Merge branch 'origin' into devel

Conflicts:
	sound/soc/pxa/pxa2xx-i2s.c

109 files changed, 8349 insertions, 3531 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index ef025c6..3d2bb6f 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -6,6 +6,7 @@ menuconfig SND_SOC
 	tristate "ALSA for SoC audio support"
 	select SND_PCM
 	select AC97_BUS if SND_SOC_AC97_BUS
+	select SND_JACK if INPUT=y || INPUT=SND
 	---help---
 
 	  If you want ASoC support, you should say Y here and also to the
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 86a9b1f..0237879 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
-snd-soc-core-objs := soc-core.o soc-dapm.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 3dcdc4e..9ef6b96 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -347,7 +347,7 @@ static int atmel_pcm_mmap(struct snd_pcm_substream *substream,
 		       vma->vm_end - vma->vm_start, vma->vm_page_prot);
 }
 
-struct snd_pcm_ops atmel_pcm_ops = {
+static struct snd_pcm_ops atmel_pcm_ops = {
 	.open		= atmel_pcm_open,
 	.close		= atmel_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index ff0054b..e588e63 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
 #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
 			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops atmel_ssc_dai_ops = {
+	.startup	= atmel_ssc_startup,
+	.shutdown	= atmel_ssc_shutdown,
+	.prepare	= atmel_ssc_prepare,
+	.hw_params	= atmel_ssc_hw_params,
+	.set_fmt	= atmel_ssc_set_dai_fmt,
+	.set_clkdiv	= atmel_ssc_set_dai_clkdiv,
+};
+
 struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 	{	.name = "atmel-ssc0",
 		.id = 0,
@@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[0],
 	},
 #if NUM_SSC_DEVICES == 3
@@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[1],
 	},
 	{	.name = "atmel-ssc2",
@@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[2],
 	},
 #endif
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 43dd8ce..7065753 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -164,38 +164,38 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
 	 */
 	switch (params_rate(params)) {
 	case 48000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_1;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_2;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 44100:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_1;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_2;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 22050:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_2;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_4;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 16000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_3;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_6;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 11025:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_4;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_8;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 8000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_6;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_12;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 6ea04be..173a239 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -36,6 +36,7 @@
 #include <linux/timer.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
+#include <linux/i2c.h>
 
 #include <linux/atmel-ssc.h>
 
@@ -45,6 +46,7 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
+#include <asm/mach-types.h>
 #include <mach/hardware.h>
 #include <mach/gpio.h>
 
@@ -52,6 +54,9 @@
 #include "atmel-pcm.h"
 #include "atmel_ssc_dai.h"
 
+#define MCLK_RATE 12000000
+
+static struct clk *mclk;
 
 static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
 {
@@ -59,11 +64,12 @@ static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 	int ret;
 
-	/* codec system clock is supplied by PCK0, set to 12MHz */
 	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
-		12000000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
+		MCLK_RATE, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		clk_disable(mclk);
 		return ret;
+	}
 
 	return 0;
 }
@@ -189,6 +195,31 @@ static struct snd_soc_ops at91sam9g20ek_ops = {
 	.shutdown = at91sam9g20ek_shutdown,
 };
 
+static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
+					enum snd_soc_bias_level level)
+{
+	static int mclk_on;
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		if (!mclk_on)
+			ret = clk_enable(mclk);
+		if (ret == 0)
+			mclk_on = 1;
+		break;
+
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_STANDBY:
+		if (mclk_on)
+			clk_disable(mclk);
+		mclk_on = 0;
+		break;
+	}
+
+	return ret;
+}
 
 static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
 	SND_SOC_DAPM_MIC("Int Mic", NULL),
@@ -243,21 +274,48 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
 };
 
 static struct snd_soc_card snd_soc_at91sam9g20ek = {
-	.name = "WM8731",
+	.name = "AT91SAMG20-EK",
 	.platform = &atmel_soc_platform,
 	.dai_link = &at91sam9g20ek_dai,
 	.num_links = 1,
+	.set_bias_level = at91sam9g20ek_set_bias_level,
 };
 
-static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = 0x1b;
+	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter 0\n");
+		return -ENODEV;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		printk(KERN_ERR "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		return -ENODEV;
+	}
+
+	return 0;
+}
 
 static struct snd_soc_device at91sam9g20ek_snd_devdata = {
 	.card = &snd_soc_at91sam9g20ek,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &at91sam9g20ek_wm8731_setup,
 };
 
 static struct platform_device *at91sam9g20ek_snd_device;
@@ -266,23 +324,56 @@ static int __init at91sam9g20ek_init(void)
 {
 	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
 	struct ssc_device *ssc = NULL;
+	struct clk *pllb;
 	int ret;
 
+	if (!machine_is_at91sam9g20ek())
+		return -ENODEV;
+
+	/*
+	 * Codec MCLK is supplied by PCK0 - set it up.
+	 */
+	mclk = clk_get(NULL, "pck0");
+	if (IS_ERR(mclk)) {
+		printk(KERN_ERR "ASoC: Failed to get MCLK\n");
+		ret = PTR_ERR(mclk);
+		goto err;
+	}
+
+	pllb = clk_get(NULL, "pllb");
+	if (IS_ERR(mclk)) {
+		printk(KERN_ERR "ASoC: Failed to get PLLB\n");
+		ret = PTR_ERR(mclk);
+		goto err_mclk;
+	}
+	ret = clk_set_parent(mclk, pllb);
+	clk_put(pllb);
+	if (ret != 0) {
+		printk(KERN_ERR "ASoC: Failed to set MCLK parent\n");
+		goto err_mclk;
+	}
+
+	clk_set_rate(mclk, MCLK_RATE);
+
 	/*
 	 * Request SSC device
 	 */
 	ssc = ssc_request(0);
 	if (IS_ERR(ssc)) {
+		printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
 		ret = PTR_ERR(ssc);
 		ssc = NULL;
 		goto err_ssc;
 	}
 	ssc_p->ssc = ssc;
 
+	ret = wm8731_i2c_register();
+	if (ret != 0)
+		goto err_ssc;
+
 	at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!at91sam9g20ek_snd_device) {
-		printk(KERN_DEBUG
-				"platform device allocation failed\n");
+		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
 		ret = -ENOMEM;
 	}
 
@@ -292,14 +383,19 @@ static int __init at91sam9g20ek_init(void)
 
 	ret = platform_device_add(at91sam9g20ek_snd_device);
 	if (ret) {
-		printk(KERN_DEBUG
-				"platform device allocation failed\n");
+		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
 		platform_device_put(at91sam9g20ek_snd_device);
 	}
 
 	return ret;
 
 err_ssc:
+	ssc_free(ssc);
+	ssc_p->ssc = NULL;
+err_mclk:
+	clk_put(mclk);
+	mclk = NULL;
+err:
 	return ret;
 }
 
@@ -317,6 +413,8 @@ static void __exit at91sam9g20ek_exit(void)
 
 	platform_device_unregister(at91sam9g20ek_snd_device);
 	at91sam9g20ek_snd_device = NULL;
+	clk_put(mclk);
+	mclk = NULL;
 }
 
 module_init(at91sam9g20ek_init);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index bc8d654..30490a2 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -305,7 +305,7 @@ static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-struct snd_pcm_ops au1xpsc_pcm_ops = {
+static struct snd_pcm_ops au1xpsc_pcm_ops = {
 	.open		= au1xpsc_pcm_open,
 	.close		= au1xpsc_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index f0e30ae..479d7bd 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.trigger	= au1xpsc_ac97_trigger,
+	.hw_params	= au1xpsc_ac97_hw_params,
+};
+
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
 	.ac97_control		= 1,
@@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.trigger	= au1xpsc_ac97_trigger,
-		.hw_params	= au1xpsc_ac97_hw_params,
-	},
+	.ops = &au1xpsc_ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index f916de4..bb58932 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.trigger	= au1xpsc_i2s_trigger,
+	.hw_params	= au1xpsc_i2s_hw_params,
+	.set_fmt	= au1xpsc_i2s_set_fmt,
+};
+
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
 	.probe			= au1xpsc_i2s_probe,
@@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 		.channels_min	= 2,
 		.channels_max	= 8,	/* 2 without external help */
 	},
-	.ops = {
-		.trigger	= au1xpsc_i2s_trigger,
-		.hw_params	= au1xpsc_i2s_hw_params,
-		.set_fmt	= au1xpsc_i2s_set_fmt,
-	},
+	.ops = &au1xpsc_i2s_dai_ops,
 };
 EXPORT_SYMBOL(au1xpsc_i2s_dai);
 
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 8067cfa..8cfed1a 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -297,7 +297,7 @@ static	int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 }
 #endif
 
-struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
+static struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
 	.open		= bf5xx_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 3be2be6..8a935f2 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -31,72 +31,46 @@
 #include "bf5xx-sport.h"
 #include "bf5xx-ac97.h"
 
-#if defined(CONFIG_BF54x)
-#define PIN_REQ_SPORT_0 {P_SPORT0_TFS, P_SPORT0_DTPRI, P_SPORT0_TSCLK, \
-		P_SPORT0_RFS, P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}
-
-#define PIN_REQ_SPORT_1 {P_SPORT1_TFS, P_SPORT1_DTPRI, P_SPORT1_TSCLK, \
-		P_SPORT1_RFS, P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}
-
-#define PIN_REQ_SPORT_2 {P_SPORT2_TFS, P_SPORT2_DTPRI, P_SPORT2_TSCLK, \
-		P_SPORT2_RFS, P_SPORT2_DRPRI, P_SPORT2_RSCLK, 0}
-
-#define PIN_REQ_SPORT_3 {P_SPORT3_TFS, P_SPORT3_DTPRI, P_SPORT3_TSCLK, \
-		P_SPORT3_RFS, P_SPORT3_DRPRI, P_SPORT3_RSCLK, 0}
-#else
-#define PIN_REQ_SPORT_0 {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, \
-		 P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}
-
-#define PIN_REQ_SPORT_1 {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, \
-		 P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}
-#endif
-
 static int *cmd_count;
 static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
 
+#define SPORT_REQ(x) \
+	[x] = {P_SPORT##x##_TFS, P_SPORT##x##_DTPRI, P_SPORT##x##_TSCLK, \
+	       P_SPORT##x##_RFS, P_SPORT##x##_DRPRI, P_SPORT##x##_RSCLK, 0}
 static u16 sport_req[][7] = {
-		PIN_REQ_SPORT_0,
-#ifdef PIN_REQ_SPORT_1
-		PIN_REQ_SPORT_1,
+#ifdef SPORT0_TCR1
+	SPORT_REQ(0),
+#endif
+#ifdef SPORT1_TCR1
+	SPORT_REQ(1),
 #endif
-#ifdef PIN_REQ_SPORT_2
-		PIN_REQ_SPORT_2,
+#ifdef SPORT2_TCR1
+	SPORT_REQ(2),
 #endif
-#ifdef PIN_REQ_SPORT_3
-		PIN_REQ_SPORT_3,
+#ifdef SPORT3_TCR1
+	SPORT_REQ(3),
 #endif
-	};
+};
 
+#define SPORT_PARAMS(x) \
+	[x] = { \
+		.dma_rx_chan = CH_SPORT##x##_RX, \
+		.dma_tx_chan = CH_SPORT##x##_TX, \
+		.err_irq     = IRQ_SPORT##x##_ERROR, \
+		.regs        = (struct sport_register *)SPORT##x##_TCR1, \
+	}
 static struct sport_param sport_params[4] = {
-	{
-		.dma_rx_chan	= CH_SPORT0_RX,
-		.dma_tx_chan	= CH_SPORT0_TX,
-		.err_irq	= IRQ_SPORT0_ERROR,
-		.regs		= (struct sport_register *)SPORT0_TCR1,
-	},
-#ifdef PIN_REQ_SPORT_1
-	{
-		.dma_rx_chan	= CH_SPORT1_RX,
-		.dma_tx_chan	= CH_SPORT1_TX,
-		.err_irq	= IRQ_SPORT1_ERROR,
-		.regs		= (struct sport_register *)SPORT1_TCR1,
-	},
+#ifdef SPORT0_TCR1
+	SPORT_PARAMS(0),
 #endif
-#ifdef PIN_REQ_SPORT_2
-	{
-		.dma_rx_chan	= CH_SPORT2_RX,
-		.dma_tx_chan	= CH_SPORT2_TX,
-		.err_irq	= IRQ_SPORT2_ERROR,
-		.regs		= (struct sport_register *)SPORT2_TCR1,
-	},
+#ifdef SPORT1_TCR1
+	SPORT_PARAMS(1),
 #endif
-#ifdef PIN_REQ_SPORT_3
-	{
-		.dma_rx_chan	= CH_SPORT3_RX,
-		.dma_tx_chan	= CH_SPORT3_TX,
-		.err_irq	= IRQ_SPORT3_ERROR,
-		.regs		= (struct sport_register *)SPORT3_TCR1,
-	}
+#ifdef SPORT2_TCR1
+	SPORT_PARAMS(2),
+#endif
+#ifdef SPORT3_TCR1
+	SPORT_PARAMS(3),
 #endif
 };
 
@@ -332,11 +306,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
 	if (cmd_count == NULL)
 		return -ENOMEM;
 
-	if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
+	if (peripheral_request_list(sport_req[sport_num], "soc-audio")) {
 		pr_err("Requesting Peripherals failed\n");
 		ret =  -EFAULT;
 		goto peripheral_err;
-		}
+	}
 
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	/* Request PB3 as reset pin */
@@ -383,9 +357,9 @@ sport_config_err:
 sport_err:
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
 gpio_err:
-	peripheral_free_list(&sport_req[sport_num][0]);
+#endif
+	peripheral_free_list(sport_req[sport_num]);
 peripheral_err:
 	free_page((unsigned long)cmd_count);
 	cmd_count = NULL;
@@ -398,7 +372,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
 {
 	free_page((unsigned long)cmd_count);
 	cmd_count = NULL;
-	peripheral_free_list(&sport_req[sport_num][0]);
+	peripheral_free_list(sport_req[sport_num]);
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
 #endif
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 7f2a5e1..edfbdc0 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -114,7 +114,7 @@ static int snd_ad73311_configure(void)
 	SSYNC();
 
 	/* When TUVF is set, the data is already send out */
-	while (!(status & TUVF) && count++ < 10000) {
+	while (!(status & TUVF) && ++count < 10000) {
 		udelay(1);
 		status = bfin_read_SPORT_STAT();
 		SSYNC();
@@ -123,7 +123,7 @@ static int snd_ad73311_configure(void)
 	SSYNC();
 	local_irq_enable();
 
-	if (count == 10000) {
+	if (count >= 10000) {
 		printk(KERN_ERR "ad73311: failed to configure codec\n");
 		return -1;
 	}
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 53d290b..1318c4f 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -184,7 +184,7 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
 	return 0 ;
 }
 
-struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
+static struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
 	.open		= bf5xx_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index d1d95d2..9648244 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev,
 #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\
 	SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
+	.startup	= bf5xx_i2s_startup,
+	.shutdown	= bf5xx_i2s_shutdown,
+	.hw_params	= bf5xx_i2s_hw_params,
+	.set_fmt	= bf5xx_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai bf5xx_i2s_dai = {
 	.name = "bf5xx-i2s",
 	.id = 0,
@@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
-	.ops = {
-		.startup   = bf5xx_i2s_startup,
-		.shutdown  = bf5xx_i2s_shutdown,
-		.hw_params = bf5xx_i2s_hw_params,
-		.set_fmt = bf5xx_i2s_set_dai_fmt,
-	},
+	.ops = &bf5xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index 3b99e48..b7953c8 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -133,7 +133,7 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
 	int i;
 
 	for (i = 0; i < fragcount; ++i) {
-		desc[i].next_desc_addr  = (unsigned long)&(desc[i + 1]);
+		desc[i].next_desc_addr  = &(desc[i + 1]);
 		desc[i].start_addr = (unsigned long)buf + i*fragsize;
 		desc[i].cfg = cfg;
 		desc[i].x_count = x_count;
@@ -143,12 +143,12 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
 	}
 
 	/* make circular */
-	desc[fragcount-1].next_desc_addr = (unsigned long)desc;
+	desc[fragcount-1].next_desc_addr = desc;
 
-	pr_debug("setup desc: desc0=%p, next0=%lx, desc1=%p,"
-		"next1=%lx\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n",
-		&(desc[0]), desc[0].next_desc_addr,
-		&(desc[1]), desc[1].next_desc_addr,
+	pr_debug("setup desc: desc0=%p, next0=%p, desc1=%p,"
+		"next1=%p\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n",
+		desc, desc[0].next_desc_addr,
+		desc+1, desc[1].next_desc_addr,
 		desc[0].x_count, desc[0].y_count,
 		desc[0].start_addr, desc[0].cfg);
 }
@@ -184,22 +184,20 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport)
 	BUG_ON(sport->curr_rx_desc == sport->dummy_rx_desc);
 
 	/* Maybe the dummy buffer descriptor ring is damaged */
-	sport->dummy_rx_desc->next_desc_addr = \
-			(unsigned long)(sport->dummy_rx_desc+1);
+	sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc + 1;
 
 	local_irq_save(flags);
-	desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_rx_chan);
+	desc = get_dma_next_desc_ptr(sport->dma_rx_chan);
 	/* Copy the descriptor which will be damaged to backup */
 	temp_desc = *desc;
 	desc->x_count = 0xa;
 	desc->y_count = 0;
-	desc->next_desc_addr = (unsigned long)(sport->dummy_rx_desc);
+	desc->next_desc_addr = sport->dummy_rx_desc;
 	local_irq_restore(flags);
 	/* Waiting for dummy buffer descriptor is already hooked*/
 	while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) -
-			sizeof(struct dmasg)) !=
-			(unsigned long)sport->dummy_rx_desc)
-		;
+			sizeof(struct dmasg)) != sport->dummy_rx_desc)
+		continue;
 	sport->curr_rx_desc = sport->dummy_rx_desc;
 	/* Restore the damaged descriptor */
 	*desc = temp_desc;
@@ -210,14 +208,12 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport)
 static inline int sport_rx_dma_start(struct sport_device *sport, int dummy)
 {
 	if (dummy) {
-		sport->dummy_rx_desc->next_desc_addr = \
-				(unsigned long) sport->dummy_rx_desc;
+		sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc;
 		sport->curr_rx_desc = sport->dummy_rx_desc;
 	} else
 		sport->curr_rx_desc = sport->dma_rx_desc;
 
-	set_dma_next_desc_addr(sport->dma_rx_chan, \
-			(unsigned long)(sport->curr_rx_desc));
+	set_dma_next_desc_addr(sport->dma_rx_chan, sport->curr_rx_desc);
 	set_dma_x_count(sport->dma_rx_chan, 0);
 	set_dma_x_modify(sport->dma_rx_chan, 0);
 	set_dma_config(sport->dma_rx_chan, (DMAFLOW_LARGE | NDSIZE_9 | \
@@ -231,14 +227,12 @@ static inline int sport_rx_dma_start(struct sport_device *sport, int dummy)
 static inline int sport_tx_dma_start(struct sport_device *sport, int dummy)
 {
 	if (dummy) {
-		sport->dummy_tx_desc->next_desc_addr = \
-				(unsigned long) sport->dummy_tx_desc;
+		sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc;
 		sport->curr_tx_desc = sport->dummy_tx_desc;
 	} else
 		sport->curr_tx_desc = sport->dma_tx_desc;
 
-	set_dma_next_desc_addr(sport->dma_tx_chan, \
-			(unsigned long)(sport->curr_tx_desc));
+	set_dma_next_desc_addr(sport->dma_tx_chan, sport->curr_tx_desc);
 	set_dma_x_count(sport->dma_tx_chan, 0);
 	set_dma_x_modify(sport->dma_tx_chan, 0);
 	set_dma_config(sport->dma_tx_chan,
@@ -261,11 +255,9 @@ int sport_rx_start(struct sport_device *sport)
 		BUG_ON(sport->curr_rx_desc != sport->dummy_rx_desc);
 		local_irq_save(flags);
 		while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) -
-			sizeof(struct dmasg)) !=
-			(unsigned long)sport->dummy_rx_desc)
-			;
-		sport->dummy_rx_desc->next_desc_addr =
-				(unsigned long)(sport->dma_rx_desc);
+			sizeof(struct dmasg)) != sport->dummy_rx_desc)
+			continue;
+		sport->dummy_rx_desc->next_desc_addr = sport->dma_rx_desc;
 		local_irq_restore(flags);
 		sport->curr_rx_desc = sport->dma_rx_desc;
 	} else {
@@ -310,23 +302,21 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport)
 	BUG_ON(sport->dummy_tx_desc == NULL);
 	BUG_ON(sport->curr_tx_desc == sport->dummy_tx_desc);
 
-	sport->dummy_tx_desc->next_desc_addr = \
-			(unsigned long)(sport->dummy_tx_desc+1);
+	sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc + 1;
 
 	/* Shorten the time on last normal descriptor */
 	local_irq_save(flags);
-	desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_tx_chan);
+	desc = get_dma_next_desc_ptr(sport->dma_tx_chan);
 	/* Store the descriptor which will be damaged */
 	temp_desc = *desc;
 	desc->x_count = 0xa;
 	desc->y_count = 0;
-	desc->next_desc_addr = (unsigned long)(sport->dummy_tx_desc);
+	desc->next_desc_addr = sport->dummy_tx_desc;
 	local_irq_restore(flags);
 	/* Waiting for dummy buffer descriptor is already hooked*/
 	while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - \
-			sizeof(struct dmasg)) != \
-			(unsigned long)sport->dummy_tx_desc)
-		;
+			sizeof(struct dmasg)) != sport->dummy_tx_desc)
+		continue;
 	sport->curr_tx_desc = sport->dummy_tx_desc;
 	/* Restore the damaged descriptor */
 	*desc = temp_desc;
@@ -347,11 +337,9 @@ int sport_tx_start(struct sport_device *sport)
 		/* Hook the normal buffer descriptor */
 		local_irq_save(flags);
 		while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) -
-			sizeof(struct dmasg)) !=
-			(unsigned long)sport->dummy_tx_desc)
-			;
-		sport->dummy_tx_desc->next_desc_addr =
-				(unsigned long)(sport->dma_tx_desc);
+			sizeof(struct dmasg)) != sport->dummy_tx_desc)
+			continue;
+		sport->dummy_tx_desc->next_desc_addr = sport->dma_tx_desc;
 		local_irq_restore(flags);
 		sport->curr_tx_desc = sport->dma_tx_desc;
 	} else {
@@ -536,19 +524,17 @@ static int sport_config_rx_dummy(struct sport_device *sport)
 	unsigned config;
 
 	pr_debug("%s entered\n", __func__);
-#if L1_DATA_A_LENGTH != 0
-	desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc));
-#else
-	{
+	if (L1_DATA_A_LENGTH)
+		desc = l1_data_sram_zalloc(2 * sizeof(*desc));
+	else {
 		dma_addr_t addr;
 		desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0);
+		memset(desc, 0, 2 * sizeof(*desc));
 	}
-#endif
 	if (desc == NULL) {
 		pr_err("Failed to allocate memory for dummy rx desc\n");
 		return -ENOMEM;
 	}
-	memset(desc, 0, 2 * sizeof(*desc));
 	sport->dummy_rx_desc = desc;
 	desc->start_addr = (unsigned long)sport->dummy_buf;
 	config = DMAFLOW_LARGE | NDSIZE_9 | compute_wdsize(sport->wdsize)
@@ -559,8 +545,8 @@ static int sport_config_rx_dummy(struct sport_device *sport)
 	desc->y_count = 0;
 	desc->y_modify = 0;
 	memcpy(desc+1, desc, sizeof(*desc));
-	desc->next_desc_addr = (unsigned long)(desc+1);
-	desc[1].next_desc_addr = (unsigned long)desc;
+	desc->next_desc_addr = desc + 1;
+	desc[1].next_desc_addr = desc;
 	return 0;
 }
 
@@ -571,19 +557,17 @@ static int sport_config_tx_dummy(struct sport_device *sport)
 
 	pr_debug("%s entered\n", __func__);
 
-#if L1_DATA_A_LENGTH != 0
-	desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc));
-#else
-	{
+	if (L1_DATA_A_LENGTH)
+		desc = l1_data_sram_zalloc(2 * sizeof(*desc));
+	else {
 		dma_addr_t addr;
 		desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0);
+		memset(desc, 0, 2 * sizeof(*desc));
 	}
-#endif
 	if (!desc) {
 		pr_err("Failed to allocate memory for dummy tx desc\n");
 		return -ENOMEM;
 	}
-	memset(desc, 0, 2 * sizeof(*desc));
 	sport->dummy_tx_desc = desc;
 	desc->start_addr = (unsigned long)sport->dummy_buf + \
 		sport->dummy_count;
@@ -595,8 +579,8 @@ static int sport_config_tx_dummy(struct sport_device *sport)
 	desc->y_count = 0;
 	desc->y_modify = 0;
 	memcpy(desc+1, desc, sizeof(*desc));
-	desc->next_desc_addr = (unsigned long)(desc+1);
-	desc[1].next_desc_addr = (unsigned long)desc;
+	desc->next_desc_addr = desc + 1;
+	desc[1].next_desc_addr = desc;
 	return 0;
 }
 
@@ -872,17 +856,15 @@ struct sport_device *sport_init(struct sport_param *param, unsigned wdsize,
 	sport->wdsize = wdsize;
 	sport->dummy_count = dummy_count;
 
-#if L1_DATA_A_LENGTH != 0
-	sport->dummy_buf = l1_data_sram_alloc(dummy_count * 2);
-#else
-	sport->dummy_buf = kmalloc(dummy_count * 2, GFP_KERNEL);
-#endif
+	if (L1_DATA_A_LENGTH)
+		sport->dummy_buf = l1_data_sram_zalloc(dummy_count * 2);
+	else
+		sport->dummy_buf = kzalloc(dummy_count * 2, GFP_KERNEL);
 	if (sport->dummy_buf == NULL) {
 		pr_err("Failed to allocate dummy buffer\n");
 		goto __error;
 	}
 
-	memset(sport->dummy_buf, 0, dummy_count * 2);
 	ret = sport_config_rx_dummy(sport);
 	if (ret) {
 		pr_err("Failed to config rx dummy ring\n");
@@ -939,6 +921,7 @@ void sport_done(struct sport_device *sport)
 		sport = NULL;
 }
 EXPORT_SYMBOL(sport_done);
+
 /*
 * It is only used to send several bytes when dma is not enabled
  * sport controller is configured but not enabled.
@@ -1029,4 +1012,3 @@ EXPORT_SYMBOL(sport_send_and_recv);
 MODULE_AUTHOR("Roy Huang");
 MODULE_DESCRIPTION("SPORT driver for ADI Blackfin");
 MODULE_LICENSE("GPL");
-
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d0e0d69..b6c7f7a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,9 +10,11 @@ config SND_SOC_I2C_AND_SPI
 
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	select SND_SOC_L3
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311 if I2C
+	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_PCM3008
@@ -24,6 +26,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
 	select SND_SOC_WM8350 if MFD_WM8350
+	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8580 if I2C
 	select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
@@ -34,6 +37,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8903 if I2C
 	select SND_SOC_WM8971 if I2C
 	select SND_SOC_WM8990 if I2C
+	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9713 if SND_SOC_AC97_BUS
         help
@@ -58,6 +62,9 @@ config SND_SOC_AD1980
 config SND_SOC_AD73311
 	tristate
 
+config SND_SOC_AK4104
+	tristate
+
 config SND_SOC_AK4535
 	tristate
 
@@ -65,12 +72,6 @@ config SND_SOC_AK4535
 config SND_SOC_CS4270
 	tristate
 
-# Cirrus Logic CS4270 Codec Hardware Mute Support
-# Select if you have external muting circuitry attached to your CS4270.
-config SND_SOC_CS4270_HWMUTE
-	bool
-	depends on SND_SOC_CS4270
-
 # 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
@@ -90,7 +91,6 @@ config SND_SOC_SSM2602
 
 config SND_SOC_TLV320AIC23
 	tristate
-	depends on I2C
 
 config SND_SOC_TLV320AIC26
 	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
@@ -98,15 +98,12 @@ config SND_SOC_TLV320AIC26
 
 config SND_SOC_TLV320AIC3X
 	tristate
-	depends on I2C
 
 config SND_SOC_TWL4030
 	tristate
-	depends on TWL4030_CORE
 
 config SND_SOC_UDA134X
        tristate
-       select SND_SOC_L3
 
 config SND_SOC_UDA1380
         tristate
@@ -114,6 +111,9 @@ config SND_SOC_UDA1380
 config SND_SOC_WM8350
 	tristate
 
+config SND_SOC_WM8400
+	tristate
+
 config SND_SOC_WM8510
 	tristate
 
@@ -144,6 +144,9 @@ config SND_SOC_WM8971
 config SND_SOC_WM8990
 	tristate
 
+config SND_SOC_WM9705
+	tristate
+
 config SND_SOC_WM9712
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index c4ddc9a..030d245 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,6 +1,7 @@
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
+snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
@@ -13,6 +14,7 @@ snd-soc-twl4030-objs := twl4030.o
 snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
 snd-soc-wm8350-objs := wm8350.o
+snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8580-objs := wm8580.o
 snd-soc-wm8728-objs := wm8728.o
@@ -23,12 +25,14 @@ snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
 snd-soc-wm8971-objs := wm8971.o
 snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
@@ -41,6 +45,7 @@ obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
 obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
+obj-$(CONFIG_SND_SOC_WM8400)	+= snd-soc-wm8400.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
 obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
@@ -51,5 +56,7 @@ obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
+obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index fb53e65..b0d4af1 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -30,7 +30,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		  AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
@@ -41,6 +41,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops ac97_dai_ops = {
+	.prepare	= ac97_prepare,
+};
+
 struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
 	.ac97_control = 1,
@@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = {
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ac97_dai);
 
@@ -84,10 +87,10 @@ static int ac97_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (!socdev->codec)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->card->codec)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->name = "AC97";
@@ -123,23 +126,21 @@ bus_err:
 	snd_soc_free_pcms(socdev);
 
 err:
-	kfree(socdev->codec->reg_cache);
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ac97_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (!codec)
 		return 0;
 
 	snd_soc_free_pcms(socdev);
-	kfree(socdev->codec->reg_cache);
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return 0;
 }
@@ -149,7 +150,7 @@ static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	snd_ac97_suspend(socdev->codec->ac97);
+	snd_ac97_suspend(socdev->card->codec->ac97);
 
 	return 0;
 }
@@ -158,7 +159,7 @@ static int ac97_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	snd_ac97_resume(socdev->codec->ac97);
+	snd_ac97_resume(socdev->card->codec->ac97);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 73fdbb4..ddb3b08 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src),
 SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
 };
 
-/* add non dapm controls */
-static int ad1980_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card, snd_soc_cnew(
-				&ad1980_snd_ac97_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 static unsigned int ac97_read(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
@@ -123,7 +109,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	default:
 		reg = reg >> 1;
 
-		if (reg >= (ARRAY_SIZE(ad1980_reg)))
+		if (reg >= ARRAY_SIZE(ad1980_reg))
 			return -EINVAL;
 
 		return cache[reg];
@@ -137,7 +123,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg < (ARRAY_SIZE(ad1980_reg)))
+	if (reg < ARRAY_SIZE(ad1980_reg))
 		cache[reg] = val;
 
 	return 0;
@@ -200,10 +186,10 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache =
@@ -269,7 +255,8 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 	ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
 	ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
 
-	ad1980_add_controls(codec);
+	snd_soc_add_controls(codec, ad1980_snd_ac97_controls,
+				ARRAY_SIZE(ad1980_snd_ac97_controls));
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ad1980: failed to register card\n");
@@ -288,15 +275,15 @@ codec_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ad1980_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index b09289a..e61dac5 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -53,7 +53,7 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 	codec->owner = THIS_MODULE;
 	codec->dai = &ad73311_dai;
 	codec->num_dai = 1;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -75,15 +75,15 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 register_err:
 	snd_soc_free_pcms(socdev);
 pcm_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ad73311_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
index 507ce0c..569573d 100644
--- a/sound/soc/codecs/ad73311.h
+++ b/sound/soc/codecs/ad73311.h
@@ -70,7 +70,7 @@
 #define REGD_IGS(x)		(x & 0x7)
 #define REGD_RMOD		(1 << 3)
 #define REGD_OGS(x)		((x & 0x7) << 4)
-#define REGD_MUTE		(x << 7)
+#define REGD_MUTE		(1 << 7)
 
 /* Control register E */
 #define CTRL_REG_E	(4 << 8)
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
new file mode 100644
index 0000000..4d47bc4
--- /dev/null
+++ b/sound/soc/codecs/ak4104.c
@@ -0,0 +1,365 @@
+/*
+ * AK4104 ALSA SoC (ASoC) driver
+ *
+ * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <linux/spi/spi.h>
+#include <sound/asoundef.h>
+
+#include "ak4104.h"
+
+/* AK4104 registers addresses */
+#define AK4104_REG_CONTROL1		0x00
+#define AK4104_REG_RESERVED		0x01
+#define AK4104_REG_CONTROL2		0x02
+#define AK4104_REG_TX			0x03
+#define AK4104_REG_CHN_STATUS(x)	((x) + 0x04)
+#define AK4104_NUM_REGS			10
+
+#define AK4104_REG_MASK			0x1f
+#define AK4104_READ			0xc0
+#define AK4104_WRITE			0xe0
+#define AK4104_RESERVED_VAL		0x5b
+
+/* Bit masks for AK4104 registers */
+#define AK4104_CONTROL1_RSTN		(1 << 0)
+#define AK4104_CONTROL1_PW		(1 << 1)
+#define AK4104_CONTROL1_DIF0		(1 << 2)
+#define AK4104_CONTROL1_DIF1		(1 << 3)
+
+#define AK4104_CONTROL2_SEL0		(1 << 0)
+#define AK4104_CONTROL2_SEL1		(1 << 1)
+#define AK4104_CONTROL2_MODE		(1 << 2)
+
+#define AK4104_TX_TXE			(1 << 0)
+#define AK4104_TX_V			(1 << 1)
+
+#define DRV_NAME "ak4104"
+
+struct ak4104_private {
+	struct snd_soc_codec codec;
+	u8 reg_cache[AK4104_NUM_REGS];
+};
+
+static int ak4104_fill_cache(struct snd_soc_codec *codec)
+{
+	int i;
+	u8 *reg_cache = codec->reg_cache;
+	struct spi_device *spi = codec->control_data;
+
+	for (i = 0; i < codec->reg_cache_size; i++) {
+		int ret = spi_w8r8(spi, i | AK4104_READ);
+		if (ret < 0) {
+			dev_err(&spi->dev, "SPI write failure\n");
+			return ret;
+		}
+
+		reg_cache[i] = ret;
+	}
+
+	return 0;
+}
+
+static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec,
+					  unsigned int reg)
+{
+	u8 *reg_cache = codec->reg_cache;
+
+	if (reg >= codec->reg_cache_size)
+		return -EINVAL;
+
+	return reg_cache[reg];
+}
+
+static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg,
+			    unsigned int value)
+{
+	u8 *cache = codec->reg_cache;
+	struct spi_device *spi = codec->control_data;
+
+	if (reg >= codec->reg_cache_size)
+		return -EINVAL;
+
+	reg &= AK4104_REG_MASK;
+	reg |= AK4104_WRITE;
+
+	/* only write to the hardware if value has changed */
+	if (cache[reg] != value) {
+		u8 tmp[2] = { reg, value };
+		if (spi_write(spi, tmp, sizeof(tmp))) {
+			dev_err(&spi->dev, "SPI write failed\n");
+			return -EIO;
+		}
+
+		cache[reg] = value;
+	}
+
+	return 0;
+}
+
+static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int val = 0;
+
+	val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+	if (val < 0)
+		return val;
+
+	val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1);
+
+	/* set DAI format */
+	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val |= AK4104_CONTROL1_DIF0;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1;
+		break;
+	default:
+		dev_err(codec->dev, "invalid dai format\n");
+		return -EINVAL;
+	}
+
+	/* This device can only be slave */
+	if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
+
+	return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+}
+
+static int ak4104_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;
+	int val = 0;
+
+	/* set the IEC958 bits: consumer mode, no copyright bit */
+	val |= IEC958_AES0_CON_NOT_COPYRIGHT;
+	ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val);
+
+	val = 0;
+
+	switch (params_rate(params)) {
+	case 44100:
+		val |= IEC958_AES3_CON_FS_44100;
+		break;
+	case 48000:
+		val |= IEC958_AES3_CON_FS_48000;
+		break;
+	case 32000:
+		val |= IEC958_AES3_CON_FS_32000;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported sampling rate\n");
+		return -EINVAL;
+	}
+
+	return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val);
+}
+
+static struct snd_soc_dai_ops ak4101_dai_ops = {
+	.hw_params = ak4104_hw_params,
+	.set_fmt = ak4104_set_dai_fmt,
+};
+
+struct snd_soc_dai ak4104_dai = {
+	.name = DRV_NAME,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000 |
+			 SNDRV_PCM_RATE_32000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE  |
+			   SNDRV_PCM_FMTBIT_S24_3LE |
+			   SNDRV_PCM_FMTBIT_S24_LE
+	},
+	.ops = &ak4101_dai_ops,
+};
+
+static struct snd_soc_codec *ak4104_codec;
+
+static int ak4104_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_codec *codec;
+	struct ak4104_private *ak4104;
+	int ret, val;
+
+	spi->bits_per_word = 8;
+	spi->mode = SPI_MODE_0;
+	ret = spi_setup(spi);
+	if (ret < 0)
+		return ret;
+
+	ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL);
+	if (!ak4104) {
+		dev_err(&spi->dev, "could not allocate codec\n");
+		return -ENOMEM;
+	}
+
+	codec = &ak4104->codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->dev = &spi->dev;
+	codec->name = DRV_NAME;
+	codec->owner = THIS_MODULE;
+	codec->dai = &ak4104_dai;
+	codec->num_dai = 1;
+	codec->private_data = ak4104;
+	codec->control_data = spi;
+	codec->reg_cache = ak4104->reg_cache;
+	codec->reg_cache_size = AK4104_NUM_REGS;
+
+	/* read all regs and fill the cache */
+	ret = ak4104_fill_cache(codec);
+	if (ret < 0) {
+		dev_err(&spi->dev, "failed to fill register cache\n");
+		return ret;
+	}
+
+	/* read the 'reserved' register - according to the datasheet, it
+	 * should contain 0x5b. Not a good way to verify the presence of
+	 * the device, but there is no hardware ID register. */
+	if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) !=
+					 AK4104_RESERVED_VAL) {
+		ret = -ENODEV;
+		goto error_free_codec;
+	}
+
+	/* set power-up and non-reset bits */
+	val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+	val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN;
+	ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+	if (ret < 0)
+		goto error_free_codec;
+
+	/* enable transmitter */
+	val = ak4104_read_reg_cache(codec, AK4104_REG_TX);
+	val |= AK4104_TX_TXE;
+	ret = ak4104_spi_write(codec, AK4104_REG_TX, val);
+	if (ret < 0)
+		goto error_free_codec;
+
+	ak4104_codec = codec;
+	ret = snd_soc_register_dai(&ak4104_dai);
+	if (ret < 0) {
+		dev_err(&spi->dev, "failed to register DAI\n");
+		goto error_free_codec;
+	}
+
+	spi_set_drvdata(spi, ak4104);
+	dev_info(&spi->dev, "SPI device initialized\n");
+	return 0;
+
+error_free_codec:
+	kfree(ak4104);
+	ak4104_dai.dev = NULL;
+	return ret;
+}
+
+static int __devexit ak4104_spi_remove(struct spi_device *spi)
+{
+	int ret, val;
+	struct ak4104_private *ak4104 = spi_get_drvdata(spi);
+
+	val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1);
+	if (val < 0)
+		return val;
+
+	/* clear power-up and non-reset bits */
+	val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
+	ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val);
+	if (ret < 0)
+		return ret;
+
+	ak4104_codec = NULL;
+	kfree(ak4104);
+	return 0;
+}
+
+static int ak4104_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = ak4104_codec;
+	int ret;
+
+	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
+	socdev->card->codec = 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\n");
+		return ret;
+	}
+
+	/* Register the socdev */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card\n");
+		snd_soc_free_pcms(socdev);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int ak4104_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_device_ak4104 = {
+	.probe = 	ak4104_probe,
+	.remove = 	ak4104_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_ak4104);
+
+static struct spi_driver ak4104_spi_driver = {
+	.driver  = {
+		.name   = DRV_NAME,
+		.owner  = THIS_MODULE,
+	},
+	.probe  = ak4104_spi_probe,
+	.remove = __devexit_p(ak4104_spi_remove),
+};
+
+static int __init ak4104_init(void)
+{
+	pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n");
+	return spi_register_driver(&ak4104_spi_driver);
+}
+module_init(ak4104_init);
+
+static void __exit ak4104_exit(void)
+{
+	spi_unregister_driver(&ak4104_spi_driver);
+}
+module_exit(ak4104_exit);
+
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h
new file mode 100644
index 0000000..eb88fe7
--- /dev/null
+++ b/sound/soc/codecs/ak4104.h
@@ -0,0 +1,7 @@
+#ifndef _AK4104_H
+#define _AK4104_H
+
+extern struct snd_soc_dai ak4104_dai;
+extern struct snd_soc_codec_device soc_codec_device_ak4104;
+
+#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 81300d8d..1f63d38 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = {
 	SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
 };
 
-/* add non dapm controls */
-static int ak4535_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Mono 1 Mixer */
 static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
@@ -344,7 +329,7 @@ static int ak4535_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ak4535_priv *ak4535 = codec->private_data;
 	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
 	int rate = params_rate(params), fs = 256;
@@ -436,6 +421,13 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops ak4535_dai_ops = {
+	.hw_params	= ak4535_hw_params,
+	.set_fmt	= ak4535_set_dai_fmt,
+	.digital_mute	= ak4535_mute,
+	.set_sysclk	= ak4535_set_dai_sysclk,
+};
+
 struct snd_soc_dai ak4535_dai = {
 	.name = "AK4535",
 	.playback = {
@@ -450,19 +442,14 @@ struct snd_soc_dai ak4535_dai = {
 		.channels_max = 2,
 		.rates = AK4535_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = ak4535_hw_params,
-		.set_fmt = ak4535_set_dai_fmt,
-		.digital_mute = ak4535_mute,
-		.set_sysclk = ak4535_set_dai_sysclk,
-	},
+	.ops = &ak4535_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ak4535_dai);
 
 static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -471,7 +458,7 @@ static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
 static int ak4535_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	ak4535_sync(codec);
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	ak4535_set_bias_level(codec, codec->suspend_bias_level);
@@ -484,7 +471,7 @@ static int ak4535_resume(struct platform_device *pdev)
  */
 static int ak4535_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "AK4535";
@@ -510,7 +497,8 @@ static int ak4535_init(struct snd_soc_device *socdev)
 	/* power on device */
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	ak4535_add_controls(codec);
+	snd_soc_add_controls(codec, ak4535_snd_controls,
+				ARRAY_SIZE(ak4535_snd_controls));
 	ak4535_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -537,7 +525,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = ak4535_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -636,7 +624,7 @@ static int ak4535_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = ak4535;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -663,7 +651,7 @@ static int ak4535_probe(struct platform_device *pdev)
 static int ak4535_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1aa0c3..7fa09a3 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -3,27 +3,22 @@
  *
  * Author: Timur Tabi <timur@freescale.com>
  *
- * Copyright 2007 Freescale Semiconductor, Inc.  This file is licensed under
- * the terms of the GNU General Public License version 2.  This program
- * is licensed "as is" without any warranty of any kind, whether express
- * or implied.
+ * Copyright 2007-2009 Freescale Semiconductor, Inc.  This file is licensed
+ * under the terms of the GNU General Public License version 2.  This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
  *
  * This is an ASoC device driver for the Cirrus Logic CS4270 codec.
  *
  * Current features/limitations:
  *
- * 1) Software mode is supported.  Stand-alone mode is automatically
- *    selected if I2C is disabled or if a CS4270 is not found on the I2C
- *    bus.  However, stand-alone mode is only partially implemented because
- *    there is no mechanism yet for this driver and the machine driver to
- *    communicate the values of the M0, M1, MCLK1, and MCLK2 pins.
- * 2) Only I2C is supported, not SPI
- * 3) Only Master mode is supported, not Slave.
- * 4) The machine driver's 'startup' function must call
- *    cs4270_set_dai_sysclk() with the value of MCLK.
- * 5) Only I2S and left-justified modes are supported
- * 6) Power management is not supported
- * 7) The only supported control is volume and hardware mute (if enabled)
+ * - Software mode is supported.  Stand-alone mode is not supported.
+ * - Only I2C is supported, not SPI
+ * - Support for master and slave mode
+ * - The machine driver's 'startup' function must call
+ *   cs4270_set_dai_sysclk() with the value of MCLK.
+ * - Only I2S and left-justified modes are supported
+ * - Power management is not supported
  */
 
 #include <linux/module.h>
@@ -35,18 +30,6 @@
 
 #include "cs4270.h"
 
-/* If I2C is defined, then we support software mode.  However, if we're
-   not compiled as module but I2C is, then we can't use I2C calls. */
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-#define USE_I2C
-#endif
-
-/* Private data for the CS4270 */
-struct cs4270_private {
-	unsigned int mclk; /* Input frequency of the MCLK pin */
-	unsigned int mode; /* The mode (I2S or left-justified) */
-};
-
 /*
  * The codec isn't really big-endian or little-endian, since the I2S
  * interface requires data to be sent serially with the MSbit first.
@@ -60,8 +43,6 @@ struct cs4270_private {
 			SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \
 			SNDRV_PCM_FMTBIT_S24_LE  | SNDRV_PCM_FMTBIT_S24_BE)
 
-#ifdef USE_I2C
-
 /* CS4270 registers addresses */
 #define CS4270_CHIPID	0x01	/* Chip ID */
 #define CS4270_PWRCTL	0x02	/* Power Control */
@@ -121,8 +102,22 @@ struct cs4270_private {
 #define CS4270_MUTE_DAC_A	0x01
 #define CS4270_MUTE_DAC_B	0x02
 
-/*
- * Clock Ratio Selection for Master Mode with I2C enabled
+/* Private data for the CS4270 */
+struct cs4270_private {
+	struct snd_soc_codec codec;
+	u8 reg_cache[CS4270_NUMREGS];
+	unsigned int mclk; /* Input frequency of the MCLK pin */
+	unsigned int mode; /* The mode (I2S or left-justified) */
+	unsigned int slave_mode;
+};
+
+/**
+ * struct cs4270_mode_ratios - clock ratio tables
+ * @ratio: the ratio of MCLK to the sample rate
+ * @speed_mode: the Speed Mode bits to set in the Mode Control register for
+ *              this ratio
+ * @mclk: the Ratio Select bits to set in the Mode Control register for this
+ *        ratio
  *
  * The data for this chart is taken from Table 5 of the CS4270 reference
  * manual.
@@ -131,31 +126,30 @@ struct cs4270_private {
  * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling
  * rates the CS4270 currently supports.
  *
- * Each element in this array corresponds to the ratios in mclk_ratios[].
- * These two arrays need to be in sync.
- *
- * 'speed_mode' is the corresponding bit pattern to be written to the
+ * @speed_mode is the corresponding bit pattern to be written to the
  * MODE bits of the Mode Control Register
  *
- * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of
+ * @mclk is the corresponding bit pattern to be wirten to the MCLK bits of
  * the Mode Control Register.
  *
  * In situations where a single ratio is represented by multiple speed
  * modes, we favor the slowest speed.  E.g, for a ratio of 128, we pick
  * double-speed instead of quad-speed.  However, the CS4270 errata states
- * that Divide-By-1.5 can cause failures, so we avoid that mode where
+ * that divide-By-1.5 can cause failures, so we avoid that mode where
  * possible.
  *
- * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not
- * work if VD = 3.3V.  If this effects you, select the
+ * Errata: There is an errata for the CS4270 where divide-by-1.5 does not
+ * work if Vd is 3.3V.  If this effects you, select the
  * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will
  * never select any sample rates that require divide-by-1.5.
  */
-static struct {
+struct cs4270_mode_ratios {
 	unsigned int ratio;
 	u8 speed_mode;
 	u8 mclk;
-} cs4270_mode_ratios[] = {
+};
+
+static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
 	{64, CS4270_MODE_4X, CS4270_MODE_DIV1},
 #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA
 	{96, CS4270_MODE_4X, CS4270_MODE_DIV15},
@@ -172,34 +166,27 @@ static struct {
 /* The number of MCLK/LRCK ratios supported by the CS4270 */
 #define NUM_MCLK_RATIOS		ARRAY_SIZE(cs4270_mode_ratios)
 
-/*
- * Determine the CS4270 samples rates.
+/**
+ * cs4270_set_dai_sysclk - determine the CS4270 samples rates.
+ * @codec_dai: the codec DAI
+ * @clk_id: the clock ID (ignored)
+ * @freq: the MCLK input frequency
+ * @dir: the clock direction (ignored)
  *
- * 'freq' is the input frequency to MCLK.  The other parameters are ignored.
+ * This function is used to tell the codec driver what the input MCLK
+ * frequency is.
  *
  * The value of MCLK is used to determine which sample rates are supported
  * by the CS4270.  The ratio of MCLK / Fs must be equal to one of nine
- * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024.
+ * supported values - 64, 96, 128, 192, 256, 384, 512, 768, and 1024.
  *
  * This function calculates the nine ratios and determines which ones match
  * a standard sample rate.  If there's a match, then it is added to the list
- * of support sample rates.
+ * of supported sample rates.
  *
  * This function must be called by the machine driver's 'startup' function,
  * otherwise the list of supported sample rates will not be available in
  * time for ALSA.
- *
- * Note that in stand-alone mode, the sample rate is determined by input
- * pins M0, M1, MDIV1, and MDIV2.  Also in stand-alone mode, divide-by-3
- * is not a programmable option.  However, divide-by-3 is not an available
- * option in stand-alone mode.  This cases two problems: a ratio of 768 is
- * not available (it requires divide-by-3) and B) ratios 192 and 384 can
- * only be selected with divide-by-1.5, but there is an errate that make
- * this selection difficult.
- *
- * In addition, there is no mechanism for communicating with the machine
- * driver what the input settings can be.  This would need to be implemented
- * for stand-alone mode to work.
  */
 static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
@@ -225,7 +212,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	rates &= ~SNDRV_PCM_RATE_KNOT;
 
 	if (!rates) {
-		printk(KERN_ERR "cs4270: could not find a valid sample rate\n");
+		dev_err(codec->dev, "could not find a valid sample rate\n");
 		return -EINVAL;
 	}
 
@@ -240,8 +227,10 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-/*
- * Configure the codec for the selected audio format
+/**
+ * cs4270_set_dai_fmt - configure the codec for the selected audio format
+ * @codec_dai: the codec DAI
+ * @format: a SND_SOC_DAIFMT_x value indicating the data format
  *
  * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the
  * codec accordingly.
@@ -258,32 +247,43 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct cs4270_private *cs4270 = codec->private_data;
 	int ret = 0;
 
+	/* set DAI format */
 	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 	case SND_SOC_DAIFMT_LEFT_J:
 		cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
 		break;
 	default:
-		printk(KERN_ERR "cs4270: invalid DAI format\n");
+		dev_err(codec->dev, "invalid dai format\n");
+		ret = -EINVAL;
+	}
+
+	/* set master/slave audio interface */
+	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs4270->slave_mode = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs4270->slave_mode = 0;
+		break;
+	default:
+		/* all other modes are unsupported by the hardware */
 		ret = -EINVAL;
 	}
 
 	return ret;
 }
 
-/*
- * A list of addresses on which this CS4270 could use.  I2C addresses are
- * 7 bits.  For the CS4270, the upper four bits are always 1001, and the
- * lower three bits are determined via the AD2, AD1, and AD0 pins
- * (respectively).
- */
-static const unsigned short normal_i2c[] = {
-	0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END
-};
-I2C_CLIENT_INSMOD;
-
-/*
- * Pre-fill the CS4270 register cache.
+/**
+ * cs4270_fill_cache - pre-fill the CS4270 register cache.
+ * @codec: the codec for this CS4270
+ *
+ * This function fills in the CS4270 register cache by reading the register
+ * values from the hardware.
+ *
+ * This CS4270 registers are cached to avoid excessive I2C I/O operations.
+ * After the initial read to pre-fill the cache, the CS4270 never updates
+ * the register values, so we won't have a cache coherency problem.
  *
  * We use the auto-increment feature of the CS4270 to read all registers in
  * one shot.
@@ -298,7 +298,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 		CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
 
 	if (length != CS4270_NUMREGS) {
-		printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n",
+		dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
 		       i2c_client->addr);
 		return -EIO;
 	}
@@ -306,12 +306,17 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 	return 0;
 }
 
-/*
- * Read from the CS4270 register cache.
+/**
+ * cs4270_read_reg_cache - read from the CS4270 register cache.
+ * @codec: the codec for this CS4270
+ * @reg: the register to read
+ *
+ * This function returns the value for a given register.  It reads only from
+ * the register cache, not the hardware itself.
  *
  * This CS4270 registers are cached to avoid excessive I2C I/O operations.
  * After the initial read to pre-fill the cache, the CS4270 never updates
- * the register values, so we won't have a cache coherncy problem.
+ * the register values, so we won't have a cache coherency problem.
  */
 static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
@@ -324,8 +329,11 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
 	return cache[reg - CS4270_FIRSTREG];
 }
 
-/*
- * Write to a CS4270 register via the I2C bus.
+/**
+ * cs4270_i2c_write - write to a CS4270 register via the I2C bus.
+ * @codec: the codec for this CS4270
+ * @reg: the register to write
+ * @value: the value to write to the register
  *
  * This function writes the given value to the given CS4270 register, and
  * also updates the register cache.
@@ -346,7 +354,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (cache[reg - CS4270_FIRSTREG] != value) {
 		struct i2c_client *client = codec->control_data;
 		if (i2c_smbus_write_byte_data(client, reg, value)) {
-			printk(KERN_ERR "cs4270: I2C write failed\n");
+			dev_err(codec->dev, "i2c write failed\n");
 			return -EIO;
 		}
 
@@ -357,11 +365,17 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 	return 0;
 }
 
-/*
- * Program the CS4270 with the given hardware parameters.
+/**
+ * cs4270_hw_params - program the CS4270 with the given hardware parameters.
+ * @substream: the audio stream
+ * @params: the hardware parameters to set
+ * @dai: the SOC DAI (ignored)
+ *
+ * This function programs the hardware with the values provided.
+ * Specifically, the sample rate and the data format.
  *
- * The .ops functions are used to provide board-specific data, like
- * input frequencies, to this driver.  This function takes that information,
+ * The .ops functions are used to provide board-specific data, like input
+ * frequencies, to this driver.  This function takes that information,
  * combines it with the hardware parameters provided, and programs the
  * hardware accordingly.
  */
@@ -371,7 +385,7 @@ static int cs4270_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct cs4270_private *cs4270 = codec->private_data;
 	int ret;
 	unsigned int i;
@@ -391,33 +405,28 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 
 	if (i == NUM_MCLK_RATIOS) {
 		/* We did not find a matching ratio */
-		printk(KERN_ERR "cs4270: could not find matching ratio\n");
+		dev_err(codec->dev, "could not find matching ratio\n");
 		return -EINVAL;
 	}
 
-	/* Freeze and power-down the codec */
-
-	ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE |
-			    CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC |
-			    CS4270_PWRCTL_PDN);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
-		return ret;
-	}
-
-	/* Program the mode control register */
+	/* Set the sample rate */
 
 	reg = snd_soc_read(codec, CS4270_MODE);
 	reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK);
-	reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk;
+	reg |= cs4270_mode_ratios[i].mclk;
+
+	if (cs4270->slave_mode)
+		reg |= CS4270_MODE_SLAVE;
+	else
+		reg |= cs4270_mode_ratios[i].speed_mode;
 
 	ret = snd_soc_write(codec, CS4270_MODE, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
-	/* Program the format register */
+	/* Set the DAI format */
 
 	reg = snd_soc_read(codec, CS4270_FORMAT);
 	reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK);
@@ -430,55 +439,23 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ;
 		break;
 	default:
-		printk(KERN_ERR "cs4270: unknown format\n");
+		dev_err(codec->dev, "unknown dai format\n");
 		return -EINVAL;
 	}
 
 	ret = snd_soc_write(codec, CS4270_FORMAT, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
-		return ret;
-	}
-
-	/* Disable auto-mute.  This feature appears to be buggy, because in
-	   some situations, auto-mute will not deactivate when it should. */
-
-	reg = snd_soc_read(codec, CS4270_MUTE);
-	reg &= ~CS4270_MUTE_AUTO;
-	ret = snd_soc_write(codec, CS4270_MUTE, reg);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
-		return ret;
-	}
-
-	/* Disable automatic volume control.  It's enabled by default, and
-	 * it causes volume change commands to be delayed, sometimes until
-	 * after playback has started.
-	 */
-
-	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
-	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
-	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
-	if (ret < 0) {
-		printk(KERN_ERR "I2C write failed\n");
-		return ret;
-	}
-
-	/* Thaw and power-up the codec */
-
-	ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
 	return ret;
 }
 
-#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-
-/*
- * Set the CS4270 external mute
+/**
+ * cs4270_mute - enable/disable the CS4270 external mute
+ * @dai: the SOC DAI
+ * @mute: 0 = disable mute, 1 = enable mute
  *
  * This function toggles the mute bits in the MUTE register.  The CS4270's
  * mute capability is intended for external muting circuitry, so if the
@@ -493,276 +470,306 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute)
 	reg6 = snd_soc_read(codec, CS4270_MUTE);
 
 	if (mute)
-		reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
-			CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
+		reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
 	else
-		reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
-			  CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+		reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
 
 	return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
 
-#endif
-
-static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *);
-
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_DOUBLE_R("Master Playback Volume",
-		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1)
-};
-
-static const struct i2c_device_id cs4270_id[] = {
-	{"cs4270", 0},
-	{}
-};
-MODULE_DEVICE_TABLE(i2c, cs4270_id);
-
-static struct i2c_driver cs4270_i2c_driver = {
-	.driver = {
-		.name = "CS4270 I2C",
-		.owner = THIS_MODULE,
-	},
-	.id_table = cs4270_id,
-	.probe = cs4270_i2c_probe,
+		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1),
+	SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0),
+	SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0),
+	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
+	SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
+	SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
+	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
 };
 
 /*
- * Global variable to store socdev for i2c probe function.
+ * cs4270_codec - global variable to store codec for the ASoC probe function
  *
  * If struct i2c_driver had a private_data field, we wouldn't need to use
- * cs4270_socdec.  This is the only way to pass the socdev structure to
- * cs4270_i2c_probe().
- *
- * The real solution to cs4270_socdev is to create a mechanism
- * that maps I2C addresses to snd_soc_device structures.  Perhaps the
- * creation of the snd_soc_device object should be moved out of
- * cs4270_probe() and into cs4270_i2c_probe(), but that would make this
- * driver dependent on I2C.  The CS4270 supports "stand-alone" mode, whereby
- * the chip is *not* connected to the I2C bus, but is instead configured via
- * input pins.
+ * cs4270_codec.  This is the only way to pass the codec structure from
+ * cs4270_i2c_probe() to cs4270_probe().  Unfortunately, there is no good
+ * way to synchronize these two functions.  cs4270_i2c_probe() can be called
+ * multiple times before cs4270_probe() is called even once.  So for now, we
+ * also only allow cs4270_i2c_probe() to be run once.  That means that we do
+ * not support more than one cs4270 device in the system, at least for now.
  */
-static struct snd_soc_device *cs4270_socdev;
+static struct snd_soc_codec *cs4270_codec;
 
-/*
- * Initialize the I2C interface of the CS4270
- *
- * This function is called for whenever the I2C subsystem finds a device
- * at a particular address.
+static struct snd_soc_dai_ops cs4270_dai_ops = {
+	.hw_params	= cs4270_hw_params,
+	.set_sysclk	= cs4270_set_dai_sysclk,
+	.set_fmt	= cs4270_set_dai_fmt,
+	.digital_mute	= cs4270_mute,
+};
+
+struct snd_soc_dai cs4270_dai = {
+	.name = "cs4270",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = 0,
+		.formats = CS4270_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = 0,
+		.formats = CS4270_FORMATS,
+	},
+	.ops = &cs4270_dai_ops,
+};
+EXPORT_SYMBOL_GPL(cs4270_dai);
+
+/**
+ * cs4270_probe - ASoC probe function
+ * @pdev: platform device
  *
- * Note: snd_soc_new_pcms() must be called before this function can be called,
- * because of snd_ctl_add().
+ * This function is called when ASoC has all the pieces it needs to
+ * instantiate a sound driver.
  */
-static int cs4270_i2c_probe(struct i2c_client *i2c_client,
-	const struct i2c_device_id *id)
+static int cs4270_probe(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = cs4270_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int i;
-	int ret = 0;
-
-	/* Probing all possible addresses has one drawback: if there are
-	   multiple CS4270s on the bus, then you cannot specify which
-	   socdev is matched with which CS4270.  For now, we just reject
-	   this I2C device if the socdev already has one attached. */
-	if (codec->control_data)
-		return -ENODEV;
-
-	/* Note: codec_dai->codec is NULL here */
-
-	codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL);
-	if (!codec->reg_cache) {
-		printk(KERN_ERR "cs4270: could not allocate register cache\n");
-		ret = -ENOMEM;
-		goto error;
-	}
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = cs4270_codec;
+	int ret;
 
-	/* Verify that we have a CS4270 */
+	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
+	socdev->card->codec = codec;
 
-	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
+	/* Register PCMs */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to read I2C\n");
-		goto error;
-	}
-	/* The top four bits of the chip ID should be 1100. */
-	if ((ret & 0xF0) != 0xC0) {
-		/* The device at this address is not a CS4270 codec */
-		ret = -ENODEV;
-		goto error;
+		dev_err(codec->dev, "failed to create pcms\n");
+		return ret;
 	}
 
-	printk(KERN_INFO "cs4270: found device at I2C address %X\n",
-		i2c_client->addr);
-	printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF);
-
-	codec->control_data = i2c_client;
-	codec->read = cs4270_read_reg_cache;
-	codec->write = cs4270_i2c_write;
-	codec->reg_cache_size = CS4270_NUMREGS;
-
-	/* The I2C interface is set up, so pre-fill our register cache */
-
-	ret = cs4270_fill_cache(codec);
+	/* Add the non-DAPM controls */
+	ret = snd_soc_add_controls(codec, cs4270_snd_controls,
+				ARRAY_SIZE(cs4270_snd_controls));
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to fill register cache\n");
-		goto error;
+		dev_err(codec->dev, "failed to add controls\n");
+		goto error_free_pcms;
 	}
 
-	/* Add the non-DAPM controls */
-
-	for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) {
-		struct snd_kcontrol *kctrl =
-		snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
-
-		ret = snd_ctl_add(codec->card, kctrl);
-		if (ret < 0)
-			goto error;
+	/* And finally, register the socdev */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card\n");
+		goto error_free_pcms;
 	}
 
-	i2c_set_clientdata(i2c_client, codec);
-
 	return 0;
 
-error:
-	codec->control_data = NULL;
-
-	kfree(codec->reg_cache);
-	codec->reg_cache = NULL;
-	codec->reg_cache_size = 0;
+error_free_pcms:
+	snd_soc_free_pcms(socdev);
 
 	return ret;
 }
 
-#endif /* USE_I2C*/
+/**
+ * cs4270_remove - ASoC remove function
+ * @pdev: platform device
+ *
+ * This function is the counterpart to cs4270_probe().
+ */
+static int cs4270_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-struct snd_soc_dai cs4270_dai = {
-	.name = "CS4270",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = 0,
-		.formats = CS4270_FORMATS,
-	},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = 0,
-		.formats = CS4270_FORMATS,
-	},
+	snd_soc_free_pcms(socdev);
+
+	return 0;
 };
-EXPORT_SYMBOL_GPL(cs4270_dai);
 
-/*
- * ASoC probe function
+/**
+ * cs4270_i2c_probe - initialize the I2C interface of the CS4270
+ * @i2c_client: the I2C client object
+ * @id: the I2C device ID (ignored)
  *
- * This function is called when the machine driver calls
- * platform_device_add().
+ * This function is called whenever the I2C subsystem finds a device that
+ * matches the device ID given via a prior call to i2c_add_driver().
  */
-static int cs4270_probe(struct platform_device *pdev)
+static int cs4270_i2c_probe(struct i2c_client *i2c_client,
+	const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
-	int ret = 0;
+	struct cs4270_private *cs4270;
+	unsigned int reg;
+	int ret;
 
-	printk(KERN_INFO "CS4270 ALSA SoC Codec\n");
+	/* For now, we only support one cs4270 device in the system.  See the
+	 * comment for cs4270_codec.
+	 */
+	if (cs4270_codec) {
+		dev_err(&i2c_client->dev, "ignoring CS4270 at addr %X\n",
+		       i2c_client->addr);
+		dev_err(&i2c_client->dev, "only one per board allowed\n");
+		/* Should we return something other than ENODEV here? */
+		return -ENODEV;
+	}
+
+	/* Verify that we have a CS4270 */
+
+	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "failed to read i2c at addr %X\n",
+		       i2c_client->addr);
+		return ret;
+	}
+	/* The top four bits of the chip ID should be 1100. */
+	if ((ret & 0xF0) != 0xC0) {
+		dev_err(&i2c_client->dev, "device at addr %X is not a CS4270\n",
+		       i2c_client->addr);
+		return -ENODEV;
+	}
+
+	dev_info(&i2c_client->dev, "found device at i2c address %X\n",
+		i2c_client->addr);
+	dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF);
 
 	/* Allocate enough space for the snd_soc_codec structure
 	   and our private data together. */
-	codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) +
-			sizeof(struct cs4270_private), GFP_KERNEL);
-	if (!codec) {
-		printk(KERN_ERR "cs4270: Could not allocate codec structure\n");
+	cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL);
+	if (!cs4270) {
+		dev_err(&i2c_client->dev, "could not allocate codec\n");
 		return -ENOMEM;
 	}
+	codec = &cs4270->codec;
 
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
+	codec->dev = &i2c_client->dev;
 	codec->name = "CS4270";
 	codec->owner = THIS_MODULE;
 	codec->dai = &cs4270_dai;
 	codec->num_dai = 1;
-	codec->private_data = (void *) codec +
-		ALIGN(sizeof(struct snd_soc_codec), 4);
-
-	socdev->codec = codec;
+	codec->private_data = cs4270;
+	codec->control_data = i2c_client;
+	codec->read = cs4270_read_reg_cache;
+	codec->write = cs4270_i2c_write;
+	codec->reg_cache = cs4270->reg_cache;
+	codec->reg_cache_size = CS4270_NUMREGS;
 
-	/* Register PCMs */
+	/* The I2C interface is set up, so pre-fill our register cache */
 
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	ret = cs4270_fill_cache(codec);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to create PCMs\n");
+		dev_err(&i2c_client->dev, "failed to fill register cache\n");
 		goto error_free_codec;
 	}
 
-#ifdef USE_I2C
-	cs4270_socdev = socdev;
+	/* Disable auto-mute.  This feature appears to be buggy.  In some
+	 * situations, auto-mute will not deactivate when it should, so we want
+	 * this feature disabled by default.  An application (e.g. alsactl) can
+	 * re-enabled it by using the controls.
+	 */
 
-	ret = i2c_add_driver(&cs4270_i2c_driver);
-	if (ret) {
-		printk(KERN_ERR "cs4270: failed to attach driver");
-		goto error_free_pcms;
+	reg = cs4270_read_reg_cache(codec, CS4270_MUTE);
+	reg &= ~CS4270_MUTE_AUTO;
+	ret = cs4270_i2c_write(codec, CS4270_MUTE, reg);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "i2c write failed\n");
+		return ret;
 	}
 
-	/* Did we find a CS4270 on the I2C bus? */
-	if (codec->control_data) {
-		/* Initialize codec ops */
-		cs4270_dai.ops.hw_params = cs4270_hw_params;
-		cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
-		cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
-#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-		cs4270_dai.ops.digital_mute = cs4270_mute;
-#endif
-	} else
-		printk(KERN_INFO "cs4270: no I2C device found, "
-			"using stand-alone mode\n");
-#else
-	printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
-#endif
+	/* Disable automatic volume control.  The hardware enables, and it
+	 * causes volume change commands to be delayed, sometimes until after
+	 * playback has started.  An application (e.g. alsactl) can
+	 * re-enabled it by using the controls.
+	 */
 
-	ret = snd_soc_init_card(socdev);
+	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
+	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
+	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to register card\n");
-		goto error_del_driver;
+		dev_err(&i2c_client->dev, "i2c write failed\n");
+		return ret;
 	}
 
-	return 0;
+	/* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI
+	 * structure for each CS4270 device, but the machine driver needs to
+	 * have a pointer to the DAI structure, so for now it must be a global
+	 * variable.
+	 */
+	cs4270_dai.dev = &i2c_client->dev;
 
-error_del_driver:
-#ifdef USE_I2C
-	i2c_del_driver(&cs4270_i2c_driver);
+	/* Register the DAI.  If all the other ASoC driver have already
+	 * registered, then this will call our probe function, so
+	 * cs4270_codec needs to be ready.
+	 */
+	cs4270_codec = codec;
+	ret = snd_soc_register_dai(&cs4270_dai);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "failed to register DAIe\n");
+		goto error_free_codec;
+	}
 
-error_free_pcms:
-#endif
-	snd_soc_free_pcms(socdev);
+	i2c_set_clientdata(i2c_client, cs4270);
+
+	return 0;
 
 error_free_codec:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(cs4270);
+	cs4270_codec = NULL;
+	cs4270_dai.dev = NULL;
 
 	return ret;
 }
 
-static int cs4270_remove(struct platform_device *pdev)
+/**
+ * cs4270_i2c_remove - remove an I2C device
+ * @i2c_client: the I2C client object
+ *
+ * This function is the counterpart to cs4270_i2c_probe().
+ */
+static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-
-	snd_soc_free_pcms(socdev);
-
-#ifdef USE_I2C
-	i2c_del_driver(&cs4270_i2c_driver);
-#endif
+	struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client);
 
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(cs4270);
+	cs4270_codec = NULL;
+	cs4270_dai.dev = NULL;
 
 	return 0;
 }
 
 /*
+ * cs4270_id - I2C device IDs supported by this driver
+ */
+static struct i2c_device_id cs4270_id[] = {
+	{"cs4270", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, cs4270_id);
+
+/*
+ * cs4270_i2c_driver - I2C device identification
+ *
+ * This structure tells the I2C subsystem how to identify and support a
+ * given I2C device type.
+ */
+static struct i2c_driver cs4270_i2c_driver = {
+	.driver = {
+		.name = "cs4270",
+		.owner = THIS_MODULE,
+	},
+	.id_table = cs4270_id,
+	.probe = cs4270_i2c_probe,
+	.remove = cs4270_i2c_remove,
+};
+
+/*
  * ASoC codec device structure
  *
  * Assign this variable to the codec_dev field of the machine driver's
@@ -776,13 +783,15 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
 static int __init cs4270_init(void)
 {
-	return snd_soc_register_dai(&cs4270_dai);
+	pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
+
+	return i2c_add_driver(&cs4270_i2c_driver);
 }
 module_init(cs4270_init);
 
 static void __exit cs4270_exit(void)
 {
-	snd_soc_unregister_dai(&cs4270_dai);
+	i2c_del_driver(&cs4270_i2c_driver);
 }
 module_exit(cs4270_exit);
 
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index 9a3e67e..5cda9e6 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -67,11 +67,11 @@ static int pcm3008_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (!socdev->codec)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->card->codec)
 		return -ENOMEM;
 
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->name = "PCM3008";
@@ -139,7 +139,7 @@ gpio_err:
 card_err:
 	snd_soc_free_pcms(socdev);
 pcm_err:
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return ret;
 }
@@ -147,7 +147,7 @@ pcm_err:
 static int pcm3008_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct pcm3008_setup_data *setup = socdev->codec_data;
 
 	if (!codec)
@@ -155,7 +155,7 @@ static int pcm3008_soc_remove(struct platform_device *pdev)
 
 	pcm3008_gpio_free(setup);
 	snd_soc_free_pcms(socdev);
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index cac3736..87f606c7 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]),
 SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
 };
 
-/* add non dapm controls */
-static int ssm2602_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Output Mixer */
 static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
@@ -291,7 +276,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	u16 srate;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
@@ -336,7 +321,7 @@ static int ssm2602_startup(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
@@ -373,7 +358,7 @@ static int ssm2602_pcm_prepare(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	/* set active */
 	ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
 
@@ -385,7 +370,7 @@ static void ssm2602_shutdown(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	/* deactivate */
 	if (!codec->active)
@@ -521,6 +506,16 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 #define SSM2602_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 ssm2602_dai_ops = {
+	.startup	= ssm2602_startup,
+	.prepare	= ssm2602_pcm_prepare,
+	.hw_params	= ssm2602_hw_params,
+	.shutdown	= ssm2602_shutdown,
+	.digital_mute	= ssm2602_mute,
+	.set_sysclk	= ssm2602_set_dai_sysclk,
+	.set_fmt	= ssm2602_set_dai_fmt,
+};
+
 struct snd_soc_dai ssm2602_dai = {
 	.name = "SSM2602",
 	.playback = {
@@ -535,22 +530,14 @@ struct snd_soc_dai ssm2602_dai = {
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
 		.formats = SSM2602_FORMATS,},
-	.ops = {
-		.startup = ssm2602_startup,
-		.prepare = ssm2602_pcm_prepare,
-		.hw_params = ssm2602_hw_params,
-		.shutdown = ssm2602_shutdown,
-		.digital_mute = ssm2602_mute,
-		.set_sysclk = ssm2602_set_dai_sysclk,
-		.set_fmt = ssm2602_set_dai_fmt,
-	}
+	.ops = &ssm2602_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ssm2602_dai);
 
 static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -559,7 +546,7 @@ static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state)
 static int ssm2602_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -581,7 +568,7 @@ static int ssm2602_resume(struct platform_device *pdev)
  */
 static int ssm2602_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "SSM2602";
@@ -622,7 +609,8 @@ static int ssm2602_init(struct snd_soc_device *socdev)
 			APANA_ENABLE_MIC_BOOST);
 	ssm2602_write(codec, SSM2602_PWR, 0);
 
-	ssm2602_add_controls(codec);
+	snd_soc_add_controls(codec, ssm2602_snd_controls,
+				ARRAY_SIZE(ssm2602_snd_controls));
 	ssm2602_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -653,7 +641,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = ssm2602_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -747,7 +735,7 @@ static int ssm2602_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = ssm2602;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -768,7 +756,7 @@ static int ssm2602_probe(struct platform_device *pdev)
 static int ssm2602_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index cfdea00..c3f4afb 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
 	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
 };
 
-/* add non dapm controls */
-static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
-{
-
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&tlv320aic23_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-
-}
-
 /* PGA Mixer controls for Line and Mic switch */
 static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
@@ -423,7 +405,7 @@ static int tlv320aic23_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 iface_reg;
 	int ret;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
@@ -471,7 +453,7 @@ static int tlv320aic23_pcm_prepare(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* set active */
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
@@ -484,7 +466,7 @@ static void tlv320aic23_shutdown(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	/* deactivate */
@@ -598,6 +580,15 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 #define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops tlv320aic23_dai_ops = {
+	.prepare	= tlv320aic23_pcm_prepare,
+	.hw_params	= tlv320aic23_hw_params,
+	.shutdown	= tlv320aic23_shutdown,
+	.digital_mute	= tlv320aic23_mute,
+	.set_fmt	= tlv320aic23_set_dai_fmt,
+	.set_sysclk	= tlv320aic23_set_dai_sysclk,
+};
+
 struct snd_soc_dai tlv320aic23_dai = {
 	.name = "tlv320aic23",
 	.playback = {
@@ -612,14 +603,7 @@ struct snd_soc_dai tlv320aic23_dai = {
 		    .channels_max = 2,
 		    .rates = AIC23_RATES,
 		    .formats = AIC23_FORMATS,},
-	.ops = {
-		.prepare = tlv320aic23_pcm_prepare,
-		.hw_params = tlv320aic23_hw_params,
-		.shutdown = tlv320aic23_shutdown,
-		.digital_mute = tlv320aic23_mute,
-		.set_fmt = tlv320aic23_set_dai_fmt,
-		.set_sysclk = tlv320aic23_set_dai_sysclk,
-	}
+	.ops = &tlv320aic23_dai_ops,
 };
 EXPORT_SYMBOL_GPL(tlv320aic23_dai);
 
@@ -627,7 +611,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
 			       pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -638,7 +622,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
 static int tlv320aic23_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u16 reg;
 
@@ -660,7 +644,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
  */
 static int tlv320aic23_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 	u16 reg;
 
@@ -718,7 +702,8 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
 
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
 
-	tlv320aic23_add_controls(codec);
+	snd_soc_add_controls(codec, tlv320aic23_snd_controls,
+				ARRAY_SIZE(tlv320aic23_snd_controls));
 	tlv320aic23_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -746,7 +731,7 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c,
 				   const struct i2c_device_id *i2c_id)
 {
 	struct snd_soc_device *socdev = tlv320aic23_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
@@ -804,7 +789,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 	if (aic23 == NULL)
 		return -ENOMEM;
 	codec = &aic23->codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -823,7 +808,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 static int tlv320aic23_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	if (codec->control_data)
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 29f2f1a..3387d9e 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -130,7 +130,7 @@ static int aic26_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic26 *aic26 = codec->private_data;
 	int fsref, divisor, wlen, pval, jval, dval, qval;
 	u16 reg;
@@ -270,6 +270,13 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 #define AIC26_FORMATS	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
 			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
 
+static struct snd_soc_dai_ops aic26_dai_ops = {
+	.hw_params	= aic26_hw_params,
+	.digital_mute	= aic26_mute,
+	.set_sysclk	= aic26_set_sysclk,
+	.set_fmt	= aic26_set_fmt,
+};
+
 struct snd_soc_dai aic26_dai = {
 	.name = "tlv320aic26",
 	.playback = {
@@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = {
 		.rates = AIC26_RATES,
 		.formats = AIC26_FORMATS,
 	},
-	.ops = {
-		.hw_params = aic26_hw_params,
-		.digital_mute = aic26_mute,
-		.set_sysclk = aic26_set_sysclk,
-		.set_fmt = aic26_set_fmt,
-	},
+	.ops = &aic26_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic26_dai);
 
@@ -322,9 +324,8 @@ static int aic26_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
-	struct snd_kcontrol *kcontrol;
 	struct aic26 *aic26;
-	int i, ret, err;
+	int ret, err;
 
 	dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
 	dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
@@ -338,7 +339,7 @@ static int aic26_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 	codec = &aic26->codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 
 	dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
 		&pdev->dev, socdev->dev);
@@ -351,11 +352,9 @@ static int aic26_probe(struct platform_device *pdev)
 
 	/* register controls */
 	dev_dbg(&pdev->dev, "Registering controls\n");
-	for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
-		kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
-		err = snd_ctl_add(codec->card, kcontrol);
-		WARN_ON(err < 0);
-	}
+	err = snd_soc_add_controls(codec, aic26_snd_controls,
+			ARRAY_SIZE(aic26_snd_controls));
+	WARN_ON(err < 0);
 
 	/* CODEC is setup, we can register the card now */
 	dev_dbg(&pdev->dev, "Registering card\n");
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index aea0cb7..ab099f4 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -45,6 +45,7 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
+#include <sound/tlv.h>
 
 #include "tlv320aic3x.h"
 
@@ -250,56 +251,86 @@ static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
 };
 
+/*
+ * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0);
+/* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */
+static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0);
+/*
+ * Output stage volumes. From -78.3 to 0 dB. Muted below -78.3 dB.
+ * Step size is approximately 0.5 dB over most of the scale but increasing
+ * near the very low levels.
+ * Define dB scale so that it is mostly correct for range about -55 to 0 dB
+ * but having increasing dB difference below that (and where it doesn't count
+ * so much). This setting shows -50 dB (actual is -50.3 dB) for register
+ * value 100 and -58.5 dB (actual is -78.3 dB) for register value 117.
+ */
+static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1);
+
 static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	/* Output */
-	SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("PCM Playback Volume",
+			 LDAC_VOL, RDAC_VOL, 0, 0x7f, 1, dac_tlv),
 
-	SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("Line DAC Playback Volume",
+			 DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
 	SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
-	SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
-	SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
-		     0, 0x7f, 1),
-	SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
-		     0, 0x7f, 1),
-	SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
-		     LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
-		     LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
-
-	SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
-		     DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("LineL DAC Playback Volume",
+			 DACL1_2_LLOPM_VOL, DACR1_2_LLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("LineL Left PGA Bypass Playback Volume",
+		       PGAL_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("LineR Right PGA Bypass Playback Volume",
+		       PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("LineL Line2 Bypass Playback Volume",
+			 LINE2L_2_LLOPM_VOL, LINE2R_2_LLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("LineR Line2 Bypass Playback Volume",
+			 LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("Mono DAC Playback Volume",
+			 DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
-	SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL,
-		     PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL,
-		     LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1),
-
-	SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL,
-		     DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("Mono PGA Bypass Playback Volume",
+			 PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("Mono Line2 Bypass Playback Volume",
+			 LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("HP DAC Playback Volume",
+			 DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
-	SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
-		     PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
-	SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
-		     0, 0x7f, 1),
-	SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
-		     0, 0x7f, 1),
-	SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
-		     LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
-
-	SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL,
-		     DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("HP Right PGA Bypass Playback Volume",
+			 PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPL PGA Bypass Playback Volume",
+		       PGAL_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPR PGA Bypass Playback Volume",
+		       PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("HP Line2 Bypass Playback Volume",
+			 LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("HPCOM DAC Playback Volume",
+			 DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
 		     0x01, 0),
-	SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
-		     0, 0x7f, 1),
-	SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
-		     0, 0x7f, 1),
-	SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
-		     LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
+	SOC_SINGLE_TLV("HPLCOM PGA Bypass Playback Volume",
+		       PGAL_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPRCOM PGA Bypass Playback Volume",
+		       PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Playback Volume",
+			 LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL,
+			 0, 118, 1, output_stage_tlv),
 
 	/*
 	 * Note: enable Automatic input Gain Controller with care. It can
@@ -308,28 +339,13 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
 
 	/* Input */
-	SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0),
+	SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL,
+			 0, 119, 0, adc_tlv),
 	SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
 
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
 };
 
-/* add non dapm controls */
-static int aic3x_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&aic3x_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
 SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
@@ -746,7 +762,7 @@ static int aic3x_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
@@ -1072,6 +1088,13 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed);
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops aic3x_dai_ops = {
+	.hw_params	= aic3x_hw_params,
+	.digital_mute	= aic3x_mute,
+	.set_sysclk	= aic3x_set_dai_sysclk,
+	.set_fmt	= aic3x_set_dai_fmt,
+};
+
 struct snd_soc_dai aic3x_dai = {
 	.name = "tlv320aic3x",
 	.playback = {
@@ -1086,19 +1109,14 @@ struct snd_soc_dai aic3x_dai = {
 		.channels_max = 2,
 		.rates = AIC3X_RATES,
 		.formats = AIC3X_FORMATS,},
-	.ops = {
-		.hw_params = aic3x_hw_params,
-		.digital_mute = aic3x_mute,
-		.set_sysclk = aic3x_set_dai_sysclk,
-		.set_fmt = aic3x_set_dai_fmt,
-	}
+	.ops = &aic3x_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic3x_dai);
 
 static int aic3x_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1108,7 +1126,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state)
 static int aic3x_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u8 *cache = codec->reg_cache;
@@ -1131,7 +1149,7 @@ static int aic3x_resume(struct platform_device *pdev)
  */
 static int aic3x_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
@@ -1227,7 +1245,8 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
 	aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
 
-	aic3x_add_controls(codec);
+	snd_soc_add_controls(codec, aic3x_snd_controls,
+				ARRAY_SIZE(aic3x_snd_controls));
 	aic3x_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1261,7 +1280,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 			   const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = aic3x_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -1366,7 +1385,7 @@ static int aic3x_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = aic3x;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1392,7 +1411,7 @@ static int aic3x_probe(struct platform_device *pdev)
 static int aic3x_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* power down chip */
 	if (codec->control_data)
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index ea370a4..97738e2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -42,7 +42,7 @@
  */
 static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* this register not used		*/
-	0x93, /* REG_CODEC_MODE		(0x1)	*/
+	0x91, /* REG_CODEC_MODE		(0x1)	*/
 	0xc3, /* REG_OPTION		(0x2)	*/
 	0x00, /* REG_UNKNOWN		(0x3)	*/
 	0x00, /* REG_MICBIAS_CTL	(0x4)	*/
@@ -117,6 +117,13 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* REG_MISC_SET_2		(0x49)	*/
 };
 
+/* codec private data */
+struct twl4030_priv {
+	unsigned int bypass_state;
+	unsigned int codec_powered;
+	unsigned int codec_muted;
+};
+
 /*
  * read twl4030 register cache
  */
@@ -125,6 +132,9 @@ static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
 {
 	u8 *cache = codec->reg_cache;
 
+	if (reg >= TWL4030_CACHEREGNUM)
+		return -EIO;
+
 	return cache[reg];
 }
 
@@ -151,26 +161,22 @@ static int twl4030_write(struct snd_soc_codec *codec,
 	return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
 }
 
-static void twl4030_clear_codecpdz(struct snd_soc_codec *codec)
+static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 {
+	struct twl4030_priv *twl4030 = codec->private_data;
 	u8 mode;
 
-	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
-	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
-		mode & ~TWL4030_CODECPDZ);
-
-	/* REVISIT: this delay is present in TI sample drivers */
-	/* but there seems to be no TRM requirement for it     */
-	udelay(10);
-}
-
-static void twl4030_set_codecpdz(struct snd_soc_codec *codec)
-{
-	u8 mode;
+	if (enable == twl4030->codec_powered)
+		return;
 
 	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
-	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
-		mode | TWL4030_CODECPDZ);
+	if (enable)
+		mode |= TWL4030_CODECPDZ;
+	else
+		mode &= ~TWL4030_CODECPDZ;
+
+	twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+	twl4030->codec_powered = enable;
 
 	/* REVISIT: this delay is present in TI sample drivers */
 	/* but there seems to be no TRM requirement for it     */
@@ -182,7 +188,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 	int i;
 
 	/* clear CODECPDZ prior to setting register defaults */
-	twl4030_clear_codecpdz(codec);
+	twl4030_codec_enable(codec, 0);
 
 	/* set all audio section registers to reasonable defaults */
 	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
@@ -190,6 +196,122 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 }
 
+static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute)
+{
+	struct twl4030_priv *twl4030 = codec->private_data;
+	u8 reg_val;
+
+	if (mute == twl4030->codec_muted)
+		return;
+
+	if (mute) {
+		/* Bypass the reg_cache and mute the volumes
+		 * Headset mute is done in it's own event handler
+		 * Things to mute:  Earpiece, PreDrivL/R, CarkitL/R
+		 */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_EAR_GAIN),
+					TWL4030_REG_EAR_CTL);
+
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PREDL_GAIN),
+					TWL4030_REG_PREDL_CTL);
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PREDR_GAIN),
+					TWL4030_REG_PREDL_CTL);
+
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PRECKL_GAIN),
+					TWL4030_REG_PRECKL_CTL);
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PRECKL_GAIN),
+					TWL4030_REG_PRECKR_CTL);
+
+		/* Disable PLL */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
+		reg_val &= ~TWL4030_APLL_EN;
+		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
+	} else {
+		/* Restore the volumes
+		 * Headset mute is done in it's own event handler
+		 * Things to restore:  Earpiece, PreDrivL/R, CarkitL/R
+		 */
+		twl4030_write(codec, TWL4030_REG_EAR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL));
+
+		twl4030_write(codec, TWL4030_REG_PREDL_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL));
+		twl4030_write(codec, TWL4030_REG_PREDR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL));
+
+		twl4030_write(codec, TWL4030_REG_PRECKL_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL));
+		twl4030_write(codec, TWL4030_REG_PRECKR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL));
+
+		/* Enable PLL */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
+		reg_val |= TWL4030_APLL_EN;
+		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
+	}
+
+	twl4030->codec_muted = mute;
+}
+
+static void twl4030_power_up(struct snd_soc_codec *codec)
+{
+	struct twl4030_priv *twl4030 = codec->private_data;
+	u8 anamicl, regmisc1, byte;
+	int i = 0;
+
+	if (twl4030->codec_powered)
+		return;
+
+	/* set CODECPDZ to turn on codec */
+	twl4030_codec_enable(codec, 1);
+
+	/* initiate offset cancellation */
+	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	twl4030_write(codec, TWL4030_REG_ANAMICL,
+		anamicl | TWL4030_CNCL_OFFSET_START);
+
+	/* wait for offset cancellation to complete */
+	do {
+		/* this takes a little while, so don't slam i2c */
+		udelay(2000);
+		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+				    TWL4030_REG_ANAMICL);
+	} while ((i++ < 100) &&
+		 ((byte & TWL4030_CNCL_OFFSET_START) ==
+		  TWL4030_CNCL_OFFSET_START));
+
+	/* 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)
+{
+	/* power down */
+	twl4030_codec_enable(codec, 0);
+}
+
 /* Earpiece */
 static const char *twl4030_earpiece_texts[] =
 		{"Off", "DACL1", "DACL2", "DACR1"};
@@ -366,6 +488,41 @@ static const struct soc_enum twl4030_micpathtx2_enum =
 static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
 SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
 
+/* Analog bypass for AudioR1 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassr1_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR1_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioL1 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassl1_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL1_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioR2 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR2_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioL2 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
+
+/* Digital bypass gain, 0 mutes the bypass */
+static const unsigned int twl4030_dapm_dbypass_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1),
+	4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0),
+};
+
+/* Digital bypass left (TX1L -> RX2L) */
+static const struct snd_kcontrol_new twl4030_dapm_dbypassl_control =
+	SOC_DAPM_SINGLE_TLV("Volume",
+			TWL4030_REG_ATX2ARXPGA, 3, 7, 0,
+			twl4030_dapm_dbypass_tlv);
+
+/* Digital bypass right (TX1R -> RX2R) */
+static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control =
+	SOC_DAPM_SINGLE_TLV("Volume",
+			TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
+			twl4030_dapm_dbypass_tlv);
+
 static int micpath_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -420,6 +577,79 @@ static int handsfree_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int headsetl_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	unsigned char hs_gain, hs_pop;
+
+	/* Save the current volume */
+	hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
+	hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* Do the anti-pop/bias ramp enable according to the TRM */
+		hs_pop |= TWL4030_VMID_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		/* Is this needed? Can we just use whatever gain here? */
+		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET,
+				(hs_gain & (~0x0f)) | 0x0a);
+		hs_pop |= TWL4030_RAMP_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+
+		/* Restore the original volume */
+		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* Do the anti-pop/bias ramp disable according to the TRM */
+		hs_pop &= ~TWL4030_RAMP_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		/* Bypass the reg_cache to mute the headset */
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					hs_gain & (~0x0f),
+					TWL4030_REG_HS_GAIN_SET);
+		hs_pop &= ~TWL4030_VMID_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		break;
+	}
+	return 0;
+}
+
+static int bypass_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_mixer_control *m =
+		(struct soc_mixer_control *)w->kcontrols->private_value;
+	struct twl4030_priv *twl4030 = w->codec->private_data;
+	unsigned char reg;
+
+	reg = twl4030_read_reg_cache(w->codec, m->reg);
+
+	if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
+		/* Analog bypass */
+		if (reg & (1 << m->shift))
+			twl4030->bypass_state |=
+				(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+		else
+			twl4030->bypass_state &=
+				~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+	} else {
+		/* Digital bypass */
+		if (reg & (0x7 << m->shift))
+			twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+		else
+			twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+	}
+
+	if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(w->codec, 0);
+		else
+			twl4030_codec_mute(w->codec, 1);
+	}
+	return 0;
+}
+
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
@@ -614,6 +844,17 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
  */
 static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
 
+static const char *twl4030_rampdelay_texts[] = {
+	"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
+	"437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms",
+	"3495/2581/1748 ms"
+};
+
+static const struct soc_enum twl4030_rampdelay_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2,
+			ARRAY_SIZE(twl4030_rampdelay_texts),
+			twl4030_rampdelay_texts);
+
 static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 	/* Common playback gain controls */
 	SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
@@ -668,23 +909,9 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 
 	SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
 		0, 3, 5, 0, input_gain_tlv),
-};
-
-/* add non dapm controls */
-static int twl4030_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&twl4030_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
 
-	return 0;
-}
+	SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
+};
 
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	/* Left channel inputs */
@@ -714,13 +941,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 
 	/* DACs */
 	SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
-			TWL4030_REG_AVDAC_CTL, 0, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
-			TWL4030_REG_AVDAC_CTL, 1, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
-			TWL4030_REG_AVDAC_CTL, 2, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
-			TWL4030_REG_AVDAC_CTL, 3, 0),
+			SND_SOC_NOPM, 0, 0),
 
 	/* Analog PGAs */
 	SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
@@ -732,6 +959,37 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
 			0, 0, NULL, 0),
 
+	/* Analog bypasses */
+	SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr1_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl1_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr2_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl2_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+
+	/* Digital bypasses */
+	SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassl_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassr_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+
+	SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			3, 0, NULL, 0),
+
 	/* Output MUX controls */
 	/* Earpiece */
 	SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
@@ -742,8 +1000,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_predriver_control),
 	/* HeadsetL/R */
-	SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_hsol_control),
+	SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsol_control, headsetl_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_hsor_control),
 	/* CarkitL/R */
@@ -782,16 +1041,16 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
 		SND_SOC_DAPM_POST_REG),
 
-	/* Analog input muxes with power switch for the physical ADCL/R */
+	/* Analog input muxes with switch for the capture amplifiers */
 	SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
-		TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control),
+		TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
 	SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
-		TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control),
+		TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
 
-	SND_SOC_DAPM_PGA("Analog Left Amplifier",
-		TWL4030_REG_ANAMICL, 4, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("Analog Right Amplifier",
-		TWL4030_REG_ANAMICR, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC Physical Left",
+		TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
+	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),
@@ -801,13 +1060,19 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	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),
 	SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0),
+
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
-	{"ARXL1_APGA", NULL, "DAC Left1"},
-	{"ARXR1_APGA", NULL, "DAC Right1"},
-	{"ARXL2_APGA", NULL, "DAC Left2"},
-	{"ARXR2_APGA", NULL, "DAC Right2"},
+	{"Analog L1 Playback Mixer", NULL, "DAC Left1"},
+	{"Analog R1 Playback Mixer", NULL, "DAC Right1"},
+	{"Analog L2 Playback Mixer", NULL, "DAC Left2"},
+	{"Analog R2 Playback Mixer", NULL, "DAC Right2"},
+
+	{"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
+	{"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
+	{"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
+	{"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
 
 	/* Internal playback routings */
 	/* Earpiece */
@@ -865,23 +1130,23 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Analog Right Capture Route", "Sub mic", "SUBMIC"},
 	{"Analog Right Capture Route", "AUXR", "AUXR"},
 
-	{"Analog Left Amplifier", NULL, "Analog Left Capture Route"},
-	{"Analog Right Amplifier", NULL, "Analog Right Capture Route"},
+	{"ADC Physical Left", NULL, "Analog Left Capture Route"},
+	{"ADC Physical Right", NULL, "Analog Right Capture Route"},
 
 	{"Digimic0 Enable", NULL, "DIGIMIC0"},
 	{"Digimic1 Enable", NULL, "DIGIMIC1"},
 
 	/* TX1 Left capture path */
-	{"TX1 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX1 Capture Route", "Analog", "ADC Physical Left"},
 	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
 	/* TX1 Right capture path */
-	{"TX1 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX1 Capture Route", "Analog", "ADC Physical Right"},
 	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
 	/* TX2 Left capture path */
-	{"TX2 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX2 Capture Route", "Analog", "ADC Physical Left"},
 	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
 	/* TX2 Right capture path */
-	{"TX2 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX2 Capture Route", "Analog", "ADC Physical Right"},
 	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
 
 	{"ADC Virtual Left1", NULL, "TX1 Capture Route"},
@@ -889,6 +1154,24 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
 	{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
 
+	/* Analog bypass routes */
+	{"Right1 Analog Loopback", "Switch", "Analog Right Capture Route"},
+	{"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
+	{"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
+	{"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+
+	{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
+	{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
+	{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
+	{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+
+	/* Digital bypass routes */
+	{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
+	{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+
+	{"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
+	{"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+
 };
 
 static int twl4030_add_widgets(struct snd_soc_codec *codec)
@@ -902,82 +1185,28 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static void twl4030_power_up(struct snd_soc_codec *codec)
-{
-	u8 anamicl, regmisc1, byte, popn;
-	int i = 0;
-
-	/* set CODECPDZ to turn on codec */
-	twl4030_set_codecpdz(codec);
-
-	/* initiate offset cancellation */
-	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
-	twl4030_write(codec, TWL4030_REG_ANAMICL,
-		anamicl | TWL4030_CNCL_OFFSET_START);
-
-
-	/* wait for offset cancellation to complete */
-	do {
-		/* this takes a little while, so don't slam i2c */
-		udelay(2000);
-		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
-				    TWL4030_REG_ANAMICL);
-	} while ((i++ < 100) &&
-		 ((byte & TWL4030_CNCL_OFFSET_START) ==
-		  TWL4030_CNCL_OFFSET_START));
-
-	/* 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_clear_codecpdz(codec);
-	twl4030_set_codecpdz(codec);
-
-	/* program anti-pop with bias ramp delay */
-	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-	popn &= TWL4030_RAMP_DELAY;
-	popn |=	TWL4030_RAMP_DELAY_645MS;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-	popn |=	TWL4030_VMID_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* enable anti-pop ramp */
-	popn |= TWL4030_RAMP_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-}
-
-static void twl4030_power_down(struct snd_soc_codec *codec)
-{
-	u8 popn;
-
-	/* disable anti-pop ramp */
-	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-	popn &= ~TWL4030_RAMP_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* disable bias out */
-	popn &= ~TWL4030_VMID_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* power down */
-	twl4030_clear_codecpdz(codec);
-}
-
 static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
+	struct twl4030_priv *twl4030 = codec->private_data;
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		twl4030_power_up(codec);
+		twl4030_codec_mute(codec, 0);
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		/* TODO: develop a twl4030_prepare function */
+		twl4030_power_up(codec);
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(codec, 0);
+		else
+			twl4030_codec_mute(codec, 1);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		/* TODO: develop a twl4030_standby function */
-		twl4030_power_down(codec);
+		twl4030_power_up(codec);
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(codec, 0);
+		else
+			twl4030_codec_mute(codec, 1);
 		break;
 	case SND_SOC_BIAS_OFF:
 		twl4030_power_down(codec);
@@ -994,10 +1223,9 @@ static int twl4030_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u8 mode, old_mode, format, old_format;
 
-
 	/* bit rate */
 	old_mode = twl4030_read_reg_cache(codec,
 			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1039,8 +1267,9 @@ static int twl4030_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_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 
 	/* sample size */
@@ -1063,13 +1292,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	if (format != old_format) {
 
 		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_clear_codecpdz(codec);
+		twl4030_codec_enable(codec, 0);
 
 		/* change format */
 		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
 
 		/* set CODECPDZ afterwards */
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 	return 0;
 }
@@ -1139,13 +1368,13 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if (format != old_format) {
 
 		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_clear_codecpdz(codec);
+		twl4030_codec_enable(codec, 0);
 
 		/* change format */
 		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
 
 		/* set CODECPDZ afterwards */
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 
 	return 0;
@@ -1154,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 #define TWL4030_RATES	 (SNDRV_PCM_RATE_8000_48000)
 #define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops twl4030_dai_ops = {
+	.hw_params	= twl4030_hw_params,
+	.set_sysclk	= twl4030_set_dai_sysclk,
+	.set_fmt	= twl4030_set_dai_fmt,
+};
+
 struct snd_soc_dai twl4030_dai = {
 	.name = "twl4030",
 	.playback = {
@@ -1168,18 +1403,14 @@ struct snd_soc_dai twl4030_dai = {
 		.channels_max = 2,
 		.rates = TWL4030_RATES,
 		.formats = TWL4030_FORMATS,},
-	.ops = {
-		.hw_params = twl4030_hw_params,
-		.set_sysclk = twl4030_set_dai_sysclk,
-		.set_fmt = twl4030_set_dai_fmt,
-	}
+	.ops = &twl4030_dai_ops,
 };
 EXPORT_SYMBOL_GPL(twl4030_dai);
 
 static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1189,7 +1420,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
 static int twl4030_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	twl4030_set_bias_level(codec, codec->suspend_bias_level);
@@ -1203,7 +1434,7 @@ static int twl4030_resume(struct platform_device *pdev)
 
 static int twl4030_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	printk(KERN_INFO "TWL4030 Audio Codec init \n");
@@ -1233,7 +1464,8 @@ static int twl4030_init(struct snd_soc_device *socdev)
 	/* power on device */
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	twl4030_add_controls(codec);
+	snd_soc_add_controls(codec, twl4030_snd_controls,
+				ARRAY_SIZE(twl4030_snd_controls));
 	twl4030_add_widgets(codec);
 
 	ret = snd_soc_init_card(socdev);
@@ -1258,12 +1490,20 @@ static int twl4030_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
+	struct twl4030_priv *twl4030;
 
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL);
+	if (twl4030 == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+
+	codec->private_data = twl4030;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1277,11 +1517,13 @@ static int twl4030_probe(struct platform_device *pdev)
 static int twl4030_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	printk(KERN_INFO "TWL4030 Audio Codec remove\n");
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
+	kfree(codec->private_data);
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 442e5a8..33dbb14 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -170,6 +170,9 @@
 #define TWL4030_CLK256FS_EN		0x02
 #define TWL4030_AIF_EN			0x01
 
+/* EAR_CTL (0x21) */
+#define TWL4030_EAR_GAIN		0x30
+
 /* HS_GAIN_SET (0x23) Fields */
 
 #define TWL4030_HSR_GAIN		0x0C
@@ -198,6 +201,18 @@
 #define TWL4030_RAMP_DELAY_2581MS	0x1C
 #define TWL4030_RAMP_EN			0x02
 
+/* PREDL_CTL (0x25) */
+#define TWL4030_PREDL_GAIN		0x30
+
+/* PREDR_CTL (0x26) */
+#define TWL4030_PREDR_GAIN		0x30
+
+/* PRECKL_CTL (0x27) */
+#define TWL4030_PRECKL_GAIN		0x30
+
+/* PRECKR_CTL (0x28) */
+#define TWL4030_PRECKR_GAIN		0x30
+
 /* HFL_CTL (0x29, 0x2A) Fields */
 #define TWL4030_HF_CTL_HB_EN		0x04
 #define TWL4030_HF_CTL_LOOP_EN		0x08
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index a2c5064..ddefb8f 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -173,7 +173,7 @@ static int uda134x_startup(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 	struct snd_pcm_runtime *master_runtime;
 
@@ -206,7 +206,7 @@ static void uda134x_shutdown(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 
 	if (uda134x->master_substream == substream)
@@ -221,7 +221,7 @@ static int uda134x_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 	u8 hw_params;
 
@@ -431,38 +431,14 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
 SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
 };
 
-static int uda134x_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i, n;
-	const struct snd_kcontrol_new *ctrls;
-	struct uda134x_platform_data *pd = codec->control_data;
-
-	switch (pd->model) {
-	case UDA134X_UDA1340:
-	case UDA134X_UDA1344:
-		n = ARRAY_SIZE(uda1340_snd_controls);
-		ctrls = uda1340_snd_controls;
-		break;
-	case UDA134X_UDA1341:
-		n = ARRAY_SIZE(uda1341_snd_controls);
-		ctrls = uda1341_snd_controls;
-		break;
-	default:
-		printk(KERN_ERR "%s unkown codec type: %d",
-		       __func__, pd->model);
-		return -EINVAL;
-	}
-
-	for (i = 0; i < n; i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&ctrls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
+static struct snd_soc_dai_ops uda134x_dai_ops = {
+	.startup	= uda134x_startup,
+	.shutdown	= uda134x_shutdown,
+	.hw_params	= uda134x_hw_params,
+	.digital_mute	= uda134x_mute,
+	.set_sysclk	= uda134x_set_dai_sysclk,
+	.set_fmt	= uda134x_set_dai_fmt,
+};
 
 struct snd_soc_dai uda134x_dai = {
 	.name = "UDA134X",
@@ -483,14 +459,7 @@ struct snd_soc_dai uda134x_dai = {
 		.formats = UDA134X_FORMATS,
 	},
 	/* pcm operations */
-	.ops = {
-		.startup = uda134x_startup,
-		.shutdown = uda134x_shutdown,
-		.hw_params = uda134x_hw_params,
-		.digital_mute = uda134x_mute,
-		.set_sysclk = uda134x_set_dai_sysclk,
-		.set_fmt = uda134x_set_dai_fmt,
-	}
+	.ops = &uda134x_dai_ops,
 };
 EXPORT_SYMBOL(uda134x_dai);
 
@@ -525,11 +494,11 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return ret;
 
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 
 	uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
 	if (uda134x == NULL)
@@ -572,7 +541,22 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = uda134x_add_controls(codec);
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1344:
+		ret = snd_soc_add_controls(codec, uda1340_snd_controls,
+					ARRAY_SIZE(uda1340_snd_controls));
+	break;
+	case UDA134X_UDA1341:
+		ret = snd_soc_add_controls(codec, uda1341_snd_controls,
+					ARRAY_SIZE(uda1341_snd_controls));
+	break;
+	default:
+		printk(KERN_ERR "%s unkown codec type: %d",
+			__func__, pd->model);
+	return -EINVAL;
+	}
+
 	if (ret < 0) {
 		printk(KERN_ERR "UDA134X: failed to register controls\n");
 		goto pcm_err;
@@ -602,7 +586,7 @@ priv_err:
 static int uda134x_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -622,7 +606,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev,
 						pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -632,7 +616,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev,
 static int uda134x_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index e6bf084..5b21594 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -25,6 +25,7 @@
 #include <linux/ioctl.h>
 #include <linux/delay.h>
 #include <linux/i2c.h>
+#include <linux/workqueue.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/initval.h>
@@ -35,7 +36,8 @@
 
 #include "uda1380.h"
 
-#define UDA1380_VERSION "0.6"
+static struct work_struct uda1380_work;
+static struct snd_soc_codec *uda1380_codec;
 
 /*
  * uda1380 register cache
@@ -52,6 +54,8 @@ static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
 	0x0000, 0x8000, 0x0002, 0x0000,
 };
 
+static unsigned long uda1380_cache_dirty;
+
 /*
  * read uda1380 register cache
  */
@@ -73,8 +77,11 @@ static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
 	u16 reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
+
 	if (reg >= UDA1380_CACHEREGNUM)
 		return;
+	if ((reg >= 0x10) && (cache[reg] != value))
+		set_bit(reg - 0x10, &uda1380_cache_dirty);
 	cache[reg] = value;
 }
 
@@ -113,6 +120,8 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
 					(data[0]<<8) | data[1]);
 			return -EIO;
 		}
+		if (reg >= 0x10)
+			clear_bit(reg - 0x10, &uda1380_cache_dirty);
 		return 0;
 	} else
 		return -EIO;
@@ -120,6 +129,20 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
 
 #define uda1380_reset(c)	uda1380_write(c, UDA1380_RESET, 0)
 
+static void uda1380_flush_work(struct work_struct *work)
+{
+	int bit, reg;
+
+	for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
+		reg = 0x10 + bit;
+		pr_debug("uda1380: flush reg %x val %x:\n", reg,
+				uda1380_read_reg_cache(uda1380_codec, reg));
+		uda1380_write(uda1380_codec, reg,
+				uda1380_read_reg_cache(uda1380_codec, reg));
+		clear_bit(bit, &uda1380_cache_dirty);
+	}
+}
+
 /* declarations of ALSA reg_elem_REAL controls */
 static const char *uda1380_deemp[] = {
 	"None",
@@ -254,7 +277,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
 	SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0),	/* DA_POL_INV */
 	SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum),				/* SEL_NS */
 	SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum),		/* MIX_POS, MIX */
-	SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0),			/* SILENCE, force DAC output to silence */
 	SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0),		/* SDET_ON */
 	SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum),		/* SD_VALUE */
 	SOC_ENUM("Oversampling Input", uda1380_os_enum),			/* OS */
@@ -271,21 +293,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
 	SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
 };
 
-/* add non dapm controls */
-static int uda1380_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Input mux */
 static const struct snd_kcontrol_new uda1380_input_mux_control =
 	SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
@@ -371,7 +378,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -381,61 +388,107 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
 	iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
 
-	/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		iface |= R01_SFORI_I2S | R01_SFORO_I2S;
 		break;
 	case SND_SOC_DAIFMT_LSB:
-		iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
+		iface |= R01_SFORI_LSB16 | R01_SFORO_LSB16;
 		break;
 	case SND_SOC_DAIFMT_MSB:
-		iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+		iface |= R01_SFORI_MSB | R01_SFORO_MSB;
 	}
 
-	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
-		iface |= R01_SIM;
+	/* DATAI is slave only, so in single-link mode, this has to be slave */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
 
 	uda1380_write(codec, UDA1380_IFACE, iface);
 
 	return 0;
 }
 
-/*
- * Flush reg cache
- * We can only write the interpolator and decimator registers
- * when the DAI is being clocked by the CPU DAI. It's up to the
- * machine and cpu DAI driver to do this before we are called.
- */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
+static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int reg, reg_start, reg_end, clk;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		reg_start = UDA1380_MVOL;
-		reg_end = UDA1380_MIXER;
-	} else {
-		reg_start = UDA1380_DEC;
-		reg_end = UDA1380_AGC;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int iface;
+
+	/* set up DAI based upon fmt */
+	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+	iface &= ~R01_SFORI_MASK;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= R01_SFORI_I2S;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		iface |= R01_SFORI_LSB16;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		iface |= R01_SFORI_MSB;
 	}
 
-	/* FIXME disable DAC_CLK */
-	clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-	uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
+	/* DATAI is slave only, so this has to be slave */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
+
+	uda1380_write(codec, UDA1380_IFACE, iface);
+
+	return 0;
+}
+
+static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int iface;
+
+	/* set up DAI based upon fmt */
+	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+	iface &= ~(R01_SIM | R01_SFORO_MASK);
 
-	for (reg = reg_start; reg <= reg_end; reg++) {
-		pr_debug("uda1380: flush reg %x val %x:", reg,
-				uda1380_read_reg_cache(codec, reg));
-		uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= R01_SFORO_I2S;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		iface |= R01_SFORO_LSB16;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		iface |= R01_SFORO_MSB;
 	}
 
-	/* FIXME enable DAC_CLK */
-	uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+		iface |= R01_SIM;
 
+	uda1380_write(codec, UDA1380_IFACE, iface);
+
+	return 0;
+}
+
+static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
+		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;
+	int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		uda1380_write_reg_cache(codec, UDA1380_MIXER,
+					mixer & ~R14_SILENCE);
+		schedule_work(&uda1380_work);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		uda1380_write_reg_cache(codec, UDA1380_MIXER,
+					mixer | R14_SILENCE);
+		schedule_work(&uda1380_work);
+		break;
+	}
 	return 0;
 }
 
@@ -445,7 +498,7 @@ static int uda1380_pcm_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
 
 	/* set WSPLL power and divider if running from this clock */
@@ -484,7 +537,7 @@ static void uda1380_pcm_shutdown(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
 
 	/* shut down WSPLL power if running from this clock */
@@ -501,24 +554,6 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
 	uda1380_write(codec, UDA1380_CLK, clk);
 }
 
-static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
-{
-	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
-
-	/* FIXME: mute(codec,0) is called when the magician clock is already
-	 * set to WSPLL, but for some unknown reason writing to interpolator
-	 * registers works only when clocked by SYSCLK */
-	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-	uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
-	if (mute)
-		uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
-	else
-		uda1380_write(codec, UDA1380_DEEMP, mute_reg);
-	uda1380_write(codec, UDA1380_CLK, clk);
-	return 0;
-}
-
 static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
@@ -544,6 +579,27 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 		       SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		       SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops uda1380_dai_ops = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.trigger	= uda1380_trigger,
+	.set_fmt	= uda1380_set_dai_fmt_both,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_playback = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.trigger	= uda1380_trigger,
+	.set_fmt	= uda1380_set_dai_fmt_playback,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_capture = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.trigger	= uda1380_trigger,
+	.set_fmt	= uda1380_set_dai_fmt_capture,
+};
+
 struct snd_soc_dai uda1380_dai[] = {
 {
 	.name = "UDA1380",
@@ -559,13 +615,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.channels_max = 2,
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt,
-	},
+	.ops = &uda1380_dai_ops,
 },
 { /* playback only - dual interface */
 	.name = "UDA1380",
@@ -576,13 +626,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt,
-	},
+	.ops = &uda1380_dai_ops_playback,
 },
 { /* capture only - dual interface*/
 	.name = "UDA1380",
@@ -593,12 +637,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.set_fmt = uda1380_set_dai_fmt,
-	},
+	.ops = &uda1380_dai_ops_capture,
 },
 };
 EXPORT_SYMBOL_GPL(uda1380_dai);
@@ -606,7 +645,7 @@ EXPORT_SYMBOL_GPL(uda1380_dai);
 static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -615,7 +654,7 @@ static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
 static int uda1380_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -637,7 +676,7 @@ static int uda1380_resume(struct platform_device *pdev)
  */
 static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "UDA1380";
@@ -655,6 +694,9 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	codec->reg_cache_step = 1;
 	uda1380_reset(codec);
 
+	uda1380_codec = codec;
+	INIT_WORK(&uda1380_work, uda1380_flush_work);
+
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -675,7 +717,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	}
 
 	/* uda1380 init */
-	uda1380_add_controls(codec);
+	snd_soc_add_controls(codec, uda1380_snd_controls,
+				ARRAY_SIZE(uda1380_snd_controls));
 	uda1380_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -702,7 +745,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c,
 {
 	struct snd_soc_device *socdev = uda1380_socdev;
 	struct uda1380_setup_data *setup = socdev->codec_data;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -786,14 +829,12 @@ static int uda1380_probe(struct platform_device *pdev)
 	struct snd_soc_codec *codec;
 	int ret;
 
-	pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
-
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -817,7 +858,7 @@ static int uda1380_probe(struct platform_device *pdev)
 static int uda1380_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 35d9975..3b1d099 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -51,10 +51,17 @@ struct wm8350_output {
 	u16 mute;
 };
 
+struct wm8350_jack_data {
+	struct snd_soc_jack *jack;
+	int report;
+};
+
 struct wm8350_data {
 	struct snd_soc_codec codec;
 	struct wm8350_output out1;
 	struct wm8350_output out2;
+	struct wm8350_jack_data hpl;
+	struct wm8350_jack_data hpr;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 };
 
@@ -775,21 +782,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Beep", NULL, "IN3R PGA"},
 };
 
-static int wm8350_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8350_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static int wm8350_add_widgets(struct snd_soc_codec *codec)
 {
 	int ret;
@@ -1309,7 +1301,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -1318,7 +1310,7 @@ static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8350_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1328,6 +1320,95 @@ static int wm8350_resume(struct platform_device *pdev)
 	return 0;
 }
 
+static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
+{
+	struct wm8350_data *priv = data;
+	u16 reg;
+	int report;
+	int mask;
+	struct wm8350_jack_data *jack = NULL;
+
+	switch (irq) {
+	case WM8350_IRQ_CODEC_JCK_DET_L:
+		jack = &priv->hpl;
+		mask = WM8350_JACK_L_LVL;
+		break;
+
+	case WM8350_IRQ_CODEC_JCK_DET_R:
+		jack = &priv->hpr;
+		mask = WM8350_JACK_R_LVL;
+		break;
+
+	default:
+		BUG();
+	}
+
+	if (!jack->jack) {
+		dev_warn(wm8350->dev, "Jack interrupt called with no jack\n");
+		return;
+	}
+
+	/* Debounce */
+	msleep(200);
+
+	reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS);
+	if (reg & mask)
+		report = jack->report;
+	else
+		report = 0;
+
+	snd_soc_jack_report(jack->jack, report, jack->report);
+}
+
+/**
+ * wm8350_hp_jack_detect - Enable headphone jack detection.
+ *
+ * @codec:  WM8350 codec
+ * @which:  left or right jack detect signal
+ * @jack:   jack to report detection events on
+ * @report: value to report
+ *
+ * Enables the headphone jack detection of the WM8350.
+ */
+int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
+			  struct snd_soc_jack *jack, int report)
+{
+	struct wm8350_data *priv = codec->private_data;
+	struct wm8350 *wm8350 = codec->control_data;
+	int irq;
+	int ena;
+
+	switch (which) {
+	case WM8350_JDL:
+		priv->hpl.jack = jack;
+		priv->hpl.report = report;
+		irq = WM8350_IRQ_CODEC_JCK_DET_L;
+		ena = WM8350_JDL_ENA;
+		break;
+
+	case WM8350_JDR:
+		priv->hpr.jack = jack;
+		priv->hpr.report = report;
+		irq = WM8350_IRQ_CODEC_JCK_DET_R;
+		ena = WM8350_JDR_ENA;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+	wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
+
+	/* Sync status */
+	wm8350_hp_jack_handler(wm8350, irq, priv);
+
+	wm8350_unmask_irq(wm8350, irq);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
+
 static struct snd_soc_codec *wm8350_codec;
 
 static int wm8350_probe(struct platform_device *pdev)
@@ -1342,8 +1423,8 @@ static int wm8350_probe(struct platform_device *pdev)
 
 	BUG_ON(!wm8350_codec);
 
-	socdev->codec = wm8350_codec;
-	codec = socdev->codec;
+	socdev->card->codec = wm8350_codec;
+	codec = socdev->card->codec;
 	wm8350 = codec->control_data;
 	priv = codec->private_data;
 
@@ -1381,13 +1462,21 @@ static int wm8350_probe(struct platform_device *pdev)
 	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
 			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
 
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
+			    wm8350_hp_jack_handler, priv);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
+			    wm8350_hp_jack_handler, priv);
+
 	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;
 	}
 
-	wm8350_add_controls(codec);
+	snd_soc_add_controls(codec, wm8350_snd_controls,
+				ARRAY_SIZE(wm8350_snd_controls));
 	wm8350_add_widgets(codec);
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1409,10 +1498,23 @@ card_err:
 static int wm8350_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350_data *priv = codec->private_data;
 	int ret;
 
+	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
+			  WM8350_JDL_ENA | WM8350_JDR_ENA);
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+
+	priv->hpl.jack = NULL;
+	priv->hpr.jack = NULL;
+
 	/* cancel any work waiting to be queued. */
 	ret = cancel_delayed_work(&codec->delayed_work);
 
@@ -1436,6 +1538,16 @@ static int wm8350_remove(struct platform_device *pdev)
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8350_dai_ops = {
+	 .hw_params	= wm8350_pcm_hw_params,
+	 .digital_mute	= wm8350_mute,
+	 .trigger	= wm8350_pcm_trigger,
+	 .set_fmt	= wm8350_set_dai_fmt,
+	 .set_sysclk	= wm8350_set_dai_sysclk,
+	 .set_pll	= wm8350_set_fll,
+	 .set_clkdiv	= wm8350_set_clkdiv,
+};
+
 struct snd_soc_dai wm8350_dai = {
 	.name = "WM8350",
 	.playback = {
@@ -1452,15 +1564,7 @@ struct snd_soc_dai wm8350_dai = {
 		 .rates = WM8350_RATES,
 		 .formats = WM8350_FORMATS,
 	 },
-	.ops = {
-		 .hw_params = wm8350_pcm_hw_params,
-		 .digital_mute = wm8350_mute,
-		 .trigger = wm8350_pcm_trigger,
-		 .set_fmt = wm8350_set_dai_fmt,
-		 .set_sysclk = wm8350_set_dai_sysclk,
-		 .set_pll = wm8350_set_fll,
-		 .set_clkdiv = wm8350_set_clkdiv,
-	 },
+	.ops = &wm8350_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8350_dai);
 
@@ -1472,7 +1576,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8350 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
 
-static int wm8350_codec_probe(struct platform_device *pdev)
+static __devinit int wm8350_codec_probe(struct platform_device *pdev)
 {
 	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
 	struct wm8350_data *priv;
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index cc2887a..d11bd92 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -17,4 +17,12 @@
 extern struct snd_soc_dai wm8350_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8350;
 
+enum wm8350_jack {
+	WM8350_JDL = 1,
+	WM8350_JDR = 2,
+};
+
+int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
+			  struct snd_soc_jack *jack, int report);
+
 #endif
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
new file mode 100644
index 0000000..510efa6
--- /dev/null
+++ b/sound/soc/codecs/wm8400.c
@@ -0,0 +1,1582 @@
+/*
+ * wm8400.c  --  WM8400 ALSA Soc Audio driver
+ *
+ * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mfd/wm8400-audio.h>
+#include <linux/mfd/wm8400-private.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 "wm8400.h"
+
+/* Fake register for internal state */
+#define WM8400_INTDRIVBITS      (WM8400_REGISTER_COUNT + 1)
+#define WM8400_INMIXL_PWR			0
+#define WM8400_AINLMUX_PWR			1
+#define WM8400_INMIXR_PWR			2
+#define WM8400_AINRMUX_PWR			3
+
+static struct regulator_bulk_data power[] = {
+	{
+		.supply = "I2S1VDD",
+	},
+	{
+		.supply = "I2S2VDD",
+	},
+	{
+		.supply = "DCVDD",
+	},
+	{
+		.supply = "AVDD",
+	},
+	{
+		.supply = "FLLVDD",
+	},
+	{
+		.supply = "HPVDD",
+	},
+	{
+		.supply = "SPKVDD",
+	},
+};
+
+/* codec private data */
+struct wm8400_priv {
+	struct snd_soc_codec codec;
+	struct wm8400 *wm8400;
+	u16 fake_register;
+	unsigned int sysclk;
+	unsigned int pcmclk;
+	struct work_struct work;
+	int fll_in, fll_out;
+};
+
+static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
+				       unsigned int reg)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	if (reg == WM8400_INTDRIVBITS)
+		return wm8400->fake_register;
+	else
+		return wm8400_reg_read(wm8400->wm8400, reg);
+}
+
+/*
+ * write to the wm8400 register space
+ */
+static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	if (reg == WM8400_INTDRIVBITS) {
+		wm8400->fake_register = value;
+		return 0;
+	} else
+		return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value);
+}
+
+static void wm8400_codec_reset(struct snd_soc_codec *codec)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	wm8400_reset_codec_reg_cache(wm8400->wm8400);
+}
+
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+
+static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int reg = mc->reg;
+        int ret;
+        u16 val;
+
+        ret = snd_soc_put_volsw(kcontrol, ucontrol);
+        if (ret < 0)
+                return ret;
+
+        /* now hit the volume update bits (always bit 8) */
+        val = wm8400_read(codec, reg);
+        return wm8400_write(codec, reg, val | 0x0100);
+}
+
+#define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw, .put = wm8400_outpga_put_volsw_vu, \
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+static const char *wm8400_digital_sidetone[] =
+	{"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8400_left_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+		WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone);
+
+static const struct soc_enum wm8400_right_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+		WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone);
+
+static const char *wm8400_adcmode[] =
+	{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8400_right_adcmode_enum =
+SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode);
+
+static const struct snd_kcontrol_new wm8400_snd_controls[] = {
+/* INMIXL */
+SOC_SINGLE("LIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L12MNBST_SHIFT,
+	   1, 0),
+SOC_SINGLE("LIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L34MNBST_SHIFT,
+	   1, 0),
+/* INMIXR */
+SOC_SINGLE("RIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R12MNBST_SHIFT,
+	   1, 0),
+SOC_SINGLE("RIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R34MNBST_SHIFT,
+	   1, 0),
+
+/* LOMIX */
+SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LLI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LR12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LL12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* ROMIX */
+SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RRI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RL12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RR12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RLI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RLBROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RRBROVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* LOUT */
+WM8400_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8400_LEFT_OUTPUT_VOLUME,
+	WM8400_LOUTVOL_SHIFT, WM8400_LOUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOUT ZC", WM8400_LEFT_OUTPUT_VOLUME, WM8400_LOZC_SHIFT, 1, 0),
+
+/* ROUT */
+WM8400_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8400_RIGHT_OUTPUT_VOLUME,
+	WM8400_ROUTVOL_SHIFT, WM8400_ROUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROUT ZC", WM8400_RIGHT_OUTPUT_VOLUME, WM8400_ROZC_SHIFT, 1, 0),
+
+/* LOPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8400_LEFT_OPGA_VOLUME,
+	WM8400_LOPGAVOL_SHIFT, WM8400_LOPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOPGA ZC Switch", WM8400_LEFT_OPGA_VOLUME,
+	WM8400_LOPGAZC_SHIFT, 1, 0),
+
+/* ROPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8400_RIGHT_OPGA_VOLUME,
+	WM8400_ROPGAVOL_SHIFT, WM8400_ROPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROPGA ZC Switch", WM8400_RIGHT_OPGA_VOLUME,
+	WM8400_ROPGAZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LOPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LOATTN_SHIFT, 1, 0),
+SOC_SINGLE("RON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_RONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_ROPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_ROATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT3 Mute Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT3MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT3 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT3ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT4 Mute Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT4MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT4 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT4ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Mode Switch", WM8400_CLASSD1,
+	WM8400_CDMODE_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Output Attenuation Volume", WM8400_SPEAKER_VOLUME,
+	WM8400_SPKATTN_SHIFT, WM8400_SPKATTN_MASK, 0),
+SOC_SINGLE("Speaker DC Boost Volume", WM8400_CLASSD3,
+	WM8400_DCGAIN_SHIFT, 6, 0),
+SOC_SINGLE("Speaker AC Boost Volume", WM8400_CLASSD3,
+	WM8400_ACGAIN_SHIFT, 6, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+	WM8400_LEFT_DAC_DIGITAL_VOLUME, WM8400_DACL_VOL_SHIFT,
+	127, 0, out_dac_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+	WM8400_RIGHT_DAC_DIGITAL_VOLUME, WM8400_DACR_VOL_SHIFT,
+	127, 0, out_dac_tlv),
+
+SOC_ENUM("Left Digital Sidetone", wm8400_left_digital_sidetone_enum),
+SOC_ENUM("Right Digital Sidetone", wm8400_right_digital_sidetone_enum),
+
+SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+	WM8400_ADCL_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+	WM8400_ADCR_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+
+SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8400_ADC_CTRL,
+	WM8400_ADC_HPF_ENA_SHIFT, 1, 0),
+
+SOC_ENUM("ADC HPF Mode", wm8400_right_adcmode_enum),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+	WM8400_LEFT_ADC_DIGITAL_VOLUME,
+	WM8400_ADCL_VOL_SHIFT,
+	WM8400_ADCL_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+	WM8400_RIGHT_ADC_DIGITAL_VOLUME,
+	WM8400_ADCR_VOL_SHIFT,
+	WM8400_ADCR_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+	WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LIN12VOL_SHIFT,
+	WM8400_LIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN12 ZC Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN12 Mute Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+	WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LIN34VOL_SHIFT,
+	WM8400_LIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN34 ZC Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN34 Mute Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LI34MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+	WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RIN12VOL_SHIFT,
+	WM8400_RIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN12 ZC Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN12 Mute Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+	WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RIN34VOL_SHIFT,
+	WM8400_RIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN34 ZC Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RI34MUTE_SHIFT, 1, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8400_add_controls(struct snd_soc_codec *codec)
+{
+	return snd_soc_add_controls(codec, wm8400_snd_controls,
+				ARRAY_SIZE(wm8400_snd_controls));
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+static int inmixer_event (struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	u16 reg, fakepower;
+
+	reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2);
+	fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS);
+
+	if (fakepower & ((1 << WM8400_INMIXL_PWR) |
+		(1 << WM8400_AINLMUX_PWR))) {
+		reg |= WM8400_AINL_ENA;
+	} else {
+		reg &= ~WM8400_AINL_ENA;
+	}
+
+	if (fakepower & ((1 << WM8400_INMIXR_PWR) |
+		(1 << WM8400_AINRMUX_PWR))) {
+		reg |= WM8400_AINR_ENA;
+	} else {
+		reg &= ~WM8400_AINL_ENA;
+	}
+	wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
+
+	return 0;
+}
+
+static int outmixer_event (struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol * kcontrol, int event)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 reg_shift = mc->shift;
+	int ret = 0;
+	u16 reg;
+
+	switch (reg_shift) {
+	case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) :
+		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1);
+		if (reg & WM8400_LDLO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 1 LDLO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8):
+		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2);
+		if (reg & WM8400_RDRO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 2 RDRO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8):
+		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		if (reg & WM8400_LDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer LDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8):
+		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		if (reg & WM8400_RDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer RDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0,7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_lin12_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN1 Switch", WM8400_INPUT_MIXER2, WM8400_LMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN2 Switch", WM8400_INPUT_MIXER2, WM8400_LMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_lin34_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN3 Switch", WM8400_INPUT_MIXER2, WM8400_LMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN4 Switch", WM8400_INPUT_MIXER2, WM8400_LMP4_SHIFT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_rin12_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN1 Switch", WM8400_INPUT_MIXER2, WM8400_RMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN2 Switch", WM8400_INPUT_MIXER2, WM8400_RMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_rin34_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN3 Switch", WM8400_INPUT_MIXER2, WM8400_RMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN4 Switch", WM8400_INPUT_MIXER2, WM8400_RMP4_SHIFT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8400_dapm_inmixl_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8400_INPUT_MIXER3,
+	WM8400_LDBVOL_SHIFT, WM8400_LDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8400_INPUT_MIXER5, WM8400_LI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("LINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+		1, 0),
+SOC_DAPM_SINGLE("LINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+		1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8400_dapm_inmixr_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8400_INPUT_MIXER4,
+	WM8400_RDBVOL_SHIFT, WM8400_RDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8400_INPUT_MIXER6, WM8400_RI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("RINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+	1, 0),
+SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+	1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8400_ainlmux[] =
+	{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8400_ainlmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT,
+	ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8400_ainrmux[] =
+	{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8400_ainrmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT,
+	ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = {
+SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT,
+			WM8400_LR4BVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT,
+			WM8400_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = {
+SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LRBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LLBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LRI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LLI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LR12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LL12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LDLO_SHIFT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8400_dapm_romix_controls[] = {
+SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RLBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RRBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RLI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RRI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RL12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RR12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RDRO_SHIFT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lonmix_controls[] = {
+SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LLOPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LROPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8400_LINE_MIXER1,
+	WM8400_LOPLON_SHIFT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lopmix_controls[] = {
+SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER1,
+	WM8400_LR12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER1,
+	WM8400_LL12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LLOPGALOP_SHIFT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ronmix_controls[] = {
+SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RROPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RLOPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8400_LINE_MIXER2,
+	WM8400_ROPRON_SHIFT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ropmix_controls[] = {
+SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER2,
+	WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER2,
+	WM8400_RR12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RROPGAROP_SHIFT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out3mix_controls[] = {
+SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+	WM8400_LI4O3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8400_OUT3_4_MIXER,
+	WM8400_LPGAO3_SHIFT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out4mix_controls[] = {
+SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8400_OUT3_4_MIXER,
+	WM8400_RPGAO4_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+	WM8400_RI4O4_SHIFT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8400_dapm_spkmix_controls[] = {
+SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LI2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LB2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LOPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+	WM8400_ROPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RI2SPK_SHIFT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8400_dapm_widgets[] = {
+/* Input Side */
+/* Input Lines */
+SND_SOC_DAPM_INPUT("LIN1"),
+SND_SOC_DAPM_INPUT("LIN2"),
+SND_SOC_DAPM_INPUT("LIN3"),
+SND_SOC_DAPM_INPUT("LIN4/RXN"),
+SND_SOC_DAPM_INPUT("RIN3"),
+SND_SOC_DAPM_INPUT("RIN4/RXP"),
+SND_SOC_DAPM_INPUT("RIN1"),
+SND_SOC_DAPM_INPUT("RIN2"),
+SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+/* DACs */
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8400_POWER_MANAGEMENT_2,
+	WM8400_ADCL_ENA_SHIFT, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8400_POWER_MANAGEMENT_2,
+	WM8400_ADCR_ENA_SHIFT, 0),
+
+/* Input PGAs */
+SND_SOC_DAPM_MIXER("LIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_LIN12_ENA_SHIFT,
+		   0, &wm8400_dapm_lin12_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lin12_pga_controls)),
+SND_SOC_DAPM_MIXER("LIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_LIN34_ENA_SHIFT,
+		   0, &wm8400_dapm_lin34_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lin34_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_RIN12_ENA_SHIFT,
+		   0, &wm8400_dapm_rin12_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_rin12_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_RIN34_ENA_SHIFT,
+		   0, &wm8400_dapm_rin34_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)),
+
+/* INMIXL */
+SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0,
+	&wm8400_dapm_inmixl_controls[0],
+	ARRAY_SIZE(wm8400_dapm_inmixl_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINLMUX */
+SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0,
+	&wm8400_dapm_ainlmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* INMIXR */
+SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0,
+	&wm8400_dapm_inmixr_controls[0],
+	ARRAY_SIZE(wm8400_dapm_inmixr_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINRMUX */
+SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0,
+	&wm8400_dapm_ainrmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* Output Side */
+/* DACs */
+SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8400_POWER_MANAGEMENT_3,
+	WM8400_DACL_ENA_SHIFT, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8400_POWER_MANAGEMENT_3,
+	WM8400_DACR_ENA_SHIFT, 0),
+
+/* LOMIX */
+SND_SOC_DAPM_MIXER_E("LOMIX", WM8400_POWER_MANAGEMENT_3,
+		     WM8400_LOMIX_ENA_SHIFT,
+		     0, &wm8400_dapm_lomix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_lomix_controls),
+		     outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LONMIX */
+SND_SOC_DAPM_MIXER("LONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LON_ENA_SHIFT,
+		   0, &wm8400_dapm_lonmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lonmix_controls)),
+
+/* LOPMIX */
+SND_SOC_DAPM_MIXER("LOPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LOP_ENA_SHIFT,
+		   0, &wm8400_dapm_lopmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lopmix_controls)),
+
+/* OUT3MIX */
+SND_SOC_DAPM_MIXER("OUT3MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT3_ENA_SHIFT,
+		   0, &wm8400_dapm_out3mix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_out3mix_controls)),
+
+/* SPKMIX */
+SND_SOC_DAPM_MIXER_E("SPKMIX", WM8400_POWER_MANAGEMENT_1, WM8400_SPK_ENA_SHIFT,
+		     0, &wm8400_dapm_spkmix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_spkmix_controls), outmixer_event,
+		     SND_SOC_DAPM_PRE_REG),
+
+/* OUT4MIX */
+SND_SOC_DAPM_MIXER("OUT4MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT4_ENA_SHIFT,
+	0, &wm8400_dapm_out4mix_controls[0],
+	ARRAY_SIZE(wm8400_dapm_out4mix_controls)),
+
+/* ROPMIX */
+SND_SOC_DAPM_MIXER("ROPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_ROP_ENA_SHIFT,
+		   0, &wm8400_dapm_ropmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_ropmix_controls)),
+
+/* RONMIX */
+SND_SOC_DAPM_MIXER("RONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_RON_ENA_SHIFT,
+		   0, &wm8400_dapm_ronmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_ronmix_controls)),
+
+/* ROMIX */
+SND_SOC_DAPM_MIXER_E("ROMIX", WM8400_POWER_MANAGEMENT_3,
+		     WM8400_ROMIX_ENA_SHIFT,
+		     0, &wm8400_dapm_romix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_romix_controls),
+		     outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LOUT PGA */
+SND_SOC_DAPM_PGA("LOUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_LOUT_ENA_SHIFT,
+		 0, NULL, 0),
+
+/* ROUT PGA */
+SND_SOC_DAPM_PGA("ROUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_ROUT_ENA_SHIFT,
+		 0, NULL, 0),
+
+/* LOPGA */
+SND_SOC_DAPM_PGA("LOPGA", WM8400_POWER_MANAGEMENT_3, WM8400_LOPGA_ENA_SHIFT, 0,
+	NULL, 0),
+
+/* ROPGA */
+SND_SOC_DAPM_PGA("ROPGA", WM8400_POWER_MANAGEMENT_3, WM8400_ROPGA_ENA_SHIFT, 0,
+	NULL, 0),
+
+/* MICBIAS */
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8400_POWER_MANAGEMENT_1,
+	WM8400_MIC1BIAS_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Make DACs turn on when playing even if not mixed into any outputs */
+	{"Internal DAC Sink", NULL, "Left DAC"},
+	{"Internal DAC Sink", NULL, "Right DAC"},
+
+	/* Make ADCs turn on when recording
+	 * even if not mixed from any inputs */
+	{"Left ADC", NULL, "Internal ADC Source"},
+	{"Right ADC", NULL, "Internal ADC Source"},
+
+	/* Input Side */
+	/* LIN12 PGA */
+	{"LIN12 PGA", "LIN1 Switch", "LIN1"},
+	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
+	/* LIN34 PGA */
+	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
+	{"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
+	/* INMIXL */
+	{"INMIXL", "Record Left Volume", "LOMIX"},
+	{"INMIXL", "LIN2 Volume", "LIN2"},
+	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+	/* AILNMUX */
+	{"AILNMUX", "INMIXL Mix", "INMIXL"},
+	{"AILNMUX", "DIFFINL Mix", "LIN12 PGA"},
+	{"AILNMUX", "DIFFINL Mix", "LIN34 PGA"},
+	{"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Left ADC", NULL, "AILNMUX"},
+
+	/* RIN12 PGA */
+	{"RIN12 PGA", "RIN1 Switch", "RIN1"},
+	{"RIN12 PGA", "RIN2 Switch", "RIN2"},
+	/* RIN34 PGA */
+	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
+	{"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
+	/* INMIXL */
+	{"INMIXR", "Record Right Volume", "ROMIX"},
+	{"INMIXR", "RIN2 Volume", "RIN2"},
+	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+	/* AIRNMUX */
+	{"AIRNMUX", "INMIXR Mix", "INMIXR"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"},
+	{"AIRNMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Right ADC", NULL, "AIRNMUX"},
+
+	/* LOMIX */
+	{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+	{"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+	{"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"LOMIX", "LOMIX Right ADC Bypass Switch", "AIRNMUX"},
+	{"LOMIX", "LOMIX Left ADC Bypass Switch", "AILNMUX"},
+	{"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+	/* ROMIX */
+	{"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+	{"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+	{"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"ROMIX", "ROMIX Right ADC Bypass Switch", "AIRNMUX"},
+	{"ROMIX", "ROMIX Left ADC Bypass Switch", "AILNMUX"},
+	{"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+	/* SPKMIX */
+	{"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+	{"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+	{"SPKMIX", "SPKMIX LADC Bypass Switch", "AILNMUX"},
+	{"SPKMIX", "SPKMIX RADC Bypass Switch", "AIRNMUX"},
+	{"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+	{"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+	{"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+	{"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+
+	/* LONMIX */
+	{"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+	/* LOPMIX */
+	{"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+	/* OUT3MIX */
+	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
+	{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+	/* OUT4MIX */
+	{"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+	{"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"},
+
+	/* RONMIX */
+	{"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+	/* ROPMIX */
+	{"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+	/* Out Mixer PGAs */
+	{"LOPGA", NULL, "LOMIX"},
+	{"ROPGA", NULL, "ROMIX"},
+
+	{"LOUT PGA", NULL, "LOMIX"},
+	{"ROUT PGA", NULL, "ROMIX"},
+
+	/* Output Pins */
+	{"LON", NULL, "LONMIX"},
+	{"LOP", NULL, "LOPMIX"},
+	{"OUT3", NULL, "OUT3MIX"},
+	{"LOUT", NULL, "LOUT PGA"},
+	{"SPKN", NULL, "SPKMIX"},
+	{"ROUT", NULL, "ROUT PGA"},
+	{"OUT4", NULL, "OUT4MIX"},
+	{"ROP", NULL, "ROPMIX"},
+	{"RON", NULL, "RONMIX"},
+};
+
+static int wm8400_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets,
+				  ARRAY_SIZE(wm8400_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+/*
+ * Clock after FLL and dividers
+ */
+static int wm8400_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 wm8400_priv *wm8400 = codec->private_data;
+
+	wm8400->sysclk = freq;
+	return 0;
+}
+
+struct fll_factors {
+	u16 n;
+	u16 k;
+	u16 outdiv;
+	u16 fratio;
+	u16 freq_ref;
+};
+
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
+		       unsigned int Fref, unsigned int Fout)
+{
+	u64 Kpart;
+	unsigned int K, Nmod, target;
+
+	factors->outdiv = 2;
+	while (Fout * factors->outdiv <  90000000 ||
+	       Fout * factors->outdiv > 100000000) {
+		factors->outdiv *= 2;
+		if (factors->outdiv > 32) {
+			dev_err(wm8400->wm8400->dev,
+				"Unsupported FLL output frequency %dHz\n",
+				Fout);
+			return -EINVAL;
+		}
+	}
+	target = Fout * factors->outdiv;
+	factors->outdiv = factors->outdiv >> 2;
+
+	if (Fref < 48000)
+		factors->freq_ref = 1;
+	else
+		factors->freq_ref = 0;
+
+	if (Fref < 1000000)
+		factors->fratio = 9;
+	else
+		factors->fratio = 0;
+
+	/* Ensure we have a fractional part */
+	do {
+		if (Fref < 1000000)
+			factors->fratio--;
+		else
+			factors->fratio++;
+
+		if (factors->fratio < 1 || factors->fratio > 8) {
+			dev_err(wm8400->wm8400->dev,
+				"Unable to calculate FRATIO\n");
+			return -EINVAL;
+		}
+
+		factors->n = target / (Fref * factors->fratio);
+		Nmod = target % (Fref * factors->fratio);
+	} while (Nmod == 0);
+
+	/* Calculate fractional part - scale up so we can round. */
+	Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, (Fref * factors->fratio));
+
+	K = Kpart & 0xFFFFFFFF;
+
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	factors->k = K / 10;
+
+	dev_dbg(wm8400->wm8400->dev,
+		"FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
+		Fref, Fout,
+		factors->n, factors->k, factors->fratio, factors->outdiv);
+
+	return 0;
+}
+
+static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+			      unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8400_priv *wm8400 = codec->private_data;
+	struct fll_factors factors;
+	int ret;
+	u16 reg;
+
+	if (freq_in == wm8400->fll_in && freq_out == wm8400->fll_out)
+		return 0;
+
+	if (freq_out != 0) {
+		ret = fll_factors(wm8400, &factors, freq_in, freq_out);
+		if (ret != 0)
+			return ret;
+	}
+
+	wm8400->fll_out = freq_out;
+	wm8400->fll_in = freq_in;
+
+	/* We *must* disable the FLL before any changes */
+	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2);
+	reg &= ~WM8400_FLL_ENA;
+	wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg);
+
+	reg = wm8400_read(codec, WM8400_FLL_CONTROL_1);
+	reg &= ~WM8400_FLL_OSC_ENA;
+	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+
+	if (freq_out == 0)
+		return 0;
+
+	reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK);
+	reg |= WM8400_FLL_FRAC | factors.fratio;
+	reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT;
+	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+
+	wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k);
+	wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n);
+
+	reg = wm8400_read(codec, WM8400_FLL_CONTROL_4);
+	reg &= WM8400_FLL_OUTDIV_MASK;
+	reg |= factors.outdiv;
+	wm8400_write(codec, WM8400_FLL_CONTROL_4, reg);
+
+	return 0;
+}
+
+/*
+ * Sets ADC and Voice DAC format.
+ */
+static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 audio1, audio3;
+
+	audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+	audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		audio3 &= ~WM8400_AIF_MSTR1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		audio3 |= WM8400_AIF_MSTR1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	audio1 &= ~WM8400_AIF_FMT_MASK;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		audio1 |= WM8400_AIF_FMT_I2S;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		audio1 |= WM8400_AIF_FMT_RIGHTJ;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		audio1 |= WM8400_AIF_FMT_LEFTJ;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		audio1 |= WM8400_AIF_FMT_DSP;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		audio1 |= WM8400_AIF_FMT_DSP | WM8400_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
+	return 0;
+}
+
+static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+		int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	switch (div_id) {
+	case WM8400_MCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_MCLK_DIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_DACCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_DAC_CLKDIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_ADCCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_ADC_CLKDIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_BCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_1) &
+			~WM8400_BCLK_DIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_1, reg | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8400_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;
+	u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+
+	audio1 &= ~WM8400_AIF_WL_MASK;
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		audio1 |= WM8400_AIF_WL_20BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		audio1 |= WM8400_AIF_WL_24BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		audio1 |= WM8400_AIF_WL_32BITS;
+		break;
+	}
+
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	return 0;
+}
+
+static int wm8400_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
+
+	if (mute)
+		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+	else
+		wm8400_write(codec, WM8400_DAC_CTRL, val);
+
+	return 0;
+}
+
+/* TODO: set bias for best performance at standby */
+static int wm8400_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+	u16 val;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VMID=2*50k */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+			~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(power),
+						    &power[0]);
+			if (ret != 0) {
+				dev_err(wm8400->wm8400->dev,
+					"Failed to enable regulators: %d\n",
+					ret);
+				return ret;
+			}
+
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+				     WM8400_CODEC_ENA | WM8400_SYSCLK_ENA);
+
+			/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+				     WM8400_BUFDCOPEN | WM8400_POBCTRL);
+
+			msleep(50);
+
+			/* Enable VREF & VMID at 2x50k */
+			val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+			val |= 0x2 | WM8400_VREF_ENA;
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+			/* Enable BUFIOEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+				     WM8400_BUFDCOPEN | WM8400_POBCTRL |
+				     WM8400_BUFIOEN);
+
+			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
+		}
+
+		/* VMID=2*300k */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+			~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* Enable POBCTRL and SOFT_ST */
+		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			WM8400_POBCTRL | WM8400_BUFIOEN);
+
+		/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			WM8400_BUFDCOPEN | WM8400_POBCTRL |
+			WM8400_BUFIOEN);
+
+		/* mute DAC */
+		val = wm8400_read(codec, WM8400_DAC_CTRL);
+		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+
+		/* Enable any disabled outputs */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+		val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
+			WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+			WM8400_ROUT_ENA;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		/* Disable VMID */
+		val &= ~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		msleep(300);
+
+		/* Enable all output discharge bits */
+		wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+			WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
+			WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
+			WM8400_DIS_ROUT);
+
+		/* Disable VREF */
+		val &= ~WM8400_VREF_ENA;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8400_write(codec, WM8400_ANTIPOP2, 0x0);
+
+		ret = regulator_bulk_disable(ARRAY_SIZE(power),
+					     &power[0]);
+		if (ret != 0)
+			return ret;
+
+		break;
+	}
+
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8400_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8400_dai_ops = {
+	.hw_params = wm8400_hw_params,
+	.digital_mute = wm8400_mute,
+	.set_fmt = wm8400_set_dai_fmt,
+	.set_clkdiv = wm8400_set_dai_clkdiv,
+	.set_sysclk = wm8400_set_dai_sysclk,
+	.set_pll = wm8400_set_dai_pll,
+};
+
+/*
+ * The WM8400 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+struct snd_soc_dai wm8400_dai = {
+/* ADC/DAC on primary */
+	.name = "WM8400 ADC/DAC Primary",
+	.id = 1,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8400_RATES,
+		.formats = WM8400_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8400_RATES,
+		.formats = WM8400_FORMATS,
+	},
+	.ops = &wm8400_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm8400_dai);
+
+static int wm8400_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;
+
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int wm8400_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+static struct snd_soc_codec *wm8400_codec;
+
+static int wm8400_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret;
+
+	if (!wm8400_codec) {
+		dev_err(&pdev->dev, "wm8400 not yet discovered\n");
+		return -ENODEV;
+	}
+	codec = wm8400_codec;
+
+	socdev->card->codec = 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");
+		goto pcm_err;
+	}
+
+	wm8400_add_controls(codec);
+	wm8400_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	return ret;
+}
+
+/* power down chip */
+static int wm8400_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_wm8400 = {
+	.probe =	wm8400_probe,
+	.remove =	wm8400_remove,
+	.suspend =	wm8400_suspend,
+	.resume =	wm8400_resume,
+};
+
+static void wm8400_probe_deferred(struct work_struct *work)
+{
+	struct wm8400_priv *priv = container_of(work, struct wm8400_priv,
+						work);
+	struct snd_soc_codec *codec = &priv->codec;
+	int ret;
+
+	/* charge output caps */
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* We're done, tell the subsystem. */
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(priv->wm8400->dev,
+			"Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8400_dai);
+	if (ret != 0) {
+		dev_err(priv->wm8400->dev,
+			"Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8400_codec_probe(struct platform_device *dev)
+{
+	struct wm8400_priv *priv;
+	int ret;
+	u16 reg;
+	struct snd_soc_codec *codec;
+
+	priv = kzalloc(sizeof(struct wm8400_priv), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	codec = &priv->codec;
+	codec->private_data = priv;
+	codec->control_data = dev->dev.driver_data;
+	priv->wm8400 = dev->dev.driver_data;
+
+	ret = regulator_bulk_get(priv->wm8400->dev,
+				 ARRAY_SIZE(power), &power[0]);
+	if (ret != 0) {
+		dev_err(&dev->dev, "Failed to get regulators: %d\n", ret);
+	        goto err;
+	}
+
+	codec->dev = &dev->dev;
+	wm8400_dai.dev = &dev->dev;
+
+	codec->name = "WM8400";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8400_read;
+	codec->write = wm8400_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8400_set_bias_level;
+	codec->dai = &wm8400_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8400_REGISTER_COUNT;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	INIT_WORK(&priv->work, wm8400_probe_deferred);
+
+	wm8400_codec_reset(codec);
+
+	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+	wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
+
+	/* Latch volume update bits */
+	reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
+	wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+		     reg & WM8400_IPVU);
+	reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
+	wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+		     reg & WM8400_IPVU);
+
+	wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+	wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+	wm8400_codec = codec;
+
+	if (!schedule_work(&priv->work)) {
+		ret = -EINVAL;
+		goto err_regulator;
+	}
+
+	return 0;
+
+err_regulator:
+	wm8400_codec = NULL;
+	regulator_bulk_free(ARRAY_SIZE(power), power);
+err:
+	kfree(priv);
+	return ret;
+}
+
+static int __exit wm8400_codec_remove(struct platform_device *dev)
+{
+	struct wm8400_priv *priv = wm8400_codec->private_data;
+	u16 reg;
+
+	snd_soc_unregister_dai(&wm8400_dai);
+	snd_soc_unregister_codec(wm8400_codec);
+
+	reg = wm8400_read(wm8400_codec, WM8400_POWER_MANAGEMENT_1);
+	wm8400_write(wm8400_codec, WM8400_POWER_MANAGEMENT_1,
+		     reg & (~WM8400_CODEC_ENA));
+
+	regulator_bulk_free(ARRAY_SIZE(power), power);
+	kfree(priv);
+
+	wm8400_codec = NULL;
+
+	return 0;
+}
+
+static struct platform_driver wm8400_codec_driver = {
+	.driver = {
+		.name = "wm8400-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8400_codec_probe,
+	.remove	= __exit_p(wm8400_codec_remove),
+};
+
+static int __init wm8400_codec_init(void)
+{
+	return platform_driver_register(&wm8400_codec_driver);
+}
+module_init(wm8400_codec_init);
+
+static void __exit wm8400_codec_exit(void)
+{
+	platform_driver_unregister(&wm8400_codec_driver);
+}
+module_exit(wm8400_codec_exit);
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8400);
+
+MODULE_DESCRIPTION("ASoC WM8400 driver");
+MODULE_AUTHOR("Mark Brown");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8400-codec");
diff --git a/sound/soc/codecs/wm8400.h b/sound/soc/codecs/wm8400.h
new file mode 100644
index 0000000..79c5934
--- /dev/null
+++ b/sound/soc/codecs/wm8400.h
@@ -0,0 +1,62 @@
+/*
+ * wm8400.h  --  audio driver for WM8400
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef _WM8400_CODEC_H
+#define _WM8400_CODEC_H
+
+#define WM8400_MCLK_DIV 0
+#define WM8400_DACCLK_DIV 1
+#define WM8400_ADCCLK_DIV 2
+#define WM8400_BCLK_DIV 3
+
+#define WM8400_MCLK_DIV_1 0x400
+#define WM8400_MCLK_DIV_2 0x800
+
+#define WM8400_DAC_CLKDIV_1    0x00
+#define WM8400_DAC_CLKDIV_1_5  0x04
+#define WM8400_DAC_CLKDIV_2    0x08
+#define WM8400_DAC_CLKDIV_3    0x0c
+#define WM8400_DAC_CLKDIV_4    0x10
+#define WM8400_DAC_CLKDIV_5_5  0x14
+#define WM8400_DAC_CLKDIV_6    0x18
+
+#define WM8400_ADC_CLKDIV_1    0x00
+#define WM8400_ADC_CLKDIV_1_5  0x20
+#define WM8400_ADC_CLKDIV_2    0x40
+#define WM8400_ADC_CLKDIV_3    0x60
+#define WM8400_ADC_CLKDIV_4    0x80
+#define WM8400_ADC_CLKDIV_5_5  0xa0
+#define WM8400_ADC_CLKDIV_6    0xc0
+
+
+#define WM8400_BCLK_DIV_1                       (0x0 << 1)
+#define WM8400_BCLK_DIV_1_5                     (0x1 << 1)
+#define WM8400_BCLK_DIV_2                       (0x2 << 1)
+#define WM8400_BCLK_DIV_3                       (0x3 << 1)
+#define WM8400_BCLK_DIV_4                       (0x4 << 1)
+#define WM8400_BCLK_DIV_5_5                     (0x5 << 1)
+#define WM8400_BCLK_DIV_6                       (0x6 << 1)
+#define WM8400_BCLK_DIV_8                       (0x7 << 1)
+#define WM8400_BCLK_DIV_11                      (0x8 << 1)
+#define WM8400_BCLK_DIV_12                      (0x9 << 1)
+#define WM8400_BCLK_DIV_16                      (0xA << 1)
+#define WM8400_BCLK_DIV_22                      (0xB << 1)
+#define WM8400_BCLK_DIV_24                      (0xC << 1)
+#define WM8400_BCLK_DIV_32                      (0xD << 1)
+#define WM8400_BCLK_DIV_44                      (0xE << 1)
+#define WM8400_BCLK_DIV_48                      (0xF << 1)
+
+extern struct snd_soc_dai wm8400_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8400;
+
+#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 40f8238..6a4cea0 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST,  8, 1, 0),
 SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1),
 };
 
-/* add non dapm controls */
-static int wm8510_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8510_snd_controls[i], codec,
-					NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Speaker Output Mixer */
 static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
@@ -352,7 +336,7 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
 		return 0;
 	}
 
-	pll_factors(freq_out*8, freq_in);
+	pll_factors(freq_out*4, freq_in);
 
 	wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
 	wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
@@ -383,7 +367,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		wm8510_write(codec, WM8510_GPIO, reg | div);
 		break;
 	case WM8510_MCLKDIV:
-		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f;
+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f;
 		wm8510_write(codec, WM8510_CLOCK, reg | div);
 		break;
 	case WM8510_ADCCLK:
@@ -468,7 +452,7 @@ static int wm8510_pcm_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f;
 	u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
 
@@ -570,6 +554,14 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 #define WM8510_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 wm8510_dai_ops = {
+	.hw_params	= wm8510_pcm_hw_params,
+	.digital_mute	= wm8510_mute,
+	.set_fmt	= wm8510_set_dai_fmt,
+	.set_clkdiv	= wm8510_set_dai_clkdiv,
+	.set_pll	= wm8510_set_dai_pll,
+};
+
 struct snd_soc_dai wm8510_dai = {
 	.name = "WM8510 HiFi",
 	.playback = {
@@ -584,20 +576,14 @@ struct snd_soc_dai wm8510_dai = {
 		.channels_max = 2,
 		.rates = WM8510_RATES,
 		.formats = WM8510_FORMATS,},
-	.ops = {
-		.hw_params = wm8510_pcm_hw_params,
-		.digital_mute = wm8510_mute,
-		.set_fmt = wm8510_set_dai_fmt,
-		.set_clkdiv = wm8510_set_dai_clkdiv,
-		.set_pll = wm8510_set_dai_pll,
-	},
+	.ops = &wm8510_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8510_dai);
 
 static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -606,7 +592,7 @@ static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8510_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -628,7 +614,7 @@ static int wm8510_resume(struct platform_device *pdev)
  */
 static int wm8510_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "WM8510";
@@ -656,7 +642,8 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	/* power on device */
 	codec->bias_level = SND_SOC_BIAS_OFF;
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8510_add_controls(codec);
+	snd_soc_add_controls(codec, wm8510_snd_controls,
+				ARRAY_SIZE(wm8510_snd_controls));
 	wm8510_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -685,7 +672,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8510_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -766,7 +753,7 @@ err_driver:
 static int __devinit wm8510_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8510_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -832,7 +819,7 @@ static int wm8510_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -862,7 +849,7 @@ static int wm8510_probe(struct platform_device *pdev)
 static int wm8510_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d004e58..442ea6f 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -1,7 +1,7 @@
 /*
  * wm8580.c  --  WM8580 ALSA Soc Audio driver
  *
- * Copyright 2008 Wolfson Microelectronics PLC.
+ * Copyright 2008, 2009 Wolfson Microelectronics PLC.
  *
  *  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
@@ -35,19 +35,6 @@
 
 #include "wm8580.h"
 
-#define WM8580_VERSION "0.1"
-
-struct pll_state {
-	unsigned int in;
-	unsigned int out;
-};
-
-/* codec private data */
-struct wm8580_priv {
-	struct pll_state a;
-	struct pll_state b;
-};
-
 /* WM8580 register space */
 #define WM8580_PLLA1                         0x00
 #define WM8580_PLLA2                         0x01
@@ -102,6 +89,8 @@ struct wm8580_priv {
 #define WM8580_READBACK                      0x34
 #define WM8580_RESET                         0x35
 
+#define WM8580_MAX_REGISTER                  0x35
+
 /* PLLB4 (register 7h) */
 #define WM8580_PLLB4_MCLKOUTSRC_MASK   0x60
 #define WM8580_PLLB4_MCLKOUTSRC_PLLA   0x20
@@ -193,6 +182,20 @@ static const u16 wm8580_reg[] = {
 	0x0000, 0x0000 /*R53*/
 };
 
+struct pll_state {
+	unsigned int in;
+	unsigned int out;
+};
+
+/* codec private data */
+struct wm8580_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8580_MAX_REGISTER + 1];
+	struct pll_state a;
+	struct pll_state b;
+};
+
+
 /*
  * read wm8580 register cache
  */
@@ -200,7 +203,7 @@ static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+	BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
 	return cache[reg];
 }
 
@@ -223,7 +226,7 @@ static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg,
 {
 	u8 data[2];
 
-	BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+	BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
 
 	/* Registers are 9 bits wide */
 	value &= 0x1ff;
@@ -330,20 +333,6 @@ SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0),
 SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0),
 };
 
-/* Add non-DAPM controls */
-static int wm8580_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8580_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
 static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = {
 SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1),
 SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1),
@@ -553,7 +542,7 @@ static int wm8580_paif_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
 
 	paifb &= ~WM8580_AIF_LENGTH_MASK;
@@ -771,8 +760,22 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
+		break;
+
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Power up and get individual control of the DACs */
+			reg = wm8580_read(codec, WM8580_PWRDN1);
+			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
+			wm8580_write(codec, WM8580_PWRDN1, reg);
+
+			/* Make VMID high impedence */
+			reg = wm8580_read(codec,  WM8580_ADC_CONTROL1);
+			reg &= ~0x100;
+			wm8580_write(codec, WM8580_ADC_CONTROL1, reg);
+		}
 		break;
+
 	case SND_SOC_BIAS_OFF:
 		reg = wm8580_read(codec, WM8580_PWRDN1);
 		wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
@@ -785,6 +788,21 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 #define WM8580_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 wm8580_dai_ops_playback = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+	.digital_mute	= wm8580_digital_mute,
+};
+
+static struct snd_soc_dai_ops wm8580_dai_ops_capture = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+};
+
 struct snd_soc_dai wm8580_dai[] = {
 	{
 		.name = "WM8580 PAIFRX",
@@ -796,13 +814,7 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-			 .digital_mute = wm8580_digital_mute,
-		 },
+		.ops = &wm8580_dai_ops_playback,
 	},
 	{
 		.name = "WM8580 PAIFTX",
@@ -814,109 +826,168 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-		 },
+		.ops = &wm8580_dai_ops_capture,
 	},
 };
 EXPORT_SYMBOL_GPL(wm8580_dai);
 
-/*
- * initialise the WM8580 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8580_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8580_codec;
+
+static int wm8580_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
 	int ret = 0;
 
-	codec->name = "WM8580";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8580_read_reg_cache;
-	codec->write = wm8580_write;
-	codec->set_bias_level = wm8580_set_bias_level;
-	codec->dai = wm8580_dai;
-	codec->num_dai = ARRAY_SIZE(wm8580_dai);
-	codec->reg_cache_size = ARRAY_SIZE(wm8580_reg);
-	codec->reg_cache = kmemdup(wm8580_reg, sizeof(wm8580_reg),
-				   GFP_KERNEL);
-
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
-
-	/* Get the codec into a known state */
-	wm8580_write(codec, WM8580_RESET, 0);
-
-	/* Power up and get individual control of the DACs */
-	wm8580_write(codec, WM8580_PWRDN1, wm8580_read(codec, WM8580_PWRDN1) &
-		     ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD));
+	if (wm8580_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
 
-	/* Make VMID high impedence */
-	wm8580_write(codec, WM8580_ADC_CONTROL1,
-		     wm8580_read(codec,  WM8580_ADC_CONTROL1) & ~0x100);
+	socdev->card->codec = wm8580_codec;
+	codec = wm8580_codec;
 
 	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1,
-			       SNDRV_DEFAULT_STR1);
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8580: failed to create pcms\n");
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
 		goto pcm_err;
 	}
 
-	wm8580_add_controls(codec);
+	snd_soc_add_controls(codec, wm8580_snd_controls,
+			     ARRAY_SIZE(wm8580_snd_controls));
 	wm8580_add_widgets(codec);
-
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8580: failed to register card\n");
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
 		goto card_err;
 	}
+
 	return ret;
 
 card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 pcm_err:
-	kfree(codec->reg_cache);
 	return ret;
 }
 
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static struct snd_soc_device *wm8580_socdev;
+/* power down chip */
+static int wm8580_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	return 0;
+}
 
-/*
- * WM8580 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
+struct snd_soc_codec_device soc_codec_dev_wm8580 = {
+	.probe = 	wm8580_probe,
+	.remove = 	wm8580_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
+
+static int wm8580_register(struct wm8580_priv *wm8580)
+{
+	int ret, i;
+	struct snd_soc_codec *codec = &wm8580->codec;
+
+	if (wm8580_codec) {
+		dev_err(codec->dev, "Another WM8580 is registered\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
+	codec->private_data = wm8580;
+	codec->name = "WM8580";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8580_read_reg_cache;
+	codec->write = wm8580_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8580_set_bias_level;
+	codec->dai = wm8580_dai;
+	codec->num_dai = ARRAY_SIZE(wm8580_dai);
+	codec->reg_cache_size = ARRAY_SIZE(wm8580->reg_cache);
+	codec->reg_cache = &wm8580->reg_cache;
+
+	memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg));
+
+	/* Get the codec into a known state */
+	ret = wm8580_write(codec, WM8580_RESET, 0);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to reset codec: %d\n", ret);
+		goto err;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++)
+		wm8580_dai[i].dev = codec->dev;
+
+	wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	wm8580_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8580);
+	return ret;
+}
+
+static void wm8580_unregister(struct wm8580_priv *wm8580)
+{
+	wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+	snd_soc_unregister_codec(&wm8580->codec);
+	kfree(wm8580);
+	wm8580_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static int wm8580_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8580_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
+	struct wm8580_priv *wm8580;
+	struct snd_soc_codec *codec;
 
-	i2c_set_clientdata(i2c, codec);
+	wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL);
+	if (wm8580 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8580->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(i2c, wm8580);
 	codec->control_data = i2c;
 
-	ret = wm8580_init(socdev);
-	if (ret < 0)
-		dev_err(&i2c->dev, "failed to initialise WM8580\n");
-	return ret;
+	codec->dev = &i2c->dev;
+
+	return wm8580_register(wm8580);
 }
 
 static int wm8580_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
+	struct wm8580_priv *wm8580 = i2c_get_clientdata(client);
+	wm8580_unregister(wm8580);
 	return 0;
 }
 
@@ -928,129 +999,35 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
 
 static struct i2c_driver wm8580_i2c_driver = {
 	.driver = {
-		.name = "WM8580 I2C Codec",
+		.name = "wm8580",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8580_i2c_probe,
 	.remove =   wm8580_i2c_remove,
 	.id_table = wm8580_i2c_id,
 };
+#endif
 
-static int wm8580_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8580_setup_data *setup)
+static int __init wm8580_modinit(void)
 {
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
 	int ret;
 
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	ret = i2c_add_driver(&wm8580_i2c_driver);
 	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8580", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
+		pr_err("Failed to register WM8580 I2C driver: %d\n", ret);
 	}
-
-	return 0;
-
-err_driver:
-	i2c_del_driver(&wm8580_i2c_driver);
-	return -ENODEV;
-}
 #endif
 
-static int wm8580_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8580_setup_data *setup;
-	struct snd_soc_codec *codec;
-	struct wm8580_priv *wm8580;
-	int ret = 0;
-
-	pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION);
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL);
-	if (wm8580 == NULL) {
-		kfree(codec);
-		return -ENOMEM;
-	}
-
-	codec->private_data = wm8580;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	wm8580_socdev = socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8580_add_i2c_device(pdev, setup);
-	}
-#else
-		/* Add other interfaces here */
-#endif
-	return ret;
-}
-
-/* power down chip */
-static int wm8580_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
-
-	if (codec->control_data)
-		wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_unregister_device(codec->control_data);
-	i2c_del_driver(&wm8580_i2c_driver);
-#endif
-	kfree(codec->private_data);
-	kfree(codec);
-
 	return 0;
 }
-
-struct snd_soc_codec_device soc_codec_dev_wm8580 = {
-	.probe = 	wm8580_probe,
-	.remove = 	wm8580_remove,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
-
-static int __init wm8580_modinit(void)
-{
-	return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
-}
 module_init(wm8580_modinit);
 
 static void __exit wm8580_exit(void)
 {
-	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8580_i2c_driver);
+#endif
 }
 module_exit(wm8580_exit);
 
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
index 09e4422..0dfb5dd 100644
--- a/sound/soc/codecs/wm8580.h
+++ b/sound/soc/codecs/wm8580.h
@@ -28,11 +28,6 @@
 #define WM8580_CLKSRC_OSC  4
 #define WM8580_CLKSRC_NONE 5
 
-struct wm8580_setup_data {
-	int i2c_bus;
-	unsigned short i2c_address;
-};
-
 #define WM8580_DAI_PAIFRX 0
 #define WM8580_DAI_PAIFTX 1
 
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 80b1198..e7ff212 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -47,7 +47,7 @@ static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
 	return cache[reg];
 }
 
@@ -55,7 +55,7 @@ static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
 	u16 reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
 	cache[reg] = value;
 }
 
@@ -92,21 +92,6 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
 SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
 };
 
-static int wm8728_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8728_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /*
  * DAPM controls.
  */
@@ -152,7 +137,7 @@ static int wm8728_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
 
 	dac &= ~0x18;
@@ -259,6 +244,12 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
 #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8728_dai_ops = {
+	.hw_params	= wm8728_hw_params,
+	.digital_mute	= wm8728_mute,
+	.set_fmt	= wm8728_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8728_dai = {
 	.name = "WM8728",
 	.playback = {
@@ -268,18 +259,14 @@ struct snd_soc_dai wm8728_dai = {
 		.rates = WM8728_RATES,
 		.formats = WM8728_FORMATS,
 	},
-	.ops = {
-		 .hw_params = wm8728_hw_params,
-		 .digital_mute = wm8728_mute,
-		 .set_fmt = wm8728_set_dai_fmt,
-	}
+	.ops = &wm8728_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8728_dai);
 
 static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -289,7 +276,7 @@ static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8728_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8728_set_bias_level(codec, codec->suspend_bias_level);
 
@@ -302,7 +289,7 @@ static int wm8728_resume(struct platform_device *pdev)
  */
 static int wm8728_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "WM8728";
@@ -330,7 +317,8 @@ static int wm8728_init(struct snd_soc_device *socdev)
 	/* power on device */
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	wm8728_add_controls(codec);
+	snd_soc_add_controls(codec, wm8728_snd_controls,
+				ARRAY_SIZE(wm8728_snd_controls));
 	wm8728_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -363,7 +351,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8728_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -444,7 +432,7 @@ err_driver:
 static int __devinit wm8728_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8728_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -508,7 +496,7 @@ static int wm8728_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -541,7 +529,7 @@ static int wm8728_probe(struct platform_device *pdev)
 static int wm8728_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index c444b9f..e043e3f 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -29,15 +29,20 @@
 
 #include "wm8731.h"
 
-#define WM8731_VERSION "0.13"
-
+static struct snd_soc_codec *wm8731_codec;
 struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 /* codec private data */
 struct wm8731_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8731_CACHEREGNUM];
 	unsigned int sysclk;
 };
 
+#ifdef CONFIG_SPI_MASTER
+static int wm8731_spi_write(struct spi_device *spi, const char *data, int len);
+#endif
+
 /*
  * wm8731 register cache
  * We can't read the WM8731 register space when we are
@@ -129,22 +134,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
 SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
 };
 
-/* add non dapm controls */
-static int wm8731_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8731_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Output Mixer */
 static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
@@ -269,7 +258,7 @@ static int wm8731_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8731_priv *wm8731 = codec->private_data;
 	u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
 	int i = get_coeff(wm8731->sysclk, params_rate(params));
@@ -299,7 +288,7 @@ static int wm8731_pcm_prepare(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* set active */
 	wm8731_write(codec, WM8731_ACTIVE, 0x0001);
@@ -312,7 +301,7 @@ static void wm8731_shutdown(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* deactivate */
 	if (!codec->active) {
@@ -414,21 +403,19 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
+	u16 reg;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		/* vref/mid, osc on, dac unmute */
-		wm8731_write(codec, WM8731_PWR, reg);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		/* everything off except vref/vmid, */
+		/* Clear PWROFF, gate CLKOUT, everything else as-is */
+		reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
 		wm8731_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
 	case SND_SOC_BIAS_OFF:
-		/* everything off, dac mute, inactive */
 		wm8731_write(codec, WM8731_ACTIVE, 0x0);
 		wm8731_write(codec, WM8731_PWR, 0xffff);
 		break;
@@ -446,6 +433,15 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8731_dai = {
 	.name = "WM8731",
 	.playback = {
@@ -460,21 +456,14 @@ struct snd_soc_dai wm8731_dai = {
 		.channels_max = 2,
 		.rates = WM8731_RATES,
 		.formats = WM8731_FORMATS,},
-	.ops = {
-		.prepare = wm8731_pcm_prepare,
-		.hw_params = wm8731_hw_params,
-		.shutdown = wm8731_shutdown,
-		.digital_mute = wm8731_mute,
-		.set_sysclk = wm8731_set_dai_sysclk,
-		.set_fmt = wm8731_set_dai_fmt,
-	}
+	.ops = &wm8731_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8731_dai);
 
 static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8731_write(codec, WM8731_ACTIVE, 0x0);
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -484,7 +473,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8731_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -500,54 +489,33 @@ static int wm8731_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8731 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8731_init(struct snd_soc_device *socdev)
+static int wm8731_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	int reg, ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
 
-	codec->name = "WM8731";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8731_read_reg_cache;
-	codec->write = wm8731_write;
-	codec->set_bias_level = wm8731_set_bias_level;
-	codec->dai = &wm8731_dai;
-	codec->num_dai = 1;
-	codec->reg_cache_size = ARRAY_SIZE(wm8731_reg);
-	codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
+	if (wm8731_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
 
-	wm8731_reset(codec);
+	socdev->card->codec = wm8731_codec;
+	codec = wm8731_codec;
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8731: failed to create pcms\n");
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
 		goto pcm_err;
 	}
 
-	/* power on device */
-	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	/* set the update bits */
-	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
-	wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
-	wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
-	wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
-	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
-
-	wm8731_add_controls(codec);
+	snd_soc_add_controls(codec, wm8731_snd_controls,
+			     ARRAY_SIZE(wm8731_snd_controls));
 	wm8731_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8731: failed to register card\n");
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
 		goto card_err;
 	}
 
@@ -557,133 +525,109 @@ card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 pcm_err:
-	kfree(codec->reg_cache);
 	return ret;
 }
 
-static struct snd_soc_device *wm8731_socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-/*
- * WM8731 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
-
-static int wm8731_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+/* power down chip */
+static int wm8731_remove(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8731\n");
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	return ret;
-}
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
 
-static int wm8731_i2c_remove(struct i2c_client *client)
-{
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
 	return 0;
 }
 
-static const struct i2c_device_id wm8731_i2c_id[] = {
-	{ "wm8731", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
-
-static struct i2c_driver wm8731_i2c_driver = {
-	.driver = {
-		.name = "WM8731 I2C Codec",
-		.owner = THIS_MODULE,
-	},
-	.probe =    wm8731_i2c_probe,
-	.remove =   wm8731_i2c_remove,
-	.id_table = wm8731_i2c_id,
+struct snd_soc_codec_device soc_codec_dev_wm8731 = {
+	.probe = 	wm8731_probe,
+	.remove = 	wm8731_remove,
+	.suspend = 	wm8731_suspend,
+	.resume =	wm8731_resume,
 };
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
-static int wm8731_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8731_setup_data *setup)
+static int wm8731_register(struct wm8731_priv *wm8731)
 {
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
 	int ret;
+	struct snd_soc_codec *codec = &wm8731->codec;
+	u16 reg;
 
-	ret = i2c_add_driver(&wm8731_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
+	if (wm8731_codec) {
+		dev_err(codec->dev, "Another WM8731 is registered\n");
+		return -EINVAL;
 	}
 
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
+	codec->private_data = wm8731;
+	codec->name = "WM8731";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8731_read_reg_cache;
+	codec->write = wm8731_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8731_set_bias_level;
+	codec->dai = &wm8731_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8731_CACHEREGNUM;
+	codec->reg_cache = &wm8731->reg_cache;
 
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
+	memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg));
+
+	ret = wm8731_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
 	}
 
-	return 0;
+	wm8731_dai.dev = codec->dev;
 
-err_driver:
-	i2c_del_driver(&wm8731_i2c_driver);
-	return -ENODEV;
-}
-#endif
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-#if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8731_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
+	/* Latch the update bits */
+	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
+	wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
+	wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
+	wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
+	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
 
-	codec->control_data = spi;
+	/* Disable bypass path by default */
+	reg = wm8731_read_reg_cache(codec, WM8731_APANA);
+	wm8731_write(codec, WM8731_APANA, reg & ~0x4);
 
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8731\n");
+	wm8731_codec = codec;
 
-	return ret;
-}
+	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(&wm8731_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
 
-static int __devexit wm8731_spi_remove(struct spi_device *spi)
-{
 	return 0;
 }
 
-static struct spi_driver wm8731_spi_driver = {
-	.driver = {
-		.name	= "wm8731",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8731_spi_probe,
-	.remove		= __devexit_p(wm8731_spi_remove),
-};
+static void wm8731_unregister(struct wm8731_priv *wm8731)
+{
+	wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm8731_dai);
+	snd_soc_unregister_codec(&wm8731->codec);
+	kfree(wm8731);
+	wm8731_codec = NULL;
+}
 
+#if defined(CONFIG_SPI_MASTER)
 static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
 {
 	struct spi_transfer t;
@@ -707,101 +651,121 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
 
 	return len;
 }
-#endif /* CONFIG_SPI_MASTER */
 
-static int wm8731_probe(struct platform_device *pdev)
+static int __devinit wm8731_spi_probe(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8731_setup_data *setup;
 	struct snd_soc_codec *codec;
 	struct wm8731_priv *wm8731;
-	int ret = 0;
-
-	pr_info("WM8731 Audio Codec %s", WM8731_VERSION);
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
 
 	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
-	if (wm8731 == NULL) {
-		kfree(codec);
+	if (wm8731 == NULL)
 		return -ENOMEM;
-	}
 
-	codec->private_data = wm8731;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
+	codec = &wm8731->codec;
+	codec->control_data = spi;
+	codec->hw_write = (hw_write_t)wm8731_spi_write;
+	codec->dev = &spi->dev;
 
-	wm8731_socdev = socdev;
-	ret = -ENODEV;
+	spi->dev.driver_data = wm8731;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8731_add_i2c_device(pdev, setup);
-	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8731_spi_write;
-		ret = spi_register_driver(&wm8731_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
-#endif
-
-	if (ret != 0) {
-		kfree(codec->private_data);
-		kfree(codec);
-	}
-	return ret;
+	return wm8731_register(wm8731);
 }
 
-/* power down chip */
-static int wm8731_remove(struct platform_device *pdev)
+static int __devexit wm8731_spi_remove(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct wm8731_priv *wm8731 = spi->dev.driver_data;
 
-	if (codec->control_data)
-		wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	wm8731_unregister(wm8731);
+
+	return 0;
+}
+
+static struct spi_driver wm8731_spi_driver = {
+	.driver = {
+		.name	= "wm8731",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8731_spi_probe,
+	.remove		= __devexit_p(wm8731_spi_remove),
+};
+#endif /* CONFIG_SPI_MASTER */
 
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_unregister_device(codec->control_data);
-	i2c_del_driver(&wm8731_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8731_spi_driver);
-#endif
-	kfree(codec->private_data);
-	kfree(codec);
+static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct wm8731_priv *wm8731;
+	struct snd_soc_codec *codec;
+
+	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+	if (wm8731 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8731->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
 
+	i2c_set_clientdata(i2c, wm8731);
+	codec->control_data = i2c;
+
+	codec->dev = &i2c->dev;
+
+	return wm8731_register(wm8731);
+}
+
+static __devexit int wm8731_i2c_remove(struct i2c_client *client)
+{
+	struct wm8731_priv *wm8731 = i2c_get_clientdata(client);
+	wm8731_unregister(wm8731);
 	return 0;
 }
 
-struct snd_soc_codec_device soc_codec_dev_wm8731 = {
-	.probe = 	wm8731_probe,
-	.remove = 	wm8731_remove,
-	.suspend = 	wm8731_suspend,
-	.resume =	wm8731_resume,
+static const struct i2c_device_id wm8731_i2c_id[] = {
+	{ "wm8731", 0 },
+	{ }
 };
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
+MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
+
+static struct i2c_driver wm8731_i2c_driver = {
+	.driver = {
+		.name = "WM8731 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8731_i2c_probe,
+	.remove =   __devexit_p(wm8731_i2c_remove),
+	.id_table = wm8731_i2c_id,
+};
+#endif
 
 static int __init wm8731_modinit(void)
 {
-	return snd_soc_register_dai(&wm8731_dai);
+	int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&wm8731_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n",
+		       ret);
+	}
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8731_spi_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n",
+		       ret);
+	}
+#endif
+	return 0;
 }
 module_init(wm8731_modinit);
 
 static void __exit wm8731_exit(void)
 {
-	snd_soc_unregister_dai(&wm8731_dai);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8731_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8731_spi_driver);
+#endif
 }
 module_exit(wm8731_exit);
 
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 95190e9..cd7b806 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -34,12 +34,6 @@
 #define WM8731_SYSCLK	0
 #define WM8731_DAI		0
 
-struct wm8731_setup_data {
-	int            spi;
-	int            i2c_bus;
-	unsigned short i2c_address;
-};
-
 extern struct snd_soc_dai wm8731_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8731;
 
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 5997fa6..b64509b 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
 
 };
 
-/* add non dapm controls */
-static int wm8750_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8750_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * DAPM Controls
  */
@@ -619,7 +604,7 @@ static int wm8750_pcm_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8750_priv *wm8750 = codec->private_data;
 	u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
 	u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
@@ -694,6 +679,13 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8750_dai_ops = {
+	.hw_params	= wm8750_pcm_hw_params,
+	.digital_mute	= wm8750_mute,
+	.set_fmt	= wm8750_set_dai_fmt,
+	.set_sysclk	= wm8750_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8750_dai = {
 	.name = "WM8750",
 	.playback = {
@@ -708,12 +700,7 @@ struct snd_soc_dai wm8750_dai = {
 		.channels_max = 2,
 		.rates = WM8750_RATES,
 		.formats = WM8750_FORMATS,},
-	.ops = {
-		.hw_params = wm8750_pcm_hw_params,
-		.digital_mute = wm8750_mute,
-		.set_fmt = wm8750_set_dai_fmt,
-		.set_sysclk = wm8750_set_dai_sysclk,
-	},
+	.ops = &wm8750_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8750_dai);
 
@@ -727,7 +714,7 @@ static void wm8750_work(struct work_struct *work)
 static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -736,7 +723,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8750_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -769,7 +756,7 @@ static int wm8750_resume(struct platform_device *pdev)
  */
 static int wm8750_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8750";
@@ -816,7 +803,8 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
 	wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
 
-	wm8750_add_controls(codec);
+	snd_soc_add_controls(codec, wm8750_snd_controls,
+				ARRAY_SIZE(wm8750_snd_controls));
 	wm8750_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -850,7 +838,7 @@ static int wm8750_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8750_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -931,7 +919,7 @@ err_driver:
 static int __devinit wm8750_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8750_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -1003,7 +991,7 @@ static int wm8750_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8750;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1057,7 +1045,7 @@ static int run_delayed_work(struct delayed_work *dwork)
 static int wm8750_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 77620ab..a6e8f3f 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -51,8 +51,6 @@
 
 #include "wm8753.h"
 
-#define WM8753_VERSION "0.16"
-
 static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
@@ -60,12 +58,6 @@ MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
 static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
 	unsigned int mode);
 
-/* codec private data */
-struct wm8753_priv {
-	unsigned int sysclk;
-	unsigned int pcmclk;
-};
-
 /*
  * wm8753 register cache
  * We can't read the WM8753 register space when we
@@ -90,6 +82,14 @@ static const u16 wm8753_reg[] = {
 	0x0000, 0x0000
 };
 
+/* codec private data */
+struct wm8753_priv {
+	unsigned int sysclk;
+	unsigned int pcmclk;
+	struct snd_soc_codec codec;
+	u16 reg_cache[ARRAY_SIZE(wm8753_reg)];
+};
+
 /*
  * read wm8753 register cache
  */
@@ -97,7 +97,7 @@ static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
+	if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
 		return -1;
 	return cache[reg - 1];
 }
@@ -109,7 +109,7 @@ static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	if (reg < 1 || reg > 0x3f)
+	if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
 		return;
 	cache[reg - 1] = value;
 }
@@ -339,21 +339,6 @@ SOC_ENUM("ADC Data Select", wm8753_enum[27]),
 SOC_ENUM("ROUT2 Phase", wm8753_enum[28]),
 };
 
-/* add non dapm controls */
-static int wm8753_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8753_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * _DAPM_ Controls
  */
@@ -927,7 +912,7 @@ static int wm8753_pcm_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8753_priv *wm8753 = codec->private_data;
 	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3;
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
@@ -1161,7 +1146,7 @@ static int wm8753_i2s_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8753_priv *wm8753 = codec->private_data;
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
 	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3;
@@ -1316,6 +1301,51 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
  * 3. Voice disabled - HIFI over HIFI
  * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
  */
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1h_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1v_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode2_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
 static const struct snd_soc_dai wm8753_all_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "WM8753 HiFi",
@@ -1332,14 +1362,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1h_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode1,
 },
 /* DAI Voice mode 1 */
 {	.name = "WM8753 Voice",
@@ -1356,14 +1379,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1v_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode1,
 },
 /* DAI HiFi mode 2 - dummy */
 {	.name = "WM8753 HiFi",
@@ -1384,14 +1400,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode2_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode2,
 },
 /* DAI HiFi mode 3 */
 {	.name = "WM8753 HiFi",
@@ -1408,14 +1417,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode3,
 },
 /* DAI Voice mode 3 - dummy */
 {	.name = "WM8753 Voice",
@@ -1436,14 +1438,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode4,
 },
 /* DAI Voice mode 4 - dummy */
 {	.name = "WM8753 Voice",
@@ -1466,30 +1461,35 @@ static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
 	if (mode < 4) {
 		int playback_active, capture_active, codec_active, pop_wait;
 		void *private_data;
+		struct list_head list;
 
 		playback_active = wm8753_dai[0].playback.active;
 		capture_active = wm8753_dai[0].capture.active;
 		codec_active = wm8753_dai[0].active;
 		private_data = wm8753_dai[0].private_data;
 		pop_wait = wm8753_dai[0].pop_wait;
+		list = wm8753_dai[0].list;
 		wm8753_dai[0] = wm8753_all_dai[mode << 1];
 		wm8753_dai[0].playback.active = playback_active;
 		wm8753_dai[0].capture.active = capture_active;
 		wm8753_dai[0].active = codec_active;
 		wm8753_dai[0].private_data = private_data;
 		wm8753_dai[0].pop_wait = pop_wait;
+		wm8753_dai[0].list = list;
 
 		playback_active = wm8753_dai[1].playback.active;
 		capture_active = wm8753_dai[1].capture.active;
 		codec_active = wm8753_dai[1].active;
 		private_data = wm8753_dai[1].private_data;
 		pop_wait = wm8753_dai[1].pop_wait;
+		list = wm8753_dai[1].list;
 		wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1];
 		wm8753_dai[1].playback.active = playback_active;
 		wm8753_dai[1].capture.active = capture_active;
 		wm8753_dai[1].active = codec_active;
 		wm8753_dai[1].private_data = private_data;
 		wm8753_dai[1].pop_wait = pop_wait;
+		wm8753_dai[1].list = list;
 	}
 	wm8753_dai[0].codec = codec;
 	wm8753_dai[1].codec = codec;
@@ -1505,7 +1505,7 @@ static void wm8753_work(struct work_struct *work)
 static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* we only need to suspend if we are a valid card */
 	if (!codec->card)
@@ -1518,7 +1518,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8753_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -1531,6 +1531,11 @@ static int wm8753_resume(struct platform_device *pdev)
 	for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
 		if (i + 1 == WM8753_RESET)
 			continue;
+
+		/* No point in writing hardware default values back */
+		if (cache[i] == wm8753_reg[i])
+			continue;
+
 		data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
 		data[1] = cache[i] & 0x00ff;
 		codec->hw_write(codec->control_data, data, 2);
@@ -1549,44 +1554,129 @@ static int wm8753_resume(struct platform_device *pdev)
 	return 0;
 }
 
+static struct snd_soc_codec *wm8753_codec;
+
+static int wm8753_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (!wm8753_codec) {
+		dev_err(&pdev->dev, "WM8753 codec not yet registered\n");
+		return -EINVAL;
+	}
+
+	socdev->card->codec = wm8753_codec;
+	codec = wm8753_codec;
+
+	wm8753_set_dai_mode(codec, 0);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8753: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, wm8753_snd_controls,
+			     ARRAY_SIZE(wm8753_snd_controls));
+	wm8753_add_widgets(codec);
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8753: failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+pcm_err:
+	return ret;
+}
+
 /*
- * initialise the WM8753 driver
- * register the mixer and dsp interfaces with the kernel
+ * This function forces any delayed work to be queued and run.
  */
-static int wm8753_init(struct snd_soc_device *socdev)
+static int run_delayed_work(struct delayed_work *dwork)
+{
+	int ret;
+
+	/* cancel any work waiting to be queued. */
+	ret = cancel_delayed_work(dwork);
+
+	/* if there was any work waiting then we run it now and
+	 * wait for it's completion */
+	if (ret) {
+		schedule_delayed_work(dwork, 0);
+		flush_scheduled_work();
+	}
+	return ret;
+}
+
+/* power down chip */
+static int wm8753_remove(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	int reg, ret = 0;
+	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_wm8753 = {
+	.probe = 	wm8753_probe,
+	.remove = 	wm8753_remove,
+	.suspend = 	wm8753_suspend,
+	.resume =	wm8753_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
+
+static int wm8753_register(struct wm8753_priv *wm8753)
+{
+	int ret, i;
+	struct snd_soc_codec *codec = &wm8753->codec;
+	u16 reg;
+
+	if (wm8753_codec) {
+		dev_err(codec->dev, "Multiple WM8753 devices not supported\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
 	codec->name = "WM8753";
 	codec->owner = THIS_MODULE;
 	codec->read = wm8753_read_reg_cache;
 	codec->write = wm8753_write;
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	codec->set_bias_level = wm8753_set_bias_level;
 	codec->dai = wm8753_dai;
 	codec->num_dai = 2;
-	codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
-	codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
-
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
-
-	wm8753_set_dai_mode(codec, 0);
+	codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache);
+	codec->reg_cache = &wm8753->reg_cache;
+	codec->private_data = wm8753;
 
-	wm8753_reset(codec);
+	memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache));
+	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
 
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	ret = wm8753_reset(codec);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8753: failed to create pcms\n");
-		goto pcm_err;
+		dev_err(codec->dev, "Failed to issue reset\n");
+		goto err;
 	}
 
 	/* charge output caps */
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
-	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	schedule_delayed_work(&codec->delayed_work,
-		msecs_to_jiffies(caps_charge));
+			      msecs_to_jiffies(caps_charge));
 
 	/* set the update bits */
 	reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
@@ -1610,59 +1700,70 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
 	wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
 
-	wm8753_add_controls(codec);
-	wm8753_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8753: failed to register card\n");
-		goto card_err;
+	wm8753_codec = codec;
+
+	for (i = 0; i < ARRAY_SIZE(wm8753_dai); i++)
+		wm8753_dai[i].dev = codec->dev;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err;
 	}
 
-	return ret;
+	ret = snd_soc_register_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai));
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAIs: %d\n", ret);
+		goto err_codec;
+	}
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-pcm_err:
-	kfree(codec->reg_cache);
+	return 0;
+
+err_codec:
+	run_delayed_work(&codec->delayed_work);
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8753);
 	return ret;
 }
 
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static struct snd_soc_device *wm8753_socdev;
+static void wm8753_unregister(struct wm8753_priv *wm8753)
+{
+	wm8753_set_bias_level(&wm8753->codec, SND_SOC_BIAS_OFF);
+	run_delayed_work(&wm8753->codec.delayed_work);
+	snd_soc_unregister_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai));
+	snd_soc_unregister_codec(&wm8753->codec);
+	kfree(wm8753);
+	wm8753_codec = NULL;
+}
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 
-/*
- * WM8753 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
-
 static int wm8753_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
+	struct snd_soc_codec *codec;
+	struct wm8753_priv *wm8753;
 
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
+	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
+	if (wm8753 == NULL)
+		return -ENOMEM;
 
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8753\n");
+        codec = &wm8753->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+        codec->control_data = i2c;
+        i2c_set_clientdata(i2c, wm8753);
 
-	return ret;
+        codec->dev = &i2c->dev;
+
+	return wm8753_register(wm8753);
 }
 
 static int wm8753_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
-	return 0;
+        struct wm8753_priv *wm8753 = i2c_get_clientdata(client);
+        wm8753_unregister(wm8753);
+        return 0;
 }
 
 static const struct i2c_device_id wm8753_i2c_id[] = {
@@ -1673,86 +1774,16 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
 
 static struct i2c_driver wm8753_i2c_driver = {
 	.driver = {
-		.name = "WM8753 I2C Codec",
+		.name = "wm8753",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8753_i2c_probe,
 	.remove =   wm8753_i2c_remove,
 	.id_table = wm8753_i2c_id,
 };
-
-static int wm8753_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8753_setup_data *setup)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&wm8753_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8753", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
-	}
-
-	return 0;
-
-err_driver:
-	i2c_del_driver(&wm8753_i2c_driver);
-	return -ENODEV;
-}
 #endif
 
 #if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8753_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	codec->control_data = spi;
-
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8753\n");
-
-	return ret;
-}
-
-static int __devexit wm8753_spi_remove(struct spi_device *spi)
-{
-	return 0;
-}
-
-static struct spi_driver wm8753_spi_driver = {
-	.driver = {
-		.name	= "wm8753",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8753_spi_probe,
-	.remove		= __devexit_p(wm8753_spi_remove),
-};
-
 static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
 {
 	struct spi_transfer t;
@@ -1776,120 +1807,69 @@ static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
 
 	return len;
 }
-#endif
 
-
-static int wm8753_probe(struct platform_device *pdev)
+static int __devinit wm8753_spi_probe(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8753_setup_data *setup;
 	struct snd_soc_codec *codec;
 	struct wm8753_priv *wm8753;
-	int ret = 0;
-
-	pr_info("WM8753 Audio Codec %s", WM8753_VERSION);
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
 
 	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
-	if (wm8753 == NULL) {
-		kfree(codec);
+	if (wm8753 == NULL)
 		return -ENOMEM;
-	}
 
-	codec->private_data = wm8753;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	wm8753_socdev = socdev;
-	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
+	codec = &wm8753->codec;
+	codec->control_data = spi;
+	codec->hw_write = (hw_write_t)wm8753_spi_write;
+	codec->dev = &spi->dev;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8753_add_i2c_device(pdev, setup);
-	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8753_spi_write;
-		ret = spi_register_driver(&wm8753_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
-#endif
+	spi->dev.driver_data = wm8753;
 
-	if (ret != 0) {
-		kfree(codec->private_data);
-		kfree(codec);
-	}
-	return ret;
+	return wm8753_register(wm8753);
 }
 
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
+static int __devexit wm8753_spi_remove(struct spi_device *spi)
 {
-	int ret;
-
-	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(dwork);
-
-	/* if there was any work waiting then we run it now and
-	 * wait for it's completion */
-	if (ret) {
-		schedule_delayed_work(dwork, 0);
-		flush_scheduled_work();
-	}
-	return ret;
+	struct wm8753_priv *wm8753 = spi->dev.driver_data;
+	wm8753_unregister(wm8753);
+	return 0;
 }
 
-/* power down chip */
-static int wm8753_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+static struct spi_driver wm8753_spi_driver = {
+	.driver = {
+		.name	= "wm8753",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8753_spi_probe,
+	.remove		= __devexit_p(wm8753_spi_remove),
+};
+#endif
 
-	if (codec->control_data)
-		wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	run_delayed_work(&codec->delayed_work);
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
+static int __init wm8753_modinit(void)
+{
+	int ret;
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_unregister_device(codec->control_data);
-	i2c_del_driver(&wm8753_i2c_driver);
+	ret = i2c_add_driver(&wm8753_i2c_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8753 I2C driver: %d\n", ret);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8753_spi_driver);
+	ret = spi_register_driver(&wm8753_spi_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8753 SPI driver: %d\n", ret);
 #endif
-	kfree(codec->private_data);
-	kfree(codec);
-
 	return 0;
 }
-
-struct snd_soc_codec_device soc_codec_dev_wm8753 = {
-	.probe = 	wm8753_probe,
-	.remove = 	wm8753_remove,
-	.suspend = 	wm8753_suspend,
-	.resume =	wm8753_resume,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
-
-static int __init wm8753_modinit(void)
-{
-	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
-}
 module_init(wm8753_modinit);
 
 static void __exit wm8753_exit(void)
 {
-	snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8753_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8753_spi_driver);
+#endif
 }
 module_exit(wm8753_exit);
 
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index f55704c..57b2ba2 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -77,12 +77,6 @@
 #define WM8753_BIASCTL		0x3d
 #define WM8753_ADCTL2		0x3f
 
-struct wm8753_setup_data {
-	int spi;
-	int i2c_bus;
-	unsigned short i2c_address;
-};
-
 #define WM8753_PLL1			0
 #define WM8753_PLL2			1
 
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 6767de1..46c5ea1 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1,
 
 };
 
-/* add non dapm controls */
-static int wm8900_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8900_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new wm8900_dapm_loutput2_control =
 SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0);
 
@@ -736,7 +720,7 @@ static int wm8900_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 
 	reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60;
@@ -1104,6 +1088,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops wm8900_dai_ops = {
+	.hw_params	= wm8900_hw_params,
+	.set_clkdiv	= wm8900_set_dai_clkdiv,
+	.set_pll	= wm8900_set_dai_pll,
+	.set_fmt	= wm8900_set_dai_fmt,
+	.digital_mute	= wm8900_digital_mute,
+};
+
 struct snd_soc_dai wm8900_dai = {
 	.name = "WM8900 HiFi",
 	.playback = {
@@ -1120,13 +1112,7 @@ struct snd_soc_dai wm8900_dai = {
 		.rates = WM8900_RATES,
 		.formats = WM8900_PCM_FORMATS,
 	 },
-	.ops = {
-		.hw_params = wm8900_hw_params,
-		 .set_clkdiv = wm8900_set_dai_clkdiv,
-		 .set_pll = wm8900_set_dai_pll,
-		 .set_fmt = wm8900_set_dai_fmt,
-		 .digital_mute = wm8900_digital_mute,
-	 },
+	.ops = &wm8900_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8900_dai);
 
@@ -1226,7 +1212,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
 static int wm8900_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8900_priv *wm8900 = codec->private_data;
 	int fll_out = wm8900->fll_out;
 	int fll_in  = wm8900->fll_in;
@@ -1250,7 +1236,7 @@ static int wm8900_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8900_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8900_priv *wm8900 = codec->private_data;
 	u16 *cache;
 	int i, ret;
@@ -1288,8 +1274,8 @@ static int wm8900_resume(struct platform_device *pdev)
 
 static struct snd_soc_codec *wm8900_codec;
 
-static int wm8900_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static __devinit int wm8900_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
 {
 	struct wm8900_priv *wm8900;
 	struct snd_soc_codec *codec;
@@ -1388,7 +1374,7 @@ err:
 	return ret;
 }
 
-static int wm8900_i2c_remove(struct i2c_client *client)
+static __devexit int wm8900_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_dai(&wm8900_dai);
 	snd_soc_unregister_codec(wm8900_codec);
@@ -1414,7 +1400,7 @@ static struct i2c_driver wm8900_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe = wm8900_i2c_probe,
-	.remove = wm8900_i2c_remove,
+	.remove = __devexit_p(wm8900_i2c_remove),
 	.id_table = wm8900_i2c_id,
 };
 
@@ -1430,7 +1416,7 @@ static int wm8900_probe(struct platform_device *pdev)
 	}
 
 	codec = wm8900_codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 
 	/* Register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -1439,7 +1425,8 @@ static int wm8900_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	wm8900_add_controls(codec);
+	snd_soc_add_controls(codec, wm8900_snd_controls,
+				ARRAY_SIZE(wm8900_snd_controls));
 	wm8900_add_widgets(codec);
 
 	ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index bde7454..8cf571f 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume",
 		 0, 63, 0, out_tlv),
 };
 
-static int wm8903_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8903_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new linput_mode_mux =
 	SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum);
 
@@ -1276,7 +1261,7 @@ static int wm8903_startup(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
@@ -1318,7 +1303,7 @@ static void wm8903_shutdown(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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1338,7 +1323,7 @@ static int wm8903_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	int fs = params_rate(params);
@@ -1512,6 +1497,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8903_dai_ops = {
+	.startup	= wm8903_startup,
+	.shutdown	= wm8903_shutdown,
+	.hw_params	= wm8903_hw_params,
+	.digital_mute	= wm8903_digital_mute,
+	.set_fmt	= wm8903_set_dai_fmt,
+	.set_sysclk	= wm8903_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8903_dai = {
 	.name = "WM8903",
 	.playback = {
@@ -1528,21 +1522,14 @@ struct snd_soc_dai wm8903_dai = {
 		 .rates = WM8903_CAPTURE_RATES,
 		 .formats = WM8903_FORMATS,
 	 },
-	.ops = {
-		 .startup = wm8903_startup,
-		 .shutdown = wm8903_shutdown,
-		 .hw_params = wm8903_hw_params,
-		 .digital_mute = wm8903_digital_mute,
-		 .set_fmt = wm8903_set_dai_fmt,
-		 .set_sysclk = wm8903_set_dai_sysclk
-	}
+	.ops = &wm8903_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8903_dai);
 
 static int wm8903_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1552,7 +1539,7 @@ static int wm8903_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8903_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct i2c_client *i2c = codec->control_data;
 	int i;
 	u16 *reg_cache = codec->reg_cache;
@@ -1577,8 +1564,8 @@ static int wm8903_resume(struct platform_device *pdev)
 
 static struct snd_soc_codec *wm8903_codec;
 
-static int wm8903_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
 {
 	struct wm8903_priv *wm8903;
 	struct snd_soc_codec *codec;
@@ -1684,7 +1671,7 @@ err:
 	return ret;
 }
 
-static int wm8903_i2c_remove(struct i2c_client *client)
+static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct snd_soc_codec *codec = i2c_get_clientdata(client);
 
@@ -1714,7 +1701,7 @@ static struct i2c_driver wm8903_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe    = wm8903_i2c_probe,
-	.remove   = wm8903_i2c_remove,
+	.remove   = __devexit_p(wm8903_i2c_remove),
 	.id_table = wm8903_i2c_id,
 };
 
@@ -1728,7 +1715,7 @@ static int wm8903_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	socdev->codec = wm8903_codec;
+	socdev->card->codec = wm8903_codec;
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -1737,8 +1724,9 @@ static int wm8903_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	wm8903_add_controls(socdev->codec);
-	wm8903_add_widgets(socdev->codec);
+	snd_soc_add_controls(socdev->card->codec, wm8903_snd_controls,
+				ARRAY_SIZE(wm8903_snd_controls));
+	wm8903_add_widgets(socdev->card->codec);
 
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1759,7 +1747,7 @@ err:
 static int wm8903_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 88ead7f..032dca2 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = {
 	SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
 };
 
-/* add non-DAPM controls */
-static int wm8971_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8971_snd_controls[i],
-					     codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * DAPM Controls
  */
@@ -546,7 +531,7 @@ static int wm8971_pcm_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8971_priv *wm8971 = codec->private_data;
 	u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3;
 	u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0;
@@ -619,6 +604,13 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
 #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8971_dai_ops = {
+	.hw_params	= wm8971_pcm_hw_params,
+	.digital_mute	= wm8971_mute,
+	.set_fmt	= wm8971_set_dai_fmt,
+	.set_sysclk	= wm8971_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8971_dai = {
 	.name = "WM8971",
 	.playback = {
@@ -633,12 +625,7 @@ struct snd_soc_dai wm8971_dai = {
 		.channels_max = 2,
 		.rates = WM8971_RATES,
 		.formats = WM8971_FORMATS,},
-	.ops = {
-		.hw_params = wm8971_pcm_hw_params,
-		.digital_mute = wm8971_mute,
-		.set_fmt = wm8971_set_dai_fmt,
-		.set_sysclk = wm8971_set_dai_sysclk,
-	},
+	.ops = &wm8971_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8971_dai);
 
@@ -652,7 +639,7 @@ static void wm8971_work(struct work_struct *work)
 static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -661,7 +648,7 @@ static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8971_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -692,7 +679,7 @@ static int wm8971_resume(struct platform_device *pdev)
 
 static int wm8971_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8971";
@@ -745,7 +732,8 @@ static int wm8971_init(struct snd_soc_device *socdev)
 	reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
 	wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
 
-	wm8971_add_controls(codec);
+	snd_soc_add_controls(codec, wm8971_snd_controls,
+				ARRAY_SIZE(wm8971_snd_controls));
 	wm8971_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -772,7 +760,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8971_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -873,7 +861,7 @@ static int wm8971_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8971;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -908,7 +896,7 @@ static int wm8971_probe(struct platform_device *pdev)
 static int wm8971_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index a5731fa..c518c3e 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -115,7 +115,7 @@ static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+	BUG_ON(reg >= ARRAY_SIZE(wm8990_reg));
 	return cache[reg];
 }
 
@@ -128,7 +128,7 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
 	u16 *cache = codec->reg_cache;
 
 	/* Reset register and reserved registers are uncached */
-	if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
+	if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg))
 		return;
 
 	cache[reg] = value;
@@ -418,21 +418,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
 
 };
 
-/* add non dapm controls */
-static int wm8990_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8990_snd_controls[i], codec,
-					NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * _DAPM_ Controls
  */
@@ -1178,7 +1163,7 @@ static int wm8990_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
 
 	audio1 &= ~WM8990_AIF_WL_MASK;
@@ -1347,6 +1332,15 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
  * 1. ADC/DAC on Primary Interface
  * 2. ADC on Primary Interface/DAC on secondary
  */
+static struct snd_soc_dai_ops wm8990_dai_ops = {
+	.hw_params	= wm8990_hw_params,
+	.digital_mute	= wm8990_mute,
+	.set_fmt	= wm8990_set_dai_fmt,
+	.set_clkdiv	= wm8990_set_dai_clkdiv,
+	.set_pll	= wm8990_set_dai_pll,
+	.set_sysclk	= wm8990_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8990_dai = {
 /* ADC/DAC on primary */
 	.name = "WM8990 ADC/DAC Primary",
@@ -1363,21 +1357,14 @@ struct snd_soc_dai wm8990_dai = {
 		.channels_max = 2,
 		.rates = WM8990_RATES,
 		.formats = WM8990_FORMATS,},
-	.ops = {
-		.hw_params = wm8990_hw_params,
-		.digital_mute = wm8990_mute,
-		.set_fmt = wm8990_set_dai_fmt,
-		.set_clkdiv = wm8990_set_dai_clkdiv,
-		.set_pll = wm8990_set_dai_pll,
-		.set_sysclk = wm8990_set_dai_sysclk,
-	},
+	.ops = &wm8990_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8990_dai);
 
 static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* we only need to suspend if we are a valid card */
 	if (!codec->card)
@@ -1390,7 +1377,7 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8990_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -1418,7 +1405,7 @@ static int wm8990_resume(struct platform_device *pdev)
  */
 static int wm8990_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 	int ret = 0;
 
@@ -1461,7 +1448,8 @@ static int wm8990_init(struct snd_soc_device *socdev)
 	wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
 	wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
 
-	wm8990_add_controls(codec);
+	snd_soc_add_controls(codec, wm8990_snd_controls,
+				ARRAY_SIZE(wm8990_snd_controls));
 	wm8990_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1495,7 +1483,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8990_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -1594,7 +1582,7 @@ static int wm8990_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8990;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1620,7 +1608,7 @@ static int wm8990_probe(struct platform_device *pdev)
 static int wm8990_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
new file mode 100644
index 0000000..3265817
--- /dev/null
+++ b/sound/soc/codecs/wm9705.c
@@ -0,0 +1,415 @@
+/*
+ * wm9705.c  --  ALSA Soc WM9705 codec support
+ *
+ * Copyright 2008 Ian Molton <spyro@f2s.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation; Version 2 of the  License only.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "wm9705.h"
+
+/*
+ * WM9705 register cache
+ */
+static const u16 wm9705_reg[] = {
+	0x6150, 0x8000, 0x8000, 0x8000, /* 0x0  */
+	0x0000, 0x8000, 0x8008, 0x8008, /* 0x8  */
+	0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */
+	0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */
+	0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */
+	0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */
+	0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */
+	0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */
+};
+
+static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = {
+	SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+	SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
+	SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+	SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
+	SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
+	SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
+	SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
+	SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+	SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1),
+	SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1),
+	SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1),
+	SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1),
+	SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1),
+	SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0),
+	SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0),
+	SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
+};
+
+static const char *wm9705_mic[] = {"Mic 1", "Mic 2"};
+static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC",
+	"Line", "Stereo Mix", "Mono Mix", "Phone"};
+
+static const struct soc_enum wm9705_enum_mic =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic);
+static const struct soc_enum wm9705_enum_rec_l =
+	SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel);
+static const struct soc_enum wm9705_enum_rec_r =
+	SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel);
+
+/* Headphone Mixer */
+static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = {
+	SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1),
+	SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1),
+	SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1),
+	SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1),
+	SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1),
+};
+
+/* Mic source */
+static const struct snd_kcontrol_new wm9705_mic_src_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_mic);
+
+/* Capture source */
+static const struct snd_kcontrol_new wm9705_capture_selectl_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_rec_l);
+static const struct snd_kcontrol_new wm9705_capture_selectr_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_rec_r);
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_mic_src_controls),
+	SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_capture_selectl_controls),
+	SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_capture_selectr_controls),
+	SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0,
+		&wm9705_hp_mixer_controls[0],
+		ARRAY_SIZE(wm9705_hp_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("HPOUTL"),
+	SND_SOC_DAPM_OUTPUT("HPOUTR"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+	SND_SOC_DAPM_OUTPUT("MONOOUT"),
+	SND_SOC_DAPM_INPUT("PHONE"),
+	SND_SOC_DAPM_INPUT("LINEINL"),
+	SND_SOC_DAPM_INPUT("LINEINR"),
+	SND_SOC_DAPM_INPUT("CDINL"),
+	SND_SOC_DAPM_INPUT("CDINR"),
+	SND_SOC_DAPM_INPUT("PCBEEP"),
+	SND_SOC_DAPM_INPUT("MIC1"),
+	SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+/* Audio map
+ * WM9705 has no switches to disable the route from the inputs to the HP mixer
+ * so in order to prevent active inputs from forcing the audio outputs to be
+ * constantly enabled, we use the mutes on those inputs to simulate such
+ * controls.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* HP mixer */
+	{"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"},
+	{"HP Mixer", "CD Playback Switch", "CD PGA"},
+	{"HP Mixer", "Mic Playback Switch", "Mic PGA"},
+	{"HP Mixer", "Phone Playback Switch", "Phone PGA"},
+	{"HP Mixer", "Line Playback Switch", "Line PGA"},
+	{"HP Mixer", NULL, "Left DAC"},
+	{"HP Mixer", NULL, "Right DAC"},
+
+	/* mono mixer */
+	{"Mono Mixer", NULL, "HP Mixer"},
+
+	/* outputs */
+	{"Headphone PGA", NULL, "HP Mixer"},
+	{"HPOUTL", NULL, "Headphone PGA"},
+	{"HPOUTR", NULL, "Headphone PGA"},
+	{"Line out PGA", NULL, "HP Mixer"},
+	{"LOUT", NULL, "Line out PGA"},
+	{"ROUT", NULL, "Line out PGA"},
+	{"Mono PGA", NULL, "Mono Mixer"},
+	{"MONOOUT", NULL, "Mono PGA"},
+
+	/* inputs */
+	{"CD PGA", NULL, "CDINL"},
+	{"CD PGA", NULL, "CDINR"},
+	{"Line PGA", NULL, "LINEINL"},
+	{"Line PGA", NULL, "LINEINR"},
+	{"Phone PGA", NULL, "PHONE"},
+	{"Mic Source", "Mic 1", "MIC1"},
+	{"Mic Source", "Mic 2", "MIC2"},
+	{"Mic PGA", NULL, "Mic Source"},
+	{"PCBEEP PGA", NULL, "PCBEEP"},
+
+	/* Left capture selector */
+	{"Left Capture Source", "Mic", "Mic Source"},
+	{"Left Capture Source", "CD", "CDINL"},
+	{"Left Capture Source", "Line", "LINEINL"},
+	{"Left Capture Source", "Stereo Mix", "HP Mixer"},
+	{"Left Capture Source", "Mono Mix", "HP Mixer"},
+	{"Left Capture Source", "Phone", "PHONE"},
+
+	/* Right capture source */
+	{"Right Capture Source", "Mic", "Mic Source"},
+	{"Right Capture Source", "CD", "CDINR"},
+	{"Right Capture Source", "Line", "LINEINR"},
+	{"Right Capture Source", "Stereo Mix", "HP Mixer"},
+	{"Right Capture Source", "Mono Mix", "HP Mixer"},
+	{"Right Capture Source", "Phone", "PHONE"},
+
+	{"ADC PGA", NULL, "Left Capture Source"},
+	{"ADC PGA", NULL, "Right Capture Source"},
+
+	/* ADC's */
+	{"Left ADC",  NULL, "ADC PGA"},
+	{"Right ADC", NULL, "ADC PGA"},
+};
+
+static int wm9705_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
+					ARRAY_SIZE(wm9705_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_widgets(codec);
+
+	return 0;
+}
+
+/* We use a register cache to enhance read performance. */
+static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	switch (reg) {
+	case AC97_RESET:
+	case AC97_VENDOR_ID1:
+	case AC97_VENDOR_ID2:
+		return soc_ac97_ops.read(codec->ac97, reg);
+	default:
+		reg = reg >> 1;
+
+		if (reg >= (ARRAY_SIZE(wm9705_reg)))
+			return -EIO;
+
+		return cache[reg];
+	}
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int val)
+{
+	u16 *cache = codec->reg_cache;
+
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	reg = reg >> 1;
+	if (reg < (ARRAY_SIZE(wm9705_reg)))
+		cache[reg] = val;
+
+	return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int reg;
+	u16 vra;
+
+	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = AC97_PCM_FRONT_DAC_RATE;
+	else
+		reg = AC97_PCM_LR_ADC_RATE;
+
+	return ac97_write(codec, reg, runtime->rate);
+}
+
+#define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
+			SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+			SNDRV_PCM_RATE_48000)
+
+static struct snd_soc_dai_ops wm9705_dai_ops = {
+	.prepare	= ac97_prepare,
+};
+
+struct snd_soc_dai wm9705_dai[] = {
+	{
+		.name = "AC97 HiFi",
+		.ac97_control = 1,
+		.playback = {
+			.stream_name = "HiFi Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.capture = {
+			.stream_name = "HiFi Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.ops = &wm9705_dai_ops,
+	},
+	{
+		.name = "AC97 Aux",
+		.playback = {
+			.stream_name = "Aux Playback",
+			.channels_min = 1,
+			.channels_max = 1,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+	}
+};
+EXPORT_SYMBOL_GPL(wm9705_dai);
+
+static int wm9705_reset(struct snd_soc_codec *codec)
+{
+	if (soc_ac97_ops.reset) {
+		soc_ac97_ops.reset(codec->ac97);
+		if (ac97_read(codec, 0) == wm9705_reg[0])
+			return 0; /* Success */
+	}
+
+	return -EIO;
+}
+
+static int wm9705_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	printk(KERN_INFO "WM9705 SoC Audio Codec\n");
+
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
+		return -ENOMEM;
+	codec = socdev->card->codec;
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto cache_err;
+	}
+	codec->reg_cache_size = sizeof(wm9705_reg);
+	codec->reg_cache_step = 2;
+
+	codec->name = "WM9705";
+	codec->owner = THIS_MODULE;
+	codec->dai = wm9705_dai;
+	codec->num_dai = ARRAY_SIZE(wm9705_dai);
+	codec->write = ac97_write;
+	codec->read = ac97_read;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9705: failed to register AC97 codec\n");
+		goto codec_err;
+	}
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0)
+		goto pcm_err;
+
+	ret = wm9705_reset(codec);
+	if (ret)
+		goto reset_err;
+
+	snd_soc_add_controls(codec, wm9705_snd_ac97_controls,
+				ARRAY_SIZE(wm9705_snd_ac97_controls));
+	wm9705_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9705: failed to register card\n");
+		goto pcm_err;
+	}
+
+	return 0;
+
+reset_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	snd_soc_free_ac97_codec(codec);
+codec_err:
+	kfree(codec->reg_cache);
+cache_err:
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
+	return ret;
+}
+
+static int wm9705_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	if (codec == NULL)
+		return 0;
+
+	snd_soc_dapm_free(socdev);
+	snd_soc_free_pcms(socdev);
+	snd_soc_free_ac97_codec(codec);
+	kfree(codec->reg_cache);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9705 = {
+	.probe = 	wm9705_soc_probe,
+	.remove = 	wm9705_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
+
+MODULE_DESCRIPTION("ASoC WM9705 driver");
+MODULE_AUTHOR("Ian Molton");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h
new file mode 100644
index 0000000..d380f11
--- /dev/null
+++ b/sound/soc/codecs/wm9705.h
@@ -0,0 +1,14 @@
+/*
+ * wm9705.h  --  WM9705 Soc Audio driver
+ */
+
+#ifndef _WM9705_H
+#define _WM9705_H
+
+#define WM9705_DAI_AC97_HIFI	0
+#define WM9705_DAI_AC97_AUX	1
+
+extern struct snd_soc_dai wm9705_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9705;
+
+#endif
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index af83d62..765cf1e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
 SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
 };
 
-/* add non dapm controls */
-static int wm9712_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm9712_snd_ac97_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path.
@@ -467,7 +452,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	else {
 		reg = reg >> 1;
 
-		if (reg > (ARRAY_SIZE(wm9712_reg)))
+		if (reg >= (ARRAY_SIZE(wm9712_reg)))
 			return -EIO;
 
 		return cache[reg];
@@ -481,7 +466,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg <= (ARRAY_SIZE(wm9712_reg)))
+	if (reg < (ARRAY_SIZE(wm9712_reg)))
 		cache[reg] = val;
 
 	return 0;
@@ -493,7 +478,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg;
 	u16 vra;
 
@@ -514,7 +499,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 vra, xsle;
 
 	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
@@ -532,6 +517,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
+	.prepare	= ac97_prepare,
+};
+
+static struct snd_soc_dai_ops wm9712_dai_ops_aux = {
+	.prepare	= ac97_aux_prepare,
+};
+
 struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -548,8 +541,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 2,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &wm9712_dai_ops_hifi,
 },
 {
 	.name = "AC97 Aux",
@@ -559,8 +551,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 1,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,},
+	.ops = &wm9712_dai_ops_aux,
 }
 };
 EXPORT_SYMBOL_GPL(wm9712_dai);
@@ -607,7 +598,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 	pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -616,7 +607,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 static int wm9712_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i, ret;
 	u16 *cache = codec->reg_cache;
 
@@ -652,10 +643,11 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache = kmemdup(wm9712_reg, sizeof(wm9712_reg), GFP_KERNEL);
@@ -698,7 +690,8 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
 
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm9712_add_controls(codec);
+	snd_soc_add_controls(codec, wm9712_snd_ac97_controls,
+				ARRAY_SIZE(wm9712_snd_ac97_controls));
 	wm9712_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -718,15 +711,15 @@ codec_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int wm9712_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index f3ca8aa..523bad0 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -32,7 +32,6 @@
 
 struct wm9713_priv {
 	u32 pll_in; /* PLL input frequency */
-	u32 pll_out; /* PLL output frequency */
 };
 
 static unsigned int ac97_read(struct snd_soc_codec *codec,
@@ -190,21 +189,6 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
 SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
 };
 
-/* add non dapm controls */
-static int wm9713_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm9713_snd_ac97_controls[i],
-					codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path using the current
@@ -636,7 +620,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	else {
 		reg = reg >> 1;
 
-		if (reg > (ARRAY_SIZE(wm9713_reg)))
+		if (reg >= (ARRAY_SIZE(wm9713_reg)))
 			return -EIO;
 
 		return cache[reg];
@@ -650,7 +634,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (reg < 0x7c)
 		soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg <= (ARRAY_SIZE(wm9713_reg)))
+	if (reg < (ARRAY_SIZE(wm9713_reg)))
 		cache[reg] = val;
 
 	return 0;
@@ -738,13 +722,13 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	struct _pll_div pll_div;
 
 	/* turn PLL off ? */
-	if (freq_in == 0 || freq_out == 0) {
+	if (freq_in == 0) {
 		/* disable PLL power and select ext source */
 		reg = ac97_read(codec, AC97_HANDSET_RATE);
 		ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
 		reg = ac97_read(codec, AC97_EXTENDED_MID);
 		ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
-		wm9713->pll_out = 0;
+		wm9713->pll_in = 0;
 		return 0;
 	}
 
@@ -788,7 +772,6 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
 	reg = ac97_read(codec, AC97_HANDSET_RATE);
 	ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
-	wm9713->pll_out = freq_out;
 	wm9713->pll_in = freq_in;
 
 	/* wait 10ms AC97 link frames for the link to stabilise */
@@ -957,13 +940,14 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u16 status;
+	u16 status, rate;
 
 	/* Gracefully shut down the voice interface. */
 	status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-	ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
+	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
 	schedule_timeout_interruptible(msecs_to_jiffies(1));
-	ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
 	ac97_write(codec, AC97_EXTENDED_MID, status);
 }
 
@@ -1021,6 +1005,27 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
+	.prepare	= ac97_hifi_prepare,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+};
+
+static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
+	.prepare	= ac97_aux_prepare,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+};
+
+static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
+	.hw_params	= wm9713_pcm_hw_params,
+	.shutdown	= wm9713_voiceshutdown,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+	.set_fmt	= wm9713_set_dai_fmt,
+	.set_tristate	= wm9713_set_dai_tristate,
+};
+
 struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -1037,10 +1042,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_hifi_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_hifi,
 	},
 	{
 	.name = "AC97 Aux",
@@ -1050,10 +1052,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 1,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_aux,
 	},
 	{
 	.name = "WM9713 Voice",
@@ -1069,14 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
-	.ops = {
-		.hw_params = wm9713_pcm_hw_params,
-		.shutdown = wm9713_voiceshutdown,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,
-		.set_fmt = wm9713_set_dai_fmt,
-		.set_tristate = wm9713_set_dai_tristate,
-	},
+	.ops = &wm9713_dai_ops_voice,
 	},
 };
 EXPORT_SYMBOL_GPL(wm9713_dai);
@@ -1132,7 +1124,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev,
 	pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 
 	/* Disable everything except touchpanel - that will be handled
@@ -1150,7 +1142,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev,
 static int wm9713_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm9713_priv *wm9713 = codec->private_data;
 	int i, ret;
 	u16 *cache = codec->reg_cache;
@@ -1164,8 +1156,8 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* do we need to re-start the PLL ? */
-	if (wm9713->pll_out)
-		wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
+	if (wm9713->pll_in)
+		wm9713_set_pll(codec, 0, wm9713->pll_in, 0);
 
 	/* only synchronise the codec if warm reset failed */
 	if (ret == 0) {
@@ -1191,10 +1183,11 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL);
@@ -1245,7 +1238,8 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	reg = ac97_read(codec, AC97_CD) & 0x7fff;
 	ac97_write(codec, AC97_CD, reg);
 
-	wm9713_add_controls(codec);
+	snd_soc_add_controls(codec, wm9713_snd_ac97_controls,
+				ARRAY_SIZE(wm9713_snd_ac97_controls));
 	wm9713_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0)
@@ -1265,15 +1259,15 @@ priv_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int wm9713_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index b502741..bd7392c 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -20,7 +20,7 @@ config SND_DAVINCI_SOC_EVM
 
 config SND_DAVINCI_SOC_SFFSDR
 	tristate "SoC Audio support for SFFSDR"
-	depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR
+	depends on SND_DAVINCI_SOC && MACH_SFFSDR
 	select SND_DAVINCI_SOC_I2S
 	select SND_SOC_PCM3008
 	select SFFSDR_FPGA
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 54851f3..9b90b34 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -186,7 +186,8 @@ static int __init evm_init(void)
 
 	platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
 	evm_snd_devdata.dev = &evm_snd_device->dev;
-	evm_snd_device->dev.platform_data = &evm_snd_data;
+	platform_device_add_data(evm_snd_device, &evm_snd_data,
+				 sizeof(evm_snd_data));
 
 	ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
 					    ARRAY_SIZE(evm_snd_resources));
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 0fee779..ffdb943 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
+static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+	.startup	= davinci_i2s_startup,
+	.trigger	= davinci_i2s_trigger,
+	.hw_params	= davinci_i2s_hw_params,
+	.set_fmt	= davinci_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
@@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = {
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = davinci_i2s_startup,
-		.trigger = davinci_i2s_trigger,
-		.hw_params = davinci_i2s_hw_params,
-		.set_fmt = davinci_i2s_set_dai_fmt,
-	},
+	.ops = &davinci_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 366049d..7af3b5b 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -286,7 +286,7 @@ static int davinci_pcm_mmap(struct snd_pcm_substream *substream,
 				     runtime->dma_bytes);
 }
 
-struct snd_pcm_ops davinci_pcm_ops = {
+static struct snd_pcm_ops davinci_pcm_ops = {
 	.open = 	davinci_pcm_open,
 	.close = 	davinci_pcm_close,
 	.ioctl = 	snd_pcm_lib_ioctl,
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 4935d1b..40eccfe 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -25,7 +25,9 @@
 
 #include <asm/dma.h>
 #include <asm/mach-types.h>
+#ifdef CONFIG_SFFSDR_FPGA
 #include <asm/plat-sffsdr/sffsdr-fpga.h>
+#endif
 
 #include <mach/mcbsp.h>
 #include <mach/edma.h>
@@ -34,31 +36,45 @@
 #include "davinci-pcm.h"
 #include "davinci-i2s.h"
 
+/*
+ * CLKX and CLKR are the inputs for the Sample Rate Generator.
+ * FSX and FSR are outputs, driven by the sample Rate Generator.
+ */
+#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B |	\
+		      SND_SOC_DAIFMT_CBM_CFS |	\
+		      SND_SOC_DAIFMT_IB_NF)
+
 static int sffsdr_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params,
-			    struct snd_soc_dai *dai)
+			    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	int fs;
 	int ret = 0;
 
-	/* Set cpu DAI configuration:
-	 * CLKX and CLKR are the inputs for the Sample Rate Generator.
-	 * FSX and FSR are outputs, driven by the sample Rate Generator. */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_RIGHT_J |
-				  SND_SOC_DAIFMT_CBM_CFS |
-				  SND_SOC_DAIFMT_IB_NF);
-	if (ret < 0)
-		return ret;
-
 	/* Fsref can be 32000, 44100 or 48000. */
 	fs = params_rate(params);
 
+#ifndef CONFIG_SFFSDR_FPGA
+	/* Without the FPGA module, the Fs is fixed at 44100 Hz */
+	if (fs != 44100) {
+		pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n");
+		return -EINVAL;
+	}
+#endif
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
+	if (ret < 0)
+		return ret;
+
 	pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
 
+#ifndef CONFIG_SFFSDR_FPGA
+	return 0;
+#else
 	return sffsdr_fpga_set_codec_fs(fs);
+#endif
 }
 
 static struct snd_soc_ops sffsdr_ops = {
@@ -127,7 +143,8 @@ static int __init sffsdr_init(void)
 
 	platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata);
 	sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev;
-	sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data;
+	platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data,
+				 sizeof(sffsdr_snd_data));
 
 	ret = platform_device_add_resources(sffsdr_snd_device,
 					    sffsdr_snd_resources,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 95c12b2..9fc9082 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,17 +1,18 @@
 config SND_SOC_OF_SIMPLE
 	tristate
 
+# ASoC platform support for the Freescale MPC8610 SOC.  This compiles drivers
+# for the SSI and the Elo DMA controller.  You will still need to select
+# a platform driver and a codec driver.
 config SND_SOC_MPC8610
-	bool "ALSA SoC support for the MPC8610 SOC"
-	depends on MPC8610_HPCD
-	default y if MPC8610
-	help
-	  Say Y if you want to add support for codecs attached to the SSI
-          device on an MPC8610.
+	tristate
+	depends on MPC8610
 
 config SND_SOC_MPC8610_HPCD
-	bool "ALSA SoC support for the Freescale MPC8610 HPCD board"
-	depends on SND_SOC_MPC8610
+	tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
+	# I2C is necessary for the CS4270 driver
+	depends on MPC8610_HPCD && I2C
+	select SND_SOC_MPC8610
 	select SND_SOC_CS4270
 	select SND_SOC_CS4270_VD33_ERRATA
 	default y if MPC8610_HPCD
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 035da4a..f85134c 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -2,10 +2,13 @@
 obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o
 
 # MPC8610 HPCD Machine Support
-obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
+snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
+obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
 
 # MPC8610 Platform Support
-obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+snd-soc-fsl-ssi-objs := fsl_ssi.o
+snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
 
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
 
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 64993ed..b3eb857 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -142,7 +142,8 @@ static const struct snd_pcm_hardware fsl_dma_hardware = {
 	.info   		= SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
-				  SNDRV_PCM_INFO_JOINT_DUPLEX,
+				  SNDRV_PCM_INFO_JOINT_DUPLEX |
+				  SNDRV_PCM_INFO_PAUSE,
 	.formats		= FSLDMA_PCM_FORMATS,
 	.rates  		= FSLDMA_PCM_RATES,
 	.rate_min       	= 5512,
@@ -464,11 +465,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 		sizeof(struct fsl_dma_link_descriptor);
 
 	for (i = 0; i < NUM_DMA_LINKS; i++) {
-		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
-
-		link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->next = cpu_to_be64(temp_link);
+		dma_private->link[i].next = cpu_to_be64(temp_link);
 
 		temp_link += sizeof(struct fsl_dma_link_descriptor);
 	}
@@ -525,79 +522,9 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
  * This function obtains hardware parameters about the opened stream and
  * programs the DMA controller accordingly.
  *
- * Note that due to a quirk of the SSI's STX register, the target address
- * for the DMA operations depends on the sample size.  So we don't program
- * the dest_addr (for playback -- source_addr for capture) fields in the
- * link descriptors here.  We do that in fsl_dma_prepare()
- */
-static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *hw_params)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct fsl_dma_private *dma_private = runtime->private_data;
-
-	dma_addr_t temp_addr;   /* Pointer to next period */
-
-	unsigned int i;
-
-	/* Get all the parameters we need */
-	size_t buffer_size = params_buffer_bytes(hw_params);
-	size_t period_size = params_period_bytes(hw_params);
-
-	/* Initialize our DMA tracking variables */
-	dma_private->period_size = period_size;
-	dma_private->num_periods = params_periods(hw_params);
-	dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
-	dma_private->dma_buf_next = dma_private->dma_buf_phys +
-		(NUM_DMA_LINKS * period_size);
-	if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
-		dma_private->dma_buf_next = dma_private->dma_buf_phys;
-
-	/*
-	 * The actual address in STX0 (destination for playback, source for
-	 * capture) is based on the sample size, but we don't know the sample
-	 * size in this function, so we'll have to adjust that later.  See
-	 * comments in fsl_dma_prepare().
-	 *
-	 * The DMA controller does not have a cache, so the CPU does not
-	 * need to tell it to flush its cache.  However, the DMA
-	 * controller does need to tell the CPU to flush its cache.
-	 * That's what the SNOOP bit does.
-	 *
-	 * Also, even though the DMA controller supports 36-bit addressing, for
-	 * simplicity we currently support only 32-bit addresses for the audio
-	 * buffer itself.
-	 */
-	temp_addr = substream->dma_buffer.addr;
-
-	for (i = 0; i < NUM_DMA_LINKS; i++) {
-		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
-
-		link->count = cpu_to_be32(period_size);
-
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			link->source_addr = cpu_to_be32(temp_addr);
-		else
-			link->dest_addr = cpu_to_be32(temp_addr);
-
-		temp_addr += period_size;
-	}
-
-	return 0;
-}
-
-/**
- * fsl_dma_prepare - prepare the DMA registers for playback.
- *
- * This function is called after the specifics of the audio data are known,
- * i.e. snd_pcm_runtime is initialized.
- *
- * In this function, we finish programming the registers of the DMA
- * controller that are dependent on the sample size.
- *
- * One of the drawbacks with big-endian is that when copying integers of
- * different sizes to a fixed-sized register, the address to which the
- * integer must be copied is dependent on the size of the integer.
+ * One drawback of big-endian is that when copying integers of different
+ * sizes to a fixed-sized register, the address to which the integer must be
+ * copied is dependent on the size of the integer.
  *
  * For example, if P is the address of a 32-bit register, and X is a 32-bit
  * integer, then X should be copied to address P.  However, if X is a 16-bit
@@ -613,22 +540,58 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
  * and 8 bytes at a time).  So we do not support packed 24-bit samples.
  * 24-bit data must be padded to 32 bits.
  */
-static int fsl_dma_prepare(struct snd_pcm_substream *substream)
+static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_dma_private *dma_private = runtime->private_data;
+
+	/* Number of bits per sample */
+	unsigned int sample_size =
+		snd_pcm_format_physical_width(params_format(hw_params));
+
+	/* Number of bytes per frame */
+	unsigned int frame_size = 2 * (sample_size / 8);
+
+	/* Bus address of SSI STX register */
+	dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
+
+	/* Size of the DMA buffer, in bytes */
+	size_t buffer_size = params_buffer_bytes(hw_params);
+
+	/* Number of bytes per period */
+	size_t period_size = params_period_bytes(hw_params);
+
+	/* Pointer to next period */
+	dma_addr_t temp_addr = substream->dma_buffer.addr;
+
+	/* Pointer to DMA controller */
 	struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
-	u32 mr;
+
+	u32 mr; /* DMA Mode Register */
+
 	unsigned int i;
-	dma_addr_t ssi_sxx_phys;	/* Bus address of SSI STX register */
-	unsigned int frame_size;	/* Number of bytes per frame */
 
-	ssi_sxx_phys = dma_private->ssi_sxx_phys;
+	/* Initialize our DMA tracking variables */
+	dma_private->period_size = period_size;
+	dma_private->num_periods = params_periods(hw_params);
+	dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
+	dma_private->dma_buf_next = dma_private->dma_buf_phys +
+		(NUM_DMA_LINKS * period_size);
+
+	if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
+		/* This happens if the number of periods == NUM_DMA_LINKS */
+		dma_private->dma_buf_next = dma_private->dma_buf_phys;
 
 	mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK |
 		  CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK);
 
-	switch (runtime->sample_bits) {
+	/* Due to a quirk of the SSI's STX register, the target address
+	 * for the DMA operations depends on the sample size.  So we calculate
+	 * that offset here.  While we're at it, also tell the DMA controller
+	 * how much data to transfer per sample.
+	 */
+	switch (sample_size) {
 	case 8:
 		mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
 		ssi_sxx_phys += 3;
@@ -641,12 +604,12 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream)
 		mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4;
 		break;
 	default:
+		/* We should never get here */
 		dev_err(substream->pcm->card->dev,
-			"unsupported sample size %u\n", runtime->sample_bits);
+			"unsupported sample size %u\n", sample_size);
 		return -EINVAL;
 	}
 
-	frame_size = runtime->frame_bits / 8;
 	/*
 	 * BWC should always be a multiple of the frame size.  BWC determines
 	 * how many bytes are sent/received before the DMA controller checks the
@@ -655,7 +618,6 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream)
 	 * capture, the receive FIFO is triggered when it contains one frame, so
 	 * we want to receive one frame at a time.
 	 */
-
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		mr |= CCSR_DMA_MR_BWC(2 * frame_size);
 	else
@@ -663,16 +625,48 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream)
 
 	out_be32(&dma_channel->mr, mr);
 
-	/*
-	 * Program the address of the DMA transfer to/from the SSI.
-	 */
 	for (i = 0; i < NUM_DMA_LINKS; i++) {
 		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
 
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		link->count = cpu_to_be32(period_size);
+
+		/* Even though the DMA controller supports 36-bit addressing,
+		 * for simplicity we allow only 32-bit addresses for the audio
+		 * buffer itself.  This was enforced in fsl_dma_new() with the
+		 * DMA mask.
+		 *
+		 * The snoop bit tells the DMA controller whether it should tell
+		 * the ECM to snoop during a read or write to an address. For
+		 * audio, we use DMA to transfer data between memory and an I/O
+		 * device (the SSI's STX0 or SRX0 register). Snooping is only
+		 * needed if there is a cache, so we need to snoop memory
+		 * addresses only.  For playback, that means we snoop the source
+		 * but not the destination.  For capture, we snoop the
+		 * destination but not the source.
+		 *
+		 * Note that failing to snoop properly is unlikely to cause
+		 * cache incoherency if the period size is larger than the
+		 * size of L1 cache.  This is because filling in one period will
+		 * flush out the data for the previous period.  So if you
+		 * increased period_bytes_min to a large enough size, you might
+		 * get more performance by not snooping, and you'll still be
+		 * okay.
+		 */
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			link->source_addr = cpu_to_be32(temp_addr);
+			link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+
 			link->dest_addr = cpu_to_be32(ssi_sxx_phys);
-		else
+			link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP);
+		} else {
 			link->source_addr = cpu_to_be32(ssi_sxx_phys);
+			link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP);
+
+			link->dest_addr = cpu_to_be32(temp_addr);
+			link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+		}
+
+		temp_addr += period_size;
 	}
 
 	return 0;
@@ -808,7 +802,6 @@ static struct snd_pcm_ops fsl_dma_ops = {
 	.ioctl  	= snd_pcm_lib_ioctl,
 	.hw_params      = fsl_dma_hw_params,
 	.hw_free	= fsl_dma_hw_free,
-	.prepare	= fsl_dma_prepare,
 	.pointer	= fsl_dma_pointer,
 };
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index c6d6eb7..169bca2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -72,6 +72,7 @@
  * @dev: struct device pointer
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
+ * @asynchronous: 0=synchronous mode, 1=asynchronous mode
  * @cpu_dai: the CPU DAI for this device
  * @dev_attr: the sysfs device attribute structure
  * @stats: SSI statistics
@@ -86,6 +87,7 @@ struct fsl_ssi_private {
 	struct device *dev;
 	unsigned int playback;
 	unsigned int capture;
+	int asynchronous;
 	struct snd_soc_dai cpu_dai;
 	struct device_attribute dev_attr;
 
@@ -301,9 +303,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 *
 		 * FIXME: Little-endian samples require a different shift dir
 		 */
-		clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK,
-			CCSR_SSI_SCR_TFR_CLK_DIS |
-			CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN);
+		clrsetbits_be32(&ssi->scr,
+			CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
+			CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
+			| (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN));
 
 		out_be32(&ssi->stcr,
 			 CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
@@ -382,10 +385,15 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 			SNDRV_PCM_HW_PARAM_RATE,
 			first_runtime->rate, first_runtime->rate);
 
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-			SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-			first_runtime->sample_bits,
-			first_runtime->sample_bits);
+		/* If we're in synchronous mode, then we need to constrain
+		 * the sample size as well.  We don't support independent sample
+		 * rates in asynchronous mode.
+		 */
+		if (!ssi_private->asynchronous)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+				SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+				first_runtime->sample_bits,
+				first_runtime->sample_bits);
 
 		ssi_private->second_stream = substream;
 	}
@@ -400,7 +408,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 }
 
 /**
- * fsl_ssi_prepare: prepare the SSI.
+ * fsl_ssi_hw_params - program the sample size
  *
  * Most of the SSI registers have been programmed in the startup function,
  * but the word length must be programmed here.  Unfortunately, programming
@@ -412,23 +420,27 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
  * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
  * clock master.
  */
-static int fsl_ssi_prepare(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
+static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
-
-	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	struct fsl_ssi_private *ssi_private = cpu_dai->private_data;
 
 	if (substream == ssi_private->first_stream) {
-		u32 wl;
+		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+		unsigned int sample_size =
+			snd_pcm_format_width(params_format(hw_params));
+		u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
 
 		/* The SSI should always be disabled at this points (SSIEN=0) */
-		wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
 
 		/* In synchronous mode, the SSI uses STCCR for capture */
-		clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+		if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
+		    !ssi_private->asynchronous)
+			clrsetbits_be32(&ssi->stccr,
+					CCSR_SSI_SxCCR_WL_MASK, wl);
+		else
+			clrsetbits_be32(&ssi->srccr,
+					CCSR_SSI_SxCCR_WL_MASK, wl);
 	}
 
 	return 0;
@@ -452,28 +464,33 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
+		clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
 			setbits32(&ssi->scr,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
 		} else {
-			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			long timeout = jiffies + 10;
+
 			setbits32(&ssi->scr,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
 
-			/*
-			 * I think we need this delay to allow time for the SSI
-			 * to put data into its FIFO.  Without it, ALSA starts
-			 * to complain about overruns.
+			/* Wait until the SSI has filled its FIFO. Without this
+			 * delay, ALSA complains about overruns.  When the FIFO
+			 * is full, the DMA controller initiates its first
+			 * transfer.  Until then, however, the DMA's DAR
+			 * register is zero, which translates to an
+			 * out-of-bounds pointer.  This makes ALSA think an
+			 * overrun has occurred.
 			 */
-			mdelay(1);
+			while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
+			       (jiffies < timeout));
+			if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
+				return -EIO;
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			clrbits32(&ssi->scr, CCSR_SSI_SCR_TE);
@@ -563,6 +580,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * fsl_ssi_dai_template: template CPU DAI for the SSI
  */
+static struct snd_soc_dai_ops fsl_ssi_dai_ops = {
+	.startup	= fsl_ssi_startup,
+	.hw_params	= fsl_ssi_hw_params,
+	.shutdown	= fsl_ssi_shutdown,
+	.trigger	= fsl_ssi_trigger,
+	.set_sysclk	= fsl_ssi_set_sysclk,
+	.set_fmt	= fsl_ssi_set_fmt,
+};
+
 static struct snd_soc_dai fsl_ssi_dai_template = {
 	.playback = {
 		/* The SSI does not support monaural audio. */
@@ -577,14 +603,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = fsl_ssi_startup,
-		.prepare = fsl_ssi_prepare,
-		.shutdown = fsl_ssi_shutdown,
-		.trigger = fsl_ssi_trigger,
-		.set_sysclk = fsl_ssi_set_sysclk,
-		.set_fmt = fsl_ssi_set_fmt,
-	},
+	.ops = &fsl_ssi_dai_ops,
 };
 
 /**
@@ -654,6 +673,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	ssi_private->ssi_phys = ssi_info->ssi_phys;
 	ssi_private->irq = ssi_info->irq;
 	ssi_private->dev = ssi_info->dev;
+	ssi_private->asynchronous = ssi_info->asynchronous;
 
 	ssi_private->dev->driver_data = fsl_ssi_dai;
 
@@ -704,6 +724,14 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 }
 EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
 
+static int __init fsl_ssi_init(void)
+{
+	printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n");
+
+	return 0;
+}
+module_init(fsl_ssi_init);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index 83b44d7..eade01f 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -208,6 +208,7 @@ struct ccsr_ssi {
  * ssi_phys: physical address of the SSI registers
  * irq: IRQ of this SSI
  * dev: struct device, used to create the sysfs statistics file
+ * asynchronous: 0=synchronous mode, 1=asynchronous mode
 */
 struct fsl_ssi_info {
 	unsigned int id;
@@ -215,6 +216,7 @@ struct fsl_ssi_info {
 	dma_addr_t ssi_phys;
 	unsigned int irq;
 	struct device *dev;
+	int asynchronous;
 };
 
 struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9eb1ce1..3aa729d 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -468,6 +468,16 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
+static struct snd_soc_dai_ops psc_i2s_dai_ops = {
+	.startup	= psc_i2s_startup,
+	.hw_params	= psc_i2s_hw_params,
+	.hw_free	= psc_i2s_hw_free,
+	.shutdown	= psc_i2s_shutdown,
+	.trigger	= psc_i2s_trigger,
+	.set_sysclk	= psc_i2s_set_sysclk,
+	.set_fmt	= psc_i2s_set_fmt,
+};
+
 static struct snd_soc_dai psc_i2s_dai_template = {
 	.playback = {
 		.channels_min = 2,
@@ -481,15 +491,7 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.rates = PSC_I2S_RATES,
 		.formats = PSC_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = psc_i2s_startup,
-		.hw_params = psc_i2s_hw_params,
-		.hw_free = psc_i2s_hw_free,
-		.shutdown = psc_i2s_shutdown,
-		.trigger = psc_i2s_trigger,
-		.set_sysclk = psc_i2s_set_sysclk,
-		.set_fmt = psc_i2s_set_fmt,
-	},
+	.ops = &psc_i2s_dai_ops,
 };
 
 /* ---------------------------------------------------------------------
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index acf39a6..ef67d1c 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -353,6 +353,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 	}
 	ssi_info.irq = machine_data->ssi_irq;
 
+	/* Do we want to use asynchronous mode? */
+	ssi_info.asynchronous =
+		of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0;
+	if (ssi_info.asynchronous)
+		dev_info(&ofdev->dev, "using asynchronous mode\n");
 
 	/* Map the global utilities registers. */
 	guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts");
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 4f7f040..675732e 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -8,7 +8,7 @@ config SND_OMAP_SOC_MCBSP
 
 config SND_OMAP_SOC_N810
 	tristate "SoC Audio support for Nokia N810"
-	depends on SND_OMAP_SOC && MACH_NOKIA_N810
+	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
 	select SND_OMAP_SOC_MCBSP
 	select OMAP_MUX
 	select SND_SOC_TLV320AIC3X
@@ -17,7 +17,7 @@ config SND_OMAP_SOC_N810
 
 config SND_OMAP_SOC_OSK5912
 	tristate "SoC Audio support for omap osk5912"
-	depends on SND_OMAP_SOC && MACH_OMAP_OSK
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TLV320AIC23
 	help
@@ -55,3 +55,13 @@ config SND_OMAP_SOC_OMAP3_PANDORA
 	select SND_SOC_TWL4030
 	help
 	  Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
+
+config SND_OMAP_SOC_OMAP3_BEAGLE
+	tristate "SoC Audio support for OMAP3 Beagle"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the Beagleboard.
+
+
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 76fedd9..0c9e4ac 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -12,6 +12,7 @@ snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
+snd-soc-omap3beagle-objs := omap3beagle.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
@@ -19,3 +20,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 25593fe..a6d1178 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -40,6 +40,13 @@
 #define N810_HEADSET_AMP_GPIO	10
 #define N810_SPEAKER_AMP_GPIO	101
 
+enum {
+	N810_JACK_DISABLED,
+	N810_JACK_HP,
+	N810_JACK_HS,
+	N810_JACK_MIC,
+};
+
 static struct clk *sys_clkout2;
 static struct clk *sys_clkout2_src;
 static struct clk *func96m_clk;
@@ -50,15 +57,32 @@ static int n810_dmic_func;
 
 static void n810_ext_control(struct snd_soc_codec *codec)
 {
+	int hp = 0, line1l = 0;
+
+	switch (n810_jack_func) {
+	case N810_JACK_HS:
+		line1l = 1;
+	case N810_JACK_HP:
+		hp = 1;
+		break;
+	case N810_JACK_MIC:
+		line1l = 1;
+		break;
+	}
+
 	if (n810_spk_func)
 		snd_soc_dapm_enable_pin(codec, "Ext Spk");
 	else
 		snd_soc_dapm_disable_pin(codec, "Ext Spk");
 
-	if (n810_jack_func)
+	if (hp)
 		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
 	else
 		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+	if (line1l)
+		snd_soc_dapm_enable_pin(codec, "LINE1L");
+	else
+		snd_soc_dapm_disable_pin(codec, "LINE1L");
 
 	if (n810_dmic_func)
 		snd_soc_dapm_enable_pin(codec, "DMic");
@@ -72,7 +96,7 @@ static int n810_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	snd_pcm_hw_constraint_minmax(runtime,
 				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
@@ -229,7 +253,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 };
 
 static const char *spk_function[] = {"Off", "On"};
-static const char *jack_function[] = {"Off", "Headphone"};
+static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
 static const char *input_function[] = {"ADC", "Digital Mic"};
 static const struct soc_enum n810_enum[] = {
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
@@ -248,20 +272,23 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = {
 
 static int n810_aic33_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	/* Not connected */
 	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
 	snd_soc_dapm_nc_pin(codec, "HPLCOM");
 	snd_soc_dapm_nc_pin(codec, "HPRCOM");
+	snd_soc_dapm_nc_pin(codec, "MIC3L");
+	snd_soc_dapm_nc_pin(codec, "MIC3R");
+	snd_soc_dapm_nc_pin(codec, "LINE1R");
+	snd_soc_dapm_nc_pin(codec, "LINE2L");
+	snd_soc_dapm_nc_pin(codec, "LINE2R");
 
 	/* Add N810 specific controls */
-	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&aic33_n810_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, aic33_n810_controls,
+				ARRAY_SIZE(aic33_n810_controls));
+	if (err < 0)
+		return err;
 
 	/* Add N810 specific widgets */
 	snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 05dd5ab..d6882be 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -461,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
+static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
+	.startup	= omap_mcbsp_dai_startup,
+	.shutdown	= omap_mcbsp_dai_shutdown,
+	.trigger	= omap_mcbsp_dai_trigger,
+	.hw_params	= omap_mcbsp_dai_hw_params,
+	.set_fmt	= omap_mcbsp_dai_set_dai_fmt,
+	.set_clkdiv	= omap_mcbsp_dai_set_clkdiv,
+	.set_sysclk	= omap_mcbsp_dai_set_dai_sysclk,
+};
+
 #define OMAP_MCBSP_DAI_BUILDER(link_id)				\
 {								\
 	.name = "omap-mcbsp-dai-"#link_id,			\
@@ -477,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
-	.ops = {						\
-		.startup = omap_mcbsp_dai_startup,		\
-		.shutdown = omap_mcbsp_dai_shutdown,		\
-		.trigger = omap_mcbsp_dai_trigger,		\
-		.hw_params = omap_mcbsp_dai_hw_params,		\
-		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
-		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
-		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
-	},							\
+	.ops = &omap_mcbsp_dai_ops,				\
 	.private_data = &mcbsp_data[(link_id)].bus_id,		\
 }
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index dd3bb29..8e1431c 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -265,7 +265,7 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream,
 				     runtime->dma_bytes);
 }
 
-struct snd_pcm_ops omap_pcm_ops = {
+static struct snd_pcm_ops omap_pcm_ops = {
 	.open		= omap_pcm_open,
 	.close		= omap_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index fcc2f5d..fe282d4 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -143,7 +143,7 @@ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_MIC("Mic (internal)", NULL),
 	SND_SOC_DAPM_MIC("Mic (external)", NULL),
 	SND_SOC_DAPM_LINE("Line In", NULL),
 };
@@ -155,16 +155,33 @@ static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
 };
 
 static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
-	{"INL", NULL, "Line In"},
-	{"INR", NULL, "Line In"},
-	{"INL", NULL, "Mic (Internal)"},
-	{"INR", NULL, "Mic (external)"},
+	{"AUXL", NULL, "Line In"},
+	{"AUXR", NULL, "Line In"},
+
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"Mic Bias 1", NULL, "Mic (internal)"},
+
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 2", NULL, "Mic (external)"},
 };
 
 static int omap3pandora_out_init(struct snd_soc_codec *codec)
 {
 	int ret;
 
+	/* All TWL4030 output pins are floating */
+	snd_soc_dapm_nc_pin(codec, "OUTL");
+	snd_soc_dapm_nc_pin(codec, "OUTR");
+	snd_soc_dapm_nc_pin(codec, "EARPIECE");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+	snd_soc_dapm_nc_pin(codec, "HSOL");
+	snd_soc_dapm_nc_pin(codec, "HSOR");
+	snd_soc_dapm_nc_pin(codec, "CARKITL");
+	snd_soc_dapm_nc_pin(codec, "CARKITR");
+	snd_soc_dapm_nc_pin(codec, "HFL");
+	snd_soc_dapm_nc_pin(codec, "HFR");
+
 	ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
 	if (ret < 0)
@@ -180,18 +197,11 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec)
 {
 	int ret;
 
-	/* All TWL4030 output pins are floating */
-	snd_soc_dapm_nc_pin(codec, "OUTL"),
-	snd_soc_dapm_nc_pin(codec, "OUTR"),
-	snd_soc_dapm_nc_pin(codec, "EARPIECE"),
-	snd_soc_dapm_nc_pin(codec, "PREDRIVEL"),
-	snd_soc_dapm_nc_pin(codec, "PREDRIVER"),
-	snd_soc_dapm_nc_pin(codec, "HSOL"),
-	snd_soc_dapm_nc_pin(codec, "HSOR"),
-	snd_soc_dapm_nc_pin(codec, "CARKITL"),
-	snd_soc_dapm_nc_pin(codec, "CARKITR"),
-	snd_soc_dapm_nc_pin(codec, "HFL"),
-	snd_soc_dapm_nc_pin(codec, "HFR"),
+	/* Not comnnected */
+	snd_soc_dapm_nc_pin(codec, "HSMIC");
+	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
 
 	ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_in_dapm_widgets));
@@ -251,10 +261,9 @@ static int __init omap3pandora_soc_init(void)
 {
 	int ret;
 
-	if (!machine_is_omap3_pandora()) {
-		pr_debug(PREFIX "Not OMAP3 Pandora\n");
+	if (!machine_is_omap3_pandora())
 		return -ENODEV;
-	}
+
 	pr_info("OMAP3 Pandora SoC init\n");
 
 	ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index cd41a94..a952a4e 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -186,13 +186,6 @@ static int __init osk_soc_init(void)
 		return -ENODEV;
 	}
 
-	if (clk_get_usecount(tlv320aic23_mclk) > 0) {
-		/* MCLK is already in use */
-		printk(KERN_WARNING
-		       "MCLK in use at %d Hz. We change it to %d Hz\n",
-		       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
-	}
-
 	/*
 	 * Configure 12 MHz output on MCLK.
 	 */
@@ -205,9 +198,8 @@ static int __init osk_soc_init(void)
 		}
 	}
 
-	printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
-	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
-	       clk_get_usecount(tlv320aic23_mclk));
+	printk(KERN_INFO "MCLK = %d [%d]\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
 
 	return 0;
 err1:
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index e226fa7..10f1c86 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -28,6 +28,7 @@
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
+#include <sound/jack.h>
 
 #include <asm/mach-types.h>
 #include <mach/hardware.h>
@@ -38,6 +39,8 @@
 #include "omap-pcm.h"
 #include "../codecs/twl4030.h"
 
+static struct snd_soc_card snd_soc_sdp3430;
+
 static int sdp3430_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
@@ -81,12 +84,121 @@ static struct snd_soc_ops sdp3430_ops = {
 	.hw_params = sdp3430_hw_params,
 };
 
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	{
+		.gpio = (OMAP_MAX_GPIO_LINES + 2),
+		.name = "hsdet-gpio",
+		.report = SND_JACK_HEADSET,
+		.debounce_time = 200,
+	},
+};
+
+/* SDP3430 machine DAPM */
+static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 1", NULL, "Ext Mic"},
+	{"Mic Bias 2", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Headset Stereophone (Headphone): HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+};
+
+static int sdp3430_twl4030_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	/* Add SDP3430 specific widgets */
+	ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets,
+				ARRAY_SIZE(sdp3430_twl4030_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up SDP3430 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* SDP3430 connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	/* TWL4030 not connected pins */
+	snd_soc_dapm_nc_pin(codec, "AUXL");
+	snd_soc_dapm_nc_pin(codec, "AUXR");
+	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+
+	snd_soc_dapm_nc_pin(codec, "OUTL");
+	snd_soc_dapm_nc_pin(codec, "OUTR");
+	snd_soc_dapm_nc_pin(codec, "EARPIECE");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+	snd_soc_dapm_nc_pin(codec, "CARKITL");
+	snd_soc_dapm_nc_pin(codec, "CARKITR");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	ret = snd_soc_jack_new(&snd_soc_sdp3430, "Headset Jack",
+				SND_JACK_HEADSET, &hs_jack);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+				hs_jack_pins);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
+	return ret;
+}
+
 /* Digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link sdp3430_dai = {
 	.name = "TWL4030",
 	.stream_name = "TWL4030",
 	.cpu_dai = &omap_mcbsp_dai[0],
 	.codec_dai = &twl4030_dai,
+	.init = sdp3430_twl4030_init,
 	.ops = &sdp3430_ops,
 };
 
@@ -142,6 +254,9 @@ module_init(sdp3430_soc_init);
 
 static void __exit sdp3430_soc_exit(void)
 {
+	snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
 	platform_device_unregister(sdp3430_snd_device);
 }
 module_exit(sdp3430_soc_exit);
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f82e106..5998ab3 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -61,6 +61,24 @@ config SND_PXA2XX_SOC_TOSA
 	  Say Y if you want to add support for SoC audio on Sharp
 	  Zaurus SL-C6000x models (Tosa).
 
+config SND_PXA2XX_SOC_E740
+	tristate "SoC AC97 Audio support for e740"
+	depends on SND_PXA2XX_SOC && MACH_E740
+	select SND_SOC_WM9705
+	select SND_PXA2XX_SOC_AC97
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  toshiba e740 PDA
+
+config SND_PXA2XX_SOC_E750
+	tristate "SoC AC97 Audio support for e750"
+	depends on SND_PXA2XX_SOC && MACH_E750
+	select SND_SOC_WM9705
+	select SND_PXA2XX_SOC_AC97
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  toshiba e750 PDA
+
 config SND_PXA2XX_SOC_E800
 	tristate "SoC AC97 Audio support for e800"
 	depends on SND_PXA2XX_SOC && MACH_E800
@@ -97,3 +115,12 @@ config SND_SOC_ZYLONITE
 	help
 	  Say Y if you want to add support for SoC audio on the
 	  Marvell Zylonite reference platform.
+
+config SND_PXA2XX_SOC_MIOA701
+        tristate "SoC Audio support for MIO A701"
+        depends on SND_PXA2XX_SOC && MACH_MIOA701
+        select SND_PXA2XX_SOC_AC97
+        select SND_SOC_WM9713
+        help
+          Say Y if you want to add support for SoC audio on the
+          MIO A701.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 08a9f27..8ed881c 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,17 +13,23 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
 snd-soc-corgi-objs := corgi.o
 snd-soc-poodle-objs := poodle.o
 snd-soc-tosa-objs := tosa.o
+snd-soc-e740-objs := e740_wm9705.o
+snd-soc-e750-objs := e750_wm9705.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
 snd-soc-zylonite-objs := zylonite.o
+snd-soc-mioa701-objs := mioa701_wm9713.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
 obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o
 obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
 obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index ec93066..d5be2b3 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -16,6 +16,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/timer.h>
+#include <linux/i2c.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
 #include <linux/gpio.h>
@@ -98,7 +99,7 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
 static int corgi_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	corgi_ext_control(codec);
@@ -273,18 +274,16 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
  */
 static int corgi_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
 	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 
 	/* Add corgi specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8731_corgi_controls,
+				ARRAY_SIZE(wm8731_corgi_controls));
+	if (err < 0)
+		return err;
 
 	/* Add corgi specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
@@ -315,19 +314,44 @@ static struct snd_soc_card snd_soc_corgi = {
 	.num_links = 1,
 };
 
-/* corgi audio private data */
-static struct wm8731_setup_data corgi_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
-
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
 	.card = &snd_soc_corgi,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &corgi_wm8731_setup,
 };
 
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = 0x1b;
+	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter 0\n");
+		return -ENODEV;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		printk(KERN_ERR "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
 static struct platform_device *corgi_snd_device;
 
 static int __init corgi_init(void)
@@ -338,6 +362,10 @@ static int __init corgi_init(void)
 	      machine_is_husky()))
 		return -ENODEV;
 
+	ret = wm8731_i2c_register();
+	if (ret != 0)
+		return ret;
+
 	corgi_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!corgi_snd_device)
 		return -ENOMEM;
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
new file mode 100644
index 0000000..7cd2f89
--- /dev/null
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -0,0 +1,211 @@
+/*
+ * e740-wm9705.c  --  SoC audio for e740
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+
+#define E740_AUDIO_OUT 1
+#define E740_AUDIO_IN  2
+
+static int e740_audio_power;
+
+static void e740_sync_audio_power(int status)
+{
+	gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status);
+	gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0);
+	gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0);
+}
+
+static int e740_mic_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		e740_audio_power |= E740_AUDIO_IN;
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		e740_audio_power &= ~E740_AUDIO_IN;
+
+	e740_sync_audio_power(e740_audio_power);
+
+	return 0;
+}
+
+static int e740_output_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		e740_audio_power |= E740_AUDIO_OUT;
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		e740_audio_power &= ~E740_AUDIO_OUT;
+
+	e740_sync_audio_power(e740_audio_power);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e740_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e740_output_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Output Amp", NULL, "LOUT"},
+	{"Output Amp", NULL, "ROUT"},
+	{"Output Amp", NULL, "MONOOUT"},
+
+	{"Speaker", NULL, "Output Amp"},
+	{"Headphone Jack", NULL, "Output Amp"},
+
+	{"MIC1", NULL, "Mic Amp"},
+	{"Mic Amp", NULL, "Mic (Internal)"},
+};
+
+static int e740_ac97_init(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_nc_pin(codec, "HPOUTL");
+	snd_soc_dapm_nc_pin(codec, "HPOUTR");
+	snd_soc_dapm_nc_pin(codec, "PHONE");
+	snd_soc_dapm_nc_pin(codec, "LINEINL");
+	snd_soc_dapm_nc_pin(codec, "LINEINR");
+	snd_soc_dapm_nc_pin(codec, "CDINL");
+	snd_soc_dapm_nc_pin(codec, "CDINR");
+	snd_soc_dapm_nc_pin(codec, "PCBEEP");
+	snd_soc_dapm_nc_pin(codec, "MIC2");
+
+	snd_soc_dapm_new_controls(codec, e740_dapm_widgets,
+					ARRAY_SIZE(e740_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link e740_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+		.init = e740_ac97_init,
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+	},
+};
+
+static struct snd_soc_card e740 = {
+	.name = "Toshiba e740",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = e740_dai,
+	.num_links = ARRAY_SIZE(e740_dai),
+};
+
+static struct snd_soc_device e740_snd_devdata = {
+	.card = &e740,
+	.codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e740_snd_device;
+
+static int __init e740_init(void)
+{
+	int ret;
+
+	if (!machine_is_e740())
+		return -ENODEV;
+
+	ret = gpio_request(GPIO_E740_MIC_ON,  "Mic amp");
+	if (ret)
+		return ret;
+
+	ret = gpio_request(GPIO_E740_AMP_ON, "Output amp");
+	if (ret)
+		goto free_mic_amp_gpio;
+
+	ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power");
+	if (ret)
+		goto free_op_amp_gpio;
+
+	/* Disable audio */
+	ret = gpio_direction_output(GPIO_E740_MIC_ON, 0);
+	if (ret)
+		goto free_apwr_gpio;
+	ret = gpio_direction_output(GPIO_E740_AMP_ON, 0);
+	if (ret)
+		goto free_apwr_gpio;
+	ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1);
+	if (ret)
+		goto free_apwr_gpio;
+
+	e740_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!e740_snd_device) {
+		ret = -ENOMEM;
+		goto free_apwr_gpio;
+	}
+
+	platform_set_drvdata(e740_snd_device, &e740_snd_devdata);
+	e740_snd_devdata.dev = &e740_snd_device->dev;
+	ret = platform_device_add(e740_snd_device);
+
+	if (!ret)
+		return 0;
+
+/* Fail gracefully */
+	platform_device_put(e740_snd_device);
+free_apwr_gpio:
+	gpio_free(GPIO_E740_WM9705_nAVDD2);
+free_op_amp_gpio:
+	gpio_free(GPIO_E740_AMP_ON);
+free_mic_amp_gpio:
+	gpio_free(GPIO_E740_MIC_ON);
+
+	return ret;
+}
+
+static void __exit e740_exit(void)
+{
+	platform_device_unregister(e740_snd_device);
+}
+
+module_init(e740_init);
+module_exit(e740_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e740");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
new file mode 100644
index 0000000..8dceccc
--- /dev/null
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -0,0 +1,187 @@
+/*
+ * e750-wm9705.c  --  SoC audio for e750
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1);
+
+	return 0;
+}
+
+static int e750_hp_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E750_HP_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E750_HP_AMP_OFF, 1);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Amp", NULL, "HPOUTL"},
+	{"Headphone Amp", NULL, "HPOUTR"},
+	{"Headphone Jack", NULL, "Headphone Amp"},
+
+	{"Speaker Amp", NULL, "MONOOUT"},
+	{"Speaker", NULL, "Speaker Amp"},
+
+	{"MIC1", NULL, "Mic (Internal)"},
+};
+
+static int e750_ac97_init(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_nc_pin(codec, "LOUT");
+	snd_soc_dapm_nc_pin(codec, "ROUT");
+	snd_soc_dapm_nc_pin(codec, "PHONE");
+	snd_soc_dapm_nc_pin(codec, "LINEINL");
+	snd_soc_dapm_nc_pin(codec, "LINEINR");
+	snd_soc_dapm_nc_pin(codec, "CDINL");
+	snd_soc_dapm_nc_pin(codec, "CDINR");
+	snd_soc_dapm_nc_pin(codec, "PCBEEP");
+	snd_soc_dapm_nc_pin(codec, "MIC2");
+
+	snd_soc_dapm_new_controls(codec, e750_dapm_widgets,
+					ARRAY_SIZE(e750_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link e750_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+		.init = e750_ac97_init,
+		/* use ops to check startup state */
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+	},
+};
+
+static struct snd_soc_card e750 = {
+	.name = "Toshiba e750",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = e750_dai,
+	.num_links = ARRAY_SIZE(e750_dai),
+};
+
+static struct snd_soc_device e750_snd_devdata = {
+	.card = &e750,
+	.codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e750_snd_device;
+
+static int __init e750_init(void)
+{
+	int ret;
+
+	if (!machine_is_e750())
+		return -ENODEV;
+
+	ret = gpio_request(GPIO_E750_HP_AMP_OFF,  "Headphone amp");
+	if (ret)
+		return ret;
+
+	ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp");
+	if (ret)
+		goto free_hp_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
+	e750_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!e750_snd_device) {
+		ret = -ENOMEM;
+		goto free_spk_amp_gpio;
+	}
+
+	platform_set_drvdata(e750_snd_device, &e750_snd_devdata);
+	e750_snd_devdata.dev = &e750_snd_device->dev;
+	ret = platform_device_add(e750_snd_device);
+
+	if (!ret)
+		return 0;
+
+/* Fail gracefully */
+	platform_device_put(e750_snd_device);
+free_spk_amp_gpio:
+	gpio_free(GPIO_E750_SPK_AMP_OFF);
+free_hp_amp_gpio:
+	gpio_free(GPIO_E750_HP_AMP_OFF);
+
+	return ret;
+}
+
+static void __exit e750_exit(void)
+{
+	platform_device_unregister(e750_snd_device);
+	gpio_free(GPIO_E750_SPK_AMP_OFF);
+	gpio_free(GPIO_E750_HP_AMP_OFF);
+}
+
+module_init(e750_init);
+module_exit(e750_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e750");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index ac294c7..bc019cd 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -1,8 +1,6 @@
 /*
  * e800-wm9712.c  --  SoC audio for e800
  *
- * Based on tosa.c
- *
  * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -13,7 +11,7 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/device.h>
+#include <linux/gpio.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -22,20 +20,84 @@
 
 #include <asm/mach-types.h>
 #include <mach/audio.h>
+#include <mach/eseries-gpio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_card e800;
+static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E800_SPK_AMP_ON, 1);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E800_SPK_AMP_ON, 0);
 
-static struct snd_soc_dai_link e800_dai[] = {
+	return 0;
+}
+
+static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E800_HP_AMP_OFF, 1);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+	{"Headphone Jack", NULL, "Headphone Amp"},
+
+	{"Speaker Amp", NULL, "MONOOUT"},
+	{"Speaker", NULL, "Speaker Amp"},
+
+	{"MIC1", NULL, "Mic (Internal1)"},
+	{"MIC2", NULL, "Mic (Internal2)"},
+};
+
+static int e800_ac97_init(struct snd_soc_codec *codec)
 {
-	.name = "AC97 Aux",
-	.stream_name = "AC97 Aux",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
-},
+	snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
+					ARRAY_SIZE(e800_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link e800_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+		.init = e800_ac97_init,
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	},
 };
 
 static struct snd_soc_card e800 = {
@@ -59,6 +121,22 @@ static int __init e800_init(void)
 	if (!machine_is_e800())
 		return -ENODEV;
 
+	ret = gpio_request(GPIO_E800_HP_AMP_OFF,  "Headphone amp");
+	if (ret)
+		return ret;
+
+	ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp");
+	if (ret)
+		goto free_hp_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
 	e800_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!e800_snd_device)
 		return -ENOMEM;
@@ -67,8 +145,15 @@ static int __init e800_init(void)
 	e800_snd_devdata.dev = &e800_snd_device->dev;
 	ret = platform_device_add(e800_snd_device);
 
-	if (ret)
-		platform_device_put(e800_snd_device);
+	if (!ret)
+		return 0;
+
+/* Fail gracefully */
+	platform_device_put(e800_snd_device);
+free_spk_amp_gpio:
+	gpio_free(GPIO_E800_SPK_AMP_ON);
+free_hp_amp_gpio:
+	gpio_free(GPIO_E800_HP_AMP_OFF);
 
 	return ret;
 }
@@ -76,6 +161,8 @@ static int __init e800_init(void)
 static void __exit e800_exit(void)
 {
 	platform_device_unregister(e800_snd_device);
+	gpio_free(GPIO_E800_SPK_AMP_ON);
+	gpio_free(GPIO_E800_HP_AMP_OFF);
 }
 
 module_init(e800_init);
@@ -84,4 +171,4 @@ module_exit(e800_exit);
 /* Module information */
 MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
 MODULE_DESCRIPTION("ALSA SoC driver for e800");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
new file mode 100644
index 0000000..19eda8b
--- /dev/null
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -0,0 +1,250 @@
+/*
+ * Handles the Mitac mioa701 SoC system
+ *
+ * Copyright (C) 2008 Robert Jarzmik
+ *
+ * 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 in version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ * This is a little schema of the sound interconnections :
+ *
+ *    Sagem X200                 Wolfson WM9713
+ *    +--------+             +-------------------+      Rear Speaker
+ *    |        |             |                   |           /-+
+ *    |        +--->----->---+MONOIN         SPKL+--->----+-+  |
+ *    |  GSM   |             |                   |        | |  |
+ *    |        +--->----->---+PCBEEP         SPKR+--->----+-+  |
+ *    |  CHIP  |             |                   |           \-+
+ *    |        +---<-----<---+MONO               |
+ *    |        |             |                   |      Front Speaker
+ *    +--------+             |                   |           /-+
+ *                           |                HPL+--->----+-+  |
+ *                           |                   |        | |  |
+ *                           |               OUT3+--->----+-+  |
+ *                           |                   |           \-+
+ *                           |                   |
+ *                           |                   |     Front Micro
+ *                           |                   |         +
+ *                           |               MIC1+-----<--+o+
+ *                           |                   |         +
+ *                           +-------------------+        ---
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "../codecs/wm9713.h"
+
+#define ARRAY_AND_SIZE(x)	(x), ARRAY_SIZE(x)
+
+#define AC97_GPIO_PULL		0x58
+
+/* Use GPIO8 for rear speaker amplifier */
+static int rear_amp_power(struct snd_soc_codec *codec, int power)
+{
+	unsigned short reg;
+
+	if (power) {
+		reg = snd_soc_read(codec, AC97_GPIO_CFG);
+		snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100);
+		reg = snd_soc_read(codec, AC97_GPIO_PULL);
+		snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15));
+	} else {
+		reg = snd_soc_read(codec, AC97_GPIO_CFG);
+		snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100);
+		reg = snd_soc_read(codec, AC97_GPIO_PULL);
+		snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15));
+	}
+
+	return 0;
+}
+
+static int rear_amp_event(struct snd_soc_dapm_widget *widget,
+			  struct snd_kcontrol *kctl, int event)
+{
+	struct snd_soc_codec *codec = widget->codec;
+
+	return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event));
+}
+
+/* mioa701 machine dapm widgets */
+static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Front Speaker", NULL),
+	SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event),
+	SND_SOC_DAPM_MIC("Headset", NULL),
+	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line In", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Front Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Call Mic */
+	{"Mic Bias", NULL, "Front Mic"},
+	{"MIC1", NULL, "Mic Bias"},
+
+	/* Headset Mic */
+	{"LINEL", NULL, "Headset Mic"},
+	{"LINER", NULL, "Headset Mic"},
+
+	/* GSM Module */
+	{"MONOIN", NULL, "GSM Line Out"},
+	{"PCBEEP", NULL, "GSM Line Out"},
+	{"GSM Line In", NULL, "MONO"},
+
+	/* headphone connected to HPL, HPR */
+	{"Headset", NULL, "HPL"},
+	{"Headset", NULL, "HPR"},
+
+	/* front speaker connected to HPL, OUT3 */
+	{"Front Speaker", NULL, "HPL"},
+	{"Front Speaker", NULL, "OUT3"},
+
+	/* rear speaker connected to SPKL, SPKR */
+	{"Rear Speaker", NULL, "SPKL"},
+	{"Rear Speaker", NULL, "SPKR"},
+};
+
+static int mioa701_wm9713_init(struct snd_soc_codec *codec)
+{
+	unsigned short reg;
+
+	/* Add mioa701 specific widgets */
+	snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets));
+
+	/* Set up mioa701 specific audio path audio_mapnects */
+	snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map));
+
+	/* Prepare GPIO8 for rear speaker amplifier */
+	reg = codec->read(codec, AC97_GPIO_CFG);
+	codec->write(codec, AC97_GPIO_CFG, reg | 0x0100);
+
+	/* Prepare MIC input */
+	reg = codec->read(codec, AC97_3D_CONTROL);
+	codec->write(codec, AC97_3D_CONTROL, reg | 0xc000);
+
+	snd_soc_dapm_enable_pin(codec, "Front Speaker");
+	snd_soc_dapm_enable_pin(codec, "Rear Speaker");
+	snd_soc_dapm_enable_pin(codec, "Front Mic");
+	snd_soc_dapm_enable_pin(codec, "GSM Line In");
+	snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_ops mioa701_ops;
+
+static struct snd_soc_dai_link mioa701_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+		.init = mioa701_wm9713_init,
+		.ops = &mioa701_ops,
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+		.ops = &mioa701_ops,
+	},
+};
+
+static struct snd_soc_card mioa701 = {
+	.name = "MioA701",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = mioa701_dai,
+	.num_links = ARRAY_SIZE(mioa701_dai),
+};
+
+static struct snd_soc_device mioa701_snd_devdata = {
+	.card = &mioa701,
+	.codec_dev = &soc_codec_dev_wm9713,
+};
+
+static struct platform_device *mioa701_snd_device;
+
+static int mioa701_wm9713_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (!machine_is_mioa701())
+		return -ENODEV;
+
+	dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+		 "lead to overheating and possible destruction of your device."
+		 "Do not use without a good knowledge of mio's board design!\n");
+
+	mioa701_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!mioa701_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata);
+	mioa701_snd_devdata.dev = &mioa701_snd_device->dev;
+
+	ret = platform_device_add(mioa701_snd_device);
+	if (!ret)
+		return 0;
+
+	platform_device_put(mioa701_snd_device);
+	return ret;
+}
+
+static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
+{
+	platform_device_unregister(mioa701_snd_device);
+	return 0;
+}
+
+static struct platform_driver mioa701_wm9713_driver = {
+	.probe		= mioa701_wm9713_probe,
+	.remove		= __devexit_p(mioa701_wm9713_remove),
+	.driver		= {
+		.name		= "mioa701-wm9713",
+		.owner		= THIS_MODULE,
+	},
+};
+
+static int __init mioa701_asoc_init(void)
+{
+	return platform_driver_register(&mioa701_wm9713_driver);
+}
+
+static void __exit mioa701_asoc_exit(void)
+{
+	platform_driver_unregister(&mioa701_wm9713_driver);
+}
+
+module_init(mioa701_asoc_init);
+module_exit(mioa701_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
+MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 4a9cf30..48a73f6 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -55,7 +55,7 @@ static void palm27x_ext_control(struct snd_soc_codec *codec)
 static int palm27x_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	palm27x_ext_control(codec);
@@ -146,19 +146,16 @@ static const struct snd_kcontrol_new palm27x_controls[] = {
 
 static int palm27x_ac97_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "OUT3");
 	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add palm27x specific controls */
-	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&palm27x_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, palm27x_controls,
+				ARRAY_SIZE(palm27x_controls));
+	if (err < 0)
+		return err;
 
 	/* add palm27x specific widgets */
 	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index cad2c4c..a51058f 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -17,6 +17,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/timer.h>
+#include <linux/i2c.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
@@ -75,7 +76,7 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
 static int poodle_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	poodle_ext_control(codec);
@@ -238,19 +239,17 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
  */
 static int poodle_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
 	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 	snd_soc_dapm_enable_pin(codec, "MICIN");
 
 	/* Add poodle specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8731_poodle_controls,
+				ARRAY_SIZE(wm8731_poodle_controls));
+	if (err < 0)
+		return err;
 
 	/* Add poodle specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
@@ -281,17 +280,42 @@ static struct snd_soc_card snd_soc_poodle = {
 	.num_links = 1,
 };
 
-/* poodle audio private data */
-static struct wm8731_setup_data poodle_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = 0x1b;
+	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter 0\n");
+		return -ENODEV;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		printk(KERN_ERR "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		return -ENODEV;
+	}
+
+	return 0;
+}
 
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
 	.card = &snd_soc_poodle,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &poodle_wm8731_setup,
 };
 
 static struct platform_device *poodle_snd_device;
@@ -303,6 +327,10 @@ static int __init poodle_init(void)
 	if (!machine_is_poodle())
 		return -ENODEV;
 
+	ret = wm8731_i2c_register();
+	if (ret != 0)
+		return ret;
+
 	locomo_gpio_set_dir(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_AMP_ON, 0);
 	/* should we mute HP at startup - burning power ?*/
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 1dfdf66..7acd3fe 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -1,4 +1,3 @@
-#define DEBUG
 /*
  * pxa-ssp.c  --  ALSA Soc Audio Layer
  *
@@ -21,6 +20,8 @@
 #include <linux/clk.h>
 #include <linux/io.h>
 
+#include <asm/irq.h>
+
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
@@ -221,9 +222,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 	int ret = 0;
 
 	if (!cpu_dai->active) {
-		ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
-		if (ret < 0)
-			return ret;
+		priv->dev.port = cpu_dai->id + 1;
+		priv->dev.irq = NO_IRQ;
+		clk_enable(priv->dev.ssp->clk);
 		ssp_disable(&priv->dev);
 	}
 	return ret;
@@ -238,7 +239,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 
 	if (!cpu_dai->active) {
 		ssp_disable(&priv->dev);
-		ssp_exit(&priv->dev);
+		clk_disable(priv->dev.ssp->clk);
 	}
 }
 
@@ -298,7 +299,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	int val;
 
 	u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
-		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 
 	dev_dbg(&ssp->pdev->dev,
 		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
@@ -326,7 +327,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	case PXA_SSP_CLK_AUDIO:
 		priv->sysclk = 0;
 		ssp_set_scr(&priv->dev, 1);
-		sscr0 |= SSCR0_ADC;
+		sscr0 |= SSCR0_ACS;
 		break;
 	default:
 		return -ENODEV;
@@ -520,9 +521,20 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	u32 sscr1;
 	u32 sspsp;
 
+	/* check if we need to change anything at all */
+	if (priv->dai_fmt == fmt)
+		return 0;
+
+	/* we can only change the settings if the port is not in use */
+	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) {
+		dev_err(&ssp->pdev->dev,
+			"can't change hardware dai format: stream is in use");
+		return -EINVAL;
+	}
+
 	/* reset port settings */
 	sscr0 = ssp_read_reg(ssp, SSCR0) &
-		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
 	sspsp = 0;
 
@@ -545,18 +557,18 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr0 |= SSCR0_PSP;
 		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
 
+		/* See hw_params() */
 		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 		case SND_SOC_DAIFMT_NB_NF:
-			sspsp |= SSPSP_FSRT;
+			sspsp |= SSPSP_SFRMP;
 			break;
 		case SND_SOC_DAIFMT_NB_IF:
-			sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
 			break;
 		case SND_SOC_DAIFMT_IB_IF:
-			sspsp |= SSPSP_SFRMP;
+			sspsp |= SSPSP_SCMODE(3);
 			break;
 		default:
 			return -EINVAL;
@@ -642,34 +654,65 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			sscr0 |= SSCR0_FPCKE;
 #endif
 		sscr0 |= SSCR0_DataSize(16);
-		if (params_channels(params) > 1)
-			sscr0 |= SSCR0_EDSS;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
-		/* we must be in network mode (2 slots) for 24 bit stereo */
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
-		/* we must be in network mode (2 slots) for 32 bit stereo */
 		break;
 	}
 	ssp_write_reg(ssp, SSCR0, sscr0);
 
 	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		/* Cleared when the DAI format is set */
-		sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+	       sspsp = ssp_read_reg(ssp, SSPSP);
+
+		if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
+		     (width == 16)) {
+			/* This is a special case where the bitclk is 64fs
+			* and we're not dealing with 2*32 bits of audio
+			* samples.
+			*
+			* The SSP values used for that are all found out by
+			* trying and failing a lot; some of the registers
+			* needed for that mode are only available on PXA3xx.
+			*/
+
+#ifdef CONFIG_PXA3xx
+			if (!cpu_is_pxa3xx())
+				return -EINVAL;
+
+			sspsp |= SSPSP_SFRMWDTH(width * 2);
+			sspsp |= SSPSP_SFRMDLY(width * 4);
+			sspsp |= SSPSP_EDMYSTOP(3);
+			sspsp |= SSPSP_DMYSTOP(3);
+			sspsp |= SSPSP_DMYSTRT(1);
+#else
+			return -EINVAL;
+#endif
+		} else {
+			/* The frame width is the width the LRCLK is
+			 * asserted for; the delay is expressed in
+			 * half cycle units.  We need the extra cycle
+			 * because the data starts clocking out one BCLK
+			 * after LRCLK changes polarity.
+			 */
+			sspsp |= SSPSP_SFRMWDTH(width + 1);
+			sspsp |= SSPSP_SFRMDLY((width + 1) * 2);
+			sspsp |= SSPSP_DMYSTRT(1);
+		}
+
 		ssp_write_reg(ssp, SSPSP, sspsp);
 		break;
 	default:
 		break;
 	}
 
-	/* We always use a network mode so we always require TDM slots
+	/* When we use a network mode, we always require TDM slots
 	 * - complain loudly and fail if they've not been set up yet.
 	 */
-	if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+	if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
 		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
 		return -EINVAL;
 	}
@@ -751,7 +794,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
 	if (!priv)
 		return -ENOMEM;
 
-	priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+	priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio");
 	if (priv->dev.ssp == NULL) {
 		ret = -ENODEV;
 		goto err_priv;
@@ -782,6 +825,19 @@ static void pxa_ssp_remove(struct platform_device *pdev,
 			    SNDRV_PCM_FMTBIT_S24_LE |	\
 			    SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+	.startup	= pxa_ssp_startup,
+	.shutdown	= pxa_ssp_shutdown,
+	.trigger	= pxa_ssp_trigger,
+	.hw_params	= pxa_ssp_hw_params,
+	.set_sysclk	= pxa_ssp_set_dai_sysclk,
+	.set_clkdiv	= pxa_ssp_set_dai_clkdiv,
+	.set_pll	= pxa_ssp_set_dai_pll,
+	.set_fmt	= pxa_ssp_set_dai_fmt,
+	.set_tdm_slot	= pxa_ssp_set_dai_tdm_slot,
+	.set_tristate	= pxa_ssp_set_dai_tristate,
+};
+
 struct snd_soc_dai pxa_ssp_dai[] = {
 	{
 		.name = "pxa2xx-ssp1",
@@ -802,18 +858,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{	.name = "pxa2xx-ssp2",
 		.id = 1,
@@ -833,18 +878,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp3",
@@ -865,18 +899,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp4",
@@ -897,18 +920,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 };
 EXPORT_SYMBOL_GPL(pxa_ssp_dai);
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a4a655f..d9c94d7 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -106,13 +106,13 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 static int pxa2xx_ac97_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	return pxa2xx_ac97_hw_probe(pdev);
+	return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev));
 }
 
 static void pxa2xx_ac97_remove(struct platform_device *pdev,
 			       struct snd_soc_dai *dai)
 {
-	pxa2xx_ac97_hw_remove(pdev);
+	pxa2xx_ac97_hw_remove(to_platform_device(dai->dev));
 }
 
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
@@ -164,6 +164,18 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_params,
+};
+
+static struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_aux_params,
+};
+
+static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_mic_params,
+};
+
 /*
  * There is only 1 physical AC97 interface for pxa2xx, but it
  * has extra fifo's that can be used for aux DACs and ADCs.
@@ -189,8 +201,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_params,},
+	.ops = &pxa_ac97_hifi_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-aux",
@@ -208,8 +219,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_aux_params,},
+	.ops = &pxa_ac97_aux_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -221,23 +231,52 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_mic_params,},
+	.ops = &pxa_ac97_mic_dai_ops,
 },
 };
 
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
-static int __init pxa_ac97_init(void)
+static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++)
+		pxa_ac97_dai[i].dev = &pdev->dev;
+
+	/* Punt most of the init to the SoC probe; we may need the machine
+	 * driver to do interesting things with the clocking to get us up
+	 * and running.
+	 */
 	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
 }
+
+static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+
+	return 0;
+}
+
+static struct platform_driver pxa2xx_ac97_driver = {
+	.probe		= pxa2xx_ac97_dev_probe,
+	.remove		= __devexit_p(pxa2xx_ac97_dev_remove),
+	.driver		= {
+		.name	= "pxa2xx-ac97",
+		.owner	= THIS_MODULE,
+	},
+};
+
+static int __init pxa_ac97_init(void)
+{
+	return platform_driver_register(&pxa2xx_ac97_driver);
+}
 module_init(pxa_ac97_init);
 
 static void __exit pxa_ac97_exit(void)
 {
-	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+	platform_driver_unregister(&pxa2xx_ac97_driver);
 }
 module_exit(pxa_ac97_exit);
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 223de89..2f4b6e4 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -25,20 +25,11 @@
 
 #include <mach/hardware.h>
 #include <mach/dma.h>
-#include <mach/pxa2xx-gpio.h>
 #include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 
-struct pxa2xx_gpio {
-	u32 sys;
-	u32	rx;
-	u32 tx;
-	u32 clk;
-	u32 frm;
-};
-
 /*
  * I2S Controller Register and Bit Definitions
  */
@@ -106,21 +97,6 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
 
-static struct pxa2xx_gpio gpio_bus[] = {
-	{ /* I2S SoC Slave */
-		.rx = GPIO29_SDATA_IN_I2S_MD,
-		.tx = GPIO30_SDATA_OUT_I2S_MD,
-		.clk = GPIO28_BITCLK_IN_I2S_MD,
-		.frm = GPIO31_SYNC_I2S_MD,
-	},
-	{ /* I2S SoC Master */
-		.rx = GPIO29_SDATA_IN_I2S_MD,
-		.tx = GPIO30_SDATA_OUT_I2S_MD,
-		.clk = GPIO28_BITCLK_OUT_I2S_MD,
-		.frm = GPIO31_SYNC_I2S_MD,
-	},
-};
-
 static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
@@ -181,9 +157,6 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	if (clk_id != PXA2XX_I2S_SYSCLK)
 		return -ENODEV;
 
-	if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT)
-		pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
-
 	return 0;
 }
 
@@ -194,10 +167,6 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
 	BUG_ON(IS_ERR(clk_i2s));
 	clk_enable(clk_i2s);
 	pxa_i2s_wait();
@@ -335,6 +304,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops pxa_i2s_dai_ops = {
+	.startup	= pxa2xx_i2s_startup,
+	.shutdown	= pxa2xx_i2s_shutdown,
+	.trigger	= pxa2xx_i2s_trigger,
+	.hw_params	= pxa2xx_i2s_hw_params,
+	.set_fmt	= pxa2xx_i2s_set_dai_fmt,
+	.set_sysclk	= pxa2xx_i2s_set_dai_sysclk,
+};
+
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
@@ -350,14 +328,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.channels_max = 2,
 		.rates = PXA2XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = pxa2xx_i2s_startup,
-		.shutdown = pxa2xx_i2s_shutdown,
-		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,
-		.set_fmt = pxa2xx_i2s_set_dai_fmt,
-		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
-	},
+	.ops = &pxa_i2s_dai_ops,
 };
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);
@@ -398,11 +369,6 @@ static struct platform_driver pxa2xx_i2s_driver = {
 
 static int __init pxa2xx_i2s_init(void)
 {
-	if (cpu_is_pxa27x())
-		gpio_bus[1].sys = GPIO113_I2S_SYSCLK_MD;
-	else
-		gpio_bus[1].sys = GPIO32_SYSCLK_I2S_MD;
-
 	clk_i2s = ERR_PTR(-ENOENT);
 	return platform_driver_register(&pxa2xx_i2s_driver);
 }
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index de8778f..c4cd2ac 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -107,7 +107,7 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 static int spitz_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	spitz_ext_control(codec);
@@ -276,7 +276,7 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
  */
 static int spitz_wm8750_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	/* NC codec pins */
 	snd_soc_dapm_nc_pin(codec, "RINPUT1");
@@ -288,12 +288,10 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8750_spitz_controls,
+				ARRAY_SIZE(wm8750_spitz_controls));
+	if (err < 0)
+		return err;
 
 	/* Add spitz specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 050223d..dbbd3e9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -80,7 +80,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);
@@ -186,18 +186,16 @@ static const struct snd_kcontrol_new tosa_controls[] = {
 
 static int tosa_ac97_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "OUT3");
 	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add tosa specific controls */
-	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&tosa_controls[i],codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, tosa_controls,
+				ARRAY_SIZE(tosa_controls));
+	if (err < 0)
+		return err;
 
 	/* add tosa specific widgets */
 	snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index f8e9ecd..9a386b4 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -14,6 +14,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/device.h>
+#include <linux/clk.h>
 #include <linux/i2c.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -26,6 +27,17 @@
 #include "pxa2xx-ac97.h"
 #include "pxa-ssp.h"
 
+/*
+ * There is a physical switch SW15 on the board which changes the MCLK
+ * for the WM9713 between the standard AC97 master clock and the
+ * output of the CLK_POUT signal from the PXA.
+ */
+static int clk_pout;
+module_param(clk_pout, int, 0);
+MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
+
+static struct clk *pout;
+
 static struct snd_soc_card zylonite;
 
 static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
@@ -61,10 +73,8 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 static int zylonite_wm9713_init(struct snd_soc_codec *codec)
 {
-	/* Currently we only support use of the AC97 clock here.  If
-	 * CLK_POUT is selected by SW15 then the clock API will need
-	 * to be used to request and enable it here.
-	 */
+	if (clk_pout)
+		snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
 
 	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
 				  ARRAY_SIZE(zylonite_dapm_widgets));
@@ -86,40 +96,35 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	unsigned int pll_out = 0;
-	unsigned int acds = 0;
 	unsigned int wm9713_div = 0;
 	int ret = 0;
+	int rate = params_rate(params);
+	int width = snd_pcm_format_physical_width(params_format(params));
 
-	switch (params_rate(params)) {
+	/* Only support ratios that we can generate neatly from the AC97
+	 * based master clock - in particular, this excludes 44.1kHz.
+	 * In most applications the voice DAC will be used for telephony
+	 * data so multiples of 8kHz will be the common case.
+	 */
+	switch (rate) {
 	case 8000:
 		wm9713_div = 12;
-		pll_out = 2048000;
 		break;
 	case 16000:
 		wm9713_div = 6;
-		pll_out = 4096000;
 		break;
 	case 48000:
-	default:
 		wm9713_div = 2;
-		pll_out = 12288000;
-		acds = 1;
 		break;
+	default:
+		/* Don't support OSS emulation */
+		return -EINVAL;
 	}
 
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
+	/* Add 1 to the width for the leading clock cycle */
+	pll_out = rate * (width + 1) * 8;
 
-	ret = snd_soc_dai_set_tdm_slot(cpu_dai,
-				       params_channels(params),
-				       params_channels(params));
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
 	if (ret < 0)
 		return ret;
 
@@ -127,19 +132,22 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (clk_pout)
+		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
+					     WM9713_PCMDIV(wm9713_div));
+	else
+		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+					     WM9713_PCMDIV(wm9713_div));
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
-	/* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
-	 * to be set instead.
-	 */
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
-				     WM9713_PCMDIV(wm9713_div));
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
@@ -173,8 +181,72 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 },
 };
 
+static int zylonite_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (clk_pout) {
+		pout = clk_get(NULL, "CLK_POUT");
+		if (IS_ERR(pout)) {
+			dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n",
+				PTR_ERR(pout));
+			return PTR_ERR(pout);
+		}
+
+		ret = clk_enable(pout);
+		if (ret != 0) {
+			dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+				ret);
+			clk_put(pout);
+			return ret;
+		}
+
+		dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n",
+			clk_get_rate(pout));
+	}
+
+	return 0;
+}
+
+static int zylonite_remove(struct platform_device *pdev)
+{
+	if (clk_pout) {
+		clk_disable(pout);
+		clk_put(pout);
+	}
+
+	return 0;
+}
+
+static int zylonite_suspend_post(struct platform_device *pdev,
+				 pm_message_t state)
+{
+	if (clk_pout)
+		clk_disable(pout);
+
+	return 0;
+}
+
+static int zylonite_resume_pre(struct platform_device *pdev)
+{
+	int ret = 0;
+
+	if (clk_pout) {
+		ret = clk_enable(pout);
+		if (ret != 0)
+			dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+				ret);
+	}
+
+	return ret;
+}
+
 static struct snd_soc_card zylonite = {
 	.name = "Zylonite",
+	.probe = &zylonite_probe,
+	.remove = &zylonite_remove,
+	.suspend_post = &zylonite_suspend_post,
+	.resume_pre = &zylonite_resume_pre,
 	.platform = &pxa2xx_soc_platform,
 	.dai_link = zylonite_dai,
 	.num_links = ARRAY_SIZE(zylonite_dai),
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index fcd03ac..2f3a21e 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,19 +1,31 @@
 config SND_S3C24XX_SOC
-	tristate "SoC Audio for the Samsung S3C24XX chips"
-	depends on ARCH_S3C2410
+	tristate "SoC Audio for the Samsung S3CXXXX chips"
+	depends on ARCH_S3C2410 || ARCH_S3C64XX
 	help
 	  Say Y or M if you want to add support for codecs attached to
-	  the S3C24XX AC97, I2S or SSP interface. You will also need
-	  to select the audio interfaces to support below.
+	  the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will
+	  also need to select the audio interfaces to support below.
 
 config SND_S3C24XX_SOC_I2S
 	tristate
+	select S3C2410_DMA
+
+config SND_S3C_I2SV2_SOC
+	tristate
 
 config SND_S3C2412_SOC_I2S
 	tristate
+	select SND_S3C_I2SV2_SOC
+	select S3C2410_DMA
+
+config SND_S3C64XX_SOC_I2S
+	tristate
+	select SND_S3C_I2SV2_SOC
+	select S3C64XX_DMA
 
 config SND_S3C2443_SOC_AC97
 	tristate
+	select S3C2410_DMA
 	select AC97_BUS
 	select SND_SOC_AC97_BUS
 	
@@ -26,6 +38,14 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
 	  Say Y if you want to add support for SoC audio on smdk2440
 	  with the WM8753.
 
+config SND_S3C24XX_SOC_JIVE_WM8750
+	tristate "SoC I2S Audio support for Jive"
+	depends on SND_S3C24XX_SOC && MACH_JIVE
+	select SND_SOC_WM8750
+	select SND_S3C2412_SOC_I2S
+	help
+	  Sat Y if you want to add support for SoC audio on the Jive.
+
 config SND_S3C24XX_SOC_SMDK2443_WM9710
 	tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
 	depends on SND_S3C24XX_SOC && MACH_SMDK2443
@@ -48,4 +68,5 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X
 	tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
        	depends on SND_S3C24XX_SOC
        	select SND_S3C24XX_SOC_I2S
+	select SND_SOC_L3
        	select SND_SOC_UDA134X
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 96b3f3f..07a93a2 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -2,19 +2,25 @@
 snd-soc-s3c24xx-objs := s3c24xx-pcm.o
 snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
 snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
+snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o
 snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
+snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
 obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
 obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
 obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
+obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o
+obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
 
 # S3C24XX Machine Support
+snd-soc-jive-wm8750-objs := jive_wm8750.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
 snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
 
+obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
new file mode 100644
index 0000000..3206379
--- /dev/null
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -0,0 +1,201 @@
+/* sound/soc/s3c24xx/jive_wm8750.c
+ *
+ * Copyright 2007,2008 Simtec Electronics
+ *
+ * Based on sound/soc/pxa/spitz.c
+ *	Copyright 2005 Wolfson Microelectronics PLC.
+ *	Copyright 2005 Openedhand Ltd.
+ *
+ * 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/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c2412-i2s.h"
+
+#include "../codecs/wm8750.h"
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{ "Headphone Jack", NULL, "LOUT1" },
+	{ "Headphone Jack", NULL, "ROUT1" },
+	{ "Internal Speaker", NULL, "LOUT2" },
+	{ "Internal Speaker", NULL, "ROUT2" },
+	{ "LINPUT1", NULL, "Line Input" },
+	{ "RINPUT1", NULL, "Line Input" },
+};
+
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Internal Speaker", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static int jive_hw_params(struct snd_pcm_substream *substream,
+			  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct s3c_i2sv2_rate_calc div;
+	unsigned int clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
+			    s3c2412_get_iisclk());
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER,
+				     div.clk_div - 1);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops jive_ops = {
+	.hw_params	= jive_hw_params,
+};
+
+static int jive_wm8750_init(struct snd_soc_codec *codec)
+{
+	int err;
+
+	/* These endpoints are not being used. */
+	snd_soc_dapm_nc_pin(codec, "LINPUT2");
+	snd_soc_dapm_nc_pin(codec, "RINPUT2");
+	snd_soc_dapm_nc_pin(codec, "LINPUT3");
+	snd_soc_dapm_nc_pin(codec, "RINPUT3");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONO");
+
+	/* Add jive specific widgets */
+	err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+					ARRAY_SIZE(wm8750_dapm_widgets));
+	if (err) {
+		printk(KERN_ERR "%s: failed to add widgets (%d)\n",
+		       __func__, err);
+		return err;
+	}
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link jive_dai = {
+	.name		= "wm8750",
+	.stream_name	= "WM8750",
+	.cpu_dai	= &s3c2412_i2s_dai,
+	.codec_dai	= &wm8750_dai,
+	.init		= jive_wm8750_init,
+	.ops		= &jive_ops,
+};
+
+/* jive audio machine driver */
+static struct snd_soc_machine snd_soc_machine_jive = {
+	.name		= "Jive",
+	.dai_link	= &jive_dai,
+	.num_links	= 1,
+};
+
+/* jive audio private data */
+static struct wm8750_setup_data jive_wm8750_setup = {
+};
+
+/* jive audio subsystem */
+static struct snd_soc_device jive_snd_devdata = {
+	.machine	= &snd_soc_machine_jive,
+	.platform	= &s3c24xx_soc_platform,
+	.codec_dev	= &soc_codec_dev_wm8750_spi,
+	.codec_data	= &jive_wm8750_setup,
+};
+
+static struct platform_device *jive_snd_device;
+
+static int __init jive_init(void)
+{
+	int ret;
+
+	if (!machine_is_jive())
+		return 0;
+
+	printk("JIVE WM8750 Audio support\n");
+
+	jive_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!jive_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(jive_snd_device, &jive_snd_devdata);
+	jive_snd_devdata.dev = &jive_snd_device->dev;
+	ret = platform_device_add(jive_snd_device);
+
+	if (ret)
+		platform_device_put(jive_snd_device);
+
+	return ret;
+}
+
+static void __exit jive_exit(void)
+{
+	platform_device_unregister(jive_snd_device);
+}
+
+module_init(jive_init);
+module_exit(jive_exit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Jive Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 45bb12e..289fadf 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -29,25 +29,17 @@
 #include <mach/regs-clock.h>
 #include <mach/regs-gpio.h>
 #include <mach/hardware.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 #include <linux/io.h>
 #include <mach/spi-gpio.h>
 
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
 
 #include "../codecs/wm8753.h"
 #include "lm4857.h"
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
 
-/* Debugging stuff */
-#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0
-#if S3C24XX_SOC_NEO1973_WM8753_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x)
-#else
-#define DBG(x...)
-#endif
-
 /* define the scenarios */
 #define NEO_AUDIO_OFF			0
 #define NEO_GSM_CALL_AUDIO_HANDSET	1
@@ -72,7 +64,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
@@ -158,7 +150,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
@@ -181,7 +173,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
@@ -224,7 +216,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
@@ -246,7 +238,7 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
 
 static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	switch (neo1973_scenario) {
 	case NEO_AUDIO_OFF:
@@ -330,7 +322,7 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (neo1973_scenario == ucontrol->value.integer.value[0])
 		return 0;
@@ -344,7 +336,7 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
 
 static void lm4857_write_regs(void)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
 		printk(KERN_ERR "lm4857: i2c write failed\n");
@@ -357,7 +349,7 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
 	int shift = (kcontrol->private_value >> 8) & 0x0F;
 	int mask = (kcontrol->private_value >> 16) & 0xFF;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
 	return 0;
@@ -385,7 +377,7 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (value)
 		value -= 5;
@@ -399,7 +391,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = ucontrol->value.integer.value[0];
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (value)
 		value += 5;
@@ -506,9 +498,9 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
  */
 static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* set up NC codec pins */
 	snd_soc_dapm_nc_pin(codec, "LOUT2");
@@ -526,13 +518,10 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
 
 	/* add neo1973 specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8753_neo1973_controls[i],
-				codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8753_neo1973_controls,
+				ARRAY_SIZE(8753_neo1973_controls));
+	if (err < 0)
+		return err;
 
 	/* set up neo1973 specific audio routes */
 	err = snd_soc_dapm_add_routes(codec, dapm_routes,
@@ -585,21 +574,15 @@ static struct snd_soc_card neo1973 = {
 	.num_links = ARRAY_SIZE(neo1973_dai),
 };
 
-static struct wm8753_setup_data neo1973_wm8753_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1a,
-};
-
 static struct snd_soc_device neo1973_snd_devdata = {
 	.card = &neo1973,
 	.codec_dev = &soc_codec_dev_wm8753,
-	.codec_data = &neo1973_wm8753_setup,
 };
 
 static int lm4857_i2c_probe(struct i2c_client *client,
 			    const struct i2c_device_id *id)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	i2c = client;
 
@@ -609,7 +592,7 @@ static int lm4857_i2c_probe(struct i2c_client *client,
 
 static int lm4857_i2c_remove(struct i2c_client *client)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	i2c = NULL;
 
@@ -620,7 +603,7 @@ static u8 lm4857_state;
 
 static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	dev_dbg(&dev->dev, "lm4857_suspend\n");
 	lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
@@ -633,7 +616,7 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 
 static int lm4857_resume(struct i2c_client *dev)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (lm4857_state) {
 		lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
@@ -644,7 +627,7 @@ static int lm4857_resume(struct i2c_client *dev)
 
 static void lm4857_shutdown(struct i2c_client *dev)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	dev_dbg(&dev->dev, "lm4857_shutdown\n");
 	lm4857_regs[LM4857_CTRL] &= 0xf0;
@@ -675,7 +658,7 @@ static int __init neo1973_init(void)
 {
 	int ret;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (!machine_is_neo1973_gta01()) {
 		printk(KERN_INFO
@@ -706,7 +689,7 @@ static int __init neo1973_init(void)
 
 static void __exit neo1973_exit(void)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
new file mode 100644
index 0000000..295a4c9
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -0,0 +1,638 @@
+/* sound/soc/s3c24xx/s3c-i2c-v2.c
+ *
+ * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs.
+ *
+ * Copyright (c) 2006 Wolfson Microelectronics PLC.
+ *	Graeme Gregory graeme.gregory@wolfsonmicro.com
+ *	linux@wolfsonmicro.com
+ *
+ * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics
+ *	http://armlinux.simtec.co.uk/
+ *	Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/regs-s3c2412-iis.h>
+
+#include <plat/audio.h>
+#include <mach/dma.h>
+
+#include "s3c-i2s-v2.h"
+
+#define S3C2412_I2S_DEBUG_CON 0
+
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+	return cpu_dai->private_data;
+}
+
+#define bit_set(v, b) (((v) & (b)) ? 1 : 0)
+
+#if S3C2412_I2S_DEBUG_CON
+static void dbg_showcon(const char *fn, u32 con)
+{
+	printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn,
+	       bit_set(con, S3C2412_IISCON_LRINDEX),
+	       bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY),
+	       bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY),
+	       bit_set(con, S3C2412_IISCON_TXFIFO_FULL),
+	       bit_set(con, S3C2412_IISCON_RXFIFO_FULL));
+
+	printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n",
+	       fn,
+	       bit_set(con, S3C2412_IISCON_TXDMA_PAUSE),
+	       bit_set(con, S3C2412_IISCON_RXDMA_PAUSE),
+	       bit_set(con, S3C2412_IISCON_TXCH_PAUSE),
+	       bit_set(con, S3C2412_IISCON_RXCH_PAUSE));
+	printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn,
+	       bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE),
+	       bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE),
+	       bit_set(con, S3C2412_IISCON_IIS_ACTIVE));
+}
+#else
+static inline void dbg_showcon(const char *fn, u32 con)
+{
+}
+#endif
+
+
+/* Turn on or off the transmission path. */
+void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
+{
+	void __iomem *regs = i2s->regs;
+	u32 fic, con, mod;
+
+	pr_debug("%s(%d)\n", __func__, on);
+
+	fic = readl(regs + S3C2412_IISFIC);
+	con = readl(regs + S3C2412_IISCON);
+	mod = readl(regs + S3C2412_IISMOD);
+
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+	if (on) {
+		con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+		con &= ~S3C2412_IISCON_TXDMA_PAUSE;
+		con &= ~S3C2412_IISCON_TXCH_PAUSE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_TXONLY:
+		case S3C2412_IISMOD_MODE_TXRX:
+			/* do nothing, we are in the right mode */
+			break;
+
+		case S3C2412_IISMOD_MODE_RXONLY:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_TXRX;
+			break;
+
+		default:
+			dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+		}
+
+		writel(con, regs + S3C2412_IISCON);
+		writel(mod, regs + S3C2412_IISMOD);
+	} else {
+		/* Note, we do not have any indication that the FIFO problems
+		 * tha the S3C2410/2440 had apply here, so we should be able
+		 * to disable the DMA and TX without resetting the FIFOS.
+		 */
+
+		con |=  S3C2412_IISCON_TXDMA_PAUSE;
+		con |=  S3C2412_IISCON_TXCH_PAUSE;
+		con &= ~S3C2412_IISCON_TXDMA_ACTIVE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_TXRX:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_RXONLY;
+			break;
+
+		case S3C2412_IISMOD_MODE_TXONLY:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			con &= ~S3C2412_IISCON_IIS_ACTIVE;
+			break;
+
+		default:
+			dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+		}
+
+		writel(mod, regs + S3C2412_IISMOD);
+		writel(con, regs + S3C2412_IISCON);
+	}
+
+	fic = readl(regs + S3C2412_IISFIC);
+	dbg_showcon(__func__, con);
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl);
+
+void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
+{
+	void __iomem *regs = i2s->regs;
+	u32 fic, con, mod;
+
+	pr_debug("%s(%d)\n", __func__, on);
+
+	fic = readl(regs + S3C2412_IISFIC);
+	con = readl(regs + S3C2412_IISCON);
+	mod = readl(regs + S3C2412_IISMOD);
+
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+	if (on) {
+		con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+		con &= ~S3C2412_IISCON_RXDMA_PAUSE;
+		con &= ~S3C2412_IISCON_RXCH_PAUSE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_TXRX:
+		case S3C2412_IISMOD_MODE_RXONLY:
+			/* do nothing, we are in the right mode */
+			break;
+
+		case S3C2412_IISMOD_MODE_TXONLY:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_TXRX;
+			break;
+
+		default:
+			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+		}
+
+		writel(mod, regs + S3C2412_IISMOD);
+		writel(con, regs + S3C2412_IISCON);
+	} else {
+		/* See txctrl notes on FIFOs. */
+
+		con &= ~S3C2412_IISCON_RXDMA_ACTIVE;
+		con |=  S3C2412_IISCON_RXDMA_PAUSE;
+		con |=  S3C2412_IISCON_RXCH_PAUSE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_RXONLY:
+			con &= ~S3C2412_IISCON_IIS_ACTIVE;
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			break;
+
+		case S3C2412_IISMOD_MODE_TXRX:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_TXONLY;
+			break;
+
+		default:
+			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+		}
+
+		writel(con, regs + S3C2412_IISCON);
+		writel(mod, regs + S3C2412_IISMOD);
+	}
+
+	fic = readl(regs + S3C2412_IISFIC);
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl);
+
+/*
+ * Wait for the LR signal to allow synchronisation to the L/R clock
+ * from the codec. May only be needed for slave mode.
+ */
+static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s)
+{
+	u32 iiscon;
+	unsigned long timeout = jiffies + msecs_to_jiffies(5);
+
+	pr_debug("Entered %s\n", __func__);
+
+	while (1) {
+		iiscon = readl(i2s->regs + S3C2412_IISCON);
+		if (iiscon & S3C2412_IISCON_LRINDEX)
+			break;
+
+		if (timeout < jiffies) {
+			printk(KERN_ERR "%s: timeout\n", __func__);
+			return -ETIMEDOUT;
+		}
+	}
+
+	return 0;
+}
+
+/*
+ * Set S3C2412 I2S DAI format
+ */
+static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+			       unsigned int fmt)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 iismod;
+
+	pr_debug("Entered %s\n", __func__);
+
+	iismod = readl(i2s->regs + S3C2412_IISMOD);
+	pr_debug("hw_params r: IISMOD: %x \n", iismod);
+
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK
+#define IISMOD_SLAVE S3C2412_IISMOD_SLAVE
+#define IISMOD_MASTER S3C2412_IISMOD_MASTER_INTERNAL
+#endif
+
+#if defined(CONFIG_PLAT_S3C64XX)
+/* From Rev1.1 datasheet, we have two master and two slave modes:
+ * IMS[11:10]:
+ *	00 = master mode, fed from PCLK
+ *	01 = master mode, fed from CLKAUDIO
+ *	10 = slave mode, using PCLK
+ *	11 = slave mode, using I2SCLK
+ */
+#define IISMOD_MASTER_MASK (1 << 11)
+#define IISMOD_SLAVE (1 << 11)
+#define IISMOD_MASTER (0x0)
+#endif
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		i2s->master = 0;
+		iismod &= ~IISMOD_MASTER_MASK;
+		iismod |= IISMOD_SLAVE;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		i2s->master = 1;
+		iismod &= ~IISMOD_MASTER_MASK;
+		iismod |= IISMOD_MASTER;
+		break;
+	default:
+		pr_debug("unknwon master/slave format\n");
+		return -EINVAL;
+	}
+
+	iismod &= ~S3C2412_IISMOD_SDF_MASK;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iismod |= S3C2412_IISMOD_SDF_MSB;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iismod |= S3C2412_IISMOD_SDF_LSB;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		iismod |= S3C2412_IISMOD_SDF_IIS;
+		break;
+	default:
+		pr_debug("Unknown data format\n");
+		return -EINVAL;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+	pr_debug("hw_params w: IISMOD: %x \n", iismod);
+	return 0;
+}
+
+static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *socdai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai_link *dai = rtd->dai;
+	struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai);
+	u32 iismod;
+
+	pr_debug("Entered %s\n", __func__);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dai->cpu_dai->dma_data = i2s->dma_playback;
+	else
+		dai->cpu_dai->dma_data = i2s->dma_capture;
+
+	/* Working copies of register */
+	iismod = readl(i2s->regs + S3C2412_IISMOD);
+	pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		iismod |= S3C2412_IISMOD_8BIT;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		iismod &= ~S3C2412_IISMOD_8BIT;
+		break;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+	pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
+	return 0;
+}
+
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai);
+	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+	unsigned long irqs;
+	int ret = 0;
+
+	pr_debug("Entered %s\n", __func__);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		/* On start, ensure that the FIFOs are cleared and reset. */
+
+		writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH,
+		       i2s->regs + S3C2412_IISFIC);
+
+		/* clear again, just in case */
+		writel(0x0, i2s->regs + S3C2412_IISFIC);
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (!i2s->master) {
+			ret = s3c2412_snd_lrsync(i2s);
+			if (ret)
+				goto exit_err;
+		}
+
+		local_irq_save(irqs);
+
+		if (capture)
+			s3c2412_snd_rxctrl(i2s, 1);
+		else
+			s3c2412_snd_txctrl(i2s, 1);
+
+		local_irq_restore(irqs);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		local_irq_save(irqs);
+
+		if (capture)
+			s3c2412_snd_rxctrl(i2s, 0);
+		else
+			s3c2412_snd_txctrl(i2s, 0);
+
+		local_irq_restore(irqs);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+exit_err:
+	return ret;
+}
+
+/*
+ * Set S3C2412 Clock dividers
+ */
+static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
+				  int div_id, int div)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 reg;
+
+	pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div);
+
+	switch (div_id) {
+	case S3C_I2SV2_DIV_BCLK:
+		reg = readl(i2s->regs + S3C2412_IISMOD);
+		reg &= ~S3C2412_IISMOD_BCLK_MASK;
+		writel(reg | div, i2s->regs + S3C2412_IISMOD);
+
+		pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+		break;
+
+	case S3C_I2SV2_DIV_RCLK:
+		if (div > 3) {
+			/* convert value to bit field */
+
+			switch (div) {
+			case 256:
+				div = S3C2412_IISMOD_RCLK_256FS;
+				break;
+
+			case 384:
+				div = S3C2412_IISMOD_RCLK_384FS;
+				break;
+
+			case 512:
+				div = S3C2412_IISMOD_RCLK_512FS;
+				break;
+
+			case 768:
+				div = S3C2412_IISMOD_RCLK_768FS;
+				break;
+
+			default:
+				return -EINVAL;
+			}
+		}
+
+		reg = readl(i2s->regs + S3C2412_IISMOD);
+		reg &= ~S3C2412_IISMOD_RCLK_MASK;
+		writel(reg | div, i2s->regs + S3C2412_IISMOD);
+		pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+		break;
+
+	case S3C_I2SV2_DIV_PRESCALER:
+		if (div >= 0) {
+			writel((div << 8) | S3C2412_IISPSR_PSREN,
+			       i2s->regs + S3C2412_IISPSR);
+		} else {
+			writel(0x0, i2s->regs + S3C2412_IISPSR);
+		}
+		pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR));
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/* default table of all avaialable root fs divisors */
+static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
+
+int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+			  unsigned int *fstab,
+			  unsigned int rate, struct clk *clk)
+{
+	unsigned long clkrate = clk_get_rate(clk);
+	unsigned int div;
+	unsigned int fsclk;
+	unsigned int actual;
+	unsigned int fs;
+	unsigned int fsdiv;
+	signed int deviation = 0;
+	unsigned int best_fs = 0;
+	unsigned int best_div = 0;
+	unsigned int best_rate = 0;
+	unsigned int best_deviation = INT_MAX;
+
+	if (fstab == NULL)
+		fstab = iis_fs_tab;
+
+	for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) {
+		fsdiv = iis_fs_tab[fs];
+
+		fsclk = clkrate / fsdiv;
+		div = fsclk / rate;
+
+		if ((fsclk % rate) > (rate / 2))
+			div++;
+
+		if (div <= 1)
+			continue;
+
+		actual = clkrate / (fsdiv * div);
+		deviation = actual - rate;
+
+		printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
+		       fsdiv, div, actual, deviation);
+
+		deviation = abs(deviation);
+
+		if (deviation < best_deviation) {
+			best_fs = fsdiv;
+			best_div = div;
+			best_rate = actual;
+			best_deviation = deviation;
+		}
+
+		if (deviation == 0)
+			break;
+	}
+
+	printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
+	       best_fs, best_div, best_rate);
+
+	info->fs_div = best_fs;
+	info->clk_div = best_div;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+
+int s3c_i2sv2_probe(struct platform_device *pdev,
+		    struct snd_soc_dai *dai,
+		    struct s3c_i2sv2_info *i2s,
+		    unsigned long base)
+{
+	struct device *dev = &pdev->dev;
+
+	i2s->dev = dev;
+
+	/* record our i2s structure for later use in the callbacks */
+	dai->private_data = i2s;
+
+	i2s->regs = ioremap(base, 0x100);
+	if (i2s->regs == NULL) {
+		dev_err(dev, "cannot ioremap registers\n");
+		return -ENXIO;
+	}
+
+	i2s->iis_pclk = clk_get(dev, "iis");
+	if (i2s->iis_pclk == NULL) {
+		dev_err(dev, "failed to get iis_clock\n");
+		iounmap(i2s->regs);
+		return -ENOENT;
+	}
+
+	clk_enable(i2s->iis_pclk);
+
+	s3c2412_snd_txctrl(i2s, 0);
+	s3c2412_snd_rxctrl(i2s, 0);
+
+	return 0;
+}
+
+EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
+
+#ifdef CONFIG_PM
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(dai);
+	u32 iismod;
+
+	if (dai->active) {
+		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
+		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
+		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
+
+		/* some basic suspend checks */
+
+		iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
+			pr_warning("%s: RXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
+			pr_warning("%s: TXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
+			pr_warning("%s: IIS active\n", __func__);
+	}
+
+	return 0;
+}
+
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(dai);
+
+	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+
+	if (dai->active) {
+		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
+		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
+		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
+
+		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
+		       i2s->regs + S3C2412_IISFIC);
+
+		ndelay(250);
+		writel(0x0, i2s->regs + S3C2412_IISFIC);
+	}
+
+	return 0;
+}
+#else
+#define s3c2412_i2s_suspend NULL
+#define s3c2412_i2s_resume  NULL
+#endif
+
+int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
+{
+	dai->ops.trigger = s3c2412_i2s_trigger;
+	dai->ops.hw_params = s3c2412_i2s_hw_params;
+	dai->ops.set_fmt = s3c2412_i2s_set_fmt;
+	dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+
+	dai->suspend = s3c2412_i2s_suspend;
+	dai->resume = s3c2412_i2s_resume;
+
+	return snd_soc_register_dai(dai);
+}
+
+EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
new file mode 100644
index 0000000..f66854a
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -0,0 +1,90 @@
+/* sound/soc/s3c24xx/s3c-i2s-v2.h
+ *
+ * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver
+ *
+ * Copyright (c) 2007 Simtec Electronics
+ *	http://armlinux.simtec.co.uk/
+ *	Ben Dooks <ben@simtec.co.uk>
+ *
+ *  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 code is the core support for the I2S block found in a number of
+ * Samsung SoC devices which is unofficially named I2S-V2. Currently the
+ * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S
+ * channels via configurable GPIO.
+ */
+
+#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H
+#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__
+
+#define S3C_I2SV2_DIV_BCLK	(1)
+#define S3C_I2SV2_DIV_RCLK	(2)
+#define S3C_I2SV2_DIV_PRESCALER	(3)
+
+/**
+ * struct s3c_i2sv2_info - S3C I2S-V2 information
+ * @dev: The parent device passed to use from the probe.
+ * @regs: The pointer to the device registe block.
+ * @master: True if the I2S core is the I2S bit clock master.
+ * @dma_playback: DMA information for playback channel.
+ * @dma_capture: DMA information for capture channel.
+ * @suspend_iismod: PM save for the IISMOD register.
+ * @suspend_iiscon: PM save for the IISCON register.
+ * @suspend_iispsr: PM save for the IISPSR register.
+ *
+ * This is the private codec state for the hardware associated with an
+ * I2S channel such as the register mappings and clock sources.
+ */
+struct s3c_i2sv2_info {
+	struct device	*dev;
+	void __iomem	*regs;
+
+	struct clk	*iis_pclk;
+	struct clk	*iis_cclk;
+	struct clk	*iis_clk;
+
+	unsigned char	 master;
+
+	struct s3c24xx_pcm_dma_params	*dma_playback;
+	struct s3c24xx_pcm_dma_params	*dma_capture;
+
+	u32		 suspend_iismod;
+	u32		 suspend_iiscon;
+	u32		 suspend_iispsr;
+};
+
+struct s3c_i2sv2_rate_calc {
+	unsigned int	clk_div;	/* for prescaler */
+	unsigned int	fs_div;		/* for root frame clock */
+};
+
+extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+				   unsigned int *fstab,
+				   unsigned int rate, struct clk *clk);
+
+/**
+ * s3c_i2sv2_probe - probe for i2s device helper
+ * @pdev: The platform device supplied to the original probe.
+ * @dai: The ASoC DAI structure supplied to the original probe.
+ * @i2s: Our local i2s structure to fill in.
+ * @base: The base address for the registers.
+ */
+extern int s3c_i2sv2_probe(struct platform_device *pdev,
+			   struct snd_soc_dai *dai,
+			   struct s3c_i2sv2_info *i2s,
+			   unsigned long base);
+
+/**
+ * s3c_i2sv2_register_dai - register dai with soc core
+ * @dai: The snd_soc_dai structure to register
+ *
+ * Fill in any missing fields and then register the given dai with the
+ * soc core.
+ */
+extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai);
+
+#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index f3fc0ab..1ca3cda 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -22,6 +22,7 @@
 #include <linux/delay.h>
 #include <linux/clk.h>
 #include <linux/kernel.h>
+#include <linux/io.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -30,26 +31,16 @@
 #include <sound/soc.h>
 #include <mach/hardware.h>
 
-#include <linux/io.h>
-#include <asm/dma.h>
-
-#include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
+#include <plat/regs-s3c2412-iis.h>
 
-#include <mach/regs-gpio.h>
-#include <mach/audio.h>
+#include <plat/regs-gpio.h>
+#include <plat/audio.h>
 #include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c2412-i2s.h"
 
 #define S3C2412_I2S_DEBUG 0
-#define S3C2412_I2S_DEBUG_CON 0
-
-#if S3C2412_I2S_DEBUG
-#define DBG(x...) printk(KERN_INFO x)
-#else
-#define DBG(x...) do { } while (0)
-#endif
 
 static struct s3c2410_dma_client s3c2412_dma_client_out = {
 	.name		= "I2S PCM Stereo out"
@@ -73,431 +64,7 @@ static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = {
 	.dma_size	= 4,
 };
 
-struct s3c2412_i2s_info {
-	struct device	*dev;
-	void __iomem	*regs;
-	struct clk	*iis_clk;
-	struct clk	*iis_pclk;
-	struct clk	*iis_cclk;
-
-	u32		 suspend_iismod;
-	u32		 suspend_iiscon;
-	u32		 suspend_iispsr;
-};
-
-static struct s3c2412_i2s_info s3c2412_i2s;
-
-#define bit_set(v, b) (((v) & (b)) ? 1 : 0)
-
-#if S3C2412_I2S_DEBUG_CON
-static void dbg_showcon(const char *fn, u32 con)
-{
-	printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn,
-	       bit_set(con, S3C2412_IISCON_LRINDEX),
-	       bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY),
-	       bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY),
-	       bit_set(con, S3C2412_IISCON_TXFIFO_FULL),
-	       bit_set(con, S3C2412_IISCON_RXFIFO_FULL));
-
-	printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n",
-	       fn,
-	       bit_set(con, S3C2412_IISCON_TXDMA_PAUSE),
-	       bit_set(con, S3C2412_IISCON_RXDMA_PAUSE),
-	       bit_set(con, S3C2412_IISCON_TXCH_PAUSE),
-	       bit_set(con, S3C2412_IISCON_RXCH_PAUSE));
-	printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn,
-	       bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE),
-	       bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE),
-	       bit_set(con, S3C2412_IISCON_IIS_ACTIVE));
-}
-#else
-static inline void dbg_showcon(const char *fn, u32 con)
-{
-}
-#endif
-
-/* Turn on or off the transmission path. */
-static void s3c2412_snd_txctrl(int on)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	void __iomem *regs = i2s->regs;
-	u32 fic, con, mod;
-
-	DBG("%s(%d)\n", __func__, on);
-
-	fic = readl(regs + S3C2412_IISFIC);
-	con = readl(regs + S3C2412_IISCON);
-	mod = readl(regs + S3C2412_IISMOD);
-
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-
-	if (on) {
-		con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
-		con &= ~S3C2412_IISCON_TXDMA_PAUSE;
-		con &= ~S3C2412_IISCON_TXCH_PAUSE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_TXONLY:
-		case S3C2412_IISMOD_MODE_TXRX:
-			/* do nothing, we are in the right mode */
-			break;
-
-		case S3C2412_IISMOD_MODE_RXONLY:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_TXRX;
-			break;
-
-		default:
-			dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
-		}
-
-		writel(con, regs + S3C2412_IISCON);
-		writel(mod, regs + S3C2412_IISMOD);
-	} else {
-		/* Note, we do not have any indication that the FIFO problems
-		 * tha the S3C2410/2440 had apply here, so we should be able
-		 * to disable the DMA and TX without resetting the FIFOS.
-		 */
-
-		con |=  S3C2412_IISCON_TXDMA_PAUSE;
-		con |=  S3C2412_IISCON_TXCH_PAUSE;
-		con &= ~S3C2412_IISCON_TXDMA_ACTIVE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_TXRX:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_RXONLY;
-			break;
-
-		case S3C2412_IISMOD_MODE_TXONLY:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			con &= ~S3C2412_IISCON_IIS_ACTIVE;
-			break;
-
-		default:
-			dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
-		}
-
-		writel(mod, regs + S3C2412_IISMOD);
-		writel(con, regs + S3C2412_IISCON);
-	}
-
-	fic = readl(regs + S3C2412_IISFIC);
-	dbg_showcon(__func__, con);
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-}
-
-static void s3c2412_snd_rxctrl(int on)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	void __iomem *regs = i2s->regs;
-	u32 fic, con, mod;
-
-	DBG("%s(%d)\n", __func__, on);
-
-	fic = readl(regs + S3C2412_IISFIC);
-	con = readl(regs + S3C2412_IISCON);
-	mod = readl(regs + S3C2412_IISMOD);
-
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-
-	if (on) {
-		con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
-		con &= ~S3C2412_IISCON_RXDMA_PAUSE;
-		con &= ~S3C2412_IISCON_RXCH_PAUSE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_TXRX:
-		case S3C2412_IISMOD_MODE_RXONLY:
-			/* do nothing, we are in the right mode */
-			break;
-
-		case S3C2412_IISMOD_MODE_TXONLY:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_TXRX;
-			break;
-
-		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
-		}
-
-		writel(mod, regs + S3C2412_IISMOD);
-		writel(con, regs + S3C2412_IISCON);
-	} else {
-		/* See txctrl notes on FIFOs. */
-
-		con &= ~S3C2412_IISCON_RXDMA_ACTIVE;
-		con |=  S3C2412_IISCON_RXDMA_PAUSE;
-		con |=  S3C2412_IISCON_RXCH_PAUSE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_RXONLY:
-			con &= ~S3C2412_IISCON_IIS_ACTIVE;
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			break;
-
-		case S3C2412_IISMOD_MODE_TXRX:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_TXONLY;
-			break;
-
-		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
-		}
-
-		writel(con, regs + S3C2412_IISCON);
-		writel(mod, regs + S3C2412_IISMOD);
-	}
-
-	fic = readl(regs + S3C2412_IISFIC);
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-}
-
-
-/*
- * Wait for the LR signal to allow synchronisation to the L/R clock
- * from the codec. May only be needed for slave mode.
- */
-static int s3c2412_snd_lrsync(void)
-{
-	u32 iiscon;
-	unsigned long timeout = jiffies + msecs_to_jiffies(5);
-
-	DBG("Entered %s\n", __func__);
-
-	while (1) {
-		iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON);
-		if (iiscon & S3C2412_IISCON_LRINDEX)
-			break;
-
-		if (timeout < jiffies) {
-			printk(KERN_ERR "%s: timeout\n", __func__);
-			return -ETIMEDOUT;
-		}
-	}
-
-	return 0;
-}
-
-/*
- * Check whether CPU is the master or slave
- */
-static inline int s3c2412_snd_is_clkmaster(void)
-{
-	u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-
-	DBG("Entered %s\n", __func__);
-
-	iismod &= S3C2412_IISMOD_MASTER_MASK;
-	return !(iismod == S3C2412_IISMOD_SLAVE);
-}
-
-/*
- * Set S3C2412 I2S DAI format
- */
-static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
-			       unsigned int fmt)
-{
-	u32 iismod;
-
-
-	DBG("Entered %s\n", __func__);
-
-	iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("hw_params r: IISMOD: %x \n", iismod);
-
-	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBM_CFM:
-		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
-		iismod |= S3C2412_IISMOD_SLAVE;
-		break;
-	case SND_SOC_DAIFMT_CBS_CFS:
-		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
-		iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
-		break;
-	default:
-		DBG("unknwon master/slave format\n");
-		return -EINVAL;
-	}
-
-	iismod &= ~S3C2412_IISMOD_SDF_MASK;
-
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_RIGHT_J:
-		iismod |= S3C2412_IISMOD_SDF_MSB;
-		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		iismod |= S3C2412_IISMOD_SDF_LSB;
-		break;
-	case SND_SOC_DAIFMT_I2S:
-		iismod |= S3C2412_IISMOD_SDF_IIS;
-		break;
-	default:
-		DBG("Unknown data format\n");
-		return -EINVAL;
-	}
-
-	writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("hw_params w: IISMOD: %x \n", iismod);
-	return 0;
-}
-
-static int s3c2412_i2s_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;
-	u32 iismod;
-
-	DBG("Entered %s\n", __func__);
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out;
-	else
-		rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in;
-
-	/* Working copies of register */
-	iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("%s: r: IISMOD: %x\n", __func__, iismod);
-
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		iismod |= S3C2412_IISMOD_8BIT;
-		break;
-	case SNDRV_PCM_FORMAT_S16_LE:
-		iismod &= ~S3C2412_IISMOD_8BIT;
-		break;
-	}
-
-	writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("%s: w: IISMOD: %x\n", __func__, iismod);
-	return 0;
-}
-
-static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
-			       struct snd_soc_dai *dai)
-{
-	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
-	unsigned long irqs;
-	int ret = 0;
-
-	DBG("Entered %s\n", __func__);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		/* On start, ensure that the FIFOs are cleared and reset. */
-
-		writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH,
-		       s3c2412_i2s.regs + S3C2412_IISFIC);
-
-		/* clear again, just in case */
-		writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC);
-
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!s3c2412_snd_is_clkmaster()) {
-			ret = s3c2412_snd_lrsync();
-			if (ret)
-				goto exit_err;
-		}
-
-		local_irq_save(irqs);
-
-		if (capture)
-			s3c2412_snd_rxctrl(1);
-		else
-			s3c2412_snd_txctrl(1);
-
-		local_irq_restore(irqs);
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		local_irq_save(irqs);
-
-		if (capture)
-			s3c2412_snd_rxctrl(0);
-		else
-			s3c2412_snd_txctrl(0);
-
-		local_irq_restore(irqs);
-		break;
-	default:
-		ret = -EINVAL;
-		break;
-	}
-
-exit_err:
-	return ret;
-}
-
-/* default table of all avaialable root fs divisors */
-static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 };
-
-int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
-			  unsigned int *fstab,
-			  unsigned int rate, struct clk *clk)
-{
-	unsigned long clkrate = clk_get_rate(clk);
-	unsigned int div;
-	unsigned int fsclk;
-	unsigned int actual;
-	unsigned int fs;
-	unsigned int fsdiv;
-	signed int deviation = 0;
-	unsigned int best_fs = 0;
-	unsigned int best_div = 0;
-	unsigned int best_rate = 0;
-	unsigned int best_deviation = INT_MAX;
-
-
-	if (fstab == NULL)
-		fstab = s3c2412_iis_fs;
-
-	for (fs = 0;; fs++) {
-		fsdiv = s3c2412_iis_fs[fs];
-
-		if (fsdiv == 0)
-			break;
-
-		fsclk = clkrate / fsdiv;
-		div = fsclk / rate;
-
-		if ((fsclk % rate) > (rate / 2))
-			div++;
-
-		if (div <= 1)
-			continue;
-
-		actual = clkrate / (fsdiv * div);
-		deviation = actual - rate;
-
-		printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
-		       fsdiv, div, actual, deviation);
-
-		deviation = abs(deviation);
-
-		if (deviation < best_deviation) {
-			best_fs = fsdiv;
-			best_div = div;
-			best_rate = actual;
-			best_deviation = deviation;
-		}
-
-		if (deviation == 0)
-			break;
-	}
-
-	printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
-	       best_fs, best_div, best_rate);
-
-	info->fs_div = best_fs;
-	info->clk_div = best_div;
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+static struct s3c_i2sv2_info s3c2412_i2s;
 
 /*
  * Set S3C2412 Clock source
@@ -507,15 +74,17 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 {
 	u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
 
-	DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
+	pr_debug("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
 	    freq, dir);
 
 	switch (clk_id) {
 	case S3C2412_CLKSRC_PCLK:
+		s3c2412_i2s.master = 1;
 		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
 		iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
 		break;
 	case S3C2412_CLKSRC_I2SCLK:
+		s3c2412_i2s.master = 0;
 		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
 		iismod |= S3C2412_IISMOD_MASTER_EXTERNAL;
 		break;
@@ -527,74 +96,6 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
-/*
- * Set S3C2412 Clock dividers
- */
-static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
-				  int div_id, int div)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	u32 reg;
-
-	DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div);
-
-	switch (div_id) {
-	case S3C2412_DIV_BCLK:
-		reg = readl(i2s->regs + S3C2412_IISMOD);
-		reg &= ~S3C2412_IISMOD_BCLK_MASK;
-		writel(reg | div, i2s->regs + S3C2412_IISMOD);
-
-		DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
-		break;
-
-	case S3C2412_DIV_RCLK:
-		if (div > 3) {
-			/* convert value to bit field */
-
-			switch (div) {
-			case 256:
-				div = S3C2412_IISMOD_RCLK_256FS;
-				break;
-
-			case 384:
-				div = S3C2412_IISMOD_RCLK_384FS;
-				break;
-
-			case 512:
-				div = S3C2412_IISMOD_RCLK_512FS;
-				break;
-
-			case 768:
-				div = S3C2412_IISMOD_RCLK_768FS;
-				break;
-
-			default:
-				return -EINVAL;
-			}
-		}
-
-		reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-		reg &= ~S3C2412_IISMOD_RCLK_MASK;
-		writel(reg | div, i2s->regs + S3C2412_IISMOD);
-		DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
-		break;
-
-	case S3C2412_DIV_PRESCALER:
-		if (div >= 0) {
-			writel((div << 8) | S3C2412_IISPSR_PSREN,
-			       i2s->regs + S3C2412_IISPSR);
-		} else {
-			writel(0x0, i2s->regs + S3C2412_IISPSR);
-		}
-		DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR));
-		break;
-
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
 
 struct clk *s3c2412_get_iisclk(void)
 {
@@ -606,34 +107,30 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
 static int s3c2412_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	DBG("Entered %s\n", __func__);
+	int ret;
 
-	s3c2412_i2s.dev = &pdev->dev;
+	pr_debug("Entered %s\n", __func__);
 
-	s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
-	if (s3c2412_i2s.regs == NULL)
-		return -ENXIO;
+	ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS);
+	if (ret)
+		return ret;
 
-	s3c2412_i2s.iis_pclk = clk_get(&pdev->dev, "iis");
-	if (s3c2412_i2s.iis_pclk == NULL) {
-		DBG("failed to get iis_clock\n");
-		iounmap(s3c2412_i2s.regs);
-		return -ENODEV;
-	}
+	s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in;
+	s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out;
 
 	s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
 	if (s3c2412_i2s.iis_cclk == NULL) {
-		DBG("failed to get i2sclk clock\n");
+		pr_debug("failed to get i2sclk clock\n");
 		iounmap(s3c2412_i2s.regs);
 		return -ENODEV;
 	}
 
-	clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
+	/* Set MPLL as the source for IIS CLK */
 
-	clk_enable(s3c2412_i2s.iis_pclk);
+	clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
 	clk_enable(s3c2412_i2s.iis_cclk);
 
-	s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk;
+	s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
 
 	/* Configure the I2S pins in correct mode */
 	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
@@ -642,78 +139,22 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
 	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
 
-	s3c2412_snd_txctrl(0);
-	s3c2412_snd_rxctrl(0);
-
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	u32 iismod;
-
-	if (dai->active) {
-		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
-		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
-		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
-
-		/* some basic suspend checks */
-
-		iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			pr_warning("%s: RXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			pr_warning("%s: TXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			pr_warning("%s: IIS active\n", __func__);
-	}
-
-	return 0;
-}
-
-static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-
-	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
-		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
-
-	if (dai->active) {
-		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
-		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
-		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
-
-		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
-		       i2s->regs + S3C2412_IISFIC);
-
-		ndelay(250);
-		writel(0x0, i2s->regs + S3C2412_IISFIC);
-
-	}
-
-	return 0;
-}
-#else
-#define s3c2412_i2s_suspend NULL
-#define s3c2412_i2s_resume  NULL
-#endif /* CONFIG_PM */
-
 #define S3C2412_I2S_RATES \
 	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
+	.set_sysclk	= s3c2412_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c2412_i2s_dai = {
-	.name	= "s3c2412-i2s",
-	.id	= 0,
-	.probe	= s3c2412_i2s_probe,
-	.suspend = s3c2412_i2s_suspend,
-	.resume = s3c2412_i2s_resume,
+	.name		= "s3c2412-i2s",
+	.id		= 0,
+	.probe		= s3c2412_i2s_probe,
 	.playback = {
 		.channels_min	= 2,
 		.channels_max	= 2,
@@ -726,19 +167,13 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 		.rates		= S3C2412_I2S_RATES,
 		.formats	= SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.trigger	= s3c2412_i2s_trigger,
-		.hw_params	= s3c2412_i2s_hw_params,
-		.set_fmt	= s3c2412_i2s_set_fmt,
-		.set_clkdiv	= s3c2412_i2s_set_clkdiv,
-		.set_sysclk	= s3c2412_i2s_set_sysclk,
-	},
+	.ops = &s3c2412_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
 static int __init s3c2412_i2s_init(void)
 {
-	return snd_soc_register_dai(&s3c2412_i2s_dai);
+	return  s3c_i2sv2_register_dai(&s3c2412_i2s_dai);
 }
 module_init(s3c2412_i2s_init);
 
@@ -748,7 +183,6 @@ static void __exit s3c2412_i2s_exit(void)
 }
 module_exit(s3c2412_i2s_exit);
 
-
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
index aac08a2..92848e5 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.h
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -15,9 +15,11 @@
 #ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H
 #define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__
 
-#define S3C2412_DIV_BCLK	(1)
-#define S3C2412_DIV_RCLK	(2)
-#define S3C2412_DIV_PRESCALER	(3)
+#include "s3c-i2s-v2.h"
+
+#define S3C2412_DIV_BCLK	S3C_I2SV2_DIV_BCLK
+#define S3C2412_DIV_RCLK	S3C_I2SV2_DIV_RCLK
+#define S3C2412_DIV_PRESCALER	S3C_I2SV2_DIV_PRESCALER
 
 #define S3C2412_CLKSRC_PCLK	(0)
 #define S3C2412_CLKSRC_I2SCLK	(1)
@@ -26,13 +28,4 @@ extern struct clk *s3c2412_get_iisclk(void);
 
 extern struct snd_soc_dai s3c2412_i2s_dai;
 
-struct s3c2412_rate_calc {
-	unsigned int	clk_div;	/* for prescaler */
-	unsigned int	fs_div;		/* for root frame clock */
-};
-
-extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
-				 unsigned int *fstab,
-				 unsigned int rate, struct clk *clk);
-
 #endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 5822d2d..3698f70 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -31,7 +31,7 @@
 #include <plat/regs-ac97.h>
 #include <mach/regs-gpio.h>
 #include <mach/regs-clock.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 #include <asm/dma.h>
 #include <mach/dma.h>
 
@@ -355,6 +355,16 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = {
+	.hw_params	= s3c2443_ac97_hw_params,
+	.trigger	= s3c2443_ac97_trigger,
+};
+
+static struct snd_soc_dai_ops s3c2443_ac97_mic_dai_ops = {
+	.hw_params	= s3c2443_ac97_hw_mic_params,
+	.trigger	= s3c2443_ac97_mic_trigger,
+};
+
 struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
@@ -374,9 +384,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_params,
-		.trigger = s3c2443_ac97_trigger},
+	.ops = &s3c2443_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -388,9 +396,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_mic_params,
-		.trigger = s3c2443_ac97_mic_trigger,},
+	.ops = &s3c2443_ac97_mic_dai_ops,
 },
 };
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 6f4d439..cc06696 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -4,7 +4,7 @@
  * (c) 2006 Wolfson Microelectronics PLC.
  * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
  *
- * (c) 2004-2005 Simtec Electronics
+ * Copyright 2004-2005 Simtec Electronics
  *	http://armlinux.simtec.co.uk/
  *	Ben Dooks <ben@simtec.co.uk>
  *
@@ -30,22 +30,15 @@
 #include <mach/hardware.h>
 #include <mach/regs-gpio.h>
 #include <mach/regs-clock.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 #include <asm/dma.h>
 #include <mach/dma.h>
 
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
 
-#define S3C24XX_I2S_DEBUG 0
-#if S3C24XX_I2S_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x)
-#else
-#define DBG(x...)
-#endif
-
 static struct s3c2410_dma_client s3c24xx_dma_client_out = {
 	.name = "I2S PCM Stereo out"
 };
@@ -84,13 +77,13 @@ static void s3c24xx_snd_txctrl(int on)
 	u32 iiscon;
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
 	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
 	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
 
-	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 
 	if (on) {
 		iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE;
@@ -120,7 +113,7 @@ static void s3c24xx_snd_txctrl(int on)
 		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
 	}
 
-	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 }
 
 static void s3c24xx_snd_rxctrl(int on)
@@ -129,13 +122,13 @@ static void s3c24xx_snd_rxctrl(int on)
 	u32 iiscon;
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
 	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
 	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
 
-	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 
 	if (on) {
 		iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE;
@@ -165,7 +158,7 @@ static void s3c24xx_snd_rxctrl(int on)
 		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
 	}
 
-	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 }
 
 /*
@@ -177,7 +170,7 @@ static int s3c24xx_snd_lrsync(void)
 	u32 iiscon;
 	int timeout = 50; /* 5ms */
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	while (1) {
 		iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -197,7 +190,7 @@ static int s3c24xx_snd_lrsync(void)
  */
 static inline int s3c24xx_snd_is_clkmaster(void)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
 }
@@ -210,10 +203,10 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 {
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params r: IISMOD: %lx \n", iismod);
+	pr_debug("hw_params r: IISMOD: %x \n", iismod);
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
@@ -238,7 +231,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 	}
 
 	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params w: IISMOD: %lx \n", iismod);
+	pr_debug("hw_params w: IISMOD: %x \n", iismod);
 	return 0;
 }
 
@@ -249,7 +242,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
@@ -258,7 +251,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	/* Working copies of register */
 	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params r: IISMOD: %lx\n", iismod);
+	pr_debug("hw_params r: IISMOD: %x\n", iismod);
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
@@ -276,7 +269,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params w: IISMOD: %lx\n", iismod);
+	pr_debug("hw_params w: IISMOD: %x\n", iismod);
 	return 0;
 }
 
@@ -285,7 +278,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -327,7 +320,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 {
 	u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iismod &= ~S3C2440_IISMOD_MPLL;
 
@@ -353,7 +346,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 {
 	u32 reg;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	switch (div_id) {
 	case S3C24XX_DIV_BCLK:
@@ -389,7 +382,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
 static int s3c24xx_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
 	if (s3c24xx_i2s.regs == NULL)
@@ -397,7 +390,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 
 	s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis");
 	if (s3c24xx_i2s.iis_clk == NULL) {
-		DBG("failed to get iis_clock\n");
+		pr_err("failed to get iis_clock\n");
 		iounmap(s3c24xx_i2s.regs);
 		return -ENODEV;
 	}
@@ -421,7 +414,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 #ifdef CONFIG_PM
 static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
 	s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -435,7 +428,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 
 static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 	clk_enable(s3c24xx_i2s.iis_clk);
 
 	writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -456,6 +449,14 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = {
+	.trigger	= s3c24xx_i2s_trigger,
+	.hw_params	= s3c24xx_i2s_hw_params,
+	.set_fmt	= s3c24xx_i2s_set_fmt,
+	.set_clkdiv	= s3c24xx_i2s_set_clkdiv,
+	.set_sysclk	= s3c24xx_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
@@ -472,13 +473,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = S3C24XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.trigger = s3c24xx_i2s_trigger,
-		.hw_params = s3c24xx_i2s_hw_params,
-		.set_fmt = s3c24xx_i2s_set_fmt,
-		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
-		.set_sysclk = s3c24xx_i2s_set_sysclk,
-	},
+	.ops = &s3c24xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 7c64d31..a9d68fa 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -4,7 +4,7 @@
  * (c) 2006 Wolfson Microelectronics PLC.
  * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
  *
- * (c) 2004-2005 Simtec Electronics
+ * Copyright 2004-2005 Simtec Electronics
  *	http://armlinux.simtec.co.uk/
  *	Ben Dooks <ben@simtec.co.uk>
  *
@@ -29,17 +29,10 @@
 #include <asm/dma.h>
 #include <mach/hardware.h>
 #include <mach/dma.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 
 #include "s3c24xx-pcm.h"
 
-#define S3C24XX_PCM_DEBUG 0
-#if S3C24XX_PCM_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x)
-#else
-#define DBG(x...)
-#endif
-
 static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
 				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -84,16 +77,16 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
 	dma_addr_t pos = prtd->dma_pos;
 	int ret;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	while (prtd->dma_loaded < prtd->dma_limit) {
 		unsigned long len = prtd->dma_period;
 
-		DBG("dma_loaded: %d\n", prtd->dma_loaded);
+		pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
 
 		if ((pos + len) > prtd->dma_end) {
 			len  = prtd->dma_end - pos;
-			DBG(KERN_DEBUG "%s: corrected dma len %ld\n",
+			pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n",
 			       __func__, len);
 		}
 
@@ -119,7 +112,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
 	struct snd_pcm_substream *substream = dev_id;
 	struct s3c24xx_runtime_data *prtd;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR)
 		return;
@@ -148,7 +141,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	unsigned long totbytes = params_buffer_bytes(params);
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
@@ -161,14 +154,14 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 		/* prepare DMA */
 		prtd->params = dma;
 
-		DBG("params %p, client %p, channel %d\n", prtd->params,
+		pr_debug("params %p, client %p, channel %d\n", prtd->params,
 			prtd->params->client, prtd->params->channel);
 
 		ret = s3c2410_dma_request(prtd->params->channel,
 					  prtd->params->client, NULL);
 
 		if (ret < 0) {
-			DBG(KERN_ERR "failed to get dma channel\n");
+			printk(KERN_ERR "failed to get dma channel\n");
 			return ret;
 		}
 	}
@@ -196,7 +189,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* TODO - do we need to ensure DMA flushed */
 	snd_pcm_set_runtime_buffer(substream, NULL);
@@ -214,7 +207,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
@@ -259,7 +252,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	spin_lock(&prtd->lock);
 
@@ -297,7 +290,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
 	unsigned long res;
 	dma_addr_t src, dst;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	spin_lock(&prtd->lock);
 	s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
@@ -309,7 +302,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
 
 	spin_unlock(&prtd->lock);
 
-	DBG("Pointer %x %x\n", src, dst);
+	pr_debug("Pointer %x %x\n", src, dst);
 
 	/* we seem to be getting the odd error from the pcm library due
 	 * to out-of-bounds pointers. this is maybe due to the dma engine
@@ -330,7 +323,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
 
@@ -349,10 +342,10 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (!prtd)
-		DBG("s3c24xx_pcm_close called with prtd == NULL\n");
+		pr_debug("s3c24xx_pcm_close called with prtd == NULL\n");
 
 	kfree(prtd);
 
@@ -364,7 +357,7 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
 				     runtime->dma_area,
@@ -390,7 +383,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
 	size_t size = s3c24xx_pcm_hardware.buffer_bytes_max;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	buf->dev.type = SNDRV_DMA_TYPE_DEV;
 	buf->dev.dev = pcm->card->dev;
@@ -409,7 +402,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	struct snd_dma_buffer *buf;
 	int stream;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	for (stream = 0; stream < 2; stream++) {
 		substream = pcm->streams[stream].substream;
@@ -433,7 +426,7 @@ static int s3c24xx_pcm_new(struct snd_card *card,
 {
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (!card->dev->dma_mask)
 		card->dev->dma_mask = &s3c24xx_pcm_dmamask;
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index a0a4d183..8e79a41 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -22,7 +22,7 @@
 #include <sound/s3c24xx_uda134x.h>
 #include <sound/uda134x.h>
 
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
new file mode 100644
index 0000000..33c5de7
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -0,0 +1,222 @@
+/* sound/soc/s3c24xx/s3c64xx-i2s.c
+ *
+ * ALSA SoC Audio Layer - S3C64XX I2S driver
+ *
+ * Copyright 2008 Openmoko, Inc.
+ * Copyright 2008 Simtec Electronics
+ *      Ben Dooks <ben@simtec.co.uk>
+ *      http://armlinux.simtec.co.uk/
+ *
+ * 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/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/gpio.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/regs-s3c2412-iis.h>
+#include <plat/gpio-bank-d.h>
+#include <plat/gpio-bank-e.h>
+#include <plat/gpio-cfg.h>
+#include <plat/audio.h>
+
+#include <mach/map.h>
+#include <mach/dma.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c64xx-i2s.h"
+
+static struct s3c2410_dma_client s3c64xx_dma_client_out = {
+	.name		= "I2S PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c64xx_dma_client_in = {
+	.name		= "I2S PCM Stereo in"
+};
+
+static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = {
+	[0] = {
+		.channel	= DMACH_I2S0_OUT,
+		.client		= &s3c64xx_dma_client_out,
+		.dma_addr	= S3C64XX_PA_IIS0 + S3C2412_IISTXD,
+		.dma_size	= 4,
+	},
+	[1] = {
+		.channel	= DMACH_I2S1_OUT,
+		.client		= &s3c64xx_dma_client_out,
+		.dma_addr	= S3C64XX_PA_IIS1 + S3C2412_IISTXD,
+		.dma_size	= 4,
+	},
+};
+
+static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = {
+	[0] = {
+		.channel	= DMACH_I2S0_IN,
+		.client		= &s3c64xx_dma_client_in,
+		.dma_addr	= S3C64XX_PA_IIS0 + S3C2412_IISRXD,
+		.dma_size	= 4,
+	},
+	[1] = {
+		.channel	= DMACH_I2S1_IN,
+		.client		= &s3c64xx_dma_client_in,
+		.dma_addr	= S3C64XX_PA_IIS1 + S3C2412_IISRXD,
+		.dma_size	= 4,
+	},
+};
+
+static struct s3c_i2sv2_info s3c64xx_i2s[2];
+
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+	return cpu_dai->private_data;
+}
+
+static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+	switch (clk_id) {
+	case S3C64XX_CLKSRC_PCLK:
+		iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX;
+		break;
+
+	case S3C64XX_CLKSRC_MUX:
+		iismod |= S3C64XX_IISMOD_IMS_SYSMUX;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+
+	return 0;
+}
+
+
+unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(dai);
+
+	return clk_get_rate(i2s->iis_cclk);
+}
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
+
+static int s3c64xx_i2s_probe(struct platform_device *pdev,
+			     struct snd_soc_dai *dai)
+{
+	struct device *dev = &pdev->dev;
+	struct s3c_i2sv2_info *i2s;
+	int ret;
+
+	dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id);
+
+	if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) {
+		dev_err(dev, "id %d out of range\n", pdev->id);
+		return -EINVAL;
+	}
+
+	i2s = &s3c64xx_i2s[pdev->id];
+
+	ret = s3c_i2sv2_probe(pdev, dai, i2s,
+			      pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
+	if (ret)
+		return ret;
+
+	i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
+	i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
+
+	i2s->iis_cclk = clk_get(dev, "audio-bus");
+	if (IS_ERR(i2s->iis_cclk)) {
+		dev_err(dev, "failed to get audio-bus");
+		iounmap(i2s->regs);
+		return -ENODEV;
+	}
+
+	/* configure GPIO for i2s port */
+	switch (pdev->id) {
+	case 0:
+		s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK);
+		s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI);
+		s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0);
+		break;
+	case 1:
+		s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK);
+		s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI);
+		s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0);
+	}
+
+	return 0;
+}
+
+
+#define S3C64XX_I2S_RATES \
+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define S3C64XX_I2S_FMTS \
+	(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
+
+static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
+	.set_sysclk	= s3c64xx_i2s_set_sysclk,	
+};
+
+struct snd_soc_dai s3c64xx_i2s_dai = {
+	.name		= "s3c64xx-i2s",
+	.id		= 0,
+	.probe		= s3c64xx_i2s_probe,
+	.playback = {
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= S3C64XX_I2S_RATES,
+		.formats	= S3C64XX_I2S_FMTS,
+	},
+	.capture = {
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= S3C64XX_I2S_RATES,
+		.formats	= S3C64XX_I2S_FMTS,
+	},
+	.ops = &s3c64xx_i2s_dai_ops,
+};
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
+
+static int __init s3c64xx_i2s_init(void)
+{
+	return  s3c_i2sv2_register_dai(&s3c64xx_i2s_dai);
+}
+module_init(s3c64xx_i2s_init);
+
+static void __exit s3c64xx_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c64xx_i2s_dai);
+}
+module_exit(s3c64xx_i2s_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("S3C64XX I2S SoC Interface");
+MODULE_LICENSE("GPL");
+
+
+
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
new file mode 100644
index 0000000..b7ffe3c
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -0,0 +1,31 @@
+/* sound/soc/s3c24xx/s3c64xx-i2s.h
+ *
+ * ALSA SoC Audio Layer - S3C64XX I2S driver
+ *
+ * Copyright 2008 Openmoko, Inc.
+ * Copyright 2008 Simtec Electronics
+ *      Ben Dooks <ben@simtec.co.uk>
+ *      http://armlinux.simtec.co.uk/
+ *
+ * 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 __SND_SOC_S3C24XX_S3C64XX_I2S_H
+#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__
+
+#include "s3c-i2s-v2.h"
+
+#define S3C64XX_DIV_BCLK	S3C_I2SV2_DIV_BCLK
+#define S3C64XX_DIV_RCLK	S3C_I2SV2_DIV_RCLK
+#define S3C64XX_DIV_PRESCALER	S3C_I2SV2_DIV_PRESCALER
+
+#define S3C64XX_CLKSRC_PCLK	(0)
+#define S3C64XX_CLKSRC_MUX	(1)
+
+extern struct snd_soc_dai s3c64xx_i2s_dai;
+
+extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
+
+#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index eab3183..41db75a 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -267,6 +267,10 @@ static int hac_hw_params(struct snd_pcm_substream *substream,
 #define AC97_FMTS	\
 	SNDRV_PCM_FMTBIT_S16_LE
 
+static struct snd_soc_dai_ops hac_dai_ops = {
+	.hw_params	= hac_hw_params,
+};
+
 struct snd_soc_dai sh4_hac_dai[] = {
 {
 	.name			= "HAC0",
@@ -284,9 +288,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.hw_params	= hac_hw_params,
-	},
+	.ops = &hac_dai_ops,
 },
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
@@ -305,9 +307,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.hw_params	= hac_hw_params,
-	},
+	.ops = &hac_dai_ops,
 
 },
 #endif
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index d1e5390..56fa087 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -336,6 +336,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
 	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
 
+static struct snd_soc_dai_ops ssi_dai_ops = {
+	.startup	= ssi_startup,
+	.shutdown	= ssi_shutdown,
+	.trigger	= ssi_trigger,
+	.hw_params	= ssi_hw_params,
+	.set_sysclk	= ssi_set_sysclk,
+	.set_clkdiv	= ssi_set_clkdiv,
+	.set_fmt	= ssi_set_fmt,
+};
+
 struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
@@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
@@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #endif
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ec3f8bb..6e710f7 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	mutex_lock(&pcm_mutex);
 
 	/* startup the audio subsystem */
-	if (cpu_dai->ops.startup) {
-		ret = cpu_dai->ops.startup(substream, cpu_dai);
+	if (cpu_dai->ops->startup) {
+		ret = cpu_dai->ops->startup(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open interface %s\n",
 				cpu_dai->name);
@@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.startup) {
-		ret = codec_dai->ops.startup(substream, codec_dai);
+	if (codec_dai->ops->startup) {
+		ret = codec_dai->ops->startup(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open codec %s\n",
 				codec_dai->name);
@@ -234,7 +234,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		cpu_dai->capture.active = codec_dai->capture.active = 1;
 	cpu_dai->active = codec_dai->active = 1;
 	cpu_dai->runtime = runtime;
-	socdev->codec->active++;
+	card->codec->active++;
 	mutex_unlock(&pcm_mutex);
 	return 0;
 
@@ -247,8 +247,8 @@ codec_dai_err:
 		platform->pcm_ops->close(substream);
 
 platform_err:
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -264,7 +264,7 @@ static void close_delayed_work(struct work_struct *work)
 	struct snd_soc_card *card = container_of(work, struct snd_soc_card,
 						 delayed_work.work);
 	struct snd_soc_device *socdev = card->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	struct snd_soc_dai *codec_dai;
 	int i;
 
@@ -319,7 +319,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 
 	mutex_lock(&pcm_mutex);
 
@@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		snd_soc_dai_digital_mute(codec_dai, 1);
 
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 
-	if (codec_dai->ops.shutdown)
-		codec_dai->ops.shutdown(substream, codec_dai);
+	if (codec_dai->ops->shutdown)
+		codec_dai->ops->shutdown(substream, codec_dai);
 
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
@@ -387,7 +387,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	int ret = 0;
 
 	mutex_lock(&pcm_mutex);
@@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.prepare) {
-		ret = codec_dai->ops.prepare(substream, codec_dai);
+	if (codec_dai->ops->prepare) {
+		ret = codec_dai->ops->prepare(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: codec DAI prepare error\n");
 			goto out;
 		}
 	}
 
-	if (cpu_dai->ops.prepare) {
-		ret = cpu_dai->ops.prepare(substream, cpu_dai);
+	if (cpu_dai->ops->prepare) {
+		ret = cpu_dai->ops->prepare(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
 			goto out;
@@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (codec_dai->ops.hw_params) {
-		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
+	if (codec_dai->ops->hw_params) {
+		ret = codec_dai->ops->hw_params(substream, params, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
 				codec_dai->name);
@@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (cpu_dai->ops.hw_params) {
-		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
+	if (cpu_dai->ops->hw_params) {
+		ret = cpu_dai->ops->hw_params(substream, params, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
@@ -526,12 +526,12 @@ out:
 	return ret;
 
 platform_err:
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 interface_err:
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
 codec_err:
 	if (machine->ops && machine->ops->hw_free)
@@ -553,7 +553,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 
 	mutex_lock(&pcm_mutex);
 
@@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		platform->pcm_ops->hw_free(substream);
 
 	/* now free hw params for the DAI's  */
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
 
-	if (codec_dai->ops.trigger) {
-		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
+	if (codec_dai->ops->trigger) {
+		ret = codec_dai->ops->trigger(substream, cmd, codec_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 			return ret;
 	}
 
-	if (cpu_dai->ops.trigger) {
-		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
+	if (cpu_dai->ops->trigger) {
+		ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -629,7 +629,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	int i;
 
 	/* Due to the resume being scheduled into a workqueue we could
@@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	/* mute any active DAC's */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
-		if (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 1);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
@@ -705,7 +705,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	struct snd_soc_device *socdev = card->socdev;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	struct platform_device *pdev = to_platform_device(socdev->dev);
 	int i;
 
@@ -741,8 +741,8 @@ static void soc_resume_deferred(struct work_struct *work)
 	/* unmute any active DACs */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
-		if (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 0);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 0);
 	}
 
 	for (i = 0; i < card->num_links; i++) {
@@ -982,8 +982,8 @@ static struct platform_driver soc_driver = {
 static int soc_new_pcm(struct snd_soc_device *socdev,
 	struct snd_soc_dai_link *dai_link, int num)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = card->codec;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
 	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
@@ -998,7 +998,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 
 	rtd->dai = dai_link;
 	rtd->socdev = socdev;
-	codec_dai->codec = socdev->codec;
+	codec_dai->codec = card->codec;
 
 	/* check client and interface hw capabilities */
 	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
@@ -1048,9 +1048,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 }
 
 /* codec register dump */
-static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
+static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
 {
-	struct snd_soc_codec *codec = devdata->codec;
 	int i, step = 1, count = 0;
 
 	if (!codec->reg_cache_size)
@@ -1090,7 +1089,7 @@ static ssize_t codec_reg_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_device *devdata = dev_get_drvdata(dev);
-	return soc_codec_reg_show(devdata, buf);
+	return soc_codec_reg_show(devdata->card->codec, buf);
 }
 
 static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
@@ -1107,12 +1106,10 @@ static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
 {
 	ssize_t ret;
 	struct snd_soc_codec *codec = file->private_data;
-	struct device *card_dev = codec->card->dev;
-	struct snd_soc_device *devdata = card_dev->driver_data;
 	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
 	if (!buf)
 		return -ENOMEM;
-	ret = soc_codec_reg_show(devdata, buf);
+	ret = soc_codec_reg_show(codec, buf);
 	if (ret >= 0)
 		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 	kfree(buf);
@@ -1309,19 +1306,19 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits);
  */
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
-	int ret = 0, i;
+	struct snd_soc_codec *codec = card->codec;
+	int ret, i;
 
 	mutex_lock(&codec->mutex);
 
 	/* register a sound card */
-	codec->card = snd_card_new(idx, xid, codec->owner, 0);
-	if (!codec->card) {
+	ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card);
+	if (ret < 0) {
 		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
 			codec->name);
 		mutex_unlock(&codec->mutex);
-		return -ENODEV;
+		return ret;
 	}
 
 	codec->card->dev = socdev->dev;
@@ -1355,8 +1352,8 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
  */
 int snd_soc_init_card(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = card->codec;
 	int ret = 0, i, ac97 = 0, err = 0;
 
 	for (i = 0; i < card->num_links; i++) {
@@ -1407,7 +1404,7 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
 	if (err < 0)
 		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
-	soc_init_codec_debugfs(socdev->codec);
+	soc_init_codec_debugfs(codec);
 	mutex_unlock(&codec->mutex);
 
 out:
@@ -1424,18 +1421,19 @@ EXPORT_SYMBOL_GPL(snd_soc_init_card);
  */
 void snd_soc_free_pcms(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	struct snd_soc_dai *codec_dai;
 	int i;
 #endif
 
 	mutex_lock(&codec->mutex);
-	soc_cleanup_codec_debugfs(socdev->codec);
+	soc_cleanup_codec_debugfs(codec);
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
-		if (codec_dai->ac97_control && codec->ac97) {
+		if (codec_dai->ac97_control && codec->ac97 &&
+		    strcmp(codec->name, "AC97") != 0) {
 			soc_ac97_dev_unregister(codec);
 			goto free_card;
 		}
@@ -1498,6 +1496,37 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 EXPORT_SYMBOL_GPL(snd_soc_cnew);
 
 /**
+ * snd_soc_add_controls - add an array of controls to a codec.
+ * Convienience function to add a list of controls. Many codecs were
+ * duplicating this code.
+ *
+ * @codec: codec to add controls to
+ * @controls: array of controls to add
+ * @num_controls: number of elements in the array
+ *
+ * Return 0 for success, else error.
+ */
+int snd_soc_add_controls(struct snd_soc_codec *codec,
+	const struct snd_kcontrol_new *controls, int num_controls)
+{
+	struct snd_card *card = codec->card;
+	int err, i;
+
+	for (i = 0; i < num_controls; i++) {
+		const struct snd_kcontrol_new *control = &controls[i];
+		err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
+		if (err < 0) {
+			dev_err(codec->dev, "%s: Failed to add %s\n",
+				codec->name, control->name);
+			return err;
+		}
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_add_controls);
+
+/**
  * snd_soc_info_enum_double - enumerated double mixer info callback
  * @kcontrol: mixer control
  * @uinfo: control element information
@@ -2023,8 +2052,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
 }
@@ -2043,8 +2072,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->ops.set_clkdiv)
-		return dai->ops.set_clkdiv(dai, div_id, div);
+	if (dai->ops->set_clkdiv)
+		return dai->ops->set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
 }
@@ -2062,8 +2091,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->ops.set_pll)
-		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
+	if (dai->ops->set_pll)
+		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -2078,8 +2107,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->ops.set_fmt)
-		return dai->ops.set_fmt(dai, fmt);
+	if (dai->ops->set_fmt)
+		return dai->ops->set_fmt(dai, fmt);
 	else
 		return -EINVAL;
 }
@@ -2097,8 +2126,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tdm_slot(dai, mask, slots);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
 }
@@ -2113,8 +2142,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tristate(dai, tristate);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tristate(dai, tristate);
 	else
 		return -EINVAL;
 }
@@ -2129,8 +2158,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->ops.digital_mute)
-		return dai->ops.digital_mute(dai, mute);
+	if (dai->ops->digital_mute)
+		return dai->ops->digital_mute(dai, mute);
 	else
 		return -EINVAL;
 }
@@ -2183,6 +2212,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 	return 0;
 }
 
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
 /**
  * snd_soc_register_dai - Register a DAI with the ASoC core
  *
@@ -2197,6 +2229,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai)
 	if (!dai->dev)
 		printk(KERN_WARNING "No device for DAI %s\n", dai->name);
 
+	if (!dai->ops)
+		dai->ops = &null_dai_ops;
+
 	INIT_LIST_HEAD(&dai->list);
 
 	mutex_lock(&client_mutex);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a2f1da8..735903a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -54,14 +54,15 @@
 static int dapm_up_seq[] = {
 	snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
 	snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac,
-	snd_soc_dapm_mixer, snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp,
-	snd_soc_dapm_spk, snd_soc_dapm_post
+	snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga,
+	snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
 };
+
 static int dapm_down_seq[] = {
 	snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
-	snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
-	snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
-	snd_soc_dapm_post
+	snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
+	snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
+	snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post
 };
 
 static int dapm_status = 1;
@@ -101,7 +102,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 {
 	switch (w->id) {
 	case snd_soc_dapm_switch:
-	case snd_soc_dapm_mixer: {
+	case snd_soc_dapm_mixer:
+	case snd_soc_dapm_mixer_named_ctl: {
 		int val;
 		struct soc_mixer_control *mc = (struct soc_mixer_control *)
 			w->kcontrols[i].private_value;
@@ -323,15 +325,32 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
 			if (path->name != (char*)w->kcontrols[i].name)
 				continue;
 
-			/* add dapm control with long name */
-			name_len = 2 + strlen(w->name)
-				+ strlen(w->kcontrols[i].name);
+			/* add dapm control with long name.
+			 * for dapm_mixer this is the concatenation of the
+			 * mixer and kcontrol name.
+			 * for dapm_mixer_named_ctl this is simply the
+			 * kcontrol name.
+			 */
+			name_len = strlen(w->kcontrols[i].name) + 1;
+			if (w->id != snd_soc_dapm_mixer_named_ctl)
+				name_len += 1 + strlen(w->name);
+
 			path->long_name = kmalloc(name_len, GFP_KERNEL);
+
 			if (path->long_name == NULL)
 				return -ENOMEM;
 
-			snprintf(path->long_name, name_len, "%s %s",
-				 w->name, w->kcontrols[i].name);
+			switch (w->id) {
+			default:
+				snprintf(path->long_name, name_len, "%s %s",
+					 w->name, w->kcontrols[i].name);
+				break;
+			case snd_soc_dapm_mixer_named_ctl:
+				snprintf(path->long_name, name_len, "%s",
+					 w->kcontrols[i].name);
+				break;
+			}
+
 			path->long_name[name_len - 1] = '\0';
 
 			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
@@ -503,6 +522,137 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 EXPORT_SYMBOL_GPL(dapm_reg_event);
 
 /*
+ * Scan a single DAPM widget for a complete audio path and update the
+ * power status appropriately.
+ */
+static int dapm_power_widget(struct snd_soc_codec *codec, int event,
+			     struct snd_soc_dapm_widget *w)
+{
+	int in, out, power_change, power, ret;
+
+	/* vmid - no action */
+	if (w->id == snd_soc_dapm_vmid)
+		return 0;
+
+	/* active ADC */
+	if (w->id == snd_soc_dapm_adc && w->active) {
+		in = is_connected_input_ep(w);
+		dapm_clear_walk(w->codec);
+		w->power = (in != 0) ? 1 : 0;
+		dapm_update_bits(w);
+		return 0;
+	}
+
+	/* active DAC */
+	if (w->id == snd_soc_dapm_dac && w->active) {
+		out = is_connected_output_ep(w);
+		dapm_clear_walk(w->codec);
+		w->power = (out != 0) ? 1 : 0;
+		dapm_update_bits(w);
+		return 0;
+	}
+
+	/* pre and post event widgets */
+	if (w->id == snd_soc_dapm_pre) {
+		if (!w->event)
+			return 0;
+
+		if (event == SND_SOC_DAPM_STREAM_START) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_PRE_PMU);
+			if (ret < 0)
+				return ret;
+		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_PRE_PMD);
+			if (ret < 0)
+				return ret;
+		}
+		return 0;
+	}
+	if (w->id == snd_soc_dapm_post) {
+		if (!w->event)
+			return 0;
+
+		if (event == SND_SOC_DAPM_STREAM_START) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_POST_PMU);
+			if (ret < 0)
+				return ret;
+		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_POST_PMD);
+			if (ret < 0)
+				return ret;
+		}
+		return 0;
+	}
+
+	/* all other widgets */
+	in = is_connected_input_ep(w);
+	dapm_clear_walk(w->codec);
+	out = is_connected_output_ep(w);
+	dapm_clear_walk(w->codec);
+	power = (out != 0 && in != 0) ? 1 : 0;
+	power_change = (w->power == power) ? 0 : 1;
+	w->power = power;
+
+	if (!power_change)
+		return 0;
+
+	/* call any power change event handlers */
+	if (w->event)
+		pr_debug("power %s event for %s flags %x\n",
+			 w->power ? "on" : "off",
+			 w->name, w->event_flags);
+
+	/* power up pre event */
+	if (power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+		ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* power down pre event */
+	if (!power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+		ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* Lower PGA volume to reduce pops */
+	if (w->id == snd_soc_dapm_pga && !power)
+		dapm_set_pga(w, power);
+
+	dapm_update_bits(w);
+
+	/* Raise PGA volume to reduce pops */
+	if (w->id == snd_soc_dapm_pga && power)
+		dapm_set_pga(w, power);
+
+	/* power up post event */
+	if (power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+		ret = w->event(w,
+			       NULL, SND_SOC_DAPM_POST_PMU);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* power down post event */
+	if (!power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+		ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
+		if (ret < 0)
+			return ret;
+	}
+
+	return 0;
+}
+
+/*
  * Scan each dapm widget for complete audio path.
  * A complete path is a route that has valid endpoints i.e.:-
  *
@@ -514,7 +664,7 @@ EXPORT_SYMBOL_GPL(dapm_reg_event);
 static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 {
 	struct snd_soc_dapm_widget *w;
-	int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
+	int i, c = 1, *seq = NULL, ret = 0;
 
 	/* do we have a sequenced stream event */
 	if (event == SND_SOC_DAPM_STREAM_START) {
@@ -525,135 +675,20 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 		seq = dapm_down_seq;
 	}
 
-	for(i = 0; i < c; i++) {
+	for (i = 0; i < c; i++) {
 		list_for_each_entry(w, &codec->dapm_widgets, list) {
 
 			/* is widget in stream order */
 			if (seq && seq[i] && w->id != seq[i])
 				continue;
 
-			/* vmid - no action */
-			if (w->id == snd_soc_dapm_vmid)
-				continue;
-
-			/* active ADC */
-			if (w->id == snd_soc_dapm_adc && w->active) {
-				in = is_connected_input_ep(w);
-				dapm_clear_walk(w->codec);
-				w->power = (in != 0) ? 1 : 0;
-				dapm_update_bits(w);
-				continue;
-			}
-
-			/* active DAC */
-			if (w->id == snd_soc_dapm_dac && w->active) {
-				out = is_connected_output_ep(w);
-				dapm_clear_walk(w->codec);
-				w->power = (out != 0) ? 1 : 0;
-				dapm_update_bits(w);
-				continue;
-			}
-
-			/* pre and post event widgets */
-			if (w->id == snd_soc_dapm_pre) {
-				if (!w->event)
-					continue;
-
-				if (event == SND_SOC_DAPM_STREAM_START) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_PRE_PMU);
-					if (ret < 0)
-						return ret;
-				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_PRE_PMD);
-					if (ret < 0)
-						return ret;
-				}
-				continue;
-			}
-			if (w->id == snd_soc_dapm_post) {
-				if (!w->event)
-					continue;
-
-				if (event == SND_SOC_DAPM_STREAM_START) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_POST_PMU);
-					if (ret < 0)
-						return ret;
-				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_POST_PMD);
-					if (ret < 0)
-						return ret;
-				}
-				continue;
-			}
-
-			/* all other widgets */
-			in = is_connected_input_ep(w);
-			dapm_clear_walk(w->codec);
-			out = is_connected_output_ep(w);
-			dapm_clear_walk(w->codec);
-			power = (out != 0 && in != 0) ? 1 : 0;
-			power_change = (w->power == power) ? 0: 1;
-			w->power = power;
-
-			if (!power_change)
-				continue;
-
-			/* call any power change event handlers */
-			if (w->event)
-				pr_debug("power %s event for %s flags %x\n",
-					 w->power ? "on" : "off",
-					 w->name, w->event_flags);
-
-			/* power up pre event */
-			if (power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
-				ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
-				if (ret < 0)
-					return ret;
-			}
-
-			/* power down pre event */
-			if (!power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
-				ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
-				if (ret < 0)
-					return ret;
-			}
-
-			/* Lower PGA volume to reduce pops */
-			if (w->id == snd_soc_dapm_pga && !power)
-				dapm_set_pga(w, power);
-
-			dapm_update_bits(w);
-
-			/* Raise PGA volume to reduce pops */
-			if (w->id == snd_soc_dapm_pga && power)
-				dapm_set_pga(w, power);
-
-			/* power up post event */
-			if (power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
-				ret = w->event(w,
-					       NULL, SND_SOC_DAPM_POST_PMU);
-				if (ret < 0)
-					return ret;
-			}
-
-			/* power down post event */
-			if (!power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
-				ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
-				if (ret < 0)
-					return ret;
-			}
+			ret = dapm_power_widget(codec, event, w);
+			if (ret != 0)
+				return ret;
 		}
 	}
 
-	return ret;
+	return 0;
 }
 
 #ifdef DEBUG
@@ -687,6 +722,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_mixer:
+		case snd_soc_dapm_mixer_named_ctl:
 			if (w->name) {
 				in = is_connected_input_ep(w);
 				dapm_clear_walk(w->codec);
@@ -760,6 +796,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
 	int found = 0;
 
 	if (widget->id != snd_soc_dapm_mixer &&
+	    widget->id != snd_soc_dapm_mixer_named_ctl &&
 	    widget->id != snd_soc_dapm_switch)
 		return -ENODEV;
 
@@ -795,7 +832,7 @@ static ssize_t dapm_widget_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_device *devdata = dev_get_drvdata(dev);
-	struct snd_soc_codec *codec = devdata->codec;
+	struct snd_soc_codec *codec = devdata->card->codec;
 	struct snd_soc_dapm_widget *w;
 	int count = 0;
 	char *state = "not set";
@@ -813,6 +850,7 @@ static ssize_t dapm_widget_show(struct device *dev,
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_mixer:
+		case snd_soc_dapm_mixer_named_ctl:
 			if (w->name)
 				count += sprintf(buf + count, "%s: %s\n",
 					w->name, w->power ? "On":"Off");
@@ -876,7 +914,7 @@ static void dapm_free_widgets(struct snd_soc_codec *codec)
 }
 
 static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
-	char *pin, int status)
+				const char *pin, int status)
 {
 	struct snd_soc_dapm_widget *w;
 
@@ -991,6 +1029,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
 		break;
 	case snd_soc_dapm_switch:
 	case snd_soc_dapm_mixer:
+	case snd_soc_dapm_mixer_named_ctl:
 		ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
 		if (ret != 0)
 			goto err;
@@ -1068,6 +1107,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
 		switch(w->id) {
 		case snd_soc_dapm_switch:
 		case snd_soc_dapm_mixer:
+		case snd_soc_dapm_mixer_named_ctl:
 			dapm_new_mixer(codec, w);
 			break;
 		case snd_soc_dapm_mux:
@@ -1396,6 +1436,76 @@ out:
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double);
 
 /**
+ * snd_soc_dapm_info_pin_switch - Info for a pin switch
+ *
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a pin switch control.
+ */
+int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_info_pin_switch);
+
+/**
+ * snd_soc_dapm_get_pin_switch - Get information for a pin switch
+ *
+ * @kcontrol: mixer control
+ * @ucontrol: Value
+ */
+int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	const char *pin = (const char *)kcontrol->private_value;
+
+	mutex_lock(&codec->mutex);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_pin_status(codec, pin);
+
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_switch);
+
+/**
+ * snd_soc_dapm_put_pin_switch - Set information for a pin switch
+ *
+ * @kcontrol: mixer control
+ * @ucontrol: Value
+ */
+int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	const char *pin = (const char *)kcontrol->private_value;
+
+	mutex_lock(&codec->mutex);
+
+	if (ucontrol->value.integer.value[0])
+		snd_soc_dapm_enable_pin(codec, pin);
+	else
+		snd_soc_dapm_disable_pin(codec, pin);
+
+	snd_soc_dapm_sync(codec);
+
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
+
+/**
  * snd_soc_dapm_new_control - create new dapm control
  * @codec: audio codec
  * @widget: widget template
@@ -1527,8 +1637,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 				enum snd_soc_bias_level level)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	if (card->set_bias_level)
@@ -1549,7 +1659,7 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin)
 {
 	return snd_soc_dapm_set_pin(codec, pin, 1);
 }
@@ -1564,7 +1674,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin)
 {
 	return snd_soc_dapm_set_pin(codec, pin, 0);
 }
@@ -1584,7 +1694,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin)
 {
 	return snd_soc_dapm_set_pin(codec, pin, 0);
 }
@@ -1599,7 +1709,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
  *
  * Returns 1 for connected otherwise 0.
  */
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin)
 {
 	struct snd_soc_dapm_widget *w;
 
@@ -1620,7 +1730,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
  */
 void snd_soc_dapm_free(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	snd_soc_dapm_sys_remove(socdev->dev);
 	dapm_free_widgets(codec);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
new file mode 100644
index 0000000..28346fb
--- /dev/null
+++ b/sound/soc/soc-jack.c
@@ -0,0 +1,267 @@
+/*
+ * soc-jack.c  --  ALSA SoC jack handling
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/workqueue.h>
+#include <linux/delay.h>
+
+/**
+ * snd_soc_jack_new - Create a new jack
+ * @card:  ASoC card
+ * @id:    an identifying string for this jack
+ * @type:  a bitmask of enum snd_jack_type values that can be detected by
+ *         this jack
+ * @jack:  structure to use for the jack
+ *
+ * Creates a new jack object.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ * On success jack will be initialised.
+ */
+int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type,
+		     struct snd_soc_jack *jack)
+{
+	jack->card = card;
+	INIT_LIST_HEAD(&jack->pins);
+
+	return snd_jack_new(card->codec->card, id, type, &jack->jack);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_new);
+
+/**
+ * snd_soc_jack_report - Report the current status for a jack
+ *
+ * @jack:   the jack
+ * @status: a bitmask of enum snd_jack_type values that are currently detected.
+ * @mask:   a bitmask of enum snd_jack_type values that being reported.
+ *
+ * If configured using snd_soc_jack_add_pins() then the associated
+ * DAPM pins will be enabled or disabled as appropriate and DAPM
+ * synchronised.
+ *
+ * Note: This function uses mutexes and should be called from a
+ * context which can sleep (such as a workqueue).
+ */
+void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
+{
+	struct snd_soc_codec *codec = jack->card->codec;
+	struct snd_soc_jack_pin *pin;
+	int enable;
+	int oldstatus;
+
+	if (!jack) {
+		WARN_ON_ONCE(!jack);
+		return;
+	}
+
+	mutex_lock(&codec->mutex);
+
+	oldstatus = jack->status;
+
+	jack->status &= ~mask;
+	jack->status |= status;
+
+	/* The DAPM sync is expensive enough to be worth skipping */
+	if (jack->status == oldstatus)
+		goto out;
+
+	list_for_each_entry(pin, &jack->pins, list) {
+		enable = pin->mask & status;
+
+		if (pin->invert)
+			enable = !enable;
+
+		if (enable)
+			snd_soc_dapm_enable_pin(codec, pin->pin);
+		else
+			snd_soc_dapm_disable_pin(codec, pin->pin);
+	}
+
+	snd_soc_dapm_sync(codec);
+
+	snd_jack_report(jack->jack, status);
+
+out:
+	mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_report);
+
+/**
+ * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack
+ *
+ * @jack:  ASoC jack
+ * @count: Number of pins
+ * @pins:  Array of pins
+ *
+ * After this function has been called the DAPM pins specified in the
+ * pins array will have their status updated to reflect the current
+ * state of the jack whenever the jack status is updated.
+ */
+int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
+			  struct snd_soc_jack_pin *pins)
+{
+	int i;
+
+	for (i = 0; i < count; i++) {
+		if (!pins[i].pin) {
+			printk(KERN_ERR "No name for pin %d\n", i);
+			return -EINVAL;
+		}
+		if (!pins[i].mask) {
+			printk(KERN_ERR "No mask for pin %d (%s)\n", i,
+			       pins[i].pin);
+			return -EINVAL;
+		}
+
+		INIT_LIST_HEAD(&pins[i].list);
+		list_add(&(pins[i].list), &jack->pins);
+	}
+
+	/* Update to reflect the last reported status; canned jack
+	 * implementations are likely to set their state before the
+	 * card has an opportunity to associate pins.
+	 */
+	snd_soc_jack_report(jack, 0, 0);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins);
+
+#ifdef CONFIG_GPIOLIB
+/* gpio detect */
+static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
+{
+	struct snd_soc_jack *jack = gpio->jack;
+	int enable;
+	int report;
+
+	if (gpio->debounce_time > 0)
+		mdelay(gpio->debounce_time);
+
+	enable = gpio_get_value(gpio->gpio);
+	if (gpio->invert)
+		enable = !enable;
+
+	if (enable)
+		report = gpio->report;
+	else
+		report = 0;
+
+	snd_soc_jack_report(jack, report, gpio->report);
+}
+
+/* irq handler for gpio pin */
+static irqreturn_t gpio_handler(int irq, void *data)
+{
+	struct snd_soc_jack_gpio *gpio = data;
+
+	schedule_work(&gpio->work);
+
+	return IRQ_HANDLED;
+}
+
+/* gpio work */
+static void gpio_work(struct work_struct *work)
+{
+	struct snd_soc_jack_gpio *gpio;
+
+	gpio = container_of(work, struct snd_soc_jack_gpio, work);
+	snd_soc_jack_gpio_detect(gpio);
+}
+
+/**
+ * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack
+ *
+ * @jack:  ASoC jack
+ * @count: number of pins
+ * @gpios: array of gpio pins
+ *
+ * This function will request gpio, set data direction and request irq
+ * for each gpio in the array.
+ */
+int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
+			struct snd_soc_jack_gpio *gpios)
+{
+	int i, ret;
+
+	for (i = 0; i < count; i++) {
+		if (!gpio_is_valid(gpios[i].gpio)) {
+			printk(KERN_ERR "Invalid gpio %d\n",
+				gpios[i].gpio);
+			ret = -EINVAL;
+			goto undo;
+		}
+		if (!gpios[i].name) {
+			printk(KERN_ERR "No name for gpio %d\n",
+				gpios[i].gpio);
+			ret = -EINVAL;
+			goto undo;
+		}
+
+		ret = gpio_request(gpios[i].gpio, gpios[i].name);
+		if (ret)
+			goto undo;
+
+		ret = gpio_direction_input(gpios[i].gpio);
+		if (ret)
+			goto err;
+
+		ret = request_irq(gpio_to_irq(gpios[i].gpio),
+				gpio_handler,
+				IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+				jack->card->dev->driver->name,
+				&gpios[i]);
+		if (ret)
+			goto err;
+
+		INIT_WORK(&gpios[i].work, gpio_work);
+		gpios[i].jack = jack;
+	}
+
+	return 0;
+
+err:
+	gpio_free(gpios[i].gpio);
+undo:
+	snd_soc_jack_free_gpios(jack, i, gpios);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios);
+
+/**
+ * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack
+ *
+ * @jack:  ASoC jack
+ * @count: number of pins
+ * @gpios: array of gpio pins
+ *
+ * Release gpio and irq resources for gpio pins associated with an ASoC jack.
+ */
+void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
+			struct snd_soc_jack_gpio *gpios)
+{
+	int i;
+
+	for (i = 0; i < count; i++) {
+		free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]);
+		gpio_free(gpios[i].gpio);
+		gpios[i].jack = NULL;
+	}
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios);
+#endif	/* CONFIG_GPIOLIB */