver = {
 	.driver		= {
 		.name		= "soc-audio",
 		.owner		= THIS_MODULE,
-		.pm		= &soc_pm_ops,
+		.pm		= &snd_soc_pm_ops,
 	},
 	.probe		= soc_probe,
 	.remove		= soc_remove,
@@ -2032,10 +2136,11 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
  *
  * Boolean function indiciating if a CODEC register is volatile.
  */
-int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg)
+int snd_soc_codec_volatile_register(struct snd_soc_codec *codec,
+				    unsigned int reg)
 {
-	if (codec->driver->volatile_register)
-		return codec->driver->volatile_register(reg);
+	if (codec->volatile_register)
+		return codec->volatile_register(codec, reg);
 	else
 		return 0;
 }
@@ -2132,19 +2237,27 @@ EXPORT_SYMBOL_GPL(snd_soc_write);
  *
  * Writes new register value.
  *
- * Returns 1 for change else 0.
+ * Returns 1 for change, 0 for no change, or negative error code.
  */
 int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
 				unsigned int mask, unsigned int value)
 {
 	int change;
 	unsigned int old, new;
+	int ret;
 
-	old = snd_soc_read(codec, reg);
+	ret = snd_soc_read(codec, reg);
+	if (ret < 0)
+		return ret;
+
+	old = ret;
 	new = (old & ~mask) | value;
 	change = old != new;
-	if (change)
-		snd_soc_write(codec, reg, new);
+	if (change) {
+		ret = snd_soc_write(codec, reg, new);
+		if (ret < 0)
+			return ret;
+	}
 
 	return change;
 }
@@ -2229,22 +2342,45 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
  * @_template: control template
  * @data: control private data
  * @long_name: control long name
+ * @prefix: control name prefix
  *
  * Create a new mixer control from a template control.
  *
  * Returns 0 for success, else error.
  */
 struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
-	void *data, char *long_name)
+				  void *data, char *long_name,
+				  const char *prefix)
 {
 	struct snd_kcontrol_new template;
+	struct snd_kcontrol *kcontrol;
+	char *name = NULL;
+	int name_len;
 
 	memcpy(&template, _template, sizeof(template));
-	if (long_name)
-		template.name = long_name;
 	template.index = 0;
 
-	return snd_ctl_new1(&template, data);
+	if (!long_name)
+		long_name = template.name;
+
+	if (prefix) {
+		name_len = strlen(long_name) + strlen(prefix) + 2;
+		name = kmalloc(name_len, GFP_ATOMIC);
+		if (!name)
+			return NULL;
+
+		snprintf(name, name_len, "%s %s", prefix, long_name);
+
+		template.name = name;
+	} else {
+		template.name = long_name;
+	}
+
+	kcontrol = snd_ctl_new1(&template, data);
+
+	kfree(name);
+
+	return kcontrol;
 }
 EXPORT_SYMBOL_GPL(snd_soc_cnew);
 
@@ -2263,22 +2399,16 @@ int snd_soc_add_controls(struct snd_soc_codec *codec,
 	const struct snd_kcontrol_new *controls, int num_controls)
 {
 	struct snd_card *card = codec->card->snd_card;
-	char prefixed_name[44], *name;
 	int err, i;
 
 	for (i = 0; i < num_controls; i++) {
 		const struct snd_kcontrol_new *control = &controls[i];
-		if (codec->name_prefix) {
-			snprintf(prefixed_name, sizeof(prefixed_name), "%s %s",
-				 codec->name_prefix, control->name);
-			name = prefixed_name;
-		} else {
-			name = control->name;
-		}
-		err = snd_ctl_add(card, snd_soc_cnew(control, codec, name));
+		err = snd_ctl_add(card, snd_soc_cnew(control, codec,
+						     control->name,
+						     codec->name_prefix));
 		if (err < 0) {
 			dev_err(codec->dev, "%s: Failed to add %s: %d\n",
-				codec->name, name, err);
+				codec->name, control->name, err);
 			return err;
 		}
 	}
@@ -2959,12 +3089,34 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 {
 	if (dai->driver && dai->driver->ops->set_sysclk)
 		return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
+	else if (dai->codec && dai->codec->driver->set_sysclk)
+		return dai->codec->driver->set_sysclk(dai->codec, clk_id,
+						      freq, dir);
 	else
 		return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 
 /**
+ * snd_soc_codec_set_sysclk - configure CODEC system or master clock.
+ * @codec: CODEC
+ * @clk_id: DAI specific clock ID
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+	unsigned int freq, int dir)
+{
+	if (codec->driver->set_sysclk)
+		return codec->driver->set_sysclk(codec, clk_id, freq, dir);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk);
+
+/**
  * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
  * @dai: DAI
  * @div_id: DAI specific clock divider ID
@@ -3000,11 +3152,35 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
 	if (dai->driver && dai->driver->ops->set_pll)
 		return dai->driver->ops->set_pll(dai, pll_id, source,
 					 freq_in, freq_out);
+	else if (dai->codec && dai->codec->driver->set_pll)
+		return dai->codec->driver->set_pll(dai->codec, pll_id, source,
+						   freq_in, freq_out);
 	else
 		return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
 
+/*
+ * snd_soc_codec_set_pll - configure codec PLL.
+ * @codec: CODEC
+ * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+			  unsigned int freq_in, unsigned int freq_out)
+{
+	if (codec->driver->set_pll)
+		return codec->driver->set_pll(codec, pll_id, source,
+					      freq_in, freq_out);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll);
+
 /**
  * snd_soc_dai_set_fmt - configure DAI hardware audio format.
  * @dai: DAI
@@ -3104,17 +3280,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
  *
  * @card: Card to register
  *
- * Note that currently this is an internal only function: it will be
- * exposed to machine drivers after further backporting of ASoC v2
- * registration APIs.
  */
-static int snd_soc_register_card(struct snd_soc_card *card)
+int snd_soc_register_card(struct snd_soc_card *card)
 {
 	int i;
 
 	if (!card->name || !card->dev)
 		return -EINVAL;
 
+	snd_soc_initialize_card_lists(card);
+
+	soc_init_card_debugfs(card);
+
 	card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) *
 			    (card->num_links + card->num_aux_devs),
 			    GFP_KERNEL);
@@ -3138,18 +3315,18 @@ static int snd_soc_register_card(struct snd_soc_card *card)
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
 /**
  * snd_soc_unregister_card - Unregister a card with the ASoC core
  *
  * @card: Card to unregister
  *
- * Note that currently this is an internal only function: it will be
- * exposed to machine drivers after further backporting of ASoC v2
- * registration APIs.
  */
-static int snd_soc_unregister_card(struct snd_soc_card *card)
+int snd_soc_unregister_card(struct snd_soc_card *card)
 {
+	if (card->instantiated)
+		soc_cleanup_card_resources(card);
 	mutex_lock(&client_mutex);
 	list_del(&card->list);
 	mutex_unlock(&client_mutex);
@@ -3157,6 +3334,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_unregister_card);
 
 /*
  * Simplify DAI link configuration by removing ".-1" from device names
@@ -3486,9 +3664,12 @@ int snd_soc_register_codec(struct device *dev,
 
 	codec->write = codec_drv->write;
 	codec->read = codec_drv->read;
+	codec->volatile_register = codec_drv->volatile_register;
+	codec->readable_register = codec_drv->readable_register;
 	codec->dapm.bias_level = SND_SOC_BIAS_OFF;
 	codec->dapm.dev = dev;
 	codec->dapm.codec = codec;
+	codec->dapm.seq_notifier = codec_drv->seq_notifier;
 	codec->dev = dev;
 	codec->driver = codec_drv;
 	codec->num_dai = num_dai;
@@ -3497,20 +3678,30 @@ int snd_soc_register_codec(struct device *dev,
 	/* allocate CODEC register cache */
 	if (codec_drv->reg_cache_size && codec_drv->reg_word_size) {
 		reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+		codec->reg_size = reg_size;
 		/* it is necessary to make a copy of the default register cache
 		 * because in the case of using a compression type that requires
 		 * the default register cache to be marked as __devinitconst the
 		 * kernel might have freed the array by the time we initialize
 		 * the cache.
 		 */
-		codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default,
-					      reg_size, GFP_KERNEL);
-		if (!codec->reg_def_copy) {
-			ret = -ENOMEM;
-			goto fail;
+		if (codec_drv->reg_cache_default) {
+			codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default,
+						      reg_size, GFP_KERNEL);
+			if (!codec->reg_def_copy) {
+				ret = -ENOMEM;
+				goto fail;
+			}
 		}
 	}
 
+	if (codec_drv->reg_access_size && codec_drv->reg_access_default) {
+		if (!codec->volatile_register)
+			codec->volatile_register = snd_soc_default_volatile_register;
+		if (!codec->readable_register)
+			codec->readable_register = snd_soc_default_readable_register;
+	}
+
 	for (i = 0; i < num_dai; i++) {
 		fixup_codec_formats(&dai_drv[i].playback);
 		fixup_codec_formats(&dai_drv[i].capture);
@@ -3577,22 +3768,22 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
 static int __init snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
-	debugfs_root = debugfs_create_dir("asoc", NULL);
-	if (IS_ERR(debugfs_root) || !debugfs_root) {
+	snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL);
+	if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) {
 		printk(KERN_WARNING
 		       "ASoC: Failed to create debugfs directory\n");
-		debugfs_root = NULL;
+		snd_soc_debugfs_root = NULL;
 	}
 
-	if (!debugfs_create_file("codecs", 0444, debugfs_root, NULL,
+	if (!debugfs_create_file("codecs", 0444, snd_soc_debugfs_root, NULL,
 				 &codec_list_fops))
 		pr_warn("ASoC: Failed to create CODEC list debugfs file\n");
 
-	if (!debugfs_create_file("dais", 0444, debugfs_root, NULL,
+	if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL,
 				 &dai_list_fops))
 		pr_warn("ASoC: Failed to create DAI list debugfs file\n");
 
-	if (!debugfs_create_file("platforms", 0444, debugfs_root, NULL,
+	if (!debugfs_create_file("platforms", 0444, snd_soc_debugfs_root, NULL,
 				 &platform_list_fops))
 		pr_warn("ASoC: Failed to create platform list debugfs file\n");
 #endif
@@ -3604,7 +3795,7 @@ module_init(snd_soc_init);
 static void __exit snd_soc_exit(void)
 {
 #ifdef CONFIG_DEBUG_FS
-	debugfs_remove_recursive(debugfs_root);
+	debugfs_remove_recursive(snd_soc_debugfs_root);
 #endif
 	platform_driver_unregister(&soc_driver);
 }
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730ab563..81c4052c127 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -32,6 +32,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
+#include <linux/async.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/bitops.h>
@@ -125,17 +126,17 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
 
 /**
  * snd_soc_dapm_set_bias_level - set the bias level for the system
- * @card: audio device
+ * @dapm: DAPM context
  * @level: level to configure
  *
  * Configure the bias (power) levels for the SoC audio device.
  *
  * Returns 0 for success else error.
  */
-static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card,
-				       struct snd_soc_dapm_context *dapm,
+static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 				       enum snd_soc_bias_level level)
 {
+	struct snd_soc_card *card = dapm->card;
 	int ret = 0;
 
 	switch (level) {
@@ -365,9 +366,20 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_dapm_widget *w)
 {
 	int i, ret = 0;
-	size_t name_len;
+	size_t name_len, prefix_len;
 	struct snd_soc_dapm_path *path;
-	struct snd_card *card = dapm->codec->card->snd_card;
+	struct snd_card *card = dapm->card->snd_card;
+	const char *prefix;
+
+	if (dapm->codec)
+		prefix = dapm->codec->name_prefix;
+	else
+		prefix = NULL;
+
+	if (prefix)
+		prefix_len = strlen(prefix) + 1;
+	else
+		prefix_len = 0;
 
 	/* add kcontrol */
 	for (i = 0; i < w->num_kcontrols; i++) {
@@ -396,8 +408,15 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
 
 			switch (w->id) {
 			default:
+				/* The control will get a prefix from
+				 * the control creation process but
+				 * we're also using the same prefix
+				 * for widgets so cut the prefix off
+				 * the front of the widget name.
+				 */
 				snprintf(path->long_name, name_len, "%s %s",
-					 w->name, w->kcontrols[i].name);
+					 w->name + prefix_len,
+					 w->kcontrols[i].name);
 				break;
 			case snd_soc_dapm_mixer_named_ctl:
 				snprintf(path->long_name, name_len, "%s",
@@ -408,7 +427,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
 			path->long_name[name_len - 1] = '\0';
 
 			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
-				path->long_name);
+						      path->long_name, prefix);
 			ret = snd_ctl_add(card, path->kcontrol);
 			if (ret < 0) {
 				dev_err(dapm->dev,
@@ -429,7 +448,9 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
 {
 	struct snd_soc_dapm_path *path = NULL;
 	struct snd_kcontrol *kcontrol;
-	struct snd_card *card = dapm->codec->card->snd_card;
+	struct snd_card *card = dapm->card->snd_card;
+	const char *prefix;
+	size_t prefix_len;
 	int ret = 0;
 
 	if (!w->num_kcontrols) {
@@ -437,7 +458,22 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
 		return -EINVAL;
 	}
 
-	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+	if (dapm->codec)
+		prefix = dapm->codec->name_prefix;
+	else
+		prefix = NULL;
+
+	if (prefix)
+		prefix_len = strlen(prefix) + 1;
+	else
+		prefix_len = 0;
+
+	/* The control will get a prefix from the control creation
+	 * process but we're also using the same prefix for widgets so
+	 * cut the prefix off the front of the widget name.
+	 */
+	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name + prefix_len,
+				prefix);
 	ret = snd_ctl_add(card, kcontrol);
 
 	if (ret < 0)
@@ -479,7 +515,7 @@ static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm)
  */
 static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
 {
-	int level = snd_power_get_state(widget->dapm->codec->card->snd_card);
+	int level = snd_power_get_state(widget->dapm->card->snd_card);
 
 	switch (level) {
 	case SNDRV_CTL_POWER_D3hot:
@@ -712,7 +748,15 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
 		    !path->connected(path->source, path->sink))
 			continue;
 
-		if (path->sink && path->sink->power_check &&
+		if (!path->sink)
+			continue;
+
+		if (path->sink->force) {
+			power = 1;
+			break;
+		}
+
+		if (path->sink->power_check &&
 		    path->sink->power_check(path->sink)) {
 			power = 1;
 			break;
@@ -726,10 +770,23 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
 
 static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
 			    struct snd_soc_dapm_widget *b,
-			    int sort[])
+			    bool power_up)
 {
+	int *sort;
+
+	if (power_up)
+		sort = dapm_up_seq;
+	else
+		sort = dapm_down_seq;
+
 	if (sort[a->id] != sort[b->id])
 		return sort[a->id] - sort[b->id];
+	if (a->subseq != b->subseq) {
+		if (power_up)
+			return a->subseq - b->subseq;
+		else
+			return b->subseq - a->subseq;
+	}
 	if (a->reg != b->reg)
 		return a->reg - b->reg;
 	if (a->dapm != b->dapm)
@@ -741,12 +798,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
 /* Insert a widget in order into a DAPM power sequence. */
 static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget,
 			    struct list_head *list,
-			    int sort[])
+			    bool power_up)
 {
 	struct snd_soc_dapm_widget *w;
 
 	list_for_each_entry(w, list, power_list)
-		if (dapm_seq_compare(new_widget, w, sort) < 0) {
+		if (dapm_seq_compare(new_widget, w, power_up) < 0) {
 			list_add_tail(&new_widget->power_list, &w->power_list);
 			return;
 		}
@@ -857,26 +914,42 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
  * handled.
  */
 static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
-			 struct list_head *list, int event, int sort[])
+			 struct list_head *list, int event, bool power_up)
 {
 	struct snd_soc_dapm_widget *w, *n;
 	LIST_HEAD(pending);
 	int cur_sort = -1;
+	int cur_subseq = -1;
 	int cur_reg = SND_SOC_NOPM;
 	struct snd_soc_dapm_context *cur_dapm = NULL;
-	int ret;
+	int ret, i;
+	int *sort;
+
+	if (power_up)
+		sort = dapm_up_seq;
+	else
+		sort = dapm_down_seq;
 
 	list_for_each_entry_safe(w, n, list, power_list) {
 		ret = 0;
 
 		/* Do we need to apply any queued changes? */
 		if (sort[w->id] != cur_sort || w->reg != cur_reg ||
-		    w->dapm != cur_dapm) {
+		    w->dapm != cur_dapm || w->subseq != cur_subseq) {
 			if (!list_empty(&pending))
 				dapm_seq_run_coalesced(cur_dapm, &pending);
 
+			if (cur_dapm && cur_dapm->seq_notifier) {
+				for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
+					if (sort[i] == cur_sort)
+						cur_dapm->seq_notifier(cur_dapm,
+								       i,
+								       cur_subseq);
+			}
+
 			INIT_LIST_HEAD(&pending);
 			cur_sort = -1;
+			cur_subseq = -1;
 			cur_reg = SND_SOC_NOPM;
 			cur_dapm = NULL;
 		}
@@ -921,6 +994,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
 		default:
 			/* Queue it up for application */
 			cur_sort = sort[w->id];
+			cur_subseq = w->subseq;
 			cur_reg = w->reg;
 			cur_dapm = w->dapm;
 			list_move(&w->power_list, &pending);
@@ -933,7 +1007,14 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
 	}
 
 	if (!list_empty(&pending))
-		dapm_seq_run_coalesced(dapm, &pending);
+		dapm_seq_run_coalesced(cur_dapm, &pending);
+
+	if (cur_dapm && cur_dapm->seq_notifier) {
+		for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
+			if (sort[i] == cur_sort)
+				cur_dapm->seq_notifier(cur_dapm,
+						       i, cur_subseq);
+	}
 }
 
 static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
@@ -969,7 +1050,62 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
 	}
 }
 
+/* Async callback run prior to DAPM sequences - brings to _PREPARE if
+ * they're changing state.
+ */
+static void dapm_pre_sequence_async(void *data, async_cookie_t cookie)
+{
+	struct snd_soc_dapm_context *d = data;
+	int ret;
 
+	if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
+		if (ret != 0)
+			dev_err(d->dev,
+				"Failed to turn on bias: %d\n", ret);
+	}
+
+	/* If we're changing to all on or all off then prepare */
+	if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
+	    (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
+		if (ret != 0)
+			dev_err(d->dev,
+				"Failed to prepare bias: %d\n", ret);
+	}
+}
+
+/* Async callback run prior to DAPM sequences - brings to their final
+ * state.
+ */
+static void dapm_post_sequence_async(void *data, async_cookie_t cookie)
+{
+	struct snd_soc_dapm_context *d = data;
+	int ret;
+
+	/* If we just powered the last thing off drop to standby bias */
+	if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
+		if (ret != 0)
+			dev_err(d->dev, "Failed to apply standby bias: %d\n",
+				ret);
+	}
+
+	/* If we're in standby and can support bias off then do that */
+	if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF);
+		if (ret != 0)
+			dev_err(d->dev, "Failed to turn off bias: %d\n", ret);
+	}
+
+	/* If we just powered up then move to active bias */
+	if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON);
+		if (ret != 0)
+			dev_err(d->dev, "Failed to apply active bias: %d\n",
+				ret);
+	}
+}
 
 /*
  * Scan each dapm widget for complete audio path.
@@ -982,12 +1118,12 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
  */
 static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 {
-	struct snd_soc_card *card = dapm->codec->card;
+	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	struct snd_soc_dapm_context *d;
 	LIST_HEAD(up_list);
 	LIST_HEAD(down_list);
-	int ret = 0;
+	LIST_HEAD(async_domain);
 	int power;
 
 	trace_snd_soc_dapm_start(card);
@@ -1002,10 +1138,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 	list_for_each_entry(w, &card->widgets, list) {
 		switch (w->id) {
 		case snd_soc_dapm_pre:
-			dapm_seq_insert(w, &down_list, dapm_down_seq);
+			dapm_seq_insert(w, &down_list, false);
 			break;
 		case snd_soc_dapm_post:
-			dapm_seq_insert(w, &up_list, dapm_up_seq);
+			dapm_seq_insert(w, &up_list, true);
 			break;
 
 		default:
@@ -1025,9 +1161,9 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 			trace_snd_soc_dapm_widget_power(w, power);
 
 			if (power)
-				dapm_seq_insert(w, &up_list, dapm_up_seq);
+				dapm_seq_insert(w, &up_list, true);
 			else
-				dapm_seq_insert(w, &down_list, dapm_down_seq);
+				dapm_seq_insert(w, &down_list, false);
 
 			w->power = power;
 			break;
@@ -1065,65 +1201,25 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 		}
 	}
 
-	list_for_each_entry(d, &dapm->card->dapm_list, list) {
-		if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_STANDBY);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to turn on bias: %d\n", ret);
-		}
-
-		/* If we're changing to all on or all off then prepare */
-		if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
-		    (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_PREPARE);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to prepare bias: %d\n", ret);
-		}
-	}
+	/* Run all the bias changes in parallel */
+	list_for_each_entry(d, &dapm->card->dapm_list, list)
+		async_schedule_domain(dapm_pre_sequence_async, d,
+					&async_domain);
+	async_synchronize_full_domain(&async_domain);
 
 	/* Power down widgets first; try to avoid amplifying pops. */
-	dapm_seq_run(dapm, &down_list, event, dapm_down_seq);
+	dapm_seq_run(dapm, &down_list, event, false);
 
 	dapm_widget_update(dapm);
 
 	/* Now power up. */
-	dapm_seq_run(dapm, &up_list, event, dapm_up_seq);
-
-	list_for_each_entry(d, &dapm->card->dapm_list, list) {
-		/* If we just powered the last thing off drop to standby bias */
-		if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_STANDBY);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to apply standby bias: %d\n",
-					ret);
-		}
+	dapm_seq_run(dapm, &up_list, event, true);
 
-		/* If we're in standby and can support bias off then do that */
-		if (d->bias_level == SND_SOC_BIAS_STANDBY &&
-		    d->idle_bias_off) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_OFF);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to turn off bias: %d\n", ret);
-		}
-
-		/* If we just powered up then move to active bias */
-		if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_ON);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to apply active bias: %d\n",
-					ret);
-		}
-	}
+	/* Run all the bias changes in parallel */
+	list_for_each_entry(d, &dapm->card->dapm_list, list)
+		async_schedule_domain(dapm_post_sequence_async, d,
+					&async_domain);
+	async_synchronize_full_domain(&async_domain);
 
 	pop_dbg(dapm->dev, card->pop_time,
 		"DAPM sequencing finished, waiting %dms\n", card->pop_time);
@@ -1181,7 +1277,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 
 		if (p->connect)
 			ret += snprintf(buf + ret, PAGE_SIZE - ret,
-					" in  %s %s\n",
+					" in  \"%s\" \"%s\"\n",
 					p->name ? p->name : "static",
 					p->source->name);
 	}
@@ -1191,7 +1287,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 
 		if (p->connect)
 			ret += snprintf(buf + ret, PAGE_SIZE - ret,
-					" out %s %s\n",
+					" out \"%s\" \"%s\"\n",
 					p->name ? p->name : "static",
 					p->sink->name);
 	}
@@ -1456,7 +1552,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	char prefixed_source[80];
 	int ret = 0;
 
-	if (dapm->codec->name_prefix) {
+	if (dapm->codec && dapm->codec->name_prefix) {
 		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
 			 dapm->codec->name_prefix, route->sink);
 		sink = prefixed_sink;
@@ -1627,6 +1723,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
 int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_widget *w;
+	unsigned int val;
 
 	list_for_each_entry(w, &dapm->card->widgets, list)
 	{
@@ -1675,6 +1772,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 		case snd_soc_dapm_post:
 			break;
 		}
+
+		/* Read the initial power state from the device */
+		if (w->reg >= 0) {
+			val = snd_soc_read(w->codec, w->reg);
+			val &= 1 << w->shift;
+			if (w->invert)
+				val = !val;
+
+			if (val)
+				w->power = 1;
+		}
+
 		w->new = 1;
 	}
 
@@ -1742,7 +1851,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
-	unsigned int val, val_mask;
+	unsigned int val;
 	int connect, change;
 	struct snd_soc_dapm_update update;
 
@@ -1750,13 +1859,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 
 	if (invert)
 		val = max - val;
-	val_mask = mask << shift;
+	mask = mask << shift;
 	val = val << shift;
 
 	mutex_lock(&widget->codec->mutex);
 	widget->value = val;
 
-	change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+	change = snd_soc_test_bits(widget->codec, reg, mask, val);
 	if (change) {
 		if (val)
 			/* new connection */
@@ -2093,14 +2202,14 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		return -ENOMEM;
 
 	name_len = strlen(widget->name) + 1;
-	if (dapm->codec->name_prefix)
+	if (dapm->codec && dapm->codec->name_prefix)
 		name_len += 1 + strlen(dapm->codec->name_prefix);
 	w->name = kmalloc(name_len, GFP_KERNEL);
 	if (w->name == NULL) {
 		kfree(w);
 		return -ENOMEM;
 	}
-	if (dapm->codec->name_prefix)
+	if (dapm->codec && dapm->codec->name_prefix)
 		snprintf(w->name, name_len, "%s %s",
 			dapm->codec->name_prefix, widget->name);
 	else
@@ -2205,7 +2314,6 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
 	mutex_unlock(&codec->mutex);
 	return 0;
 }
-EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 
 /**
  * snd_soc_dapm_enable_pin - enable pin.
@@ -2372,7 +2480,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 		if (w->dapm != dapm)
 			continue;
 		if (w->power) {
-			dapm_seq_insert(w, &down_list, dapm_down_seq);
+			dapm_seq_insert(w, &down_list, false);
 			w->power = 0;
 			powerdown = 1;
 		}
@@ -2382,9 +2490,9 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 	 * standby.
 	 */
 	if (powerdown) {
-		snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE);
-		dapm_seq_run(dapm, &down_list, 0, dapm_down_seq);
-		snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY);
+		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+		dapm_seq_run(dapm, &down_list, 0, false);
+		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
 	}
 }
 
@@ -2397,7 +2505,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
 
 	list_for_each_entry(codec, &card->codec_dev_list, list) {
 		soc_dapm_shutdown_codec(&codec->dapm);
-		snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF);
+		snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
 	}
 }
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index ac5a5bc7375..fcab80b36a3 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
 {
 	jack->codec = codec;
 	INIT_LIST_HEAD(&jack->pins);
+	INIT_LIST_HEAD(&jack->jack_zones);
 	BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier);
 
 	return snd_jack_new(codec->card->snd_card, id, type, &jack->jack);
@@ -100,7 +101,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	}
 
 	/* Report before the DAPM sync to help users updating micbias status */
-	blocking_notifier_call_chain(&jack->notifier, status, NULL);
+	blocking_notifier_call_chain(&jack->notifier, status, jack);
 
 	snd_soc_dapm_sync(dapm);
 
@@ -112,6 +113,51 @@ out:
 EXPORT_SYMBOL_GPL(snd_soc_jack_report);
 
 /**
+ * snd_soc_jack_add_zones - Associate voltage zones with jack
+ *
+ * @jack:  ASoC jack
+ * @count: Number of zones
+ * @zone:  Array of zones
+ *
+ * After this function has been called the zones specified in the
+ * array will be associated with the jack.
+ */
+int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count,
+			  struct snd_soc_jack_zone *zones)
+{
+	int i;
+
+	for (i = 0; i < count; i++) {
+		INIT_LIST_HEAD(&zones[i].list);
+		list_add(&(zones[i].list), &jack->jack_zones);
+	}
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones);
+
+/**
+ * snd_soc_jack_get_type - Based on the mic bias value, this function returns
+ * the type of jack from the zones delcared in the jack type
+ *
+ * @micbias_voltage:  mic bias voltage at adc channel when jack is plugged in
+ *
+ * Based on the mic bias value passed, this function helps identify
+ * the type of jack from the already delcared jack zones
+ */
+int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage)
+{
+	struct snd_soc_jack_zone *zone;
+
+	list_for_each_entry(zone, &jack->jack_zones, list) {
+		if (micbias_voltage >= zone->min_mv &&
+			micbias_voltage < zone->max_mv)
+				return zone->jack_type;
+	}
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_get_type);
+
+/**
  * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack
  *
  * @jack:  ASoC jack
@@ -194,7 +240,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
 	int enable;
 	int report;
 
-	enable = gpio_get_value(gpio->gpio);
+	enable = gpio_get_value_cansleep(gpio->gpio);
 	if (gpio->invert)
 		enable = !enable;
 
@@ -284,6 +330,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 		if (ret)
 			goto err;
 
+		if (gpios[i].wake) {
+			ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
+			if (ret != 0)
+				printk(KERN_ERR
+				  "Failed to mark GPIO %d as wake source: %d\n",
+					gpios[i].gpio, ret);
+		}
+
 #ifdef CONFIG_GPIO_SYSFS
 		/* Expose GPIO value over sysfs for diagnostic purposes */
 		gpio_export(gpios[i].gpio, false);
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 1d07b931f3d..3f45e6a439b 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -28,26 +28,9 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params)
 {
 	int sample_size;
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-	case SNDRV_PCM_FORMAT_S16_BE:
-		sample_size = 16;
-		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
-	case SNDRV_PCM_FORMAT_S20_3BE:
-		sample_size = 20;
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-	case SNDRV_PCM_FORMAT_S24_BE:
-		sample_size = 24;
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-	case SNDRV_PCM_FORMAT_S32_BE:
-		sample_size = 32;
-		break;
-	default:
-		return -ENOTSUPP;
-	}
+	sample_size = snd_pcm_format_width(params_format(params));
+	if (sample_size < 0)
+		return sample_size;
 
 	return snd_soc_calc_frame_size(sample_size, params_channels(params),
 				       1);
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
new file mode 100644
index 00000000000..66b504f06c2
--- /dev/null
+++ b/sound/soc/tegra/Kconfig
@@ -0,0 +1,26 @@
+config SND_TEGRA_SOC
+	tristate "SoC Audio for the Tegra System-on-Chip"
+	depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA
+	default m
+	help
+	  Say Y or M here if you want support for SoC audio on Tegra.
+
+config SND_TEGRA_SOC_I2S
+	tristate
+	depends on SND_TEGRA_SOC
+	default m
+	help
+	  Say Y or M if you want to add support for codecs attached to the
+	  Tegra I2S interface. You will also need to select the individual
+	  machine drivers to support below.
+
+config SND_TEGRA_SOC_HARMONY
+	tristate "SoC Audio support for Tegra Harmony reference board"
+	depends on SND_TEGRA_SOC && MACH_HARMONY && I2C
+	default m
+	select SND_TEGRA_SOC_I2S
+	select SND_SOC_WM8903
+	help
+	  Say Y or M here if you want to add support for SoC audio on the
+	  Tegra Harmony reference board.
+
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
new file mode 100644
index 00000000000..fd183d3ab4f
--- /dev/null
+++ b/sound/soc/tegra/Makefile
@@ -0,0 +1,15 @@
+# Tegra platform Support
+snd-soc-tegra-das-objs := tegra_das.o
+snd-soc-tegra-pcm-objs := tegra_pcm.o
+snd-soc-tegra-i2s-objs := tegra_i2s.o
+snd-soc-tegra-utils-objs += tegra_asoc_utils.o
+
+obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-utils.o
+obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-das.o
+obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o
+obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o
+
+# Tegra machine Support
+snd-soc-tegra-harmony-objs := harmony.o
+
+obj-$(CONFIG_SND_TEGRA_SOC_HARMONY) += snd-soc-tegra-harmony.o
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
new file mode 100644
index 00000000000..8585957477e
--- /dev/null
+++ b/sound/soc/tegra/harmony.c
@@ -0,0 +1,393 @@
+/*
+ * harmony.c - Harmony machine ASoC driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010-2011 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd.
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <asm/mach-types.h>
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+
+#include <mach/harmony_audio.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "../codecs/wm8903.h"
+
+#include "tegra_das.h"
+#include "tegra_i2s.h"
+#include "tegra_pcm.h"
+#include "tegra_asoc_utils.h"
+
+#define DRV_NAME "tegra-snd-harmony"
+
+#define GPIO_SPKR_EN    BIT(0)
+#define GPIO_INT_MIC_EN BIT(1)
+#define GPIO_EXT_MIC_EN BIT(2)
+
+struct tegra_harmony {
+	struct tegra_asoc_utils_data util_data;
+	struct harmony_audio_platform_data *pdata;
+	int gpio_requested;
+};
+
+static int harmony_asoc_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;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = codec->card;
+	struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
+	int srate, mclk, mclk_change;
+	int err;
+
+	srate = params_rate(params);
+	switch (srate) {
+	case 64000:
+	case 88200:
+	case 96000:
+		mclk = 128 * srate;
+		break;
+	default:
+		mclk = 256 * srate;
+		break;
+	}
+	/* FIXME: Codec only requires >= 3MHz if OSR==0 */
+	while (mclk < 6000000)
+		mclk *= 2;
+
+	err = tegra_asoc_utils_set_rate(&harmony->util_data, srate, mclk,
+					&mclk_change);
+	if (err < 0) {
+		dev_err(card->dev, "Can't configure clocks\n");
+		return err;
+	}
+
+	err = snd_soc_dai_set_fmt(codec_dai,
+					SND_SOC_DAIFMT_I2S |
+					SND_SOC_DAIFMT_NB_NF |
+					SND_SOC_DAIFMT_CBS_CFS);
+	if (err < 0) {
+		dev_err(card->dev, "codec_dai fmt not set\n");
+		return err;
+	}
+
+	err = snd_soc_dai_set_fmt(cpu_dai,
+					SND_SOC_DAIFMT_I2S |
+					SND_SOC_DAIFMT_NB_NF |
+					SND_SOC_DAIFMT_CBS_CFS);
+	if (err < 0) {
+		dev_err(card->dev, "cpu_dai fmt not set\n");
+		return err;
+	}
+
+	if (mclk_change) {
+		err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+					     SND_SOC_CLOCK_IN);
+		if (err < 0) {
+			dev_err(card->dev, "codec_dai clock not set\n");
+			return err;
+		}
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops harmony_asoc_ops = {
+	.hw_params = harmony_asoc_hw_params,
+};
+
+static struct snd_soc_jack harmony_hp_jack;
+
+static struct snd_soc_jack_pin harmony_hp_jack_pins[] = {
+	{
+		.pin = "Headphone Jack",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static struct snd_soc_jack_gpio harmony_hp_jack_gpios[] = {
+	{
+		.name = "headphone detect",
+		.report = SND_JACK_HEADPHONE,
+		.debounce_time = 150,
+		.invert = 1,
+	}
+};
+
+static struct snd_soc_jack harmony_mic_jack;
+
+static struct snd_soc_jack_pin harmony_mic_jack_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+static int harmony_event_int_spk(struct snd_soc_dapm_widget *w,
+					struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct snd_soc_card *card = codec->card;
+	struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
+	struct harmony_audio_platform_data *pdata = harmony->pdata;
+
+	gpio_set_value_cansleep(pdata->gpio_spkr_en,
+				SND_SOC_DAPM_EVENT_ON(event));
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Int Spk", harmony_event_int_spk),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route harmony_audio_map[] = {
+	{"Headphone Jack", NULL, "HPOUTR"},
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Int Spk", NULL, "ROP"},
+	{"Int Spk", NULL, "RON"},
+	{"Int Spk", NULL, "LOP"},
+	{"Int Spk", NULL, "LON"},
+	{"Mic Bias", NULL, "Mic Jack"},
+	{"IN1L", NULL, "Mic Bias"},
+};
+
+static const struct snd_kcontrol_new harmony_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Int Spk"),
+};
+
+static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_card *card = codec->card;
+	struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
+	struct harmony_audio_platform_data *pdata = harmony->pdata;
+	int ret;
+
+	ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
+	if (ret) {
+		dev_err(card->dev, "cannot get spkr_en gpio\n");
+		return ret;
+	}
+	harmony->gpio_requested |= GPIO_SPKR_EN;
+
+	gpio_direction_output(pdata->gpio_spkr_en, 0);
+
+	ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en");
+	if (ret) {
+		dev_err(card->dev, "cannot get int_mic_en gpio\n");
+		return ret;
+	}
+	harmony->gpio_requested |= GPIO_INT_MIC_EN;
+
+	/* Disable int mic; enable signal is active-high */
+	gpio_direction_output(pdata->gpio_int_mic_en, 0);
+
+	ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en");
+	if (ret) {
+		dev_err(card->dev, "cannot get ext_mic_en gpio\n");
+		return ret;
+	}
+	harmony->gpio_requested |= GPIO_EXT_MIC_EN;
+
+	/* Enable ext mic; enable signal is active-low */
+	gpio_direction_output(pdata->gpio_ext_mic_en, 0);
+
+	ret = snd_soc_add_controls(codec, harmony_controls,
+				   ARRAY_SIZE(harmony_controls));
+	if (ret < 0)
+		return ret;
+
+	snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets,
+					ARRAY_SIZE(harmony_dapm_widgets));
+
+	snd_soc_dapm_add_routes(dapm, harmony_audio_map,
+				ARRAY_SIZE(harmony_audio_map));
+
+	harmony_hp_jack_gpios[0].gpio = pdata->gpio_hp_det;
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
+			 &harmony_hp_jack);
+	snd_soc_jack_add_pins(&harmony_hp_jack,
+			      ARRAY_SIZE(harmony_hp_jack_pins),
+			      harmony_hp_jack_pins);
+	snd_soc_jack_add_gpios(&harmony_hp_jack,
+			       ARRAY_SIZE(harmony_hp_jack_gpios),
+			       harmony_hp_jack_gpios);
+
+	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
+			 &harmony_mic_jack);
+	snd_soc_jack_add_pins(&harmony_mic_jack,
+			      ARRAY_SIZE(harmony_mic_jack_pins),
+			      harmony_mic_jack_pins);
+	wm8903_mic_detect(codec, &harmony_mic_jack, SND_JACK_MICROPHONE, 0);
+
+	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+
+	snd_soc_dapm_nc_pin(dapm, "IN3L");
+	snd_soc_dapm_nc_pin(dapm, "IN3R");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUTL");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUTR");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link harmony_wm8903_dai = {
+	.name = "WM8903",
+	.stream_name = "WM8903 PCM",
+	.codec_name = "wm8903.0-001a",
+	.platform_name = "tegra-pcm-audio",
+	.cpu_dai_name = "tegra-i2s.0",
+	.codec_dai_name = "wm8903-hifi",
+	.init = harmony_asoc_init,
+	.ops = &harmony_asoc_ops,
+};
+
+static struct snd_soc_card snd_soc_harmony = {
+	.name = "tegra-harmony",
+	.dai_link = &harmony_wm8903_dai,
+	.num_links = 1,
+};
+
+static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &snd_soc_harmony;
+	struct tegra_harmony *harmony;
+	struct harmony_audio_platform_data *pdata;
+	int ret;
+
+	if (!machine_is_harmony()) {
+		dev_err(&pdev->dev, "Not running on Tegra Harmony!\n");
+		return -ENODEV;
+	}
+
+	pdata = pdev->dev.platform_data;
+	if (!pdata) {
+		dev_err(&pdev->dev, "no platform data supplied\n");
+		return -EINVAL;
+	}
+
+	harmony = kzalloc(sizeof(struct tegra_harmony), GFP_KERNEL);
+	if (!harmony) {
+		dev_err(&pdev->dev, "Can't allocate tegra_harmony\n");
+		return -ENOMEM;
+	}
+
+	harmony->pdata = pdata;
+
+	ret = tegra_asoc_utils_init(&harmony->util_data, &pdev->dev);
+	if (ret)
+		goto err_free_harmony;
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, harmony);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+			ret);
+		goto err_clear_drvdata;
+	}
+
+	return 0;
+
+err_clear_drvdata:
+	snd_soc_card_set_drvdata(card, NULL);
+	platform_set_drvdata(pdev, NULL);
+	card->dev = NULL;
+	tegra_asoc_utils_fini(&harmony->util_data);
+err_free_harmony:
+	kfree(harmony);
+	return ret;
+}
+
+static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
+	struct harmony_audio_platform_data *pdata = harmony->pdata;
+
+	snd_soc_unregister_card(card);
+
+	snd_soc_card_set_drvdata(card, NULL);
+	platform_set_drvdata(pdev, NULL);
+	card->dev = NULL;
+
+	tegra_asoc_utils_fini(&harmony->util_data);
+
+	if (harmony->gpio_requested & GPIO_EXT_MIC_EN)
+		gpio_free(pdata->gpio_ext_mic_en);
+	if (harmony->gpio_requested & GPIO_INT_MIC_EN)
+		gpio_free(pdata->gpio_int_mic_en);
+	if (harmony->gpio_requested & GPIO_SPKR_EN)
+		gpio_free(pdata->gpio_spkr_en);
+
+	kfree(harmony);
+
+	return 0;
+}
+
+static struct platform_driver tegra_snd_harmony_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = tegra_snd_harmony_probe,
+	.remove = __devexit_p(tegra_snd_harmony_remove),
+};
+
+static int __init snd_tegra_harmony_init(void)
+{
+	return platform_driver_register(&tegra_snd_harmony_driver);
+}
+module_init(snd_tegra_harmony_init);
+
+static void __exit snd_tegra_harmony_exit(void)
+{
+	platform_driver_unregister(&tegra_snd_harmony_driver);
+}
+module_exit(snd_tegra_harmony_exit);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Harmony machine ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c
new file mode 100644
index 00000000000..52f0a3f9ce4
--- /dev/null
+++ b/sound/soc/tegra/tegra_asoc_utils.c
@@ -0,0 +1,155 @@
+/*
+ * tegra_asoc_utils.c - Harmony machine ASoC driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/err.h>
+#include <linux/kernel.h>
+
+#include "tegra_asoc_utils.h"
+
+int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate,
+			      int mclk, int *mclk_change)
+{
+	int new_baseclock;
+	int err;
+
+	switch (srate) {
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+		new_baseclock = 56448000;
+		break;
+	case 8000:
+	case 16000:
+	case 32000:
+	case 48000:
+	case 64000:
+	case 96000:
+		new_baseclock = 73728000;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	*mclk_change = ((new_baseclock != data->set_baseclock) ||
+			(mclk != data->set_mclk));
+	if (!*mclk_change)
+	    return 0;
+
+	data->set_baseclock = 0;
+	data->set_mclk = 0;
+
+	clk_disable(data->clk_cdev1);
+	clk_disable(data->clk_pll_a_out0);
+	clk_disable(data->clk_pll_a);
+
+	err = clk_set_rate(data->clk_pll_a, new_baseclock);
+	if (err) {
+		dev_err(data->dev, "Can't set pll_a rate: %d\n", err);
+		return err;
+	}
+
+	err = clk_set_rate(data->clk_pll_a_out0, mclk);
+	if (err) {
+		dev_err(data->dev, "Can't set pll_a_out0 rate: %d\n", err);
+		return err;
+	}
+
+	/* Don't set cdev1 rate; its locked to pll_a_out0 */
+
+	err = clk_enable(data->clk_pll_a);
+	if (err) {
+		dev_err(data->dev, "Can't enable pll_a: %d\n", err);
+		return err;
+	}
+
+	err = clk_enable(data->clk_pll_a_out0);
+	if (err) {
+		dev_err(data->dev, "Can't enable pll_a_out0: %d\n", err);
+		return err;
+	}
+
+	err = clk_enable(data->clk_cdev1);
+	if (err) {
+		dev_err(data->dev, "Can't enable cdev1: %d\n", err);
+		return err;
+	}
+
+	data->set_baseclock = new_baseclock;
+	data->set_mclk = mclk;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_rate);
+
+int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
+			  struct device *dev)
+{
+	int ret;
+
+	data->dev = dev;
+
+	data->clk_pll_a = clk_get_sys(NULL, "pll_a");
+	if (IS_ERR(data->clk_pll_a)) {
+		dev_err(data->dev, "Can't retrieve clk pll_a\n");
+		ret = PTR_ERR(data->clk_pll_a);
+		goto err;
+	}
+
+	data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0");
+	if (IS_ERR(data->clk_pll_a_out0)) {
+		dev_err(data->dev, "Can't retrieve clk pll_a_out0\n");
+		ret = PTR_ERR(data->clk_pll_a_out0);
+		goto err_put_pll_a;
+	}
+
+	data->clk_cdev1 = clk_get_sys(NULL, "cdev1");
+	if (IS_ERR(data->clk_cdev1)) {
+		dev_err(data->dev, "Can't retrieve clk cdev1\n");
+		ret = PTR_ERR(data->clk_cdev1);
+		goto err_put_pll_a_out0;
+	}
+
+	return 0;
+
+err_put_pll_a_out0:
+	clk_put(data->clk_pll_a_out0);
+err_put_pll_a:
+	clk_put(data->clk_pll_a);
+err:
+	return ret;
+}
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_init);
+
+void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data)
+{
+	clk_put(data->clk_cdev1);
+	clk_put(data->clk_pll_a_out0);
+	clk_put(data->clk_pll_a);
+}
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_fini);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra ASoC utility code");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
new file mode 100644
index 00000000000..bbba7afdfc2
--- /dev/null
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -0,0 +1,45 @@
+/*
+ * tegra_asoc_utils.h - Definitions for Tegra DAS driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TEGRA_ASOC_UTILS_H__
+#define __TEGRA_ASOC_UTILS_H_
+
+struct clk;
+struct device;
+
+struct tegra_asoc_utils_data {
+	struct device *dev;
+	struct clk *clk_pll_a;
+	struct clk *clk_pll_a_out0;
+	struct clk *clk_cdev1;
+	int set_baseclock;
+	int set_mclk;
+};
+
+int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate,
+			      int mclk, int *mclk_change);
+int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
+			  struct device *dev);
+void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data);
+
+#endif
+
diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c
new file mode 100644
index 00000000000..9f24ef73f2c
--- /dev/null
+++ b/sound/soc/tegra/tegra_das.c
@@ -0,0 +1,265 @@
+/*
+ * tegra_das.c - Tegra DAS driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/debugfs.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/seq_file.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <mach/iomap.h>
+#include <sound/soc.h>
+#include "tegra_das.h"
+
+#define DRV_NAME "tegra-das"
+
+static struct tegra_das *das;
+
+static inline void tegra_das_write(u32 reg, u32 val)
+{
+	__raw_writel(val, das->regs + reg);
+}
+
+static inline u32 tegra_das_read(u32 reg)
+{
+	return __raw_readl(das->regs + reg);
+}
+
+int tegra_das_connect_dap_to_dac(int dap, int dac)
+{
+	u32 addr;
+	u32 reg;
+
+	if (!das)
+		return -ENODEV;
+
+	addr = TEGRA_DAS_DAP_CTRL_SEL +
+		(dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE);
+	reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P;
+
+	tegra_das_write(addr, reg);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac);
+
+int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master,
+					int sdata1rx, int sdata2rx)
+{
+	u32 addr;
+	u32 reg;
+
+	if (!das)
+		return -ENODEV;
+
+	addr = TEGRA_DAS_DAP_CTRL_SEL +
+		(dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE);
+	reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P |
+		!!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P |
+		!!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P |
+		!!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P;
+
+	tegra_das_write(addr, reg);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap);
+
+int tegra_das_connect_dac_to_dap(int dac, int dap)
+{
+	u32 addr;
+	u32 reg;
+
+	if (!das)
+		return -ENODEV;
+
+	addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL +
+		(dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE);
+	reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P |
+		dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P |
+		dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P;
+
+	tegra_das_write(addr, reg);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap);
+
+#ifdef CONFIG_DEBUG_FS
+static int tegra_das_show(struct seq_file *s, void *unused)
+{
+	int i;
+	u32 addr;
+	u32 reg;
+
+	for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) {
+		addr = TEGRA_DAS_DAP_CTRL_SEL +
+			(i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE);
+		reg = tegra_das_read(addr);
+		seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg);
+	}
+
+	for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) {
+		addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL +
+			(i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE);
+		reg = tegra_das_read(addr);
+		seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n",
+				 i, reg);
+	}
+
+	return 0;
+}
+
+static int tegra_das_debug_open(struct inode *inode, struct file *file)
+{
+	return single_open(file, tegra_das_show, inode->i_private);
+}
+
+static const struct file_operations tegra_das_debug_fops = {
+	.open    = tegra_das_debug_open,
+	.read    = seq_read,
+	.llseek  = seq_lseek,
+	.release = single_release,
+};
+
+static void tegra_das_debug_add(struct tegra_das *das)
+{
+	das->debug = debugfs_create_file(DRV_NAME, S_IRUGO,
+					 snd_soc_debugfs_root, das,
+					 &tegra_das_debug_fops);
+}
+
+static void tegra_das_debug_remove(struct tegra_das *das)
+{
+	if (das->debug)
+		debugfs_remove(das->debug);
+}
+#else
+static inline void tegra_das_debug_add(struct tegra_das *das)
+{
+}
+
+static inline void tegra_das_debug_remove(struct tegra_das *das)
+{
+}
+#endif
+
+static int __devinit tegra_das_probe(struct platform_device *pdev)
+{
+	struct resource *res, *region;
+	int ret = 0;
+
+	if (das)
+		return -ENODEV;
+
+	das = kzalloc(sizeof(struct tegra_das), GFP_KERNEL);
+	if (!das) {
+		dev_err(&pdev->dev, "Can't allocate tegra_das\n");
+		ret = -ENOMEM;
+		goto exit;
+	}
+	das->dev = &pdev->dev;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		dev_err(&pdev->dev, "No memory resource\n");
+		ret = -ENODEV;
+		goto err_free;
+	}
+
+	region = request_mem_region(res->start, resource_size(res),
+					pdev->name);
+	if (!region) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		ret = -EBUSY;
+		goto err_free;
+	}
+
+	das->regs = ioremap(res->start, resource_size(res));
+	if (!das->regs) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_release;
+	}
+
+	tegra_das_debug_add(das);
+
+	platform_set_drvdata(pdev, das);
+
+	return 0;
+
+err_release:
+	release_mem_region(res->start, resource_size(res));
+err_free:
+	kfree(das);
+	das = 0;
+exit:
+	return ret;
+}
+
+static int __devexit tegra_das_remove(struct platform_device *pdev)
+{
+	struct resource *res;
+
+	if (!das)
+		return -ENODEV;
+
+	platform_set_drvdata(pdev, NULL);
+
+	tegra_das_debug_remove(das);
+
+	iounmap(das->regs);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	release_mem_region(res->start, resource_size(res));
+
+	kfree(das);
+	das = 0;
+
+	return 0;
+}
+
+static struct platform_driver tegra_das_driver = {
+	.probe = tegra_das_probe,
+	.remove = __devexit_p(tegra_das_remove),
+	.driver = {
+		.name = DRV_NAME,
+	},
+};
+
+static int __init tegra_das_modinit(void)
+{
+	return platform_driver_register(&tegra_das_driver);
+}
+module_init(tegra_das_modinit);
+
+static void __exit tegra_das_modexit(void)
+{
+	platform_driver_unregister(&tegra_das_driver);
+}
+module_exit(tegra_das_modexit);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra DAS driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra_das.h
new file mode 100644
index 00000000000..2c96c7b3c45
--- /dev/null
+++ b/sound/soc/tegra/tegra_das.h
@@ -0,0 +1,135 @@
+/*
+ * tegra_das.h - Definitions for Tegra DAS driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TEGRA_DAS_H__
+#define __TEGRA_DAS_H__
+
+/* Register TEGRA_DAS_DAP_CTRL_SEL */
+#define TEGRA_DAS_DAP_CTRL_SEL				0x00
+#define TEGRA_DAS_DAP_CTRL_SEL_COUNT			5
+#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE			4
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P		31
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S		1
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P	30
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S	1
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P	29
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S	1
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P		0
+#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S		5
+
+/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */
+#define TEGRA_DAS_DAP_SEL_DAC1	0
+#define TEGRA_DAS_DAP_SEL_DAC2	1
+#define TEGRA_DAS_DAP_SEL_DAC3	2
+#define TEGRA_DAS_DAP_SEL_DAP1	16
+#define TEGRA_DAS_DAP_SEL_DAP2	17
+#define TEGRA_DAS_DAP_SEL_DAP3	18
+#define TEGRA_DAS_DAP_SEL_DAP4	19
+#define TEGRA_DAS_DAP_SEL_DAP5	20
+
+/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL			0x40
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT			3
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE			4
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P	28
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S	4
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P	24
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S	4
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P		0
+#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S		4
+
+/*
+ * Values for:
+ * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL
+ * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL
+ * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL
+ */
+#define TEGRA_DAS_DAC_SEL_DAP1	0
+#define TEGRA_DAS_DAC_SEL_DAP2	1
+#define TEGRA_DAS_DAC_SEL_DAP3	2
+#define TEGRA_DAS_DAC_SEL_DAP4	3
+#define TEGRA_DAS_DAC_SEL_DAP5	4
+
+/*
+ * Names/IDs of the DACs/DAPs.
+ */
+
+#define TEGRA_DAS_DAP_ID_1 0
+#define TEGRA_DAS_DAP_ID_2 1
+#define TEGRA_DAS_DAP_ID_3 2
+#define TEGRA_DAS_DAP_ID_4 3
+#define TEGRA_DAS_DAP_ID_5 4
+
+#define TEGRA_DAS_DAC_ID_1 0
+#define TEGRA_DAS_DAC_ID_2 1
+#define TEGRA_DAS_DAC_ID_3 2
+
+struct tegra_das {
+	struct device *dev;
+	void __iomem *regs;
+	struct dentry *debug;
+};
+
+/*
+ * Terminology:
+ * DAS: Digital audio switch (HW module controlled by this driver)
+ * DAP: Digital audio port (port/pins on Tegra device)
+ * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere)
+ * 
+ * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific
+ * DAC, or another DAP. When DAPs are connected, one must be the master and
+ * one the slave. Each DAC allows selection of a specific DAP for input, to
+ * cater for the case where N DAPs are connected to 1 DAC for broadcast
+ * output.
+ *
+ * This driver is dumb; no attempt is made to ensure that a valid routing
+ * configuration is programmed.
+ */
+
+/*
+ * Connect a DAP to to a DAC
+ * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_*
+ * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC*
+ */
+extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel);
+
+/*
+ * Connect a DAP to to another DAP
+ * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_*
+ * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP*
+ * master: Is this DAP the master (1) or slave (0)
+ * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0)
+ * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0)
+ */
+extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel,
+					int master, int sdata1rx,
+					int sdata2rx);
+
+/*
+ * Connect a DAC's input to a DAP
+ * (DAC outputs are selected by the DAP)
+ * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_*
+ * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP*
+ */
+extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel);
+
+#endif
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
new file mode 100644
index 00000000000..4f5e2c90b02
--- /dev/null
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -0,0 +1,503 @@
+/*
+ * tegra_i2s.c - Tegra I2S driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (c) 2009-2010, NVIDIA Corporation.
+ * Scott Peterson <speterson@nvidia.com>
+ *
+ * Copyright (C) 2010 Google, Inc.
+ * Iliyan Malchev <malchev@google.com>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/debugfs.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/seq_file.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <mach/iomap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "tegra_das.h"
+#include "tegra_i2s.h"
+
+#define DRV_NAME "tegra-i2s"
+
+static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val)
+{
+	__raw_writel(val, i2s->regs + reg);
+}
+
+static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg)
+{
+	return __raw_readl(i2s->regs + reg);
+}
+
+#ifdef CONFIG_DEBUG_FS
+static int tegra_i2s_show(struct seq_file *s, void *unused)
+{
+#define REG(r) { r, #r }
+	static const struct {
+		int offset;
+		const char *name;
+	} regs[] = {
+		REG(TEGRA_I2S_CTRL),
+		REG(TEGRA_I2S_STATUS),
+		REG(TEGRA_I2S_TIMING),
+		REG(TEGRA_I2S_FIFO_SCR),
+		REG(TEGRA_I2S_PCM_CTRL),
+		REG(TEGRA_I2S_NW_CTRL),
+		REG(TEGRA_I2S_TDM_CTRL),
+		REG(TEGRA_I2S_TDM_TX_RX_CTRL),
+	};
+#undef REG
+
+	struct tegra_i2s *i2s = s->private;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(regs); i++) {
+		u32 val = tegra_i2s_read(i2s, regs[i].offset);
+		seq_printf(s, "%s = %08x\n", regs[i].name, val);
+	}
+
+	return 0;
+}
+
+static int tegra_i2s_debug_open(struct inode *inode, struct file *file)
+{
+	return single_open(file, tegra_i2s_show, inode->i_private);
+}
+
+static const struct file_operations tegra_i2s_debug_fops = {
+	.open    = tegra_i2s_debug_open,
+	.read    = seq_read,
+	.llseek  = seq_lseek,
+	.release = single_release,
+};
+
+static void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id)
+{
+	char name[] = DRV_NAME ".0";
+
+	snprintf(name, sizeof(name), DRV_NAME".%1d", id);
+	i2s->debug = debugfs_create_file(name, S_IRUGO, snd_soc_debugfs_root,
+						i2s, &tegra_i2s_debug_fops);
+}
+
+static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
+{
+	if (i2s->debug)
+		debugfs_remove(i2s->debug);
+}
+#else
+static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s)
+{
+}
+
+static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
+{
+}
+#endif
+
+static int tegra_i2s_set_fmt(struct snd_soc_dai *dai,
+				unsigned int fmt)
+{
+	struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE;
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | 
+				TEGRA_I2S_CTRL_LRCK_MASK);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP;
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP;
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S;
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM;
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM;
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+        struct device *dev = substream->pcm->card->dev;
+	struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	u32 reg;
+	int ret, sample_size, srate, i2sclock, bitcnt;
+
+	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16;
+		sample_size = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24;
+		sample_size = 24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32;
+		sample_size = 32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	srate = params_rate(params);
+
+	/* Final "* 2" required by Tegra hardware */
+	i2sclock = srate * params_channels(params) * sample_size * 2;
+
+	ret = clk_set_rate(i2s->clk_i2s, i2sclock);
+	if (ret) {
+		dev_err(dev, "Can't set I2S clock rate: %d\n", ret);
+		return ret;
+	}
+
+	bitcnt = (i2sclock / (2 * srate)) - 1;
+	if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US)
+		return -EINVAL;
+	reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
+
+	if (i2sclock % (2 * srate))
+		reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
+
+	tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
+
+	tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
+		TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
+		TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+
+	return 0;
+}
+
+static void tegra_i2s_start_playback(struct tegra_i2s *i2s)
+{
+	i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE;
+	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra_i2s_stop_playback(struct tegra_i2s *i2s)
+{
+	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE;
+	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra_i2s_start_capture(struct tegra_i2s *i2s)
+{
+	i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE;
+	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static void tegra_i2s_stop_capture(struct tegra_i2s *i2s)
+{
+	i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE;
+	tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl);
+}
+
+static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		if (!i2s->clk_refs)
+			clk_enable(i2s->clk_i2s);
+		i2s->clk_refs++;
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			tegra_i2s_start_playback(i2s);
+		else
+			tegra_i2s_start_capture(i2s);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			tegra_i2s_stop_playback(i2s);
+		else
+			tegra_i2s_stop_capture(i2s);
+		i2s->clk_refs--;
+		if (!i2s->clk_refs)
+			clk_disable(i2s->clk_i2s);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra_i2s_probe(struct snd_soc_dai *dai)
+{
+	struct tegra_i2s * i2s = snd_soc_dai_get_drvdata(dai);
+
+	dai->capture_dma_data = &i2s->capture_dma_data;
+	dai->playback_dma_data = &i2s->playback_dma_data;
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops tegra_i2s_dai_ops = {
+	.set_fmt	= tegra_i2s_set_fmt,
+	.hw_params	= tegra_i2s_hw_params,
+	.trigger	= tegra_i2s_trigger,
+};
+
+struct snd_soc_dai_driver tegra_i2s_dai[] = {
+	{
+		.name = DRV_NAME ".0",
+		.probe = tegra_i2s_probe,
+		.playback = {
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.capture = {
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.ops = &tegra_i2s_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = DRV_NAME ".1",
+		.probe = tegra_i2s_probe,
+		.playback = {
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.capture = {
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.ops = &tegra_i2s_dai_ops,
+		.symmetric_rates = 1,
+	},
+};
+
+static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
+{
+	struct tegra_i2s * i2s;
+	char clk_name[12]; /* tegra-i2s.0 */
+	struct resource *mem, *memregion, *dmareq;
+	int ret;
+
+	if ((pdev->id < 0) ||
+		(pdev->id >= ARRAY_SIZE(tegra_i2s_dai))) {
+		dev_err(&pdev->dev, "ID %d out of range\n", pdev->id);
+		return -EINVAL;
+	}
+
+	/*
+	 * FIXME: Until a codec driver exists for the tegra DAS, hard-code a
+	 * 1:1 mapping between audio controllers and audio ports.
+	 */
+	ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1 + pdev->id,
+					TEGRA_DAS_DAP_SEL_DAC1 + pdev->id);
+	if (ret) {
+		dev_err(&pdev->dev, "Can't set up DAP connection\n");
+		return ret;
+	}
+	ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1 + pdev->id,
+					TEGRA_DAS_DAC_SEL_DAP1 + pdev->id);
+	if (ret) {
+		dev_err(&pdev->dev, "Can't set up DAC connection\n");
+		return ret;
+	}
+
+	i2s = kzalloc(sizeof(struct tegra_i2s), GFP_KERNEL);
+	if (!i2s) {
+		dev_err(&pdev->dev, "Can't allocate tegra_i2s\n");
+		ret = -ENOMEM;
+		goto exit;
+	}
+	dev_set_drvdata(&pdev->dev, i2s);
+
+	snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id);
+	i2s->clk_i2s = clk_get_sys(clk_name, NULL);
+	if (IS_ERR(i2s->clk_i2s)) {
+		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
+		ret = PTR_ERR(i2s->clk_i2s);
+		goto err_free;
+	}
+
+	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem) {
+		dev_err(&pdev->dev, "No memory resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmareq) {
+		dev_err(&pdev->dev, "No DMA resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	memregion = request_mem_region(mem->start, resource_size(mem),
+					DRV_NAME);
+	if (!memregion) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		ret = -EBUSY;
+		goto err_clk_put;
+	}
+
+	i2s->regs = ioremap(mem->start, resource_size(mem));
+	if (!i2s->regs) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_release;
+	}
+
+	i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2;
+	i2s->capture_dma_data.wrap = 4;
+	i2s->capture_dma_data.width = 32;
+	i2s->capture_dma_data.req_sel = dmareq->start;
+
+	i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1;
+	i2s->playback_dma_data.wrap = 4;
+	i2s->playback_dma_data.width = 32;
+	i2s->playback_dma_data.req_sel = dmareq->start;
+
+	i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED;
+
+	ret = snd_soc_register_dai(&pdev->dev, &tegra_i2s_dai[pdev->id]);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
+		ret = -ENOMEM;
+		goto err_unmap;
+	}
+
+	tegra_i2s_debug_add(i2s, pdev->id);
+
+	return 0;
+
+err_unmap:
+	iounmap(i2s->regs);
+err_release:
+	release_mem_region(mem->start, resource_size(mem));
+err_clk_put:
+	clk_put(i2s->clk_i2s);
+err_free:
+	kfree(i2s);
+exit:
+	return ret;
+}
+
+static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev)
+{
+	struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev);
+	struct resource *res;
+
+	snd_soc_unregister_dai(&pdev->dev);
+
+	tegra_i2s_debug_remove(i2s);
+
+	iounmap(i2s->regs);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	release_mem_region(res->start, resource_size(res));
+
+	clk_put(i2s->clk_i2s);
+
+	kfree(i2s);
+
+	return 0;
+}
+
+static struct platform_driver tegra_i2s_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = tegra_i2s_platform_probe,
+	.remove = __devexit_p(tegra_i2s_platform_remove),
+};
+
+static int __init snd_tegra_i2s_init(void)
+{
+	return platform_driver_register(&tegra_i2s_driver);
+}
+module_init(snd_tegra_i2s_init);
+
+static void __exit snd_tegra_i2s_exit(void)
+{
+	platform_driver_unregister(&tegra_i2s_driver);
+}
+module_exit(snd_tegra_i2s_exit);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra I2S ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h
new file mode 100644
index 00000000000..2b38a096f46
--- /dev/null
+++ b/sound/soc/tegra/tegra_i2s.h
@@ -0,0 +1,165 @@
+/*
+ * tegra_i2s.h - Definitions for Tegra I2S driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (c) 2009-2010, NVIDIA Corporation.
+ * Scott Peterson <speterson@nvidia.com>
+ *
+ * Copyright (C) 2010 Google, Inc.
+ * Iliyan Malchev <malchev@google.com>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TEGRA_I2S_H__
+#define __TEGRA_I2S_H__
+
+#include "tegra_pcm.h"
+
+/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */
+
+#define TEGRA_I2S_CTRL					0x00
+#define TEGRA_I2S_STATUS				0x04
+#define TEGRA_I2S_TIMING				0x08
+#define TEGRA_I2S_FIFO_SCR				0x0c
+#define TEGRA_I2S_PCM_CTRL				0x10
+#define TEGRA_I2S_NW_CTRL				0x14
+#define TEGRA_I2S_TDM_CTRL				0x20
+#define TEGRA_I2S_TDM_TX_RX_CTRL			0x24
+#define TEGRA_I2S_FIFO1					0x40
+#define TEGRA_I2S_FIFO2					0x80
+
+/* Fields in TEGRA_I2S_CTRL */
+
+#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE			(1 << 30)
+#define TEGRA_I2S_CTRL_FIFO1_ENABLE			(1 << 29)
+#define TEGRA_I2S_CTRL_FIFO2_ENABLE			(1 << 28)
+#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE			(1 << 27)
+#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE			(1 << 26)
+#define TEGRA_I2S_CTRL_MASTER_ENABLE			(1 << 25)
+
+#define TEGRA_I2S_LRCK_LEFT_LOW				0
+#define TEGRA_I2S_LRCK_RIGHT_LOW			1
+
+#define TEGRA_I2S_CTRL_LRCK_SHIFT			24
+#define TEGRA_I2S_CTRL_LRCK_MASK			(1                        << TEGRA_I2S_CTRL_LRCK_SHIFT)
+#define TEGRA_I2S_CTRL_LRCK_L_LOW			(TEGRA_I2S_LRCK_LEFT_LOW  << TEGRA_I2S_CTRL_LRCK_SHIFT)
+#define TEGRA_I2S_CTRL_LRCK_R_LOW			(TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT)
+
+#define TEGRA_I2S_BIT_FORMAT_I2S			0
+#define TEGRA_I2S_BIT_FORMAT_RJM			1
+#define TEGRA_I2S_BIT_FORMAT_LJM			2
+#define TEGRA_I2S_BIT_FORMAT_DSP			3
+
+#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT			10
+#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK			(3                        << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S			(TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM			(TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM			(TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP			(TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT)
+
+#define TEGRA_I2S_BIT_SIZE_16				0
+#define TEGRA_I2S_BIT_SIZE_20				1
+#define TEGRA_I2S_BIT_SIZE_24				2
+#define TEGRA_I2S_BIT_SIZE_32				3
+
+#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT			8
+#define TEGRA_I2S_CTRL_BIT_SIZE_MASK			(3                     << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_SIZE_16			(TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_SIZE_20			(TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_SIZE_24			(TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
+#define TEGRA_I2S_CTRL_BIT_SIZE_32			(TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT)
+
+#define TEGRA_I2S_FIFO_16_LSB				0
+#define TEGRA_I2S_FIFO_20_LSB				1
+#define TEGRA_I2S_FIFO_24_LSB				2
+#define TEGRA_I2S_FIFO_32				3
+#define TEGRA_I2S_FIFO_PACKED				7
+
+#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT		4
+#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK			(7                     << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB		(TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB		(TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB		(TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_FIFO_FORMAT_32			(TEGRA_I2S_FIFO_32     << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
+#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED		(TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT)
+
+#define TEGRA_I2S_CTRL_IE_FIFO1_ERR			(1 << 3)
+#define TEGRA_I2S_CTRL_IE_FIFO2_ERR			(1 << 2)
+#define TEGRA_I2S_CTRL_QE_FIFO1				(1 << 1)
+#define TEGRA_I2S_CTRL_QE_FIFO2				(1 << 0)
+
+/* Fields in TEGRA_I2S_STATUS */
+
+#define TEGRA_I2S_STATUS_FIFO1_RDY			(1 << 31)
+#define TEGRA_I2S_STATUS_FIFO2_RDY			(1 << 30)
+#define TEGRA_I2S_STATUS_FIFO1_BSY			(1 << 29)
+#define TEGRA_I2S_STATUS_FIFO2_BSY			(1 << 28)
+#define TEGRA_I2S_STATUS_FIFO1_ERR			(1 << 3)
+#define TEGRA_I2S_STATUS_FIFO2_ERR			(1 << 2)
+#define TEGRA_I2S_STATUS_QS_FIFO1			(1 << 1)
+#define TEGRA_I2S_STATUS_QS_FIFO2			(1 << 0)
+
+/* Fields in TEGRA_I2S_TIMING */
+
+#define TEGRA_I2S_TIMING_NON_SYM_ENABLE			(1 << 12)
+#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
+#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
+#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK		(TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
+
+/* Fields in TEGRA_I2S_FIFO_SCR */
+
+#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT	24
+#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT	16
+#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK	0x3f
+
+#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR			(1 << 12)
+#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR			(1 << 8)
+
+#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT			0
+#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS		1
+#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS		2
+#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS		3
+
+#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT		4
+#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK		(3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT	(TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT     << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS   << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS  << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT)
+
+#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT		0
+#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK		(3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT	(TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT     << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS   << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS  << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS	(TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT)
+
+struct tegra_i2s {
+	struct clk *clk_i2s;
+	int clk_refs;
+	struct tegra_pcm_dma_params capture_dma_data;
+	struct tegra_pcm_dma_params playback_dma_data;
+	void __iomem *regs;
+	struct dentry *debug;
+	u32 reg_ctrl;
+};
+
+#endif
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
new file mode 100644
index 00000000000..3c271f95358
--- /dev/null
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -0,0 +1,404 @@
+/*
+ * tegra_pcm.c - Tegra PCM driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (c) 2009-2010, NVIDIA Corporation.
+ * Scott Peterson <speterson@nvidia.com>
+ * Vijay Mali <vmali@nvidia.com>
+ *
+ * Copyright (C) 2010 Google, Inc.
+ * Iliyan Malchev <malchev@google.com>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "tegra_pcm.h"
+
+#define DRV_NAME "tegra-pcm-audio"
+
+static const struct snd_pcm_hardware tegra_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
+	.channels_min		= 2,
+	.channels_max		= 2,
+	.period_bytes_min	= 1024,
+	.period_bytes_max	= PAGE_SIZE,
+	.periods_min		= 2,
+	.periods_max		= 8,
+	.buffer_bytes_max	= PAGE_SIZE * 8,
+	.fifo_size		= 4,
+};
+
+static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd)
+{
+	struct snd_pcm_substream *substream = prtd->substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	struct tegra_dma_req *dma_req;
+	unsigned long addr;
+
+	dma_req = &prtd->dma_req[prtd->dma_req_idx];
+	prtd->dma_req_idx = 1 - prtd->dma_req_idx;
+
+	addr = buf->addr + prtd->dma_pos;
+	prtd->dma_pos += dma_req->size;
+	if (prtd->dma_pos >= prtd->dma_pos_end)
+		prtd->dma_pos = 0;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dma_req->source_addr = addr;
+	else
+		dma_req->dest_addr = addr;
+
+	tegra_dma_enqueue_req(prtd->dma_chan, dma_req);
+}
+
+static void dma_complete_callback(struct tegra_dma_req *req)
+{
+	struct tegra_runtime_data *prtd = (struct tegra_runtime_data *)req->dev;
+	struct snd_pcm_substream *substream = prtd->substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	spin_lock(&prtd->lock);
+
+	if (!prtd->running) {
+		spin_unlock(&prtd->lock);
+		return;
+	}
+
+	if (++prtd->period_index >= runtime->periods)
+		prtd->period_index = 0;
+
+	tegra_pcm_queue_dma(prtd);
+
+	spin_unlock(&prtd->lock);
+
+	snd_pcm_period_elapsed(substream);
+}
+
+static void setup_dma_tx_request(struct tegra_dma_req *req,
+					struct tegra_pcm_dma_params * dmap)
+{
+	req->complete = dma_complete_callback;
+	req->to_memory = false;
+	req->dest_addr = dmap->addr;
+	req->dest_wrap = dmap->wrap;
+	req->source_bus_width = 32;
+	req->source_wrap = 0;
+	req->dest_bus_width = dmap->width;
+	req->req_sel = dmap->req_sel;
+}
+
+static void setup_dma_rx_request(struct tegra_dma_req *req,
+					struct tegra_pcm_dma_params * dmap)
+{
+	req->complete = dma_complete_callback;
+	req->to_memory = true;
+	req->source_addr = dmap->addr;
+	req->dest_wrap = 0;
+	req->source_bus_width = dmap->width;
+	req->source_wrap = dmap->wrap;
+	req->dest_bus_width = 32;
+	req->req_sel = dmap->req_sel;
+}
+
+static int tegra_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct tegra_runtime_data *prtd;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct tegra_pcm_dma_params * dmap;
+	int ret = 0;
+
+	prtd = kzalloc(sizeof(struct tegra_runtime_data), GFP_KERNEL);
+	if (prtd == NULL)
+		return -ENOMEM;
+
+	runtime->private_data = prtd;
+	prtd->substream = substream;
+
+	spin_lock_init(&prtd->lock);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+		setup_dma_tx_request(&prtd->dma_req[0], dmap);
+		setup_dma_tx_request(&prtd->dma_req[1], dmap);
+	} else {
+		dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+		setup_dma_rx_request(&prtd->dma_req[0], dmap);
+		setup_dma_rx_request(&prtd->dma_req[1], dmap);
+	}
+
+	prtd->dma_req[0].dev = prtd;
+	prtd->dma_req[1].dev = prtd;
+
+	prtd->dma_chan = tegra_dma_allocate_channel(TEGRA_DMA_MODE_ONESHOT);
+	if (prtd->dma_chan == NULL) {
+		ret = -ENOMEM;
+		goto err;
+	}
+
+	/* Set HW params now that initialization is complete */
+	snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware);
+
+	/* Ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(runtime,
+						SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		goto err;
+
+	return 0;
+
+err:
+	if (prtd->dma_chan) {
+		tegra_dma_free_channel(prtd->dma_chan);
+	}
+
+	kfree(prtd);
+
+	return ret;
+}
+
+static int tegra_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct tegra_runtime_data *prtd = runtime->private_data;
+
+	tegra_dma_free_channel(prtd->dma_chan);
+
+	kfree(prtd);
+
+	return 0;
+}
+
+static int tegra_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct tegra_runtime_data *prtd = runtime->private_data;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	prtd->dma_req[0].size = params_period_bytes(params);
+	prtd->dma_req[1].size = prtd->dma_req[0].size;
+
+	return 0;
+}
+
+static int tegra_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+
+	return 0;
+}
+
+static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct tegra_runtime_data *prtd = runtime->private_data;
+	unsigned long flags;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		prtd->dma_pos = 0;
+		prtd->dma_pos_end = frames_to_bytes(runtime, runtime->periods * runtime->period_size);
+		prtd->period_index = 0;
+		prtd->dma_req_idx = 0;
+		/* Fall-through */
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		spin_lock_irqsave(&prtd->lock, flags);
+		prtd->running = 1;
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		tegra_pcm_queue_dma(prtd);
+		tegra_pcm_queue_dma(prtd);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		spin_lock_irqsave(&prtd->lock, flags);
+		prtd->running = 0;
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		tegra_dma_dequeue_req(prtd->dma_chan, &prtd->dma_req[0]);
+		tegra_dma_dequeue_req(prtd->dma_chan, &prtd->dma_req[1]);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static snd_pcm_uframes_t tegra_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct tegra_runtime_data *prtd = runtime->private_data;
+
+	return prtd->period_index * runtime->period_size;
+}
+
+
+static int tegra_pcm_mmap(struct snd_pcm_substream *substream,
+				struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+					runtime->dma_area,
+					runtime->dma_addr,
+					runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops tegra_pcm_ops = {
+	.open		= tegra_pcm_open,
+	.close		= tegra_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= tegra_pcm_hw_params,
+	.hw_free	= tegra_pcm_hw_free,
+	.trigger	= tegra_pcm_trigger,
+	.pointer	= tegra_pcm_pointer,
+	.mmap		= tegra_pcm_mmap,
+};
+
+static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = tegra_pcm_hardware.buffer_bytes_max;
+
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+						&buf->addr, GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+	buf->bytes = size;
+
+	return 0;
+}
+
+static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+	if (!buf->area)
+		return;
+
+	dma_free_writecombine(pcm->card->dev, buf->bytes,
+				buf->area, buf->addr);
+	buf->area = NULL;
+}
+
+static u64 tegra_dma_mask = DMA_BIT_MASK(32);
+
+static int tegra_pcm_new(struct snd_card *card,
+				struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &tegra_dma_mask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = 0xffffffff;
+
+	if (dai->driver->playback.channels_min) {
+		ret = tegra_pcm_preallocate_dma_buffer(pcm,
+						SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto err;
+	}
+
+	if (dai->driver->capture.channels_min) {
+		ret = tegra_pcm_preallocate_dma_buffer(pcm,
+						SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto err_free_play;
+	}
+
+	return 0;
+
+err_free_play:
+	tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK);
+err:
+	return ret;
+}
+
+static void tegra_pcm_free(struct snd_pcm *pcm)
+{
+	tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE);
+	tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK);
+}
+
+struct snd_soc_platform_driver tegra_pcm_platform = {
+	.ops		= &tegra_pcm_ops,
+	.pcm_new	= tegra_pcm_new,
+	.pcm_free	= tegra_pcm_free,
+};
+
+static int __devinit tegra_pcm_platform_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_platform(&pdev->dev, &tegra_pcm_platform);
+}
+
+static int __devexit tegra_pcm_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver tegra_pcm_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = tegra_pcm_platform_probe,
+	.remove = __devexit_p(tegra_pcm_platform_remove),
+};
+
+static int __init snd_tegra_pcm_init(void)
+{
+	return platform_driver_register(&tegra_pcm_driver);
+}
+module_init(snd_tegra_pcm_init);
+
+static void __exit snd_tegra_pcm_exit(void)
+{
+	platform_driver_unregister(&tegra_pcm_driver);
+}
+module_exit(snd_tegra_pcm_exit);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra PCM ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h
new file mode 100644
index 00000000000..dbb90339fe0
--- /dev/null
+++ b/sound/soc/tegra/tegra_pcm.h
@@ -0,0 +1,55 @@
+/*
+ * tegra_pcm.h - Definitions for Tegra PCM driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2010 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (c) 2009-2010, NVIDIA Corporation.
+ * Scott Peterson <speterson@nvidia.com>
+ *
+ * Copyright (C) 2010 Google, Inc.
+ * Iliyan Malchev <malchev@google.com>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TEGRA_PCM_H__
+#define __TEGRA_PCM_H__
+
+#include <mach/dma.h>
+
+struct tegra_pcm_dma_params {
+	unsigned long addr;
+	unsigned long wrap;
+	unsigned long width;
+	unsigned long req_sel;
+};
+
+struct tegra_runtime_data {
+	struct snd_pcm_substream *substream;
+	spinlock_t lock;
+	int running;
+	int dma_pos;
+	int dma_pos_end;
+	int period_index;
+	int dma_req_idx;
+	struct tegra_dma_req dma_req[2];
+	struct tegra_dma_channel *dma_chan;
+};
+
+#endif