Merge branch 'topic/asoc' into for-linus

38 files changed, 3122 insertions, 358 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 5da30eb6ad00..83f5c67d3c41 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -22,9 +22,11 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
+	select SND_SOC_CS42L51 if I2C
 	select SND_SOC_CS4270 if I2C
-	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_DA7210 if I2C
+	select SND_SOC_JZ4740 if SOC_JZ4740
+	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if I2C
@@ -48,6 +50,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8727
 	select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
@@ -120,13 +123,13 @@ config SND_SOC_AK4671
 config SND_SOC_CQ0093VC
 	tristate
 
+config SND_SOC_CS42L51
+	tristate
+
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
 	tristate
 
-config SND_SOC_DA7210
-        tristate
-
 # Cirrus Logic CS4270 Codec VD = 3.3V Errata
 # Select if you are affected by the errata where the part will not function
 # if MCLK divide-by-1.5 is selected and VD is set to 3.3V.  The driver will
@@ -138,9 +141,15 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_CX20442
 	tristate
 
+config SND_SOC_JZ4740_CODEC
+	tristate
+
 config SND_SOC_L3
        tristate
 
+config SND_SOC_DA7210
+        tristate
+
 config SND_SOC_PCM3008
        tristate
 
@@ -206,6 +215,9 @@ config SND_SOC_WM8728
 config SND_SOC_WM8731
 	tristate
 
+config SND_SOC_WM8741
+	tristate
+
 config SND_SOC_WM8750
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 91429eab0707..53524095759c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -9,6 +9,7 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4642-objs := ak4642.o
 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-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
@@ -34,6 +35,7 @@ snd-soc-wm8711-objs := wm8711.o
 snd-soc-wm8727-objs := wm8727.o
 snd-soc-wm8728-objs := wm8728.o
 snd-soc-wm8731-objs := wm8731.o
+snd-soc-wm8741-objs := wm8741.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8776-objs := wm8776.o
@@ -56,6 +58,7 @@ snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 snd-soc-wm-hubs-objs := wm_hubs.o
+snd-soc-jz4740-codec-objs := jz4740.o
 
 # Amp
 snd-soc-max9877-objs := max9877.o
@@ -74,10 +77,12 @@ 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_CQ0093VC) += snd-soc-cq93vc.o
+obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
@@ -99,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8711)	+= snd-soc-wm8711.o
 obj-$(CONFIG_SND_SOC_WM8727)	+= snd-soc-wm8727.o
 obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
+obj-$(CONFIG_SND_SOC_WM8741)	+= snd-soc-wm8741.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8776)	+= snd-soc-wm8776.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 217538423225..a01006c8c606 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -272,6 +272,7 @@ static int ad1836_register(struct ad1836_priv *ad1836)
 
 	if (ad1836_codec) {
 		dev_err(codec->dev, "Another ad1836 is registered\n");
+		kfree(ad1836);
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index c8ca1142b2f4..1def75e4862f 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -24,6 +24,7 @@
 
 /* codec private data */
 struct ad193x_priv {
+	unsigned int sysclk;
 	struct snd_soc_codec codec;
 	u8 reg_cache[AD193X_NUM_REGS];
 };
@@ -251,15 +252,32 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
+static int ad193x_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 ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+	switch (freq) {
+	case 12288000:
+	case 18432000:
+	case 24576000:
+	case 36864000:
+		ad193x->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
 static int ad193x_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params,
 		struct snd_soc_dai *dai)
 {
-	int word_len = 0, reg = 0;
+	int word_len = 0, reg = 0, master_rate = 0;
 
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
+	struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
 
 	/* bit size */
 	switch (params_format(params)) {
@@ -275,6 +293,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
+	switch (ad193x->sysclk) {
+	case 12288000:
+		master_rate = AD193X_PLL_INPUT_256;
+		break;
+	case 18432000:
+		master_rate = AD193X_PLL_INPUT_384;
+		break;
+	case 24576000:
+		master_rate = AD193X_PLL_INPUT_512;
+		break;
+	case 36864000:
+		master_rate = AD193X_PLL_INPUT_768;
+		break;
+	}
+
+	reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0);
+	reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate;
+	snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
+
 	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
 	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
 	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
@@ -348,6 +385,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
 	/* pll input: mclki/xi */
 	snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
 	snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04);
+	ad193x->sysclk = 12288000;
 
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
@@ -383,6 +421,7 @@ static struct snd_soc_dai_ops ad193x_dai_ops = {
 	.hw_params = ad193x_hw_params,
 	.digital_mute = ad193x_mute,
 	.set_tdm_slot = ad193x_set_tdm_slot,
+	.set_sysclk	= ad193x_set_dai_sysclk,
 	.set_fmt = ad193x_set_dai_fmt,
 };
 
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index a03c880d52f9..654ba64ae04c 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -11,6 +11,11 @@
 
 #define AD193X_PLL_CLK_CTRL0    0x800
 #define AD193X_PLL_POWERDOWN           0x01
+#define AD193X_PLL_INPUT_MASK   (~0x6)
+#define AD193X_PLL_INPUT_256    (0 << 1)
+#define AD193X_PLL_INPUT_384    (1 << 1)
+#define AD193X_PLL_INPUT_512    (2 << 1)
+#define AD193X_PLL_INPUT_768    (3 << 1)
 #define AD193X_PLL_CLK_CTRL1    0x801
 #define AD193X_DAC_CTRL0        0x802
 #define AD193X_DAC_POWERDOWN           0x01
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 7528a54102b5..3d7dc55305ec 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -22,20 +22,13 @@
  * AK4643 is tested.
  */
 
-#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/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 "ak4642.h"
 
@@ -111,6 +104,23 @@
 
 struct snd_soc_codec_device soc_codec_dev_ak4642;
 
+/*
+ * Playback Volume (table 39)
+ *
+ * max : 0x00 : +12.0 dB
+ *       ( 0.5 dB step )
+ * min : 0xFE : -115.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+
+static const struct snd_kcontrol_new ak4642_snd_controls[] = {
+
+	SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
+			 0, 0xFF, 1, out_tlv),
+};
+
+
 /* codec private data */
 struct ak4642_priv {
 	struct snd_soc_codec codec;
@@ -204,7 +214,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		 *
 		 * PLL, Master Mode
 		 * Audio I/F Format :MSB justified (ADC & DAC)
-		 * Digital Volume: -8dB
 		 * Bass Boost Level : Middle
 		 *
 		 * This operation came from example code of
@@ -214,8 +223,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		ak4642_write(codec, 0x0e, 0x19);
 		ak4642_write(codec, 0x09, 0x91);
 		ak4642_write(codec, 0x0c, 0x91);
-		ak4642_write(codec, 0x0a, 0x28);
-		ak4642_write(codec, 0x0d, 0x28);
 		ak4642_write(codec, 0x00, 0x64);
 		snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK,	PMHP);
 		snd_soc_update_bits(codec, PW_MGMT2, HPMTN,	HPMTN);
@@ -491,8 +498,10 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
 	codec->control_data = i2c;
 
 	ret = ak4642_init(ak4642);
-	if (ret < 0)
+	if (ret < 0) {
 		printk(KERN_ERR "failed to initialise AK4642\n");
+		kfree(ak4642);
+	}
 
 	return ret;
 }
@@ -548,6 +557,9 @@ static int ak4642_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
+	snd_soc_add_controls(ak4642_codec, ak4642_snd_controls,
+			     ARRAY_SIZE(ak4642_snd_controls));
+
 	dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
 	return ret;
 
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
new file mode 100644
index 000000000000..dd9b8550c402
--- /dev/null
+++ b/sound/soc/codecs/cs42l51.c
@@ -0,0 +1,763 @@
+/*
+ * cs42l51.c
+ *
+ * ASoC Driver for Cirrus Logic CS42L51 codecs
+ *
+ * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com>
+ *
+ * Based on cs4270.c - Copyright (c) Freescale Semiconductor
+ *
+ * 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.
+ *
+ * For now:
+ *  - Only I2C is support. Not SPI
+ *  - master mode *NOT* supported
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm.h>
+#include <linux/i2c.h>
+
+#include "cs42l51.h"
+
+enum master_slave_mode {
+	MODE_SLAVE,
+	MODE_SLAVE_AUTO,
+	MODE_MASTER,
+};
+
+struct cs42l51_private {
+	unsigned int mclk;
+	unsigned int audio_mode;	/* The mode (I2S or left-justified) */
+	enum master_slave_mode func;
+	struct snd_soc_codec codec;
+	u8 reg_cache[CS42L51_NUMREGS];
+};
+
+static struct snd_soc_codec *cs42l51_codec;
+
+#define CS42L51_FORMATS ( \
+		SNDRV_PCM_FMTBIT_S16_LE  | SNDRV_PCM_FMTBIT_S16_BE  | \
+		SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
+		SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
+		SNDRV_PCM_FMTBIT_S24_LE  | SNDRV_PCM_FMTBIT_S24_BE)
+
+static int cs42l51_fill_cache(struct snd_soc_codec *codec)
+{
+	u8 *cache = codec->reg_cache + 1;
+	struct i2c_client *i2c_client = codec->control_data;
+	s32 length;
+
+	length = i2c_smbus_read_i2c_block_data(i2c_client,
+			CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache);
+	if (length != CS42L51_NUMREGS) {
+		dev_err(&i2c_client->dev,
+				"I2C read failure, addr=0x%x (ret=%d vs %d)\n",
+				i2c_client->addr, length, CS42L51_NUMREGS);
+		return -EIO;
+	}
+
+	return 0;
+}
+
+static int cs42l51_i2c_probe(struct i2c_client *i2c_client,
+	const struct i2c_device_id *id)
+{
+	struct snd_soc_codec *codec;
+	struct cs42l51_private *cs42l51;
+	int ret = 0;
+	int reg;
+
+	if (cs42l51_codec)
+		return -EBUSY;
+
+	/* Verify that we have a CS42L51 */
+	ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "failed to read I2C\n");
+		goto error;
+	}
+
+	if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
+	    (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
+		dev_err(&i2c_client->dev, "Invalid chip id\n");
+		ret = -ENODEV;
+		goto error;
+	}
+
+	dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n",
+				ret & 7);
+
+	cs42l51 = kzalloc(sizeof(struct cs42l51_private), GFP_KERNEL);
+	if (!cs42l51) {
+		dev_err(&i2c_client->dev, "could not allocate codec\n");
+		return -ENOMEM;
+	}
+	codec = &cs42l51->codec;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->dev = &i2c_client->dev;
+	codec->name = "CS42L51";
+	codec->owner = THIS_MODULE;
+	codec->dai = &cs42l51_dai;
+	codec->num_dai = 1;
+	snd_soc_codec_set_drvdata(codec, cs42l51);
+
+	codec->control_data = i2c_client;
+	codec->reg_cache = cs42l51->reg_cache;
+	codec->reg_cache_size = CS42L51_NUMREGS;
+	i2c_set_clientdata(i2c_client, codec);
+
+	ret = cs42l51_fill_cache(codec);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "failed to fill register cache\n");
+		goto error_alloc;
+	}
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "Failed to set cache I/O: %d\n", ret);
+		goto error_alloc;
+	}
+
+	/*
+	 * DAC configuration
+	 * - Use signal processor
+	 * - auto mute
+	 * - vol changes immediate
+	 * - no de-emphasize
+	 */
+	reg = CS42L51_DAC_CTL_DATA_SEL(1)
+		| CS42L51_DAC_CTL_AMUTE | CS42L51_DAC_CTL_DACSZ(0);
+	ret = snd_soc_write(codec, CS42L51_DAC_CTL, reg);
+	if (ret < 0)
+		goto error_alloc;
+
+	cs42l51_dai.dev = codec->dev;
+	cs42l51_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto error_alloc;
+	}
+
+	ret = snd_soc_register_dai(&cs42l51_dai);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "failed to register DAIe\n");
+		goto error_reg;
+	}
+
+	return 0;
+
+error_reg:
+	snd_soc_unregister_codec(codec);
+error_alloc:
+	kfree(cs42l51);
+error:
+	return ret;
+}
+
+static int cs42l51_i2c_remove(struct i2c_client *client)
+{
+	struct cs42l51_private *cs42l51 = i2c_get_clientdata(client);
+	snd_soc_unregister_dai(&cs42l51_dai);
+	snd_soc_unregister_codec(cs42l51_codec);
+	cs42l51_codec = NULL;
+	kfree(cs42l51);
+	return 0;
+}
+
+
+static const struct i2c_device_id cs42l51_id[] = {
+	{"cs42l51", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, cs42l51_id);
+
+static struct i2c_driver cs42l51_i2c_driver = {
+	.driver = {
+		.name = "CS42L51 I2C",
+		.owner = THIS_MODULE,
+	},
+	.id_table = cs42l51_id,
+	.probe = cs42l51_i2c_probe,
+	.remove = cs42l51_i2c_remove,
+};
+
+static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned long value = snd_soc_read(codec, CS42L51_PCM_MIXER)&3;
+
+	switch (value) {
+	default:
+	case 0:
+		ucontrol->value.integer.value[0] = 0;
+		break;
+	/* same value : (L+R)/2 and (R+L)/2 */
+	case 1:
+	case 2:
+		ucontrol->value.integer.value[0] = 1;
+		break;
+	case 3:
+		ucontrol->value.integer.value[0] = 2;
+		break;
+	}
+
+	return 0;
+}
+
+#define CHAN_MIX_NORMAL	0x00
+#define CHAN_MIX_BOTH	0x55
+#define CHAN_MIX_SWAP	0xFF
+
+static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned char val;
+
+	switch (ucontrol->value.integer.value[0]) {
+	default:
+	case 0:
+		val = CHAN_MIX_NORMAL;
+		break;
+	case 1:
+		val = CHAN_MIX_BOTH;
+		break;
+	case 2:
+		val = CHAN_MIX_SWAP;
+		break;
+	}
+
+	snd_soc_write(codec, CS42L51_PCM_MIXER, val);
+
+	return 1;
+}
+
+static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0);
+/* This is a lie. after -102 db, it stays at -102 */
+/* maybe a range would be better */
+static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0);
+static const char *chan_mix[] = {
+	"L R",
+	"L+R",
+	"R L",
+};
+
+static const struct soc_enum cs42l51_chan_mix =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix);
+
+static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
+	SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
+			CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
+			7, 0xffffff99, 0x18, adc_pcm_tlv),
+	SOC_DOUBLE_R("PCM Playback Switch",
+			CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
+	SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
+			CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL,
+			8, 0xffffff19, 0x18, aout_tlv),
+	SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
+			CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
+			7, 0xffffff99, 0x18, adc_pcm_tlv),
+	SOC_DOUBLE_R("ADC Mixer Switch",
+			CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
+	SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
+	SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0),
+	SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0),
+	SOC_SINGLE("Zero Cross Switch", CS42L51_DAC_CTL, 0, 0, 0),
+	SOC_DOUBLE_TLV("Mic Boost Volume",
+			CS42L51_MIC_CTL, 0, 1, 1, 0, boost_tlv),
+	SOC_SINGLE_TLV("Bass Volume", CS42L51_TONE_CTL, 0, 0xf, 1, tone_tlv),
+	SOC_SINGLE_TLV("Treble Volume", CS42L51_TONE_CTL, 4, 0xf, 1, tone_tlv),
+	SOC_ENUM_EXT("PCM channel mixer",
+			cs42l51_chan_mix,
+			cs42l51_get_chan_mix, cs42l51_set_chan_mix),
+};
+
+/*
+ * to power down, one must:
+ * 1.) Enable the PDN bit
+ * 2.) enable power-down for the select channels
+ * 3.) disable the PDN bit.
+ */
+static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	unsigned long value;
+
+	value = snd_soc_read(w->codec, CS42L51_POWER_CTL1);
+	value &= ~CS42L51_POWER_CTL1_PDN;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMD:
+		value |= CS42L51_POWER_CTL1_PDN;
+		break;
+	default:
+	case SND_SOC_DAPM_POST_PMD:
+		break;
+	}
+	snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1,
+		CS42L51_POWER_CTL1_PDN, value);
+
+	return 0;
+}
+
+static const char *cs42l51_dac_names[] = {"Direct PCM",
+	"DSP PCM", "ADC"};
+static const struct soc_enum cs42l51_dac_mux_enum =
+	SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names);
+static const struct snd_kcontrol_new cs42l51_dac_mux_controls =
+	SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum);
+
+static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left",
+	"MIC Left", "MIC+preamp Left"};
+static const struct soc_enum cs42l51_adcl_mux_enum =
+	SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names);
+static const struct snd_kcontrol_new cs42l51_adcl_mux_controls =
+	SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum);
+
+static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right",
+	"MIC Right", "MIC+preamp Right"};
+static const struct soc_enum cs42l51_adcr_mux_enum =
+	SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names);
+static const struct snd_kcontrol_new cs42l51_adcr_mux_controls =
+	SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum);
+
+static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
+	SND_SOC_DAPM_MICBIAS("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1),
+	SND_SOC_DAPM_PGA_E("Left PGA", CS42L51_POWER_CTL1, 3, 1, NULL, 0,
+		cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Right PGA", CS42L51_POWER_CTL1, 4, 1, NULL, 0,
+		cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_ADC_E("Left ADC", "Left HiFi Capture",
+		CS42L51_POWER_CTL1, 1, 1,
+		cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture",
+		CS42L51_POWER_CTL1, 2, 1,
+		cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback",
+		CS42L51_POWER_CTL1, 5, 1,
+		cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback",
+		CS42L51_POWER_CTL1, 6, 1,
+		cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+
+	/* analog/mic */
+	SND_SOC_DAPM_INPUT("AIN1L"),
+	SND_SOC_DAPM_INPUT("AIN1R"),
+	SND_SOC_DAPM_INPUT("AIN2L"),
+	SND_SOC_DAPM_INPUT("AIN2R"),
+	SND_SOC_DAPM_INPUT("MICL"),
+	SND_SOC_DAPM_INPUT("MICR"),
+
+	SND_SOC_DAPM_MIXER("Mic Preamp Left",
+		CS42L51_MIC_POWER_CTL, 2, 1, NULL, 0),
+	SND_SOC_DAPM_MIXER("Mic Preamp Right",
+		CS42L51_MIC_POWER_CTL, 3, 1, NULL, 0),
+
+	/* HP */
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+
+	/* mux */
+	SND_SOC_DAPM_MUX("DAC Mux", SND_SOC_NOPM, 0, 0,
+		&cs42l51_dac_mux_controls),
+	SND_SOC_DAPM_MUX("PGA-ADC Mux Left", SND_SOC_NOPM, 0, 0,
+		&cs42l51_adcl_mux_controls),
+	SND_SOC_DAPM_MUX("PGA-ADC Mux Right", SND_SOC_NOPM, 0, 0,
+		&cs42l51_adcr_mux_controls),
+};
+
+static const struct snd_soc_dapm_route cs42l51_routes[] = {
+	{"HPL", NULL, "Left DAC"},
+	{"HPR", NULL, "Right DAC"},
+
+	{"Left ADC", NULL, "Left PGA"},
+	{"Right ADC", NULL, "Right PGA"},
+
+	{"Mic Preamp Left",  NULL,  "MICL"},
+	{"Mic Preamp Right", NULL,  "MICR"},
+
+	{"PGA-ADC Mux Left",  "AIN1 Left",        "AIN1L" },
+	{"PGA-ADC Mux Left",  "AIN2 Left",        "AIN2L" },
+	{"PGA-ADC Mux Left",  "MIC Left",         "MICL"  },
+	{"PGA-ADC Mux Left",  "MIC+preamp Left",  "Mic Preamp Left" },
+	{"PGA-ADC Mux Right", "AIN1 Right",       "AIN1R" },
+	{"PGA-ADC Mux Right", "AIN2 Right",       "AIN2R" },
+	{"PGA-ADC Mux Right", "MIC Right",        "MICR" },
+	{"PGA-ADC Mux Right", "MIC+preamp Right", "Mic Preamp Right" },
+
+	{"Left PGA", NULL, "PGA-ADC Mux Left"},
+	{"Right PGA", NULL, "PGA-ADC Mux Right"},
+};
+
+static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		cs42l51->audio_mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
+		break;
+	default:
+		dev_err(codec->dev, "invalid DAI format\n");
+		ret = -EINVAL;
+	}
+
+	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs42l51->func = MODE_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs42l51->func = MODE_SLAVE_AUTO;
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+struct cs42l51_ratios {
+	unsigned int ratio;
+	unsigned char speed_mode;
+	unsigned char mclk;
+};
+
+static struct cs42l51_ratios slave_ratios[] = {
+	{  512, CS42L51_QSM_MODE, 0 }, {  768, CS42L51_QSM_MODE, 0 },
+	{ 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
+	{ 2048, CS42L51_QSM_MODE, 0 }, { 3072, CS42L51_QSM_MODE, 0 },
+	{  256, CS42L51_HSM_MODE, 0 }, {  384, CS42L51_HSM_MODE, 0 },
+	{  512, CS42L51_HSM_MODE, 0 }, {  768, CS42L51_HSM_MODE, 0 },
+	{ 1024, CS42L51_HSM_MODE, 0 }, { 1536, CS42L51_HSM_MODE, 0 },
+	{  128, CS42L51_SSM_MODE, 0 }, {  192, CS42L51_SSM_MODE, 0 },
+	{  256, CS42L51_SSM_MODE, 0 }, {  384, CS42L51_SSM_MODE, 0 },
+	{  512, CS42L51_SSM_MODE, 0 }, {  768, CS42L51_SSM_MODE, 0 },
+	{  128, CS42L51_DSM_MODE, 0 }, {  192, CS42L51_DSM_MODE, 0 },
+	{  256, CS42L51_DSM_MODE, 0 }, {  384, CS42L51_DSM_MODE, 0 },
+};
+
+static struct cs42l51_ratios slave_auto_ratios[] = {
+	{ 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
+	{ 2048, CS42L51_QSM_MODE, 1 }, { 3072, CS42L51_QSM_MODE, 1 },
+	{  512, CS42L51_HSM_MODE, 0 }, {  768, CS42L51_HSM_MODE, 0 },
+	{ 1024, CS42L51_HSM_MODE, 1 }, { 1536, CS42L51_HSM_MODE, 1 },
+	{  256, CS42L51_SSM_MODE, 0 }, {  384, CS42L51_SSM_MODE, 0 },
+	{  512, CS42L51_SSM_MODE, 1 }, {  768, CS42L51_SSM_MODE, 1 },
+	{  128, CS42L51_DSM_MODE, 0 }, {  192, CS42L51_DSM_MODE, 0 },
+	{  256, CS42L51_DSM_MODE, 1 }, {  384, CS42L51_DSM_MODE, 1 },
+};
+
+static int cs42l51_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 cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+	struct cs42l51_ratios *ratios = NULL;
+	int nr_ratios = 0;
+	unsigned int rates = 0;
+	unsigned int rate_min = -1;
+	unsigned int rate_max = 0;
+	int i;
+
+	cs42l51->mclk = freq;
+
+	switch (cs42l51->func) {
+	case MODE_MASTER:
+		return -EINVAL;
+	case MODE_SLAVE:
+		ratios = slave_ratios;
+		nr_ratios = ARRAY_SIZE(slave_ratios);
+		break;
+	case MODE_SLAVE_AUTO:
+		ratios = slave_auto_ratios;
+		nr_ratios = ARRAY_SIZE(slave_auto_ratios);
+		break;
+	}
+
+	for (i = 0; i < nr_ratios; i++) {
+		unsigned int rate = freq / ratios[i].ratio;
+		rates |= snd_pcm_rate_to_rate_bit(rate);
+		if (rate < rate_min)
+			rate_min = rate;
+		if (rate > rate_max)
+			rate_max = rate;
+	}
+	rates &= ~SNDRV_PCM_RATE_KNOT;
+
+	if (!rates) {
+		dev_err(codec->dev, "could not find a valid sample rate\n");
+		return -EINVAL;
+	}
+
+	codec_dai->playback.rates = rates;
+	codec_dai->playback.rate_min = rate_min;
+	codec_dai->playback.rate_max = rate_max;
+
+	codec_dai->capture.rates = rates;
+	codec_dai->capture.rate_min = rate_min;
+	codec_dai->capture.rate_max = rate_max;
+
+	return 0;
+}
+
+static int cs42l51_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_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+	unsigned int i;
+	unsigned int rate;
+	unsigned int ratio;
+	struct cs42l51_ratios *ratios = NULL;
+	int nr_ratios = 0;
+	int intf_ctl, power_ctl, fmt;
+
+	switch (cs42l51->func) {
+	case MODE_MASTER:
+		return -EINVAL;
+	case MODE_SLAVE:
+		ratios = slave_ratios;
+		nr_ratios = ARRAY_SIZE(slave_ratios);
+		break;
+	case MODE_SLAVE_AUTO:
+		ratios = slave_auto_ratios;
+		nr_ratios = ARRAY_SIZE(slave_auto_ratios);
+		break;
+	}
+
+	/* Figure out which MCLK/LRCK ratio to use */
+	rate = params_rate(params);     /* Sampling rate, in Hz */
+	ratio = cs42l51->mclk / rate;    /* MCLK/LRCK ratio */
+	for (i = 0; i < nr_ratios; i++) {
+		if (ratios[i].ratio == ratio)
+			break;
+	}
+
+	if (i == nr_ratios) {
+		/* We did not find a matching ratio */
+		dev_err(codec->dev, "could not find matching ratio\n");
+		return -EINVAL;
+	}
+
+	intf_ctl = snd_soc_read(codec, CS42L51_INTF_CTL);
+	power_ctl = snd_soc_read(codec, CS42L51_MIC_POWER_CTL);
+
+	intf_ctl &= ~(CS42L51_INTF_CTL_MASTER | CS42L51_INTF_CTL_ADC_I2S
+			| CS42L51_INTF_CTL_DAC_FORMAT(7));
+	power_ctl &= ~(CS42L51_MIC_POWER_CTL_SPEED(3)
+			| CS42L51_MIC_POWER_CTL_MCLK_DIV2);
+
+	switch (cs42l51->func) {
+	case MODE_MASTER:
+		intf_ctl |= CS42L51_INTF_CTL_MASTER;
+		power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
+		break;
+	case MODE_SLAVE:
+		power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
+		break;
+	case MODE_SLAVE_AUTO:
+		power_ctl |= CS42L51_MIC_POWER_CTL_AUTO;
+		break;
+	}
+
+	switch (cs42l51->audio_mode) {
+	case SND_SOC_DAIFMT_I2S:
+		intf_ctl |= CS42L51_INTF_CTL_ADC_I2S;
+		intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_I2S);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24);
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		switch (params_format(params)) {
+		case SNDRV_PCM_FORMAT_S16_LE:
+		case SNDRV_PCM_FORMAT_S16_BE:
+			fmt = CS42L51_DAC_DIF_RJ16;
+			break;
+		case SNDRV_PCM_FORMAT_S18_3LE:
+		case SNDRV_PCM_FORMAT_S18_3BE:
+			fmt = CS42L51_DAC_DIF_RJ18;
+			break;
+		case SNDRV_PCM_FORMAT_S20_3LE:
+		case SNDRV_PCM_FORMAT_S20_3BE:
+			fmt = CS42L51_DAC_DIF_RJ20;
+			break;
+		case SNDRV_PCM_FORMAT_S24_LE:
+		case SNDRV_PCM_FORMAT_S24_BE:
+			fmt = CS42L51_DAC_DIF_RJ24;
+			break;
+		default:
+			dev_err(codec->dev, "unknown format\n");
+			return -EINVAL;
+		}
+		intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(fmt);
+		break;
+	default:
+		dev_err(codec->dev, "unknown format\n");
+		return -EINVAL;
+	}
+
+	if (ratios[i].mclk)
+		power_ctl |= CS42L51_MIC_POWER_CTL_MCLK_DIV2;
+
+	ret = snd_soc_write(codec, CS42L51_INTF_CTL, intf_ctl);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_write(codec, CS42L51_MIC_POWER_CTL, power_ctl);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int reg;
+	int mask = CS42L51_DAC_OUT_CTL_DACA_MUTE|CS42L51_DAC_OUT_CTL_DACB_MUTE;
+
+	reg = snd_soc_read(codec, CS42L51_DAC_OUT_CTL);
+
+	if (mute)
+		reg |= mask;
+	else
+		reg &= ~mask;
+
+	return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg);
+}
+
+static struct snd_soc_dai_ops cs42l51_dai_ops = {
+	.hw_params      = cs42l51_hw_params,
+	.set_sysclk     = cs42l51_set_dai_sysclk,
+	.set_fmt        = cs42l51_set_dai_fmt,
+	.digital_mute   = cs42l51_dai_mute,
+};
+
+struct snd_soc_dai cs42l51_dai = {
+	.name = "CS42L51 HiFi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = CS42L51_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = CS42L51_FORMATS,
+	},
+	.ops = &cs42l51_dai_ops,
+};
+EXPORT_SYMBOL_GPL(cs42l51_dai);
+
+
+static int cs42l51_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (!cs42l51_codec) {
+		dev_err(&pdev->dev, "CS42L51 codec not yet registered\n");
+		return -EINVAL;
+	}
+
+	socdev->card->codec = cs42l51_codec;
+	codec = socdev->card->codec;
+
+	/* Register PCMs */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create PCMs\n");
+		return ret;
+	}
+
+	snd_soc_add_controls(codec, cs42l51_snd_controls,
+		ARRAY_SIZE(cs42l51_snd_controls));
+	snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets,
+		ARRAY_SIZE(cs42l51_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, cs42l51_routes,
+		ARRAY_SIZE(cs42l51_routes));
+
+	return 0;
+}
+
+
+static int cs42l51_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_device_cs42l51 = {
+	.probe =	cs42l51_probe,
+	.remove =	cs42l51_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_cs42l51);
+
+static int __init cs42l51_init(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&cs42l51_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "%s: can't add i2c driver\n", __func__);
+		return ret;
+	}
+	return 0;
+}
+module_init(cs42l51_init);
+
+static void __exit cs42l51_exit(void)
+{
+	i2c_del_driver(&cs42l51_i2c_driver);
+}
+module_exit(cs42l51_exit);
+
+MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>");
+MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h
new file mode 100644
index 000000000000..8f0bd9786ad2
--- /dev/null
+++ b/sound/soc/codecs/cs42l51.h
@@ -0,0 +1,163 @@
+/*
+ * cs42l51.h
+ *
+ * ASoC Driver for Cirrus Logic CS42L51 codecs
+ *
+ * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.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.
+ */
+#ifndef _CS42L51_H
+#define _CS42L51_H
+
+#define CS42L51_CHIP_ID			0x1B
+#define CS42L51_CHIP_REV_A		0x00
+#define CS42L51_CHIP_REV_B		0x01
+
+#define CS42L51_CHIP_REV_ID		0x01
+#define CS42L51_MK_CHIP_REV(a, b)	((a)<<3|(b))
+
+#define CS42L51_POWER_CTL1		0x02
+#define CS42L51_POWER_CTL1_PDN_DACB	(1<<6)
+#define CS42L51_POWER_CTL1_PDN_DACA	(1<<5)
+#define CS42L51_POWER_CTL1_PDN_PGAB	(1<<4)
+#define CS42L51_POWER_CTL1_PDN_PGAA	(1<<3)
+#define CS42L51_POWER_CTL1_PDN_ADCB	(1<<2)
+#define CS42L51_POWER_CTL1_PDN_ADCA	(1<<1)
+#define CS42L51_POWER_CTL1_PDN		(1<<0)
+
+#define CS42L51_MIC_POWER_CTL		0x03
+#define CS42L51_MIC_POWER_CTL_AUTO	(1<<7)
+#define CS42L51_MIC_POWER_CTL_SPEED(x)	(((x)&3)<<5)
+#define CS42L51_QSM_MODE		3
+#define CS42L51_HSM_MODE		2
+#define	CS42L51_SSM_MODE		1
+#define CS42L51_DSM_MODE		0
+#define CS42L51_MIC_POWER_CTL_3ST_SP	(1<<4)
+#define CS42L51_MIC_POWER_CTL_PDN_MICB	(1<<3)
+#define CS42L51_MIC_POWER_CTL_PDN_MICA	(1<<2)
+#define CS42L51_MIC_POWER_CTL_PDN_BIAS	(1<<1)
+#define CS42L51_MIC_POWER_CTL_MCLK_DIV2	(1<<0)
+
+#define CS42L51_INTF_CTL		0x04
+#define CS42L51_INTF_CTL_LOOPBACK	(1<<7)
+#define CS42L51_INTF_CTL_MASTER		(1<<6)
+#define CS42L51_INTF_CTL_DAC_FORMAT(x)	(((x)&7)<<3)
+#define CS42L51_DAC_DIF_LJ24		0x00
+#define CS42L51_DAC_DIF_I2S		0x01
+#define CS42L51_DAC_DIF_RJ24		0x02
+#define CS42L51_DAC_DIF_RJ20		0x03
+#define CS42L51_DAC_DIF_RJ18		0x04
+#define CS42L51_DAC_DIF_RJ16		0x05
+#define CS42L51_INTF_CTL_ADC_I2S	(1<<2)
+#define CS42L51_INTF_CTL_DIGMIX		(1<<1)
+#define CS42L51_INTF_CTL_MICMIX		(1<<0)
+
+#define CS42L51_MIC_CTL			0x05
+#define CS42L51_MIC_CTL_ADC_SNGVOL	(1<<7)
+#define CS42L51_MIC_CTL_ADCD_DBOOST	(1<<6)
+#define CS42L51_MIC_CTL_ADCA_DBOOST	(1<<5)
+#define CS42L51_MIC_CTL_MICBIAS_SEL	(1<<4)
+#define CS42L51_MIC_CTL_MICBIAS_LVL(x)	(((x)&3)<<2)
+#define CS42L51_MIC_CTL_MICB_BOOST	(1<<1)
+#define CS42L51_MIC_CTL_MICA_BOOST	(1<<0)
+
+#define CS42L51_ADC_CTL			0x06
+#define CS42L51_ADC_CTL_ADCB_HPFEN	(1<<7)
+#define CS42L51_ADC_CTL_ADCB_HPFRZ	(1<<6)
+#define CS42L51_ADC_CTL_ADCA_HPFEN	(1<<5)
+#define CS42L51_ADC_CTL_ADCA_HPFRZ	(1<<4)
+#define CS42L51_ADC_CTL_SOFTB		(1<<3)
+#define CS42L51_ADC_CTL_ZCROSSB		(1<<2)
+#define CS42L51_ADC_CTL_SOFTA		(1<<1)
+#define CS42L51_ADC_CTL_ZCROSSA		(1<<0)
+
+#define CS42L51_ADC_INPUT		0x07
+#define CS42L51_ADC_INPUT_AINB_MUX(x)	(((x)&3)<<6)
+#define CS42L51_ADC_INPUT_AINA_MUX(x)	(((x)&3)<<4)
+#define CS42L51_ADC_INPUT_INV_ADCB	(1<<3)
+#define CS42L51_ADC_INPUT_INV_ADCA	(1<<2)
+#define CS42L51_ADC_INPUT_ADCB_MUTE	(1<<1)
+#define CS42L51_ADC_INPUT_ADCA_MUTE	(1<<0)
+
+#define CS42L51_DAC_OUT_CTL		0x08
+#define CS42L51_DAC_OUT_CTL_HP_GAIN(x)	(((x)&7)<<5)
+#define CS42L51_DAC_OUT_CTL_DAC_SNGVOL	(1<<4)
+#define CS42L51_DAC_OUT_CTL_INV_PCMB	(1<<3)
+#define CS42L51_DAC_OUT_CTL_INV_PCMA	(1<<2)
+#define CS42L51_DAC_OUT_CTL_DACB_MUTE	(1<<1)
+#define CS42L51_DAC_OUT_CTL_DACA_MUTE	(1<<0)
+
+#define CS42L51_DAC_CTL			0x09
+#define CS42L51_DAC_CTL_DATA_SEL(x)	(((x)&3)<<6)
+#define CS42L51_DAC_CTL_FREEZE		(1<<5)
+#define CS42L51_DAC_CTL_DEEMPH		(1<<3)
+#define CS42L51_DAC_CTL_AMUTE		(1<<2)
+#define CS42L51_DAC_CTL_DACSZ(x)	(((x)&3)<<0)
+
+#define CS42L51_ALC_PGA_CTL		0x0A
+#define CS42L51_ALC_PGB_CTL		0x0B
+#define CS42L51_ALC_PGX_ALCX_SRDIS	(1<<7)
+#define CS42L51_ALC_PGX_ALCX_ZCDIS	(1<<6)
+#define CS42L51_ALC_PGX_PGX_VOL(x)	(((x)&0x1f)<<0)
+
+#define CS42L51_ADCA_ATT		0x0C
+#define CS42L51_ADCB_ATT		0x0D
+
+#define CS42L51_ADCA_VOL		0x0E
+#define CS42L51_ADCB_VOL		0x0F
+#define CS42L51_PCMA_VOL		0x10
+#define CS42L51_PCMB_VOL		0x11
+#define CS42L51_MIX_MUTE_ADCMIX		(1<<7)
+#define CS42L51_MIX_VOLUME(x)		(((x)&0x7f)<<0)
+
+#define CS42L51_BEEP_FREQ		0x12
+#define CS42L51_BEEP_VOL		0x13
+#define CS42L51_BEEP_CONF		0x14
+
+#define CS42L51_TONE_CTL		0x15
+#define CS42L51_TONE_CTL_TREB(x)	(((x)&0xf)<<4)
+#define CS42L51_TONE_CTL_BASS(x)	(((x)&0xf)<<0)
+
+#define CS42L51_AOUTA_VOL		0x16
+#define CS42L51_AOUTB_VOL		0x17
+#define CS42L51_PCM_MIXER		0x18
+#define CS42L51_LIMIT_THRES_DIS		0x19
+#define CS42L51_LIMIT_REL		0x1A
+#define CS42L51_LIMIT_ATT		0x1B
+#define CS42L51_ALC_EN			0x1C
+#define CS42L51_ALC_REL			0x1D
+#define CS42L51_ALC_THRES		0x1E
+#define CS42L51_NOISE_CONF		0x1F
+
+#define CS42L51_STATUS			0x20
+#define CS42L51_STATUS_SP_CLKERR	(1<<6)
+#define CS42L51_STATUS_SPEA_OVFL	(1<<5)
+#define CS42L51_STATUS_SPEB_OVFL	(1<<4)
+#define CS42L51_STATUS_PCMA_OVFL	(1<<3)
+#define CS42L51_STATUS_PCMB_OVFL	(1<<2)
+#define CS42L51_STATUS_ADCA_OVFL	(1<<1)
+#define CS42L51_STATUS_ADCB_OVFL	(1<<0)
+
+#define CS42L51_CHARGE_FREQ		0x21
+
+#define CS42L51_FIRSTREG	0x01
+/*
+ * Hack: with register 0x21, it makes 33 registers. Looks like someone in the
+ * i2c layer doesn't like i2c smbus block read of 33 regs. Workaround by using
+ * 32 regs
+ */
+#define CS42L51_LASTREG		0x20
+#define CS42L51_NUMREGS		(CS42L51_LASTREG - CS42L51_FIRSTREG + 1)
+
+extern struct snd_soc_dai cs42l51_dai;
+extern struct snd_soc_codec_device soc_codec_device_cs42l51;
+#endif
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 75af2d6e0e78..3c51d6a57523 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -15,23 +15,15 @@
  * option) any later version.
  */
 
-#include <linux/module.h>
-#include <linux/moduleparam.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/tlv.h>
 #include <sound/initval.h>
-#include <asm/div64.h>
+#include <sound/tlv.h>
 
 #include "da7210.h"
 
@@ -145,6 +137,29 @@
 
 #define DA7210_VERSION "0.0.1"
 
+/*
+ * Playback Volume
+ *
+ * max		: 0x3F (+15.0 dB)
+ *		   (1.5 dB step)
+ * min		: 0x11 (-54.0 dB)
+ * mute		: 0x10
+ * reserved	: 0x00 - 0x0F
+ *
+ * ** FIXME **
+ *
+ * Reserved area are considered as "mute".
+ * -> min = -79.5 dB
+ */
+static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1);
+
+static const struct snd_kcontrol_new da7210_snd_controls[] = {
+
+	SOC_DOUBLE_R_TLV("HeadPhone Playback Volume",
+			 DA7210_HP_L_VOL, DA7210_HP_R_VOL,
+			 0, 0x3F, 0, hp_out_tlv),
+};
+
 /* Codec private data */
 struct da7210_priv {
 	struct snd_soc_codec codec;
@@ -227,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = dai->codec;
 
 	if (is_play) {
-		/* PlayBack Volume 40 */
-		snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40);
-		snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40);
-
 		/* Enable Out */
 		snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10);
 		snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10);
@@ -488,7 +499,7 @@ static int da7210_init(struct da7210_priv *da7210)
 	ret = snd_soc_register_dai(&da7210_dai);
 	if (ret) {
 		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		goto init_err;
+		goto codec_err;
 	}
 
 	/* FIXME
@@ -574,6 +585,8 @@ static int da7210_init(struct da7210_priv *da7210)
 
 	return ret;
 
+codec_err:
+	snd_soc_unregister_codec(codec);
 init_err:
 	kfree(codec->reg_cache);
 	codec->reg_cache = NULL;
@@ -601,8 +614,10 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
 	codec->control_data = i2c;
 
 	ret = da7210_init(da7210);
-	if (ret < 0)
+	if (ret < 0) {
 		pr_err("Failed to initialise da7210 audio codec\n");
+		kfree(da7210);
+	}
 
 	return ret;
 }
@@ -656,6 +671,9 @@ static int da7210_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto pcm_err;
 
+	snd_soc_add_controls(da7210_codec, da7210_snd_controls,
+			     ARRAY_SIZE(da7210_snd_controls));
+
 	dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
 
 pcm_err:
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
new file mode 100644
index 000000000000..66557de1e4fe
--- /dev/null
+++ b/sound/soc/codecs/jz4740.c
@@ -0,0 +1,511 @@
+/*
+ * Copyright (C) 2009-2010, 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 version 2 as
+ * published by the Free Software Foundation.
+ *
+ *  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.,
+ *  675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc.h>
+
+#define JZ4740_REG_CODEC_1 0x0
+#define JZ4740_REG_CODEC_2 0x1
+
+#define JZ4740_CODEC_1_LINE_ENABLE BIT(29)
+#define JZ4740_CODEC_1_MIC_ENABLE BIT(28)
+#define JZ4740_CODEC_1_SW1_ENABLE BIT(27)
+#define JZ4740_CODEC_1_ADC_ENABLE BIT(26)
+#define JZ4740_CODEC_1_SW2_ENABLE BIT(25)
+#define JZ4740_CODEC_1_DAC_ENABLE BIT(24)
+#define JZ4740_CODEC_1_VREF_DISABLE BIT(20)
+#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19)
+#define JZ4740_CODEC_1_VREF_PULLDOWN BIT(18)
+#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17)
+#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16)
+#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14)
+#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13)
+#define JZ4740_CODEC_1_HEADPHONE_CHARGE BIT(12)
+#define JZ4740_CODEC_1_HEADPHONE_PULLDOWN (BIT(11) | BIT(10))
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M BIT(9)
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN BIT(8)
+#define JZ4740_CODEC_1_SUSPEND BIT(1)
+#define JZ4740_CODEC_1_RESET BIT(0)
+
+#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29
+#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28
+#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27
+#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26
+#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25
+#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24
+#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET 8
+
+#define JZ4740_CODEC_2_INPUT_VOLUME_MASK		0x1f0000
+#define JZ4740_CODEC_2_SAMPLE_RATE_MASK			0x000f00
+#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK		0x000030
+#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK	0x000003
+
+#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET		16
+#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET		 8
+#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET	 4
+#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET	 0
+
+static const uint32_t jz4740_codec_regs[] = {
+	0x021b2302, 0x00170803,
+};
+
+struct jz4740_codec {
+	void __iomem *base;
+	struct resource *mem;
+
+	uint32_t reg_cache[2];
+	struct snd_soc_codec codec;
+};
+
+static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec)
+{
+	return container_of(codec, struct jz4740_codec, codec);
+}
+
+static unsigned int jz4740_codec_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
+	return readl(jz4740_codec->base + (reg << 2));
+}
+
+static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int val)
+{
+	struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
+
+	jz4740_codec->reg_cache[reg] = val;
+	writel(val, jz4740_codec->base + (reg << 2));
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new jz4740_codec_controls[] = {
+	SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2,
+			JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0),
+	SOC_SINGLE("Master Capture Volume", JZ4740_REG_CODEC_2,
+			JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0),
+	SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1),
+	SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2,
+			JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0),
+};
+
+static const struct snd_kcontrol_new jz4740_codec_output_controls[] = {
+	SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0),
+	SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0),
+};
+
+static const struct snd_kcontrol_new jz4740_codec_input_controls[] = {
+	SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0),
+	SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = {
+	SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0),
+	SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0),
+
+	SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET, 1,
+			jz4740_codec_output_controls,
+			ARRAY_SIZE(jz4740_codec_output_controls)),
+
+	SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0,
+			jz4740_codec_input_controls,
+			ARRAY_SIZE(jz4740_codec_input_controls)),
+	SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+
+	SND_SOC_DAPM_INPUT("MIC"),
+	SND_SOC_DAPM_INPUT("LIN"),
+	SND_SOC_DAPM_INPUT("RIN"),
+};
+
+static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = {
+	{"Line Input", NULL, "LIN"},
+	{"Line Input", NULL, "RIN"},
+
+	{"Input Mixer", "Line Capture Switch", "Line Input"},
+	{"Input Mixer", "Mic Capture Switch", "MIC"},
+
+	{"ADC", NULL, "Input Mixer"},
+
+	{"Output Mixer", "Bypass Switch", "Input Mixer"},
+	{"Output Mixer", "DAC Switch", "DAC"},
+
+	{"LOUT", NULL, "Output Mixer"},
+	{"ROUT", NULL, "Output Mixer"},
+};
+
+static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	uint32_t val;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	switch (params_rate(params)) {
+	case 8000:
+		val = 0;
+		break;
+	case 11025:
+		val = 1;
+		break;
+	case 12000:
+		val = 2;
+		break;
+	case 16000:
+		val = 3;
+		break;
+	case 22050:
+		val = 4;
+		break;
+	case 24000:
+		val = 5;
+		break;
+	case 32000:
+		val = 6;
+		break;
+	case 44100:
+		val = 7;
+		break;
+	case 48000:
+		val = 8;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET;
+
+	snd_soc_update_bits(codec, JZ4740_REG_CODEC_2,
+				JZ4740_CODEC_2_SAMPLE_RATE_MASK, val);
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops jz4740_codec_dai_ops = {
+	.hw_params = jz4740_codec_hw_params,
+};
+
+struct snd_soc_dai jz4740_codec_dai = {
+	.name = "jz4740",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+	},
+	.ops = &jz4740_codec_dai_ops,
+	.symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(jz4740_codec_dai);
+
+static void jz4740_codec_wakeup(struct snd_soc_codec *codec)
+{
+	int i;
+	uint32_t *cache = codec->reg_cache;
+
+	snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+		JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET);
+	udelay(2);
+
+	snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+		JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0);
+
+	for (i = 0; i < ARRAY_SIZE(jz4740_codec_regs); ++i)
+		jz4740_codec_write(codec, i, cache[i]);
+}
+
+static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	unsigned int mask;
+	unsigned int value;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		mask = JZ4740_CODEC_1_VREF_DISABLE |
+				JZ4740_CODEC_1_VREF_AMP_DISABLE |
+				JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+		value = 0;
+
+		snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* The only way to clear the suspend flag is to reset the codec */
+		if (codec->bias_level == SND_SOC_BIAS_OFF)
+			jz4740_codec_wakeup(codec);
+
+		mask = JZ4740_CODEC_1_VREF_DISABLE |
+			JZ4740_CODEC_1_VREF_AMP_DISABLE |
+			JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+		value = JZ4740_CODEC_1_VREF_DISABLE |
+			JZ4740_CODEC_1_VREF_AMP_DISABLE |
+			JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+
+		snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+		break;
+	case SND_SOC_BIAS_OFF:
+		mask = JZ4740_CODEC_1_SUSPEND;
+		value = JZ4740_CODEC_1_SUSPEND;
+
+		snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+		break;
+	default:
+		break;
+	}
+
+	codec->bias_level = level;
+
+	return 0;
+}
+
+static struct snd_soc_codec *jz4740_codec_codec;
+
+static int jz4740_codec_dev_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = jz4740_codec_codec;
+
+	BUG_ON(!codec);
+
+	socdev->card->codec = codec;
+
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret);
+		return ret;
+	}
+
+	snd_soc_add_controls(codec, jz4740_codec_controls,
+		ARRAY_SIZE(jz4740_codec_controls));
+
+	snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets,
+		ARRAY_SIZE(jz4740_codec_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes,
+		ARRAY_SIZE(jz4740_codec_dapm_routes));
+
+	snd_soc_dapm_new_widgets(codec);
+
+	return 0;
+}
+
+static int jz4740_codec_dev_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+
+static int jz4740_codec_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int jz4740_codec_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+#else
+#define jz4740_codec_suspend NULL
+#define jz4740_codec_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_jz4740_codec = {
+	.probe = jz4740_codec_dev_probe,
+	.remove = jz4740_codec_dev_remove,
+	.suspend = jz4740_codec_suspend,
+	.resume = jz4740_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec);
+
+static int __devinit jz4740_codec_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct jz4740_codec *jz4740_codec;
+	struct snd_soc_codec *codec;
+	struct resource *mem;
+
+	jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL);
+	if (!jz4740_codec)
+		return -ENOMEM;
+
+	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem) {
+		dev_err(&pdev->dev, "Failed to get mmio memory resource\n");
+		ret = -ENOENT;
+		goto err_free_codec;
+	}
+
+	mem = request_mem_region(mem->start, resource_size(mem), pdev->name);
+	if (!mem) {
+		dev_err(&pdev->dev, "Failed to request mmio memory region\n");
+		ret = -EBUSY;
+		goto err_free_codec;
+	}
+
+	jz4740_codec->base = ioremap(mem->start, resource_size(mem));
+	if (!jz4740_codec->base) {
+		dev_err(&pdev->dev, "Failed to ioremap mmio memory\n");
+		ret = -EBUSY;
+		goto err_release_mem_region;
+	}
+	jz4740_codec->mem = mem;
+
+	jz4740_codec_dai.dev = &pdev->dev;
+
+	codec = &jz4740_codec->codec;
+
+	codec->dev		= &pdev->dev;
+	codec->name		= "jz4740";
+	codec->owner		= THIS_MODULE;
+
+	codec->read		= jz4740_codec_read;
+	codec->write		= jz4740_codec_write;
+	codec->set_bias_level	= jz4740_codec_set_bias_level;
+	codec->bias_level	= SND_SOC_BIAS_OFF;
+
+	codec->dai		= &jz4740_codec_dai;
+	codec->num_dai		= 1;
+
+	codec->reg_cache	= jz4740_codec->reg_cache;
+	codec->reg_cache_size	= 2;
+	memcpy(codec->reg_cache, jz4740_codec_regs, sizeof(jz4740_codec_regs));
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	jz4740_codec_codec = codec;
+
+	snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+			JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE);
+
+	platform_set_drvdata(pdev, jz4740_codec);
+
+	ret = snd_soc_register_codec(codec);
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to register codec\n");
+		goto err_iounmap;
+	}
+
+	ret = snd_soc_register_dai(&jz4740_codec_dai);
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to register codec dai\n");
+		goto err_unregister_codec;
+	}
+
+	jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+
+err_unregister_codec:
+	snd_soc_unregister_codec(codec);
+err_iounmap:
+	iounmap(jz4740_codec->base);
+err_release_mem_region:
+	release_mem_region(mem->start, resource_size(mem));
+err_free_codec:
+	kfree(jz4740_codec);
+
+	return ret;
+}
+
+static int __devexit jz4740_codec_remove(struct platform_device *pdev)
+{
+	struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev);
+	struct resource *mem = jz4740_codec->mem;
+
+	snd_soc_unregister_dai(&jz4740_codec_dai);
+	snd_soc_unregister_codec(&jz4740_codec->codec);
+
+	iounmap(jz4740_codec->base);
+	release_mem_region(mem->start, resource_size(mem));
+
+	platform_set_drvdata(pdev, NULL);
+	kfree(jz4740_codec);
+
+	return 0;
+}
+
+static struct platform_driver jz4740_codec_driver = {
+	.probe = jz4740_codec_probe,
+	.remove = __devexit_p(jz4740_codec_remove),
+	.driver = {
+		.name = "jz4740-codec",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init jz4740_codec_init(void)
+{
+	return platform_driver_register(&jz4740_codec_driver);
+}
+module_init(jz4740_codec_init);
+
+static void __exit jz4740_codec_exit(void)
+{
+	platform_driver_unregister(&jz4740_codec_driver);
+}
+module_exit(jz4740_codec_exit);
+
+MODULE_DESCRIPTION("JZ4740 SoC internal codec driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:jz4740-codec");
diff --git a/sound/soc/codecs/jz4740.h b/sound/soc/codecs/jz4740.h
new file mode 100644
index 000000000000..b5a0691be763
--- /dev/null
+++ b/sound/soc/codecs/jz4740.h
@@ -0,0 +1,20 @@
+/*
+ * Copyright (C) 2009, 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 version 2 as
+ * published by the Free Software Foundation.
+ *
+ *  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.,
+ *  675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__
+#define __SND_SOC_CODECS_JZ4740_CODEC_H__
+
+extern struct snd_soc_dai jz4740_codec_dai;
+extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec;
+
+#endif
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
index a63191141052..9119836051a4 100644
--- a/sound/soc/codecs/spdif_transciever.c
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -16,8 +16,10 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
+#include <linux/slab.h>
 #include <sound/soc.h>
 #include <sound/pcm.h>
+#include <sound/initval.h>
 
 #include "spdif_transciever.h"
 
@@ -26,6 +28,48 @@ MODULE_LICENSE("GPL");
 #define STUB_RATES	SNDRV_PCM_RATE_8000_96000
 #define STUB_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
 
+static struct snd_soc_codec *spdif_dit_codec;
+
+static int spdif_dit_codec_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret;
+
+	if (spdif_dit_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = spdif_dit_codec;
+	codec = spdif_dit_codec;
+
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto err_create_pcms;
+	}
+
+	return 0;
+
+err_create_pcms:
+	return ret;
+}
+
+static int spdif_dit_codec_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_spdif_dit = {
+	.probe		= spdif_dit_codec_probe,
+	.remove		= spdif_dit_codec_remove,
+}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit);
+
 struct snd_soc_dai dit_stub_dai = {
 	.name		= "DIT",
 	.playback 	= {
@@ -40,13 +84,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai);
 
 static int spdif_dit_probe(struct platform_device *pdev)
 {
+	struct snd_soc_codec *codec;
+	int ret;
+
+	if (spdif_dit_codec) {
+		dev_err(&pdev->dev, "Another Codec is registered\n");
+		ret = -EINVAL;
+		goto err_reg_codec;
+	}
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	codec->dev = &pdev->dev;
+
+	mutex_init(&codec->mutex);
+
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->name = "spdif-dit";
+	codec->owner = THIS_MODULE;
+	codec->dai = &dit_stub_dai;
+	codec->num_dai = 1;
+
+	spdif_dit_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err_reg_codec;
+	}
+
 	dit_stub_dai.dev = &pdev->dev;
-	return snd_soc_register_dai(&dit_stub_dai);
+	ret = snd_soc_register_dai(&dit_stub_dai);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to register dai: %d\n", ret);
+		goto err_reg_dai;
+	}
+
+	return 0;
+
+err_reg_dai:
+	snd_soc_unregister_codec(codec);
+err_reg_codec:
+	kfree(spdif_dit_codec);
+	return ret;
 }
 
 static int spdif_dit_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_dai(&dit_stub_dai);
+	snd_soc_unregister_codec(spdif_dit_codec);
+	kfree(spdif_dit_codec);
+	spdif_dit_codec = NULL;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h
index 296f2eb6c4ef..1e102124f546 100644
--- a/sound/soc/codecs/spdif_transciever.h
+++ b/sound/soc/codecs/spdif_transciever.h
@@ -12,6 +12,7 @@
 #ifndef CODEC_STUBS_H
 #define CODEC_STUBS_H
 
+extern struct snd_soc_codec_device soc_codec_dev_spdif_dit;
 extern struct snd_soc_dai dit_stub_dai;
 
 #endif /* CODEC_STUBS_H */
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index b0bae3508b29..0a4b0fef3355 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -560,13 +560,16 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		/* vref/mid, osc on, dac unmute */
+		reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
+			TLV320AIC23_DAC_OFF);
 		tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* everything off except vref/vmid, */
-		tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
+		tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \
+			TLV320AIC23_CLK_OFF);
 		break;
 	case SND_SOC_BIAS_OFF:
 		/* everything off, dac mute, inactive */
@@ -615,7 +618,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 
-	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
@@ -632,7 +634,6 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 		u16 val = tlv320aic23_read_reg_cache(codec, reg);
 		tlv320aic23_write(codec, reg, val);
 	}
-
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 65adc77eada1..8651b01ed223 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -49,8 +49,6 @@
 
 #define NSAMPLE_MAX		5700
 
-#define LATENCY_TIME_MS		20
-
 #define MODE7_LTHR		10
 #define MODE7_UTHR		(DAC33_BUFFER_SIZE_SAMPLES - 10)
 
@@ -62,6 +60,9 @@
 #define US_TO_SAMPLES(rate, us) \
 	(rate / (1000000 / us))
 
+#define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
+	((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate)))
+
 static void dac33_calculate_times(struct snd_pcm_substream *substream);
 static int dac33_prepare_chip(struct snd_pcm_substream *substream);
 
@@ -107,6 +108,10 @@ struct tlv320dac33_priv {
 					 * this */
 	enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
 	unsigned int nsample;		/* burst read amount from host */
+	int mode1_latency;		/* latency caused by the i2c writes in
+					 * us */
+	int auto_fifo_config; 		/* Configure the FIFO based on the
+					 * period size */
 	u8 burst_bclkdiv;		/* BCLK divider value in burst mode */
 	unsigned int burst_rate;	/* Interface speed in Burst modes */
 
@@ -120,6 +125,8 @@ struct tlv320dac33_priv {
 					 * samples */
 	unsigned int mode7_us_to_lthr;	/* Time to reach lthr from uthr */
 
+	unsigned int uthr;
+
 	enum dac33_state state;
 };
 
@@ -442,6 +449,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
+static int dac33_get_uthr(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = dac33->uthr;
+
+	return 0;
+}
+
+static int dac33_set_uthr(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	if (dac33->substream)
+		return -EBUSY;
+
+	if (dac33->uthr == ucontrol->value.integer.value[0])
+		return 0;
+
+	if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) ||
+	    ucontrol->value.integer.value[0] > MODE7_UTHR)
+		ret = -EINVAL;
+	else
+		dac33->uthr = ucontrol->value.integer.value[0];
+
+	return ret;
+}
+
 static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
@@ -503,13 +543,18 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = {
 		 DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1),
 };
 
-static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = {
-	SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
-		 dac33_get_nsample, dac33_set_nsample),
+static const struct snd_kcontrol_new dac33_mode_snd_controls[] = {
 	SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum,
 		 dac33_get_fifo_mode, dac33_set_fifo_mode),
 };
 
+static const struct snd_kcontrol_new dac33_fifo_snd_controls[] = {
+	SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
+		dac33_get_nsample, dac33_set_nsample),
+	SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0,
+		 dac33_get_uthr, dac33_set_uthr),
+};
+
 /* Analog bypass */
 static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
 	SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1);
@@ -612,7 +657,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_MODE1:
 		dac33_write16(codec, DAC33_NSAMPLE_MSB,
-			DAC33_THRREG(dac33->nsample + dac33->alarm_threshold));
+			DAC33_THRREG(dac33->nsample));
 
 		/* Take the timestamps */
 		spin_lock_irq(&dac33->lock);
@@ -761,6 +806,10 @@ static void dac33_shutdown(struct snd_pcm_substream *substream,
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	dac33->substream = NULL;
+
+	/* Reset the nSample restrictions */
+	dac33->nsample_min = 0;
+	dac33->nsample_max = NSAMPLE_MAX;
 }
 
 static int dac33_hw_params(struct snd_pcm_substream *substream,
@@ -985,7 +1034,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		 * Configure the threshold levels, and leave 10 sample space
 		 * at the bottom, and also at the top of the FIFO
 		 */
-		dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR));
+		dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr));
 		dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR));
 		break;
 	default:
@@ -1003,57 +1052,71 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+	unsigned int period_size = substream->runtime->period_size;
+	unsigned int rate = substream->runtime->rate;
 	unsigned int nsample_limit;
 
 	/* In bypass mode we don't need to calculate */
 	if (!dac33->fifo_mode)
 		return;
 
-	/* Number of samples (16bit, stereo) in one period */
-	dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4;
-
-	/* Number of samples (16bit, stereo) in ALSA buffer */
-	dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4;
-	/* Subtract one period from the total */
-	dac33->nsample_max -= dac33->nsample_min;
-
-	/* Number of samples for LATENCY_TIME_MS / 2 */
-	dac33->alarm_threshold = substream->runtime->rate /
-				 (1000 / (LATENCY_TIME_MS / 2));
-
-	/* Find and fix up the lowest nsmaple limit */
-	nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS);
-
-	if (dac33->nsample_min < nsample_limit)
-		dac33->nsample_min = nsample_limit;
-
-	if (dac33->nsample < dac33->nsample_min)
-		dac33->nsample = dac33->nsample_min;
-
-	/*
-	 * Find and fix up the highest nsmaple limit
-	 * In order to not overflow the DAC33 buffer substract the
-	 * alarm_threshold value from the size of the DAC33 buffer
-	 */
-	nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold;
-
-	if (dac33->nsample_max > nsample_limit)
-		dac33->nsample_max = nsample_limit;
-
-	if (dac33->nsample > dac33->nsample_max)
-		dac33->nsample = dac33->nsample_max;
-
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_MODE1:
+		/* Number of samples under i2c latency */
+		dac33->alarm_threshold = US_TO_SAMPLES(rate,
+						dac33->mode1_latency);
+		if (dac33->auto_fifo_config) {
+			if (period_size <= dac33->alarm_threshold)
+				/*
+				 * Configure nSamaple to number of periods,
+				 * which covers the latency requironment.
+				 */
+				dac33->nsample = period_size *
+				       ((dac33->alarm_threshold / period_size) +
+				       (dac33->alarm_threshold % period_size ?
+				       1 : 0));
+			else
+				dac33->nsample = period_size;
+		} else {
+			/* nSample time shall not be shorter than i2c latency */
+			dac33->nsample_min = dac33->alarm_threshold;
+			/*
+			 * nSample should not be bigger than alsa buffer minus
+			 * size of one period to avoid overruns
+			 */
+			dac33->nsample_max = substream->runtime->buffer_size -
+						period_size;
+			nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
+					dac33->alarm_threshold;
+			if (dac33->nsample_max > nsample_limit)
+				dac33->nsample_max = nsample_limit;
+
+			/* Correct the nSample if it is outside of the ranges */
+			if (dac33->nsample < dac33->nsample_min)
+				dac33->nsample = dac33->nsample_min;
+			if (dac33->nsample > dac33->nsample_max)
+				dac33->nsample = dac33->nsample_max;
+		}
+
 		dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
 						      dac33->nsample);
 		dac33->t_stamp1 = 0;
 		dac33->t_stamp2 = 0;
 		break;
 	case DAC33_FIFO_MODE7:
+		if (dac33->auto_fifo_config) {
+			dac33->uthr = UTHR_FROM_PERIOD_SIZE(
+					period_size,
+					rate,
+					dac33->burst_rate) + 9;
+			if (dac33->uthr > MODE7_UTHR)
+				dac33->uthr = MODE7_UTHR;
+			if (dac33->uthr < (MODE7_LTHR + 10))
+				dac33->uthr = (MODE7_LTHR + 10);
+		}
 		dac33->mode7_us_to_lthr =
-					SAMPLES_TO_US(substream->runtime->rate,
-						MODE7_UTHR - MODE7_LTHR + 1);
+				SAMPLES_TO_US(substream->runtime->rate,
+					dac33->uthr - MODE7_LTHR + 1);
 		dac33->t_stamp1 = 0;
 		break;
 	default:
@@ -1104,7 +1167,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	unsigned long long t0, t1, t_now;
-	unsigned int time_delta;
+	unsigned int time_delta, uthr;
 	int samples_out, samples_in, samples;
 	snd_pcm_sframes_t delay = 0;
 
@@ -1182,6 +1245,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
 	case DAC33_FIFO_MODE7:
 		spin_lock(&dac33->lock);
 		t0 = dac33->t_stamp1;
+		uthr = dac33->uthr;
 		spin_unlock(&dac33->lock);
 		t_now = ktime_to_us(ktime_get());
 
@@ -1194,7 +1258,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
 			 * Either the timestamps are messed or equal. Report
 			 * maximum delay
 			 */
-			delay = MODE7_UTHR;
+			delay = uthr;
 			goto out;
 		}
 
@@ -1208,8 +1272,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
 					substream->runtime->rate,
 					time_delta);
 
-			if (likely(MODE7_UTHR > samples_out))
-				delay = MODE7_UTHR - samples_out;
+			if (likely(uthr > samples_out))
+				delay = uthr - samples_out;
 			else
 				delay = 0;
 		} else {
@@ -1227,8 +1291,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
 					time_delta);
 			delay = MODE7_LTHR + samples_in - samples_out;
 
-			if (unlikely(delay > MODE7_UTHR))
-				delay = MODE7_UTHR;
+			if (unlikely(delay > uthr))
+				delay = uthr;
 		}
 		break;
 	default:
@@ -1347,10 +1411,15 @@ static int dac33_soc_probe(struct platform_device *pdev)
 
 	snd_soc_add_controls(codec, dac33_snd_controls,
 			     ARRAY_SIZE(dac33_snd_controls));
-	/* Only add the nSample controls, if we have valid IRQ number */
-	if (dac33->irq >= 0)
-		snd_soc_add_controls(codec, dac33_nsample_snd_controls,
-				     ARRAY_SIZE(dac33_nsample_snd_controls));
+	/* Only add the FIFO controls, if we have valid IRQ number */
+	if (dac33->irq >= 0) {
+		snd_soc_add_controls(codec, dac33_mode_snd_controls,
+				     ARRAY_SIZE(dac33_mode_snd_controls));
+		/* FIFO usage controls only, if autoio config is not selected */
+		if (!dac33->auto_fifo_config)
+			snd_soc_add_controls(codec, dac33_fifo_snd_controls,
+					ARRAY_SIZE(dac33_fifo_snd_controls));
+	}
 
 	dac33_add_widgets(codec);
 
@@ -1481,9 +1550,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 	/* Pre calculate the burst rate */
 	dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32;
 	dac33->keep_bclk = pdata->keep_bclk;
+	dac33->auto_fifo_config = pdata->auto_fifo_config;
+	dac33->mode1_latency = pdata->mode1_latency;
+	if (!dac33->mode1_latency)
+		dac33->mode1_latency = 10000; /* 10ms */
 	dac33->irq = client->irq;
 	dac33->nsample = NSAMPLE_MAX;
 	dac33->nsample_max = NSAMPLE_MAX;
+	dac33->uthr = MODE7_UTHR;
 	/* Disable FIFO use by default */
 	dac33->fifo_mode = DAC33_FIFO_BYPASS;
 
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index b4fcdb01fc49..7b618bbff884 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -43,37 +43,37 @@
  */
 static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* this register not used		*/
-	0x91, /* REG_CODEC_MODE		(0x1)	*/
-	0xc3, /* REG_OPTION		(0x2)	*/
+	0x00, /* REG_CODEC_MODE		(0x1)	*/
+	0x00, /* REG_OPTION		(0x2)	*/
 	0x00, /* REG_UNKNOWN		(0x3)	*/
 	0x00, /* REG_MICBIAS_CTL	(0x4)	*/
-	0x20, /* REG_ANAMICL		(0x5)	*/
+	0x00, /* REG_ANAMICL		(0x5)	*/
 	0x00, /* REG_ANAMICR		(0x6)	*/
 	0x00, /* REG_AVADC_CTL		(0x7)	*/
 	0x00, /* REG_ADCMICSEL		(0x8)	*/
 	0x00, /* REG_DIGMIXING		(0x9)	*/
-	0x0c, /* REG_ATXL1PGA		(0xA)	*/
-	0x0c, /* REG_ATXR1PGA		(0xB)	*/
-	0x00, /* REG_AVTXL2PGA		(0xC)	*/
-	0x00, /* REG_AVTXR2PGA		(0xD)	*/
+	0x0f, /* REG_ATXL1PGA		(0xA)	*/
+	0x0f, /* REG_ATXR1PGA		(0xB)	*/
+	0x0f, /* REG_AVTXL2PGA		(0xC)	*/
+	0x0f, /* REG_AVTXR2PGA		(0xD)	*/
 	0x00, /* REG_AUDIO_IF		(0xE)	*/
 	0x00, /* REG_VOICE_IF		(0xF)	*/
-	0x00, /* REG_ARXR1PGA		(0x10)	*/
-	0x00, /* REG_ARXL1PGA		(0x11)	*/
-	0x6c, /* REG_ARXR2PGA		(0x12)	*/
-	0x6c, /* REG_ARXL2PGA		(0x13)	*/
-	0x00, /* REG_VRXPGA		(0x14)	*/
+	0x3f, /* REG_ARXR1PGA		(0x10)	*/
+	0x3f, /* REG_ARXL1PGA		(0x11)	*/
+	0x3f, /* REG_ARXR2PGA		(0x12)	*/
+	0x3f, /* REG_ARXL2PGA		(0x13)	*/
+	0x25, /* REG_VRXPGA		(0x14)	*/
 	0x00, /* REG_VSTPGA		(0x15)	*/
 	0x00, /* REG_VRX2ARXPGA		(0x16)	*/
 	0x00, /* REG_AVDAC_CTL		(0x17)	*/
 	0x00, /* REG_ARX2VTXPGA		(0x18)	*/
-	0x00, /* REG_ARXL1_APGA_CTL	(0x19)	*/
-	0x00, /* REG_ARXR1_APGA_CTL	(0x1A)	*/
-	0x4a, /* REG_ARXL2_APGA_CTL	(0x1B)	*/
-	0x4a, /* REG_ARXR2_APGA_CTL	(0x1C)	*/
+	0x32, /* REG_ARXL1_APGA_CTL	(0x19)	*/
+	0x32, /* REG_ARXR1_APGA_CTL	(0x1A)	*/
+	0x32, /* REG_ARXL2_APGA_CTL	(0x1B)	*/
+	0x32, /* REG_ARXR2_APGA_CTL	(0x1C)	*/
 	0x00, /* REG_ATX2ARXPGA		(0x1D)	*/
 	0x00, /* REG_BT_IF		(0x1E)	*/
-	0x00, /* REG_BTPGA		(0x1F)	*/
+	0x55, /* REG_BTPGA		(0x1F)	*/
 	0x00, /* REG_BTSTPGA		(0x20)	*/
 	0x00, /* REG_EAR_CTL		(0x21)	*/
 	0x00, /* REG_HS_SEL		(0x22)	*/
@@ -85,32 +85,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* REG_PRECKR_CTL		(0x28)	*/
 	0x00, /* REG_HFL_CTL		(0x29)	*/
 	0x00, /* REG_HFR_CTL		(0x2A)	*/
-	0x00, /* REG_ALC_CTL		(0x2B)	*/
+	0x05, /* REG_ALC_CTL		(0x2B)	*/
 	0x00, /* REG_ALC_SET1		(0x2C)	*/
 	0x00, /* REG_ALC_SET2		(0x2D)	*/
 	0x00, /* REG_BOOST_CTL		(0x2E)	*/
 	0x00, /* REG_SOFTVOL_CTL	(0x2F)	*/
-	0x00, /* REG_DTMF_FREQSEL	(0x30)	*/
+	0x13, /* REG_DTMF_FREQSEL	(0x30)	*/
 	0x00, /* REG_DTMF_TONEXT1H	(0x31)	*/
 	0x00, /* REG_DTMF_TONEXT1L	(0x32)	*/
 	0x00, /* REG_DTMF_TONEXT2H	(0x33)	*/
 	0x00, /* REG_DTMF_TONEXT2L	(0x34)	*/
-	0x00, /* REG_DTMF_TONOFF	(0x35)	*/
-	0x00, /* REG_DTMF_WANONOFF	(0x36)	*/
+	0x79, /* REG_DTMF_TONOFF	(0x35)	*/
+	0x11, /* REG_DTMF_WANONOFF	(0x36)	*/
 	0x00, /* REG_I2S_RX_SCRAMBLE_H	(0x37)	*/
 	0x00, /* REG_I2S_RX_SCRAMBLE_M	(0x38)	*/
 	0x00, /* REG_I2S_RX_SCRAMBLE_L	(0x39)	*/
 	0x06, /* REG_APLL_CTL		(0x3A)	*/
 	0x00, /* REG_DTMF_CTL		(0x3B)	*/
-	0x00, /* REG_DTMF_PGA_CTL2	(0x3C)	*/
-	0x00, /* REG_DTMF_PGA_CTL1	(0x3D)	*/
+	0x44, /* REG_DTMF_PGA_CTL2	(0x3C)	*/
+	0x69, /* REG_DTMF_PGA_CTL1	(0x3D)	*/
 	0x00, /* REG_MISC_SET_1		(0x3E)	*/
 	0x00, /* REG_PCMBTMUX		(0x3F)	*/
 	0x00, /* not used		(0x40)	*/
 	0x00, /* not used		(0x41)	*/
 	0x00, /* not used		(0x42)	*/
 	0x00, /* REG_RX_PATH_SEL	(0x43)	*/
-	0x00, /* REG_VDL_APGA_CTL	(0x44)	*/
+	0x32, /* REG_VDL_APGA_CTL	(0x44)	*/
 	0x00, /* REG_VIBRA_CTL		(0x45)	*/
 	0x00, /* REG_VIBRA_SET		(0x46)	*/
 	0x00, /* REG_VIBRA_PWM_SET	(0x47)	*/
@@ -143,6 +143,9 @@ struct twl4030_priv {
 	u8 earpiece_enabled;
 	u8 predrivel_enabled, predriver_enabled;
 	u8 carkitl_enabled, carkitr_enabled;
+
+	/* Delay needed after enabling the digimic interface */
+	unsigned int digimic_delay;
 };
 
 /*
@@ -244,58 +247,95 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 	udelay(10);
 }
 
-static void twl4030_init_chip(struct snd_soc_codec *codec)
+static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
 {
-	u8 *cache = codec->reg_cache;
-	int i;
+	int i, difference = 0;
+	u8 val;
+
+	dev_dbg(codec->dev, "Checking TWL audio default configuration\n");
+	for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) {
+		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i);
+		if (val != twl4030_reg[i]) {
+			difference++;
+			dev_dbg(codec->dev,
+				 "Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n",
+				 i, val, twl4030_reg[i]);
+		}
+	}
+	dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+		 difference, difference ? "Not OK" : "OK");
+}
 
-	/* clear CODECPDZ prior to setting register defaults */
-	twl4030_codec_enable(codec, 0);
+static inline void twl4030_reset_registers(struct snd_soc_codec *codec)
+{
+	int i;
 
 	/* set all audio section registers to reasonable defaults */
 	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
 		if (i != TWL4030_REG_APLL_CTL)
-			twl4030_write(codec, i,	cache[i]);
+			twl4030_write(codec, i, twl4030_reg[i]);
 
 }
 
-static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
+static void twl4030_init_chip(struct platform_device *pdev)
 {
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct twl4030_setup_data *setup = socdev->codec_data;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-	int status = -1;
+	u8 reg, byte;
+	int i = 0;
 
-	if (enable) {
-		twl4030->apll_enabled++;
-		if (twl4030->apll_enabled == 1)
-			status = twl4030_codec_enable_resource(
-							TWL4030_CODEC_RES_APLL);
-	} else {
-		twl4030->apll_enabled--;
-		if (!twl4030->apll_enabled)
-			status = twl4030_codec_disable_resource(
-							TWL4030_CODEC_RES_APLL);
-	}
+	/* Check defaults, if instructed before anything else */
+	if (setup && setup->check_defaults)
+		twl4030_check_defaults(codec);
 
-	if (status >= 0)
-		twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
-}
+	/* Reset registers, if no setup data or if instructed to do so */
+	if (!setup || (setup && setup->reset_registers))
+		twl4030_reset_registers(codec);
 
-static void twl4030_power_up(struct snd_soc_codec *codec)
-{
-	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-	u8 anamicl, regmisc1, byte;
-	int i = 0;
+	/* Refresh APLL_CTL register from HW */
+	twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+			    TWL4030_REG_APLL_CTL);
+	twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte);
+
+	/* anti-pop when changing analog gain */
+	reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+		reg | TWL4030_SMOOTH_ANAVOL_EN);
 
-	if (twl4030->codec_powered)
+	twl4030_write(codec, TWL4030_REG_OPTION,
+		TWL4030_ATXL1_EN | TWL4030_ATXR1_EN |
+		TWL4030_ARXL2_EN | TWL4030_ARXR2_EN);
+
+	/* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */
+	twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32);
+
+	/* Machine dependent setup */
+	if (!setup)
 		return;
 
-	/* set CODECPDZ to turn on codec */
-	twl4030_codec_enable(codec, 1);
+	twl4030->digimic_delay = setup->digimic_delay;
+
+	/* Configuration for headset ramp delay from setup data */
+	if (setup->sysclk != twl4030->sysclk)
+		dev_warn(codec->dev,
+				"Mismatch in APLL mclk: %u (configured: %u)\n",
+				setup->sysclk, twl4030->sysclk);
+
+	reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+	reg &= ~TWL4030_RAMP_DELAY;
+	reg |= (setup->ramp_delay_value << 2);
+	twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg);
 
 	/* initiate offset cancellation */
-	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	twl4030_codec_enable(codec, 1);
+
+	reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	reg &= ~TWL4030_OFFSET_CNCL_SEL;
+	reg |= setup->offset_cncl_path;
 	twl4030_write(codec, TWL4030_REG_ANAMICL,
-		anamicl | TWL4030_CNCL_OFFSET_START);
+		reg | TWL4030_CNCL_OFFSET_START);
 
 	/* wait for offset cancellation to complete */
 	do {
@@ -310,23 +350,28 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 	/* Make sure that the reg_cache has the same value as the HW */
 	twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
 
-	/* anti-pop when changing analog gain */
-	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
-	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
-		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
-
-	/* toggle CODECPDZ as per TRM */
 	twl4030_codec_enable(codec, 0);
-	twl4030_codec_enable(codec, 1);
 }
 
-/*
- * Unconditional power down
- */
-static void twl4030_power_down(struct snd_soc_codec *codec)
+static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 {
-	/* power down */
-	twl4030_codec_enable(codec, 0);
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
+	int status = -1;
+
+	if (enable) {
+		twl4030->apll_enabled++;
+		if (twl4030->apll_enabled == 1)
+			status = twl4030_codec_enable_resource(
+							TWL4030_CODEC_RES_APLL);
+	} else {
+		twl4030->apll_enabled--;
+		if (!twl4030->apll_enabled)
+			status = twl4030_codec_disable_resource(
+							TWL4030_CODEC_RES_APLL);
+	}
+
+	if (status >= 0)
+		twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
 }
 
 /* Earpiece */
@@ -500,10 +545,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
 static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
 	SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
 
-/* Digital bypass gain, 0 mutes the bypass */
+/* Digital bypass gain, mute instead of -30dB */
 static const unsigned int twl4030_dapm_dbypass_tlv[] = {
-	TLV_DB_RANGE_HEAD(2),
-	0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1),
+	TLV_DB_RANGE_HEAD(3),
+	0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1),
+	2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0),
 	4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0),
 };
 
@@ -531,36 +577,6 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
 			TWL4030_REG_VSTPGA, 0, 0x29, 0,
 			twl4030_dapm_dbypassv_tlv);
 
-static int micpath_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
-	unsigned char adcmicsel, micbias_ctl;
-
-	adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL);
-	micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL);
-	/* Prepare the bits for the given TX path:
-	 * shift_l == 0: TX1 microphone path
-	 * shift_l == 2: TX2 microphone path */
-	if (e->shift_l) {
-		/* TX2 microphone path */
-		if (adcmicsel & TWL4030_TX2IN_SEL)
-			micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */
-		else
-			micbias_ctl &= ~TWL4030_MICBIAS2_CTL;
-	} else {
-		/* TX1 microphone path */
-		if (adcmicsel & TWL4030_TX1IN_SEL)
-			micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */
-		else
-			micbias_ctl &= ~TWL4030_MICBIAS1_CTL;
-	}
-
-	twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl);
-
-	return 0;
-}
-
 /*
  * Output PGA builder:
  * Handle the muting and unmuting of the given output (turning off the
@@ -814,6 +830,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int digimic_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
+
+	if (twl4030->digimic_delay)
+		mdelay(twl4030->digimic_delay);
+	return 0;
+}
+
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
@@ -1374,14 +1400,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	/* Analog/Digital mic path selection.
 	   TX1 Left/Right: either analog Left/Right or Digimic0
 	   TX2 Left/Right: either analog Left/Right or Digimic1 */
-	SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_micpathtx1_control, micpath_event,
-		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
-		SND_SOC_DAPM_POST_REG),
-	SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_micpathtx2_control, micpath_event,
-		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
-		SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_MUX("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_micpathtx1_control),
+	SND_SOC_DAPM_MUX("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_micpathtx2_control),
 
 	/* Analog input mixers for the capture amplifiers */
 	SND_SOC_DAPM_MIXER("Analog Left",
@@ -1398,10 +1420,17 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("ADC Physical Right",
 		TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0),
 
-	SND_SOC_DAPM_PGA("Digimic0 Enable",
-		TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("Digimic1 Enable",
-		TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("Digimic0 Enable",
+		TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0,
+		digimic_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_E("Digimic1 Enable",
+		TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0,
+		digimic_event, SND_SOC_DAPM_POST_PMU),
+
+	SND_SOC_DAPM_SUPPLY("micbias1 select", TWL4030_REG_MICBIAS_CTL, 5, 0,
+			    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0,
+			    NULL, 0),
 
 	SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
 	SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
@@ -1419,8 +1448,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	/* Supply for the digital part (APLL) */
 	{"Digital Voice Playback Mixer", NULL, "APLL Enable"},
 
-	{"Digital R1 Playback Mixer", NULL, "AIF Enable"},
-	{"Digital L1 Playback Mixer", NULL, "AIF Enable"},
+	{"DAC Left1", NULL, "AIF Enable"},
+	{"DAC Right1", NULL, "AIF Enable"},
+	{"DAC Left2", NULL, "AIF Enable"},
+	{"DAC Right1", NULL, "AIF Enable"},
+
 	{"Digital R2 Playback Mixer", NULL, "AIF Enable"},
 	{"Digital L2 Playback Mixer", NULL, "AIF Enable"},
 
@@ -1491,10 +1523,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 
 	/* outputs */
 	/* Must be always connected (for AIF and APLL) */
-	{"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"},
-	{"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"},
-	{"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"},
-	{"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"},
+	{"Virtual HiFi OUT", NULL, "DAC Left1"},
+	{"Virtual HiFi OUT", NULL, "DAC Right1"},
+	{"Virtual HiFi OUT", NULL, "DAC Left2"},
+	{"Virtual HiFi OUT", NULL, "DAC Right2"},
 	/* Must be always connected (for APLL) */
 	{"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"},
 	/* Physical outputs */
@@ -1531,6 +1563,9 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Digimic0 Enable", NULL, "DIGIMIC0"},
 	{"Digimic1 Enable", NULL, "DIGIMIC1"},
 
+	{"DIGIMIC0", NULL, "micbias1 select"},
+	{"DIGIMIC1", NULL, "micbias2 select"},
+
 	/* TX1 Left capture path */
 	{"TX1 Capture Route", "Analog", "ADC Physical Left"},
 	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
@@ -1605,10 +1640,10 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->bias_level == SND_SOC_BIAS_OFF)
-			twl4030_power_up(codec);
+			twl4030_codec_enable(codec, 1);
 		break;
 	case SND_SOC_BIAS_OFF:
-		twl4030_power_down(codec);
+		twl4030_codec_enable(codec, 0);
 		break;
 	}
 	codec->bias_level = level;
@@ -1794,13 +1829,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	if (mode != old_mode) {
-		/* change rate and set CODECPDZ */
-		twl4030_codec_enable(codec, 0);
-		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
-		twl4030_codec_enable(codec, 1);
-	}
-
 	/* sample size */
 	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
 	format = old_format;
@@ -1818,16 +1846,20 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	if (format != old_format) {
-
-		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_codec_enable(codec, 0);
-
-		/* change format */
-		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
-
-		/* set CODECPDZ afterwards */
-		twl4030_codec_enable(codec, 1);
+	if (format != old_format || mode != old_mode) {
+		if (twl4030->codec_powered) {
+			/*
+			 * If the codec is powered, than we need to toggle the
+			 * codec power.
+			 */
+			twl4030_codec_enable(codec, 0);
+			twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+			twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+			twl4030_codec_enable(codec, 1);
+		} else {
+			twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+			twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+		}
 	}
 
 	/* Store the important parameters for the DAI configuration and set
@@ -1877,6 +1909,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			     unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 old_format, format;
 
 	/* get format */
@@ -1911,15 +1944,17 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	}
 
 	if (format != old_format) {
-
-		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_codec_enable(codec, 0);
-
-		/* change format */
-		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
-
-		/* set CODECPDZ afterwards */
-		twl4030_codec_enable(codec, 1);
+		if (twl4030->codec_powered) {
+			/*
+			 * If the codec is powered, than we need to toggle the
+			 * codec power.
+			 */
+			twl4030_codec_enable(codec, 0);
+			twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+			twl4030_codec_enable(codec, 1);
+		} else {
+			twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+		}
 	}
 
 	return 0;
@@ -2011,6 +2046,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 old_mode, mode;
 
 	/* Enable voice digital filters */
@@ -2035,10 +2071,17 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (mode != old_mode) {
-		/* change rate and set CODECPDZ */
-		twl4030_codec_enable(codec, 0);
-		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
-		twl4030_codec_enable(codec, 1);
+		if (twl4030->codec_powered) {
+			/*
+			 * If the codec is powered, than we need to toggle the
+			 * codec power.
+			 */
+			twl4030_codec_enable(codec, 0);
+			twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+			twl4030_codec_enable(codec, 1);
+		} else {
+			twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+		}
 	}
 
 	return 0;
@@ -2068,6 +2111,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 old_format, format;
 
 	/* get format */
@@ -2099,10 +2143,17 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	}
 
 	if (format != old_format) {
-		/* change format and set CODECPDZ */
-		twl4030_codec_enable(codec, 0);
-		twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
-		twl4030_codec_enable(codec, 1);
+		if (twl4030->codec_powered) {
+			/*
+			 * If the codec is powered, than we need to toggle the
+			 * codec power.
+			 */
+			twl4030_codec_enable(codec, 0);
+			twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+			twl4030_codec_enable(codec, 1);
+		} else {
+			twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+		}
 	}
 
 	return 0;
@@ -2202,31 +2253,15 @@ static struct snd_soc_codec *twl4030_codec;
 static int twl4030_soc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct twl4030_setup_data *setup = socdev->codec_data;
 	struct snd_soc_codec *codec;
-	struct twl4030_priv *twl4030;
 	int ret;
 
 	BUG_ON(!twl4030_codec);
 
 	codec = twl4030_codec;
-	twl4030 = snd_soc_codec_get_drvdata(codec);
 	socdev->card->codec = codec;
 
-	/* Configuration for headset ramp delay from setup data */
-	if (setup) {
-		unsigned char hs_pop;
-
-		if (setup->sysclk != twl4030->sysclk)
-			dev_warn(&pdev->dev,
-				 "Mismatch in APLL mclk: %u (configured: %u)\n",
-				 setup->sysclk, twl4030->sysclk);
-
-		hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-		hs_pop &= ~TWL4030_RAMP_DELAY;
-		hs_pop |= (setup->ramp_delay_value << 2);
-		twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-	}
+	twl4030_init_chip(pdev);
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -2247,6 +2282,8 @@ static int twl4030_soc_remove(struct platform_device *pdev)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 
+	/* Reset registers to their chip default before leaving */
+	twl4030_reset_registers(codec);
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
@@ -2287,6 +2324,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 	codec->read = twl4030_read_reg_cache;
 	codec->write = twl4030_write;
 	codec->set_bias_level = twl4030_set_bias_level;
+	codec->idle_bias_off = 1;
 	codec->dai = twl4030_dai;
 	codec->num_dai = ARRAY_SIZE(twl4030_dai);
 	codec->reg_cache_size = sizeof(twl4030_reg);
@@ -2302,9 +2340,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 
 	/* Set the defaults, and power up the codec */
 	twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
-	twl4030_init_chip(codec);
 	codec->bias_level = SND_SOC_BIAS_OFF;
-	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
@@ -2322,7 +2358,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 	return 0;
 
 error_codec:
-	twl4030_power_down(codec);
+	twl4030_codec_enable(codec, 0);
 	kfree(codec->reg_cache);
 error_cache:
 	kfree(twl4030);
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index f206d242ca31..6c57430f6e24 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -41,7 +41,11 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030;
 
 struct twl4030_setup_data {
 	unsigned int ramp_delay_value;
+	unsigned int digimic_delay; /* in ms */
 	unsigned int sysclk;
+	unsigned int offset_cncl_path;
+	unsigned int check_defaults:1;
+	unsigned int reset_registers:1;
 	unsigned int hs_extmute:1;
 	void (*set_hs_extmute)(int mute);
 };
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index af36346ff336..64a807f1a8a1 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
 	return 0;
 }
 
+static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *kcontrol, int event)
+{
+	msleep(1);
+	return 0;
+}
+
 static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
 			struct snd_kcontrol *kcontrol, int event)
 {
@@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
 	else
 		priv->non_lp--;
 
+	msleep(1);
+
 	return 0;
 }
 
@@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls =
 static const struct snd_kcontrol_new hfdacr_switch_controls =
 	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0);
 
-/* Headset driver switches */
-static const struct snd_kcontrol_new hsl_driver_switch_controls =
-	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0);
-
-static const struct snd_kcontrol_new hsr_driver_switch_controls =
-	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0);
-
-/* Handsfree driver switches */
-static const struct snd_kcontrol_new hfl_driver_switch_controls =
-	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0);
-
-static const struct snd_kcontrol_new hfr_driver_switch_controls =
-	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0);
-
 static const struct snd_kcontrol_new ep_driver_switch_controls =
 	SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0);
 
@@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 			TWL6040_REG_DMICBCTL, 4, 0),
 
 	/* DACs */
-	SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback",
-			TWL6040_REG_HSLCTL, 0, 0),
-	SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback",
-			TWL6040_REG_HSRCTL, 0, 0),
+	SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback",
+			TWL6040_REG_HSLCTL, 0, 0,
+			twl6040_hs_dac_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback",
+			TWL6040_REG_HSRCTL, 0, 0,
+			twl6040_hs_dac_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback",
 			TWL6040_REG_HFLCTL, 0, 0,
 			twl6040_power_mode_event,
@@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 	SND_SOC_DAPM_SWITCH("HFDAC Right Playback",
 			SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls),
 
-	SND_SOC_DAPM_SWITCH("Headset Left Driver",
-			SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls),
-	SND_SOC_DAPM_SWITCH("Headset Right Driver",
-			SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls),
-	SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver",
-			SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls,
+	/* Analog playback drivers */
+	SND_SOC_DAPM_PGA_E("Handsfree Left Driver",
+			TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
 			twl6040_power_mode_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
-	SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver",
-			SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls,
+	SND_SOC_DAPM_PGA_E("Handsfree Right Driver",
+			TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
 			twl6040_power_mode_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA("Headset Left Driver",
+			TWL6040_REG_HSLCTL, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Headset Right Driver",
+			TWL6040_REG_HSRCTL, 2, 0, NULL, 0),
 	SND_SOC_DAPM_SWITCH_E("Earphone Driver",
 			SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls,
 			twl6040_power_mode_event,
@@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"HSDAC Left Playback", "Switch", "HSDAC Left"},
 	{"HSDAC Right Playback", "Switch", "HSDAC Right"},
 
-	{"Headset Left Driver", "Switch", "HSDAC Left Playback"},
-	{"Headset Right Driver", "Switch", "HSDAC Right Playback"},
+	{"Headset Left Driver", NULL, "HSDAC Left Playback"},
+	{"Headset Right Driver", NULL, "HSDAC Right Playback"},
 
 	{"HSOL", NULL, "Headset Left Driver"},
 	{"HSOR", NULL, "Headset Right Driver"},
@@ -928,7 +928,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		case 19200000:
 			/* mclk input, pll disabled */
 			hppllctl |= TWL6040_MCLK_19200KHZ |
-				    TWL6040_HPLLSQRBP |
+				    TWL6040_HPLLSQRENA |
 				    TWL6040_HPLLBP;
 			break;
 		case 26000000:
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 28aac53c97bb..f3b4c1d6a82d 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -28,19 +28,6 @@
 #include "uda134x.h"
 
 
-#define POWER_OFF_ON_STANDBY 1
-/*
-  ALSA SOC usually puts the device in standby mode when it's not used
-  for sometime. If you define POWER_OFF_ON_STANDBY the driver will
-  turn off the ADC/DAC when this callback is invoked and turn it back
-  on when needed. Unfortunately this will result in a very light bump
-  (it can be audible only with good earphones). If this bothers you
-  just comment this line, you will have slightly higher power
-  consumption . Please note that sending the L3 command for ADC is
-  enough to make the bump, so it doesn't make difference if you
-  completely take off power from the codec.
- */
-
 #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
 		SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
@@ -58,7 +45,7 @@ static const char uda134x_reg[UDA134X_REGS_NUM] = {
 	/* Extended address registers */
 	0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
 	/* Status, data regs */
-	0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+	0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00,
 };
 
 /*
@@ -117,6 +104,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
 	case UDA134X_DATA000:
 	case UDA134X_DATA001:
 	case UDA134X_DATA010:
+	case UDA134X_DATA011:
 		addr = UDA134X_DATA0_ADDR;
 		break;
 	case UDA134X_DATA1:
@@ -353,8 +341,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		/* ADC, DAC on */
-		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
-		uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+		switch (pd->model) {
+		case UDA134X_UDA1340:
+		case UDA134X_UDA1344:
+		case UDA134X_UDA1345:
+			reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
+			uda134x_write(codec, UDA134X_DATA011, reg | 0x03);
+			break;
+		case UDA134X_UDA1341:
+			reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+			uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+			break;
+		default:
+			printk(KERN_ERR "UDA134X SoC codec: "
+			       "unsupported model %d\n", pd->model);
+			return -EINVAL;
+		}
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		/* power on */
@@ -367,8 +369,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* ADC, DAC power off */
-		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
-		uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+		switch (pd->model) {
+		case UDA134X_UDA1340:
+		case UDA134X_UDA1344:
+		case UDA134X_UDA1345:
+			reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
+			uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03));
+			break;
+		case UDA134X_UDA1341:
+			reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+			uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+			break;
+		default:
+			printk(KERN_ERR "UDA134X SoC codec: "
+			       "unsupported model %d\n", pd->model);
+			return -EINVAL;
+		}
 		break;
 	case SND_SOC_BIAS_OFF:
 		/* power off */
@@ -531,9 +547,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 	codec->num_dai = 1;
 	codec->read = uda134x_read_reg_cache;
 	codec->write = uda134x_write;
-#ifdef POWER_OFF_ON_STANDBY
-	codec->set_bias_level = uda134x_set_bias_level;
-#endif
+
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -544,6 +558,14 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 
 	uda134x_reset(codec);
 
+	if (pd->is_powered_on_standby) {
+		codec->set_bias_level = NULL;
+		uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+	} else {
+		codec->set_bias_level = uda134x_set_bias_level;
+		uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	}
+
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
index 94f440490b31..205f03b3eaf8 100644
--- a/sound/soc/codecs/uda134x.h
+++ b/sound/soc/codecs/uda134x.h
@@ -23,9 +23,10 @@
 #define UDA134X_DATA000 10
 #define UDA134X_DATA001 11
 #define UDA134X_DATA010 12
-#define UDA134X_DATA1	13
+#define UDA134X_DATA011 13
+#define UDA134X_DATA1   14
 
-#define UDA134X_REGS_NUM 14
+#define UDA134X_REGS_NUM 15
 
 #define STATUS0_DAIFMT_MASK (~(7<<1))
 #define STATUS0_SYSCLK_MASK (~(3<<4))
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 002e289d1255..4bcd168794e1 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -795,6 +795,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
 
 	dev_set_drvdata(&i2c->dev, wm2000);
 	wm2000->anc_eng_ena = 1;
+	wm2000->anc_active = 1;
+	wm2000->spk_ena = 1;
 	wm2000->i2c = i2c;
 
 	wm2000_reset(wm2000);
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 37242a7d3077..0ad039b4adf5 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -482,7 +482,8 @@ static int wm8523_register(struct wm8523_priv *wm8523,
 
 	if (wm8523_codec) {
 		dev_err(codec->dev, "Another WM8523 is registered\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err;
 	}
 
 	mutex_init(&codec->mutex);
@@ -570,18 +571,19 @@ static int wm8523_register(struct wm8523_priv *wm8523,
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		return ret;
+		goto err_enable;
 	}
 
 	ret = snd_soc_register_dai(&wm8523_dai);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		snd_soc_unregister_codec(codec);
-		return ret;
+		goto err_codec;
 	}
 
 	return 0;
 
+err_codec:
+	snd_soc_unregister_codec(codec);
 err_enable:
 	regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
 err_get:
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index effb14eee7d4..e2dba07f0260 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -439,7 +439,8 @@ static int wm8711_register(struct wm8711_priv *wm8711,
 
 	if (wm8711_codec) {
 		dev_err(codec->dev, "Another WM8711 is registered\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err;
 	}
 
 	mutex_init(&codec->mutex);
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
new file mode 100644
index 000000000000..b9ea8904ad4b
--- /dev/null
+++ b/sound/soc/codecs/wm8741.c
@@ -0,0 +1,579 @@
+/*
+ * wm8741.c  --  WM8741 ALSA SoC Audio driver
+ *
+ * Copyright 2010 Wolfson Microelectronics plc
+ *
+ * Author: Ian Lartey <ian@opensource.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.
+ */
+
+#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/regulator/consumer.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 "wm8741.h"
+
+static struct snd_soc_codec *wm8741_codec;
+struct snd_soc_codec_device soc_codec_dev_wm8741;
+
+#define WM8741_NUM_SUPPLIES 2
+static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = {
+	"AVDD",
+	"DVDD",
+};
+
+#define WM8741_NUM_RATES 4
+
+/* codec private data */
+struct wm8741_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8741_REGISTER_COUNT];
+	struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES];
+	unsigned int sysclk;
+	unsigned int rate_constraint_list[WM8741_NUM_RATES];
+	struct snd_pcm_hw_constraint_list rate_constraint;
+};
+
+static const u16 wm8741_reg_defaults[WM8741_REGISTER_COUNT] = {
+	0x0000,     /* R0  - DACLLSB Attenuation */
+	0x0000,     /* R1  - DACLMSB Attenuation */
+	0x0000,     /* R2  - DACRLSB Attenuation */
+	0x0000,     /* R3  - DACRMSB Attenuation */
+	0x0000,     /* R4  - Volume Control */
+	0x000A,     /* R5  - Format Control */
+	0x0000,     /* R6  - Filter Control */
+	0x0000,     /* R7  - Mode Control 1 */
+	0x0002,     /* R8  - Mode Control 2 */
+	0x0000,	    /* R9  - Reset */
+	0x0002,     /* R32 - ADDITONAL_CONTROL_1 */
+};
+
+
+static int wm8741_reset(struct snd_soc_codec *codec)
+{
+	return snd_soc_write(codec, WM8741_RESET, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0);
+
+static const struct snd_kcontrol_new wm8741_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION,
+		 WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine),
+SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION,
+		 WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv),
+};
+
+static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("VOUTLP"),
+SND_SOC_DAPM_OUTPUT("VOUTLN"),
+SND_SOC_DAPM_OUTPUT("VOUTRP"),
+SND_SOC_DAPM_OUTPUT("VOUTRN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{ "VOUTLP", NULL, "DACL" },
+	{ "VOUTLN", NULL, "DACL" },
+	{ "VOUTRP", NULL, "DACR" },
+	{ "VOUTRN", NULL, "DACR" },
+};
+
+static int wm8741_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets,
+				  ARRAY_SIZE(wm8741_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	return 0;
+}
+
+static struct {
+	int value;
+	int ratio;
+} lrclk_ratios[WM8741_NUM_RATES] = {
+	{ 1, 256 },
+	{ 2, 384 },
+	{ 3, 512 },
+	{ 4, 768 },
+};
+
+
+static int wm8741_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+
+	/* The set of sample rates that can be supported depends on the
+	 * MCLK supplied to the CODEC - enforce this.
+	 */
+	if (!wm8741->sysclk) {
+		dev_err(codec->dev,
+			"No MCLK configured, call set_sysclk() on init\n");
+		return -EINVAL;
+	}
+
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_RATE,
+				   &wm8741->rate_constraint);
+
+	return 0;
+}
+
+static int wm8741_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_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+	u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC;
+	int i;
+
+	/* Find a supported LRCLK ratio */
+	for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+		if (wm8741->sysclk / params_rate(params) ==
+		    lrclk_ratios[i].ratio)
+			break;
+	}
+
+	/* Should never happen, should be handled by constraints */
+	if (i == ARRAY_SIZE(lrclk_ratios)) {
+		dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n",
+			wm8741->sysclk / params_rate(params));
+		return -EINVAL;
+	}
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x0001;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x0002;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x0003;
+		break;
+	default:
+		dev_dbg(codec->dev, "wm8741_hw_params:    Unsupported bit size param = %d",
+			params_format(params));
+		return -EINVAL;
+	}
+
+	dev_dbg(codec->dev, "wm8741_hw_params:    bit size param = %d",
+		params_format(params));
+
+	snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface);
+	return 0;
+}
+
+static int wm8741_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 wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+	int i;
+
+	dev_dbg(codec->dev, "wm8741_set_dai_sysclk info: freq=%dHz\n", freq);
+
+	wm8741->sysclk = freq;
+
+	wm8741->rate_constraint.count = 0;
+
+	for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+		dev_dbg(codec->dev, "index = %d, ratio = %d, freq = %d",
+				i, lrclk_ratios[i].ratio, freq);
+
+		val = freq / lrclk_ratios[i].ratio;
+		/* Check that it's a standard rate since core can't
+		 * cope with others and having the odd rates confuses
+		 * constraint matching.
+		 */
+		switch (val) {
+		case 32000:
+		case 44100:
+		case 48000:
+		case 64000:
+		case 88200:
+		case 96000:
+			dev_dbg(codec->dev, "Supported sample rate: %dHz\n",
+				val);
+			wm8741->rate_constraint_list[i] = val;
+			wm8741->rate_constraint.count++;
+			break;
+		default:
+			dev_dbg(codec->dev, "Skipping sample rate: %dHz\n",
+				val);
+		}
+	}
+
+	/* Need at least one supported rate... */
+	if (wm8741->rate_constraint.count == 0)
+		return -EINVAL;
+
+	return 0;
+}
+
+static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1C3;
+
+	/* check master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x0008;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x0004;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x0003;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x0013;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0x0010;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= 0x0020;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x0030;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+
+	dev_dbg(codec->dev, "wm8741_set_dai_fmt:    Format=%x, Clock Inv=%x\n",
+				fmt & SND_SOC_DAIFMT_FORMAT_MASK,
+				((fmt & SND_SOC_DAIFMT_INV_MASK)));
+
+	snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface);
+	return 0;
+}
+
+#define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+			SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
+			SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
+			SNDRV_PCM_RATE_192000)
+
+#define WM8741_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 wm8741_dai_ops = {
+	.startup	= wm8741_startup,
+	.hw_params	= wm8741_hw_params,
+	.set_sysclk	= wm8741_set_dai_sysclk,
+	.set_fmt	= wm8741_set_dai_fmt,
+};
+
+struct snd_soc_dai wm8741_dai = {
+	.name = "WM8741",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,  /* Mono modes not yet supported */
+		.channels_max = 2,
+		.rates = WM8741_RATES,
+		.formats = WM8741_FORMATS,
+	},
+	.ops = &wm8741_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm8741_dai);
+
+#ifdef CONFIG_PM
+static int wm8741_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u16 *cache = codec->reg_cache;
+	int i;
+
+	/* RESTORE REG Cache */
+	for (i = 0; i < WM8741_REGISTER_COUNT; i++) {
+		if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i)
+			continue;
+		snd_soc_write(codec, i, cache[i]);
+	}
+	return 0;
+}
+#else
+#define wm8741_suspend NULL
+#define wm8741_resume NULL
+#endif
+
+static int wm8741_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (wm8741_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm8741_codec;
+	codec = wm8741_codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, wm8741_snd_controls,
+			     ARRAY_SIZE(wm8741_snd_controls));
+	wm8741_add_widgets(codec);
+
+	return ret;
+
+pcm_err:
+	return ret;
+}
+
+static int wm8741_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8741 = {
+	.probe =	wm8741_probe,
+	.remove =	wm8741_remove,
+	.resume =	wm8741_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8741);
+
+static int wm8741_register(struct wm8741_priv *wm8741,
+			   enum snd_soc_control_type control)
+{
+	int ret;
+	struct snd_soc_codec *codec = &wm8741->codec;
+	int i;
+
+	if (wm8741_codec) {
+		dev_err(codec->dev, "Another WM8741 is registered\n");
+		return -EINVAL;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	snd_soc_codec_set_drvdata(codec, wm8741);
+	codec->name = "WM8741";
+	codec->owner = THIS_MODULE;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = NULL;
+	codec->dai = &wm8741_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8741_REGISTER_COUNT;
+	codec->reg_cache = &wm8741->reg_cache;
+
+	wm8741->rate_constraint.list = &wm8741->rate_constraint_list[0];
+	wm8741->rate_constraint.count =
+		ARRAY_SIZE(wm8741->rate_constraint_list);
+
+	memcpy(codec->reg_cache, wm8741_reg_defaults,
+		sizeof(wm8741->reg_cache));
+
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		goto err;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++)
+		wm8741->supplies[i].supply = wm8741_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies),
+				 wm8741->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		goto err;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies),
+				    wm8741->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_get;
+	}
+
+	ret = wm8741_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		goto err_enable;
+	}
+
+	wm8741_dai.dev = codec->dev;
+
+	/* Change some default settings - latch VU */
+	wm8741->reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL;
+	wm8741->reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM;
+	wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL;
+	wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM;
+
+	wm8741_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm8741_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
+
+	dev_dbg(codec->dev, "Successful registration\n");
+	return 0;
+
+err_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+err_get:
+	regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+err:
+	kfree(wm8741);
+	return ret;
+}
+
+static void wm8741_unregister(struct wm8741_priv *wm8741)
+{
+	regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+	snd_soc_unregister_dai(&wm8741_dai);
+	snd_soc_unregister_codec(&wm8741->codec);
+	kfree(wm8741);
+	wm8741_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int wm8741_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct wm8741_priv *wm8741;
+	struct snd_soc_codec *codec;
+
+	wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL);
+	if (wm8741 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8741->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(i2c, wm8741);
+	codec->control_data = i2c;
+
+	codec->dev = &i2c->dev;
+
+	return wm8741_register(wm8741, SND_SOC_I2C);
+}
+
+static __devexit int wm8741_i2c_remove(struct i2c_client *client)
+{
+	struct wm8741_priv *wm8741 = i2c_get_clientdata(client);
+	wm8741_unregister(wm8741);
+	return 0;
+}
+
+static const struct i2c_device_id wm8741_i2c_id[] = {
+	{ "wm8741", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id);
+
+
+static struct i2c_driver wm8741_i2c_driver = {
+	.driver = {
+		.name = "WM8741",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8741_i2c_probe,
+	.remove =   __devexit_p(wm8741_i2c_remove),
+	.id_table = wm8741_i2c_id,
+};
+#endif
+
+static int __init wm8741_modinit(void)
+{
+	int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&wm8741_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8741 I2C driver: %d\n",
+		       ret);
+	}
+#endif
+	return 0;
+}
+module_init(wm8741_modinit);
+
+static void __exit wm8741_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8741_i2c_driver);
+#endif
+}
+module_exit(wm8741_exit);
+
+MODULE_DESCRIPTION("ASoC WM8741 driver");
+MODULE_AUTHOR("Ian Lartey <ian@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h
new file mode 100644
index 000000000000..fdef6ecd1f6f
--- /dev/null
+++ b/sound/soc/codecs/wm8741.h
@@ -0,0 +1,214 @@
+/*
+ * wm8741.h  --  WM8423 ASoC driver
+ *
+ * Copyright 2010 Wolfson Microelectronics, plc
+ *
+ * Author: Ian Lartey <ian@opensource.wolfsonmicro.com>
+ *
+ * Based on wm8753.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 _WM8741_H
+#define _WM8741_H
+
+/*
+ * Register values.
+ */
+#define WM8741_DACLLSB_ATTENUATION              0x00
+#define WM8741_DACLMSB_ATTENUATION              0x01
+#define WM8741_DACRLSB_ATTENUATION              0x02
+#define WM8741_DACRMSB_ATTENUATION              0x03
+#define WM8741_VOLUME_CONTROL                   0x04
+#define WM8741_FORMAT_CONTROL                   0x05
+#define WM8741_FILTER_CONTROL                   0x06
+#define WM8741_MODE_CONTROL_1                   0x07
+#define WM8741_MODE_CONTROL_2                   0x08
+#define WM8741_RESET                            0x09
+#define WM8741_ADDITIONAL_CONTROL_1             0x20
+
+#define WM8741_REGISTER_COUNT                   11
+#define WM8741_MAX_REGISTER                     0x20
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - DACLLSB_ATTENUATION
+ */
+#define WM8741_UPDATELL                         0x0020  /* UPDATELL */
+#define WM8741_UPDATELL_MASK                    0x0020  /* UPDATELL */
+#define WM8741_UPDATELL_SHIFT                        5  /* UPDATELL */
+#define WM8741_UPDATELL_WIDTH                        1  /* UPDATELL */
+#define WM8741_LAT_4_0_MASK                     0x001F  /* LAT[4:0] - [4:0] */
+#define WM8741_LAT_4_0_SHIFT                         0  /* LAT[4:0] - [4:0] */
+#define WM8741_LAT_4_0_WIDTH                         5  /* LAT[4:0] - [4:0] */
+
+/*
+ * R1 (0x01) - DACLMSB_ATTENUATION
+ */
+#define WM8741_UPDATELM                         0x0020  /* UPDATELM */
+#define WM8741_UPDATELM_MASK                    0x0020  /* UPDATELM */
+#define WM8741_UPDATELM_SHIFT                        5  /* UPDATELM */
+#define WM8741_UPDATELM_WIDTH                        1  /* UPDATELM */
+#define WM8741_LAT_9_5_0_MASK                   0x001F  /* LAT[9:5] - [4:0] */
+#define WM8741_LAT_9_5_0_SHIFT                       0  /* LAT[9:5] - [4:0] */
+#define WM8741_LAT_9_5_0_WIDTH                       5  /* LAT[9:5] - [4:0] */
+
+/*
+ * R2 (0x02) - DACRLSB_ATTENUATION
+ */
+#define WM8741_UPDATERL                         0x0020  /* UPDATERL */
+#define WM8741_UPDATERL_MASK                    0x0020  /* UPDATERL */
+#define WM8741_UPDATERL_SHIFT                        5  /* UPDATERL */
+#define WM8741_UPDATERL_WIDTH                        1  /* UPDATERL */
+#define WM8741_RAT_4_0_MASK                     0x001F  /* RAT[4:0] - [4:0] */
+#define WM8741_RAT_4_0_SHIFT                         0  /* RAT[4:0] - [4:0] */
+#define WM8741_RAT_4_0_WIDTH                         5  /* RAT[4:0] - [4:0] */
+
+/*
+ * R3 (0x03) - DACRMSB_ATTENUATION
+ */
+#define WM8741_UPDATERM                         0x0020  /* UPDATERM */
+#define WM8741_UPDATERM_MASK                    0x0020  /* UPDATERM */
+#define WM8741_UPDATERM_SHIFT                        5  /* UPDATERM */
+#define WM8741_UPDATERM_WIDTH                        1  /* UPDATERM */
+#define WM8741_RAT_9_5_0_MASK                   0x001F  /* RAT[9:5] - [4:0] */
+#define WM8741_RAT_9_5_0_SHIFT                       0  /* RAT[9:5] - [4:0] */
+#define WM8741_RAT_9_5_0_WIDTH                       5  /* RAT[9:5] - [4:0] */
+
+/*
+ * R4 (0x04) - VOLUME_CONTROL
+ */
+#define WM8741_AMUTE                            0x0080  /* AMUTE */
+#define WM8741_AMUTE_MASK                       0x0080  /* AMUTE */
+#define WM8741_AMUTE_SHIFT                           7  /* AMUTE */
+#define WM8741_AMUTE_WIDTH                           1  /* AMUTE */
+#define WM8741_ZFLAG_MASK                       0x0060  /* ZFLAG - [6:5] */
+#define WM8741_ZFLAG_SHIFT                           5  /* ZFLAG - [6:5] */
+#define WM8741_ZFLAG_WIDTH                           2  /* ZFLAG - [6:5] */
+#define WM8741_IZD                              0x0010  /* IZD */
+#define WM8741_IZD_MASK                         0x0010  /* IZD */
+#define WM8741_IZD_SHIFT                             4  /* IZD */
+#define WM8741_IZD_WIDTH                             1  /* IZD */
+#define WM8741_SOFT                             0x0008  /* SOFT MUTE */
+#define WM8741_SOFT_MASK                        0x0008  /* SOFT MUTE */
+#define WM8741_SOFT_SHIFT                            3  /* SOFT MUTE */
+#define WM8741_SOFT_WIDTH                            1  /* SOFT MUTE */
+#define WM8741_ATC                              0x0004  /* ATC */
+#define WM8741_ATC_MASK                         0x0004  /* ATC */
+#define WM8741_ATC_SHIFT                             2  /* ATC */
+#define WM8741_ATC_WIDTH                             1  /* ATC */
+#define WM8741_ATT2DB                           0x0002  /* ATT2DB */
+#define WM8741_ATT2DB_MASK                      0x0002  /* ATT2DB */
+#define WM8741_ATT2DB_SHIFT                          1  /* ATT2DB */
+#define WM8741_ATT2DB_WIDTH                          1  /* ATT2DB */
+#define WM8741_VOL_RAMP                         0x0001  /* VOL_RAMP */
+#define WM8741_VOL_RAMP_MASK                    0x0001  /* VOL_RAMP */
+#define WM8741_VOL_RAMP_SHIFT                        0  /* VOL_RAMP */
+#define WM8741_VOL_RAMP_WIDTH                        1  /* VOL_RAMP */
+
+/*
+ * R5 (0x05) - FORMAT_CONTROL
+ */
+#define WM8741_PWDN                             0x0080  /* PWDN */
+#define WM8741_PWDN_MASK                        0x0080  /* PWDN */
+#define WM8741_PWDN_SHIFT                            7  /* PWDN */
+#define WM8741_PWDN_WIDTH                            1  /* PWDN */
+#define WM8741_REV                              0x0040  /* REV */
+#define WM8741_REV_MASK                         0x0040  /* REV */
+#define WM8741_REV_SHIFT                             6  /* REV */
+#define WM8741_REV_WIDTH                             1  /* REV */
+#define WM8741_BCP                              0x0020  /* BCP */
+#define WM8741_BCP_MASK                         0x0020  /* BCP */
+#define WM8741_BCP_SHIFT                             5  /* BCP */
+#define WM8741_BCP_WIDTH                             1  /* BCP */
+#define WM8741_LRP                              0x0010  /* LRP */
+#define WM8741_LRP_MASK                         0x0010  /* LRP */
+#define WM8741_LRP_SHIFT                             4  /* LRP */
+#define WM8741_LRP_WIDTH                             1  /* LRP */
+#define WM8741_FMT_MASK                         0x000C  /* FMT - [3:2] */
+#define WM8741_FMT_SHIFT                             2  /* FMT - [3:2] */
+#define WM8741_FMT_WIDTH                             2  /* FMT - [3:2] */
+#define WM8741_IWL_MASK                         0x0003  /* IWL - [1:0] */
+#define WM8741_IWL_SHIFT                             0  /* IWL - [1:0] */
+#define WM8741_IWL_WIDTH                             2  /* IWL - [1:0] */
+
+/*
+ * R6 (0x06) - FILTER_CONTROL
+ */
+#define WM8741_ZFLAG_HI                         0x0080  /* ZFLAG_HI */
+#define WM8741_ZFLAG_HI_MASK                    0x0080  /* ZFLAG_HI */
+#define WM8741_ZFLAG_HI_SHIFT                        7  /* ZFLAG_HI */
+#define WM8741_ZFLAG_HI_WIDTH                        1  /* ZFLAG_HI */
+#define WM8741_DEEMPH_MASK                      0x0060  /* DEEMPH - [6:5] */
+#define WM8741_DEEMPH_SHIFT                          5  /* DEEMPH - [6:5] */
+#define WM8741_DEEMPH_WIDTH                          2  /* DEEMPH - [6:5] */
+#define WM8741_DSDFILT_MASK                     0x0018  /* DSDFILT - [4:3] */
+#define WM8741_DSDFILT_SHIFT                         3  /* DSDFILT - [4:3] */
+#define WM8741_DSDFILT_WIDTH                         2  /* DSDFILT - [4:3] */
+#define WM8741_FIRSEL_MASK                      0x0007  /* FIRSEL - [2:0] */
+#define WM8741_FIRSEL_SHIFT                          0  /* FIRSEL - [2:0] */
+#define WM8741_FIRSEL_WIDTH                          3  /* FIRSEL - [2:0] */
+
+/*
+ * R7 (0x07) - MODE_CONTROL_1
+ */
+#define WM8741_MODE8X                           0x0080  /* MODE8X */
+#define WM8741_MODE8X_MASK                      0x0080  /* MODE8X */
+#define WM8741_MODE8X_SHIFT                          7  /* MODE8X */
+#define WM8741_MODE8X_WIDTH                          1  /* MODE8X */
+#define WM8741_OSR_MASK                         0x0060  /* OSR - [6:5] */
+#define WM8741_OSR_SHIFT                             5  /* OSR - [6:5] */
+#define WM8741_OSR_WIDTH                             2  /* OSR - [6:5] */
+#define WM8741_SR_MASK                          0x001C  /* SR - [4:2] */
+#define WM8741_SR_SHIFT                              2  /* SR - [4:2] */
+#define WM8741_SR_WIDTH                              3  /* SR - [4:2] */
+#define WM8741_MODESEL_MASK                     0x0003  /* MODESEL - [1:0] */
+#define WM8741_MODESEL_SHIFT                         0  /* MODESEL - [1:0] */
+#define WM8741_MODESEL_WIDTH                         2  /* MODESEL - [1:0] */
+
+/*
+ * R8 (0x08) - MODE_CONTROL_2
+ */
+#define WM8741_DSD_GAIN                         0x0040  /* DSD_GAIN */
+#define WM8741_DSD_GAIN_MASK                    0x0040  /* DSD_GAIN */
+#define WM8741_DSD_GAIN_SHIFT                        6  /* DSD_GAIN */
+#define WM8741_DSD_GAIN_WIDTH                        1  /* DSD_GAIN */
+#define WM8741_SDOUT                            0x0020  /* SDOUT */
+#define WM8741_SDOUT_MASK                       0x0020  /* SDOUT */
+#define WM8741_SDOUT_SHIFT                           5  /* SDOUT */
+#define WM8741_SDOUT_WIDTH                           1  /* SDOUT */
+#define WM8741_DOUT                             0x0010  /* DOUT */
+#define WM8741_DOUT_MASK                        0x0010  /* DOUT */
+#define WM8741_DOUT_SHIFT                            4  /* DOUT */
+#define WM8741_DOUT_WIDTH                            1  /* DOUT */
+#define WM8741_DIFF_MASK                        0x000C  /* DIFF - [3:2] */
+#define WM8741_DIFF_SHIFT                            2  /* DIFF - [3:2] */
+#define WM8741_DIFF_WIDTH                            2  /* DIFF - [3:2] */
+#define WM8741_DITHER_MASK                      0x0003  /* DITHER - [1:0] */
+#define WM8741_DITHER_SHIFT                          0  /* DITHER - [1:0] */
+#define WM8741_DITHER_WIDTH                          2  /* DITHER - [1:0] */
+
+/*
+ * R32 (0x20) - ADDITONAL_CONTROL_1
+ */
+#define WM8741_DSD_LEVEL                        0x0002  /* DSD_LEVEL */
+#define WM8741_DSD_LEVEL_MASK                   0x0002  /* DSD_LEVEL */
+#define WM8741_DSD_LEVEL_SHIFT                       1  /* DSD_LEVEL */
+#define WM8741_DSD_LEVEL_WIDTH                       1  /* DSD_LEVEL */
+#define WM8741_DSD_NO_NOTCH                     0x0001  /* DSD_NO_NOTCH */
+#define WM8741_DSD_NO_NOTCH_MASK                0x0001  /* DSD_NO_NOTCH */
+#define WM8741_DSD_NO_NOTCH_SHIFT                    0  /* DSD_NO_NOTCH */
+#define WM8741_DSD_NO_NOTCH_WIDTH                    1  /* DSD_NO_NOTCH */
+
+#define  WM8741_SYSCLK 0
+
+extern struct snd_soc_dai wm8741_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8741;
+
+#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9407e193fcc3..e2c05e3e323a 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client)
 
 static const struct i2c_device_id wm8750_i2c_id[] = {
 	{ "wm8750", 0 },
+	{ "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id);
@@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
+static const struct spi_device_id wm8750_spi_id[] = {
+	{ "wm8750", 0 },
+	{ "wm8987", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(spi, wm8750_spi_id);
+
 static struct spi_driver wm8750_spi_driver = {
 	.driver = {
-		.name	= "wm8750",
+		.name	= "WM8750 SPI Codec",
 		.bus	= &spi_bus_type,
 		.owner	= THIS_MODULE,
 	},
 	.probe		= wm8750_spi_probe,
 	.remove		= __devexit_p(wm8750_spi_remove),
+	.id_table	= wm8750_spi_id,
 };
 #endif
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 87f14f8675fa..f7dcabf6283c 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2433,7 +2433,8 @@ static int wm8904_register(struct wm8904_priv *wm8904,
 
 	if (wm8904_codec) {
 		dev_err(codec->dev, "Another WM8904 is registered\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err;
 	}
 
 	mutex_init(&codec->mutex);
@@ -2462,7 +2463,8 @@ static int wm8904_register(struct wm8904_priv *wm8904,
 	default:
 		dev_err(codec->dev, "Unknown device type %d\n",
 			wm8904->devtype);
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err;
 	}
 
 	memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg));
@@ -2566,18 +2568,19 @@ static int wm8904_register(struct wm8904_priv *wm8904,
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		return ret;
+		goto err_enable;
 	}
 
 	ret = snd_soc_register_dai(&wm8904_dai);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		snd_soc_unregister_codec(codec);
-		return ret;
+		goto err_codec;
 	}
 
 	return 0;
 
+err_codec:
+	snd_soc_unregister_codec(codec);
 err_enable:
 	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
 err_get:
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index e3c4bbfaae27..f0c11138e610 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -845,6 +845,7 @@ static void wm8940_unregister(struct wm8940_priv *wm8940)
 static int wm8940_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
+	int ret;
 	struct wm8940_priv *wm8940;
 	struct snd_soc_codec *codec;
 
@@ -858,7 +859,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c,
 	codec->control_data = i2c;
 	codec->dev = &i2c->dev;
 
-	return wm8940_register(wm8940, SND_SOC_I2C);
+	ret = wm8940_register(wm8940, SND_SOC_I2C);
+	if (ret < 0)
+		kfree(wm8940);
+
+	return ret;
 }
 
 static int __devexit wm8940_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index fedb76452f1b..5f025593d84d 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -964,7 +964,8 @@ static int wm8955_register(struct wm8955_priv *wm8955,
 
 	if (wm8955_codec) {
 		dev_err(codec->dev, "Another WM8955 is registered\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err;
 	}
 
 	mutex_init(&codec->mutex);
@@ -1047,18 +1048,19 @@ static int wm8955_register(struct wm8955_priv *wm8955,
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		return ret;
+		goto err_enable;
 	}
 
 	ret = snd_soc_register_dai(&wm8955_dai);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		snd_soc_unregister_codec(codec);
-		return ret;
+		goto err_codec;
 	}
 
 	return 0;
 
+err_codec:
+	snd_soc_unregister_codec(codec);
 err_enable:
 	regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies);
 err_get:
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 7233cc68435a..3c6ee61f6c95 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -79,12 +79,13 @@ struct wm8960_priv {
 	struct snd_soc_dapm_widget *lout1;
 	struct snd_soc_dapm_widget *rout1;
 	struct snd_soc_dapm_widget *out3;
+	bool deemph;
+	int playback_fs;
 };
 
 #define wm8960_reset(c)	snd_soc_write(c, WM8960_RESET, 0)
 
 /* enumerated controls */
-static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
 static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
 	"Right Inverted", "Stereo Inversion"};
 static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
@@ -93,7 +94,6 @@ static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
 static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
 
 static const struct soc_enum wm8960_enum[] = {
-	SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
 	SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
 	SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
 	SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
@@ -102,6 +102,59 @@ static const struct soc_enum wm8960_enum[] = {
 	SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
 };
 
+static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
+
+static int wm8960_set_deemph(struct snd_soc_codec *codec)
+{
+	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+	int val, i, best;
+
+	/* If we're using deemphasis select the nearest available sample
+	 * rate.
+	 */
+	if (wm8960->deemph) {
+		best = 1;
+		for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
+			if (abs(deemph_settings[i] - wm8960->playback_fs) <
+			    abs(deemph_settings[best] - wm8960->playback_fs))
+				best = i;
+		}
+
+		val = best << 1;
+	} else {
+		val = 0;
+	}
+
+	dev_dbg(codec->dev, "Set deemphasis %d\n", val);
+
+	return snd_soc_update_bits(codec, WM8960_DACCTL1,
+				   0x6, val);
+}
+
+static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+
+	return wm8960->deemph;
+}
+
+static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+	int deemph = ucontrol->value.enumerated.item[0];
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	wm8960->deemph = deemph;
+
+	return wm8960_set_deemph(codec);
+}
+
 static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
 static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
@@ -131,23 +184,24 @@ SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
 SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
 
 SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
-SOC_ENUM("ADC Polarity", wm8960_enum[1]),
-SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
+SOC_ENUM("ADC Polarity", wm8960_enum[0]),
 SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
 
 SOC_ENUM("DAC Polarity", wm8960_enum[2]),
+SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
+		    wm8960_get_deemph, wm8960_put_deemph),
 
-SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
-SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[2]),
+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[3]),
 SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
 SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
 
-SOC_ENUM("ALC Function", wm8960_enum[5]),
+SOC_ENUM("ALC Function", wm8960_enum[4]),
 SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
 SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
 SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
 SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
-SOC_ENUM("ALC Mode", wm8960_enum[6]),
+SOC_ENUM("ALC Mode", wm8960_enum[5]),
 SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
 SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
 
@@ -433,6 +487,21 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
+static struct {
+	int rate;
+	unsigned int val;
+} alc_rates[] = {
+	{ 48000, 0 },
+	{ 44100, 0 },
+	{ 32000, 1 },
+	{ 22050, 2 },
+	{ 24000, 2 },
+	{ 16000, 3 },
+	{ 11250, 4 },
+	{ 12000, 4 },
+	{  8000, 5 },
+};
+
 static int wm8960_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_pcm_hw_params *params,
 			    struct snd_soc_dai *dai)
@@ -440,7 +509,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
+	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
+	int i;
 
 	/* bit size */
 	switch (params_format(params)) {
@@ -454,6 +525,18 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
+	/* Update filters for the new rate */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		wm8960->playback_fs = params_rate(params);
+		wm8960_set_deemph(codec);
+	} else {
+		for (i = 0; i < ARRAY_SIZE(alc_rates); i++)
+			if (alc_rates[i].rate == params_rate(params))
+				snd_soc_update_bits(codec,
+						    WM8960_ADDCTL3, 0x7,
+						    alc_rates[i].val);
+	}
+
 	/* set iface */
 	snd_soc_write(codec, WM8960_IFACE1, iface);
 	return 0;
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 5b9a756242f1..2549d3a297ab 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -1102,7 +1102,7 @@ static int wm8961_register(struct wm8961_priv *wm8961)
 	ret = wm8961_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
-		return ret;
+		goto err;
 	}
 
 	/* Enable class W */
@@ -1147,18 +1147,19 @@ static int wm8961_register(struct wm8961_priv *wm8961)
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		return ret;
+		goto err;
 	}
 
 	ret = snd_soc_register_dai(&wm8961_dai);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		snd_soc_unregister_codec(codec);
-		return ret;
+		goto err_codec;
 	}
 
 	return 0;
 
+err_codec:
+	snd_soc_unregister_codec(codec);
 err:
 	kfree(wm8961);
 	return ret;
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index a2c4b2f37cca..1468fe10cbbe 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -670,7 +670,8 @@ static __devinit int wm8974_register(struct wm8974_priv *wm8974)
 
 	if (wm8974_codec) {
 		dev_err(codec->dev, "Another WM8974 is registered\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err;
 	}
 
 	mutex_init(&codec->mutex);
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 51d5f433215c..8a1ad778e7e3 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -1076,7 +1076,6 @@ static __devinit int wm8978_register(struct wm8978_priv *wm8978)
 err_codec:
 	snd_soc_unregister_codec(codec);
 err:
-	kfree(wm8978);
 	return ret;
 }
 
@@ -1085,13 +1084,13 @@ static __devexit void wm8978_unregister(struct wm8978_priv *wm8978)
 	wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF);
 	snd_soc_unregister_dai(&wm8978_dai);
 	snd_soc_unregister_codec(&wm8978->codec);
-	kfree(wm8978);
 	wm8978_codec = NULL;
 }
 
 static __devinit int wm8978_i2c_probe(struct i2c_client *i2c,
 				      const struct i2c_device_id *id)
 {
+	int ret;
 	struct wm8978_priv *wm8978;
 	struct snd_soc_codec *codec;
 
@@ -1107,13 +1106,18 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c,
 
 	codec->dev = &i2c->dev;
 
-	return wm8978_register(wm8978);
+	ret = wm8978_register(wm8978);
+	if (ret < 0)
+		kfree(wm8978);
+
+	return ret;
 }
 
 static __devexit int wm8978_i2c_remove(struct i2c_client *client)
 {
 	struct wm8978_priv *wm8978 = i2c_get_clientdata(client);
 	wm8978_unregister(wm8978);
+	kfree(wm8978);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c018772cc430..dd8d909788c1 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -30,8 +30,6 @@
 
 #include "wm8990.h"
 
-#define WM8990_VERSION "0.2"
-
 /* codec private data */
 struct wm8990_priv {
 	unsigned int sysclk;
@@ -1511,8 +1509,6 @@ static int wm8990_probe(struct platform_device *pdev)
 	struct wm8990_priv *wm8990;
 	int ret;
 
-	pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION);
-
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index e84a1177f350..a87046a96f2a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1677,6 +1677,26 @@ static struct {
 
 static int wm8994_readable(unsigned int reg)
 {
+	switch (reg) {
+	case WM8994_GPIO_1:
+	case WM8994_GPIO_2:
+	case WM8994_GPIO_3:
+	case WM8994_GPIO_4:
+	case WM8994_GPIO_5:
+	case WM8994_GPIO_6:
+	case WM8994_GPIO_7:
+	case WM8994_GPIO_8:
+	case WM8994_GPIO_9:
+	case WM8994_GPIO_10:
+	case WM8994_GPIO_11:
+	case WM8994_INTERRUPT_STATUS_1:
+	case WM8994_INTERRUPT_STATUS_2:
+	case WM8994_INTERRUPT_RAW_STATUS_2:
+		return 1;
+	default:
+		break;
+	}
+
 	if (reg >= ARRAY_SIZE(access_masks))
 		return 0;
 	return access_masks[reg].readable != 0;
@@ -2341,6 +2361,20 @@ SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING,
 		0, 1, 0),
 };
 
+static const struct snd_kcontrol_new aif1adc2l_mix[] = {
+SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
+		1, 1, 0),
+SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
+		0, 1, 0),
+};
+
+static const struct snd_kcontrol_new aif1adc2r_mix[] = {
+SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
+		1, 1, 0),
+SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
+		0, 1, 0),
+};
+
 static const struct snd_kcontrol_new aif2dac2l_mix[] = {
 SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
 		5, 1, 0),
@@ -2472,6 +2506,7 @@ static const struct snd_kcontrol_new aif3adc_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_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
 		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -2506,6 +2541,11 @@ SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0,
 SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0,
 		   aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)),
 
+SND_SOC_DAPM_MIXER("AIF1ADC2L Mixer", SND_SOC_NOPM, 0, 0,
+		   aif1adc2l_mix, ARRAY_SIZE(aif1adc2l_mix)),
+SND_SOC_DAPM_MIXER("AIF1ADC2R Mixer", SND_SOC_NOPM, 0, 0,
+		   aif1adc2r_mix, ARRAY_SIZE(aif1adc2r_mix)),
+
 SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0,
 		   aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)),
 SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0,
@@ -2668,6 +2708,14 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" },
 	{ "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" },
 
+	{ "AIF1ADC2L", NULL, "AIF1ADC2L Mixer" },
+	{ "AIF1ADC2L Mixer", "DMIC Switch", "DMIC2L" },
+	{ "AIF1ADC2L Mixer", "AIF2 Switch", "AIF2DACL" },
+
+	{ "AIF1ADC2R", NULL, "AIF1ADC2R Mixer" },
+	{ "AIF1ADC2R Mixer", "DMIC Switch", "DMIC2R" },
+	{ "AIF1ADC2R Mixer", "AIF2 Switch", "AIF2DACR" },
+
 	/* Pin level routing for AIF3 */
 	{ "AIF1DAC1L", NULL, "AIF1DAC Mux" },
 	{ "AIF1DAC1R", NULL, "AIF1DAC Mux" },
@@ -2946,11 +2994,14 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
 	return 0;
 }
 
+static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 };
+
 static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+	int i;
 
 	switch (dai->id) {
 	case 1:
@@ -2988,6 +3039,25 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
 		dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id);
 		break;
 
+	case WM8994_SYSCLK_OPCLK:
+		/* Special case - a division (times 10) is given and
+		 * no effect on main clocking. 
+		 */
+		if (freq) {
+			for (i = 0; i < ARRAY_SIZE(opclk_divs); i++)
+				if (opclk_divs[i] == freq)
+					break;
+			if (i == ARRAY_SIZE(opclk_divs))
+				return -EINVAL;
+			snd_soc_update_bits(codec, WM8994_CLOCKING_2,
+					    WM8994_OPCLK_DIV_MASK, i);
+			snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
+					    WM8994_OPCLK_ENA, WM8994_OPCLK_ENA);
+		} else {
+			snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
+					    WM8994_OPCLK_ENA, 0);
+		}
+
 	default:
 		return -EINVAL;
 	}
@@ -4004,6 +4074,11 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 			    1 << WM8994_AIF2DAC_3D_GAIN_SHIFT,
 			    1 << WM8994_AIF2DAC_3D_GAIN_SHIFT);
 
+	/* Unconditionally enable AIF1 ADC TDM mode; it only affects
+	 * behaviour on idle TDM clock cycles. */
+	snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1,
+			    WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM);
+
 	wm8994_update_class_w(codec);
 
 	ret = snd_soc_register_codec(codec);
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 7072dc539354..2e0ca67a8df7 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -20,6 +20,9 @@ extern struct snd_soc_dai wm8994_dai[];
 #define WM8994_SYSCLK_FLL1  3
 #define WM8994_SYSCLK_FLL2  4
 
+/* OPCLK is also configured with set_dai_sysclk, specify division*10 as rate. */
+#define WM8994_SYSCLK_OPCLK 5
+
 #define WM8994_FLL1 1
 #define WM8994_FLL2 2
 
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 13186fb4dcb4..76b37ff6c264 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1356,7 +1356,7 @@ static int wm9081_register(struct wm9081_priv *wm9081,
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
+		goto err;
 	}
 
 	reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET);
@@ -1369,7 +1369,7 @@ static int wm9081_register(struct wm9081_priv *wm9081,
 	ret = wm9081_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
-		return ret;
+		goto err;
 	}
 
 	wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1388,18 +1388,19 @@ static int wm9081_register(struct wm9081_priv *wm9081,
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		return ret;
+		goto err;
 	}
 
 	ret = snd_soc_register_dai(&wm9081_dai);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		snd_soc_unregister_codec(codec);
-		return ret;
+		goto err_codec;
 	}
 
 	return 0;
 
+err_codec:
+	snd_soc_unregister_codec(codec);
 err:
 	kfree(wm9081);
 	return ret;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 16f1a57da08a..2cb81538cd91 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -410,6 +410,8 @@ static int hp_event(struct snd_soc_dapm_widget *w,
 				    WM8993_HPOUT1L_DLY |
 				    WM8993_HPOUT1R_DLY, 0);
 
+		snd_soc_write(codec, WM8993_DC_SERVO_0, 0);
+
 		snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
 				    WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
 				    0);