diff --git a/MAINTAINERS b/MAINTAINERS
index 29801f7..7bcc395 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -533,6 +533,8 @@
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 W:	http://wiki.analog.com/
 S:	Supported
+F:	sound/soc/codecs/adau*
+F:	sound/soc/codecs/adav*
 F:	sound/soc/codecs/ad1*
 F:	sound/soc/codecs/ssm*
 
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 1bafe95..5ad5f3a 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -209,6 +209,10 @@
 	struct snd_soc_pcm_stream capture;
 	struct snd_soc_pcm_stream playback;
 	unsigned int symmetric_rates:1;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c46e7d8..e09505c 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -348,6 +348,8 @@
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+			     const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
@@ -429,6 +431,7 @@
 	/* status */
 	u32 connect:1;	/* source and sink widgets are connected */
 	u32 walked:1;	/* path has been walked */
+	u32 weak:1;	/* path ignored for power management */
 
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink);
@@ -444,6 +447,7 @@
 	char *name;		/* widget name */
 	char *sname;	/* stream name */
 	struct snd_soc_codec *codec;
+	struct snd_soc_platform *platform;
 	struct list_head list;
 	struct snd_soc_dapm_context *dapm;
 
@@ -507,10 +511,11 @@
 
 	struct device *dev; /* from parent - for debug */
 	struct snd_soc_codec *codec; /* parent codec */
+	struct snd_soc_platform *platform; /* parent platform */
 	struct snd_soc_card *card; /* parent card */
 
 	/* used during DAPM updates */
-	int dev_power;
+	enum snd_soc_bias_level target_bias_level;
 	struct list_head list;
 
 #ifdef CONFIG_DEBUG_FS
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3a4bd3a..aa19f5a 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -203,6 +203,16 @@
 	SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
 
 /*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST		-2
+#define SND_SOC_COMP_ORDER_EARLY		-1
+#define SND_SOC_COMP_ORDER_NORMAL		0
+#define SND_SOC_COMP_ORDER_LATE		1
+#define SND_SOC_COMP_ORDER_LAST		2
+
+/*
  * Bias levels
  *
  * @ON:      Bias is fully on for audio playback and capture operations.
@@ -214,10 +224,10 @@
  * @OFF:     Power Off. No restrictions on transition times.
  */
 enum snd_soc_bias_level {
-	SND_SOC_BIAS_OFF,
-	SND_SOC_BIAS_STANDBY,
-	SND_SOC_BIAS_PREPARE,
-	SND_SOC_BIAS_ON,
+	SND_SOC_BIAS_OFF = 0,
+	SND_SOC_BIAS_STANDBY = 1,
+	SND_SOC_BIAS_PREPARE = 2,
+	SND_SOC_BIAS_ON = 3,
 };
 
 struct snd_jack;
@@ -258,6 +268,11 @@
 	SND_SOC_RBTREE_COMPRESSION
 };
 
+enum snd_soc_pcm_subclass {
+	SND_SOC_PCM_CLASS_PCM	= 0,
+	SND_SOC_PCM_CLASS_BE	= 1,
+};
+
 int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			     unsigned int freq, int dir);
 int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -297,6 +312,10 @@
 				      unsigned int reg);
 int snd_soc_default_writable_register(struct snd_soc_codec *codec,
 				      unsigned int reg);
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+					unsigned int reg);
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+					unsigned int reg, unsigned int val);
 
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
@@ -349,6 +368,8 @@
 				  const char *prefix);
 int snd_soc_add_controls(struct snd_soc_codec *codec,
 	const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
+	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@@ -612,6 +633,10 @@
 
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /* SoC platform interface */
@@ -623,10 +648,17 @@
 	int (*resume)(struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
-		struct snd_pcm *);
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
 	void (*pcm_free)(struct snd_pcm *);
 
+	/* Default control and setup, added after probe() is run */
+	const struct snd_kcontrol_new *controls;
+	int num_controls;
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	int num_dapm_routes;
+
 	/*
 	 * For platform caused delay reporting.
 	 * Optional.
@@ -636,6 +668,14 @@
 
 	/* platform stream ops */
 	struct snd_pcm_ops *ops;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
+
+	/* platform IO - used for platform DAPM */
+	unsigned int (*read)(struct snd_soc_platform *, unsigned int);
+	int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
 };
 
 struct snd_soc_platform {
@@ -650,6 +690,8 @@
 	struct snd_soc_card *card;
 	struct list_head list;
 	struct list_head card_list;
+
+	struct snd_soc_dapm_context dapm;
 };
 
 struct snd_soc_dai_link {
@@ -725,8 +767,10 @@
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_card *,
+			      struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 	int (*set_bias_level_post)(struct snd_soc_card *,
+				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level);
 
 	long pmdown_time;
@@ -789,6 +833,9 @@
 	struct device dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
+	struct mutex pcm_mutex;
+	enum snd_soc_pcm_subclass pcm_subclass;
+	struct snd_pcm_ops ops;
 
 	unsigned int complete:1;
 	unsigned int dev_registered:1;
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index ae973d2..603f5a0 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -9,6 +9,7 @@
 
 struct snd_soc_jack;
 struct snd_soc_codec;
+struct snd_soc_platform;
 struct snd_soc_card;
 struct snd_soc_dapm_widget;
 
@@ -59,6 +60,50 @@
 
 );
 
+DECLARE_EVENT_CLASS(snd_soc_preg,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val),
+
+	TP_STRUCT__entry(
+		__string(	name,		platform->name	)
+		__field(	int,		id		)
+		__field(	unsigned int,	reg		)
+		__field(	unsigned int,	val		)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, platform->name);
+		__entry->id = platform->id;
+		__entry->reg = reg;
+		__entry->val = val;
+	),
+
+	TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name),
+		  (int)__entry->id, (unsigned int)__entry->reg,
+		  (unsigned int)__entry->val)
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val)
+
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val)
+
+);
+
 DECLARE_EVENT_CLASS(snd_soc_card,
 
 	TP_PROTO(struct snd_soc_card *card, int val),
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 1ed61c5..4f91387 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,5 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
+snd-soc-core-objs += soc-pcm.o soc-io.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 d0e7532..f81d4c3 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -364,9 +364,11 @@
 \*--------------------------------------------------------------------------*/
 static u64 atmel_pcm_dmamask = 0xffffffff;
 
-static int atmel_pcm_new(struct snd_card *card,
-	struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
@@ -382,7 +384,7 @@
 	}
 
 	if (dai->driver->capture.channels_min) {
-		pr_debug("at32-pcm:"
+		pr_debug("atmel-pcm:"
 				"Allocating PCM capture DMA buffer\n");
 		ret = atmel_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h
index 2597329..5e0a95e 100644
--- a/sound/soc/atmel/atmel-pcm.h
+++ b/sound/soc/atmel/atmel-pcm.h
@@ -60,7 +60,7 @@
  * This structure, shared between the PCM driver and the interface,
  * contains all information required by the PCM driver to perform the
  * PDC DMA operation.  All fields except dma_intr_handler() are initialized
- * by the interface.  The dms_intr_handler() pointer is set by the PCM
+ * by the interface.  The dma_intr_handler() pointer is set by the PCM
  * driver and called by the interface SSC interrupt handler if it is
  * non-NULL.
  */
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index eda955b..7122509 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -402,7 +402,7 @@
 	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
 		&& bits > 16) {
 		printk(KERN_WARNING
-				"atmel_ssc_dai: sample size %d"
+				"atmel_ssc_dai: sample size %d "
 				"is too large for I2S\n", bits);
 		return -EINVAL;
 	}
@@ -838,10 +838,8 @@
 	}
 
 	ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id);
-	if (!ssc_pdev) {
-		ssc_free(ssc);
+	if (!ssc_pdev)
 		return -ENOMEM;
-	}
 
 	/* If we can grab the SSC briefly to parent the DAI device off it */
 	ssc = ssc_request(ssc_id);
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 95572d2..bad3aa1 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -92,6 +92,7 @@
 };
 
 static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
 					enum snd_soc_bias_level level)
 {
 	static int mclk_on;
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 10fdd28..20bb53a 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -319,10 +319,11 @@
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-static int au1xpsc_pcm_new(struct snd_card *card,
-			   struct snd_soc_dai *dai,
-			   struct snd_pcm *pcm)
+static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 		card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
 
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ae40359..fe9d548 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -10,13 +10,36 @@
 
 config SND_BF5XX_SOC_SSM2602
 	tristate "SoC SSM2602 Audio support for BF52x ezkit"
-	depends on SND_BF5XX_I2S
+	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_SSM2602
-	select I2C
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
+config SND_SOC_BFIN_EVAL_ADAU1701
+	tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
+	depends on SND_BF5XX_I2S
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_ADAU1701
+	select I2C
+	help
+	  Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ
+	  board connected to one of the Blackfin evaluation boards like the
+	  BF5XX-STAMP or BF5XX-EZKIT.
+
+config SND_SOC_BFIN_EVAL_ADAV80X
+	tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
+	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_ADAV80X
+	help
+	  Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
+	  EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
+	  like the BF5XX-STAMP or BF5XX-EZKIT.
+
+	  Note: This driver assumes that the ADAV80X digital record and playback
+	  interfaces are connected to the first SPORT port on the BF5XX board.
+
 config SND_BF5XX_SOC_AD73311
 	tristate "SoC AD73311 Audio support for Blackfin"
 	depends on SND_BF5XX_I2S
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 49af3f3..6018bf5 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -21,9 +21,13 @@
 snd-ssm2602-objs := bf5xx-ssm2602.o
 snd-ad73311-objs := bf5xx-ad73311.o
 snd-ad193x-objs := bf5xx-ad193x.o
+snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
+snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
 
 obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
 obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 98b44b3..9e59f68 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -418,9 +418,11 @@
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	pr_debug("%s enter\n", __func__);
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index f1fd95b..61ddf94 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -168,7 +168,7 @@
 
 	snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware);
 
-	ret = snd_pcm_hw_constraint_integer(runtime, \
+	ret = snd_pcm_hw_constraint_integer(runtime,
 			SNDRV_PCM_HW_PARAM_PERIODS);
 	if (ret < 0)
 		goto out;
@@ -257,9 +257,11 @@
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	pr_debug("%s enter\n", __func__);
@@ -304,8 +306,8 @@
 
 static struct platform_driver bfin_i2s_pcm_driver = {
 	.driver = {
-			.name = "bfin-i2s-pcm-audio",
-			.owner = THIS_MODULE,
+		.name = "bfin-i2s-pcm-audio",
+		.owner = THIS_MODULE,
 	},
 
 	.probe = bfin_i2s_soc_platform_probe,
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c
index 07cfc7a..c95cc03 100644
--- a/sound/soc/blackfin/bf5xx-tdm-pcm.c
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c
@@ -283,9 +283,11 @@
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
new file mode 100644
index 0000000..e5550ac
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -0,0 +1,139 @@
+/*
+ * Machine driver for EVAL-ADAU1701MINIZ on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/adau1701.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adau1701_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adau1701_dapm_routes[] = {
+	{ "Speaker", NULL, "OUT0" },
+	{ "Speaker", NULL, "OUT1" },
+	{ "Line Out", NULL, "OUT2" },
+	{ "Line Out", NULL, "OUT3" },
+
+	{ "IN0", NULL, "Line In" },
+	{ "IN1", NULL, "Line In" },
+};
+
+static int bfin_eval_adau1701_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 *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000,
+			SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+static struct snd_soc_ops bfin_eval_adau1701_ops = {
+	.hw_params = bfin_eval_adau1701_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = {
+	{
+		.name = "adau1701",
+		.stream_name = "adau1701",
+		.cpu_dai_name = "bfin-i2s.0",
+		.codec_dai_name = "adau1701",
+		.platform_name = "bfin-i2s-pcm-audio",
+		.codec_name = "adau1701.0-0034",
+		.ops = &bfin_eval_adau1701_ops,
+	},
+	{
+		.name = "adau1701",
+		.stream_name = "adau1701",
+		.cpu_dai_name = "bfin-i2s.1",
+		.codec_dai_name = "adau1701",
+		.platform_name = "bfin-i2s-pcm-audio",
+		.codec_name = "adau1701.0-0034",
+		.ops = &bfin_eval_adau1701_ops,
+	},
+};
+
+static struct snd_soc_card bfin_eval_adau1701 = {
+	.name = "bfin-eval-adau1701",
+	.dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM],
+	.num_links = 1,
+
+	.dapm_widgets		= bfin_eval_adau1701_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adau1701_dapm_widgets),
+	.dapm_routes		= bfin_eval_adau1701_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adau1701_dapm_routes),
+};
+
+static int bfin_eval_adau1701_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &bfin_eval_adau1701;
+
+	card->dev = &pdev->dev;
+
+	return snd_soc_register_card(&bfin_eval_adau1701);
+}
+
+static int __devexit bfin_eval_adau1701_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver bfin_eval_adau1701_driver = {
+	.driver = {
+		.name = "bfin-eval-adau1701",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bfin_eval_adau1701_probe,
+	.remove = __devexit_p(bfin_eval_adau1701_remove),
+};
+
+static int __init bfin_eval_adau1701_init(void)
+{
+	return platform_driver_register(&bfin_eval_adau1701_driver);
+}
+module_init(bfin_eval_adau1701_init);
+
+static void __exit bfin_eval_adau1701_exit(void)
+{
+	platform_driver_unregister(&bfin_eval_adau1701_driver);
+}
+module_exit(bfin_eval_adau1701_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bfin-eval-adau1701");
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
new file mode 100644
index 0000000..8d014d0
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -0,0 +1,173 @@
+/*
+ * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include "../codecs/adav80x.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = {
+	{ "Line Out", NULL, "VOUTL" },
+	{ "Line Out", NULL, "VOUTR" },
+
+	{ "VINL", NULL, "Line In" },
+	{ "VINR", NULL, "Line In" },
+};
+
+static int bfin_eval_adav80x_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 *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL,
+			27000000, params_rate(params) * 256);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1,
+			params_rate(params) * 256, SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0,
+	    SND_SOC_CLOCK_OUT);
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0,
+	    SND_SOC_CLOCK_OUT);
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0,
+	    SND_SOC_CLOCK_OUT);
+
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0);
+
+	return 0;
+}
+
+static struct snd_soc_ops bfin_eval_adav80x_ops = {
+	.hw_params = bfin_eval_adav80x_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = {
+	{
+		.name = "adav80x",
+		.stream_name = "ADAV80x HiFi",
+		.cpu_dai_name = "bfin-i2s.0",
+		.codec_dai_name = "adav80x-hifi",
+		.platform_name = "bfin-i2s-pcm-audio",
+		.init = bfin_eval_adav80x_codec_init,
+		.ops = &bfin_eval_adav80x_ops,
+	},
+};
+
+static struct snd_soc_card bfin_eval_adav80x = {
+	.name = "bfin-eval-adav80x",
+	.dai_link = bfin_eval_adav80x_dais,
+	.num_links = ARRAY_SIZE(bfin_eval_adav80x_dais),
+
+	.dapm_widgets		= bfin_eval_adav80x_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets),
+	.dapm_routes		= bfin_eval_adav80x_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adav80x_dapm_routes),
+};
+
+enum bfin_eval_adav80x_type {
+	BFIN_EVAL_ADAV801,
+	BFIN_EVAL_ADAV803,
+};
+
+static int bfin_eval_adav80x_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &bfin_eval_adav80x;
+	const char *codec_name;
+
+	switch (platform_get_device_id(pdev)->driver_data) {
+	case BFIN_EVAL_ADAV801:
+		codec_name = "spi0.1";
+		break;
+	case BFIN_EVAL_ADAV803:
+		codec_name = "adav803.0-0034";
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	bfin_eval_adav80x_dais[0].codec_name = codec_name;
+
+	card->dev = &pdev->dev;
+
+	return snd_soc_register_card(&bfin_eval_adav80x);
+}
+
+static int __devexit bfin_eval_adav80x_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static const struct platform_device_id bfin_eval_adav80x_ids[] = {
+	{ "bfin-eval-adav801", BFIN_EVAL_ADAV801 },
+	{ "bfin-eval-adav803", BFIN_EVAL_ADAV803 },
+	{ },
+};
+MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids);
+
+static struct platform_driver bfin_eval_adav80x_driver = {
+	.driver = {
+		.name = "bfin-eval-adav80x",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bfin_eval_adav80x_probe,
+	.remove = __devexit_p(bfin_eval_adav80x_remove),
+	.id_table = bfin_eval_adav80x_ids,
+};
+
+static int __init bfin_eval_adav80x_init(void)
+{
+	return platform_driver_register(&bfin_eval_adav80x_driver);
+}
+module_init(bfin_eval_adav80x_init);
+
+static void __exit bfin_eval_adav80x_exit(void)
+{
+	platform_driver_unregister(&bfin_eval_adav80x_driver);
+}
+module_exit(bfin_eval_adav80x_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 98175a0..ff43405 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -17,6 +17,7 @@
 	select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311
+	select SND_SOC_ADAV80X
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
@@ -42,6 +43,7 @@
 	select SND_SOC_SN95031 if INTEL_SCU_IPC
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_STA32X if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -71,6 +73,7 @@
 	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8770 if SPI_MASTER
 	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8782
 	select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
@@ -130,7 +133,14 @@
 
 config SND_SOC_AD73311
 	tristate
-	
+
+config SND_SOC_ADAU1701
+	select SIGMA
+	tristate
+
+config SND_SOC_ADAV80X
+	tristate
+
 config SND_SOC_ADS117X
 	tristate
 
@@ -216,6 +226,9 @@
 config SND_SOC_SSM2602
 	tristate
 
+config SND_SOC_STA32X
+	tristate
+
 config SND_SOC_STAC9766
 	tristate
 
@@ -299,6 +312,9 @@
 config SND_SOC_WM8776
 	tristate
 
+config SND_SOC_WM8782
+	tristate
+
 config SND_SOC_WM8804
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fd85584..4957431 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -4,6 +4,8 @@
 snd-soc-ad193x-objs := ad193x.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
+snd-soc-adau1701-objs := adau1701.o
+snd-soc-adav80x-objs := adav80x.o
 snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
@@ -28,6 +30,7 @@
 snd-soc-sn95031-objs := sn95031.o
 snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-sta32x-objs := sta32x.o
 snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -55,6 +58,7 @@
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8770-objs := wm8770.o
 snd-soc-wm8776-objs := wm8776.o
+snd-soc-wm8782-objs := wm8782.o
 snd-soc-wm8804-objs := wm8804.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
@@ -95,6 +99,8 @@
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU1701)  += snd-soc-adau1701.o
+obj-$(CONFIG_SND_SOC_ADAV80X)  += snd-soc-adav80x.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
@@ -120,6 +126,7 @@
 obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
@@ -147,6 +154,7 @@
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8770)	+= snd-soc-wm8770.o
 obj-$(CONFIG_SND_SOC_WM8776)	+= snd-soc-wm8776.o
+obj-$(CONFIG_SND_SOC_WM8782)	+= snd-soc-wm8782.o
 obj-$(CONFIG_SND_SOC_WM8804)	+= snd-soc-wm8804.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 754c496..4e5c572 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -1,19 +1,10 @@
-/*
- * File:         sound/soc/codecs/ad1836.c
- * Author:       Barry Song <Barry.Song@analog.com>
+ /*
+ * Audio Codec driver supporting:
+ *  AD1835A, AD1836, AD1837A, AD1838A, AD1839A
  *
- * Created:      Aug 04 2009
- * Description:  Driver for AD1836 sound chip
+ * Copyright 2009-2011 Analog Devices Inc.
  *
- * Modified:
- *               Copyright 2009 Analog Devices Inc.
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * Licensed under the GPL-2 or later.
  */
 
 #include <linux/init.h>
@@ -30,10 +21,15 @@
 #include <linux/spi/spi.h>
 #include "ad1836.h"
 
+enum ad1836_type {
+	AD1835,
+	AD1836,
+	AD1838,
+};
+
 /* codec private data */
 struct ad1836_priv {
-	enum snd_soc_control_type control_type;
-	void *control_data;
+	enum ad1836_type type;
 };
 
 /*
@@ -44,29 +40,60 @@
 static const struct soc_enum ad1836_deemp_enum =
 	SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
 
-static const struct snd_kcontrol_new ad1836_snd_controls[] = {
-	/* DAC volume control */
-	SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL,
-			AD1836_DAC_R1_VOL, 0, 0x3FF, 0),
-	SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL,
-			AD1836_DAC_R2_VOL, 0, 0x3FF, 0),
-	SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL,
-			AD1836_DAC_R3_VOL, 0, 0x3FF, 0),
+#define AD1836_DAC_VOLUME(x) \
+	SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \
+			AD1836_DAC_R_VOL(x), 0, 0x3FF, 0)
 
-	/* ADC switch control */
-	SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE,
-		AD1836_ADCR1_MUTE, 1, 1),
-	SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE,
-		AD1836_ADCR2_MUTE, 1, 1),
+#define AD1836_DAC_SWITCH(x) \
+	SOC_DOUBLE("DAC" #x " Playback Switch", AD1836_DAC_CTRL2, \
+			AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
 
-	/* DAC switch control */
-	SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE,
-		AD1836_DACR1_MUTE, 1, 1),
-	SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE,
-		AD1836_DACR2_MUTE, 1, 1),
-	SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE,
-		AD1836_DACR3_MUTE, 1, 1),
+#define AD1836_ADC_SWITCH(x) \
+	SOC_DOUBLE("ADC" #x " Capture Switch", AD1836_ADC_CTRL2, \
+		AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
 
+static const struct snd_kcontrol_new ad183x_dac_controls[] = {
+	AD1836_DAC_VOLUME(1),
+	AD1836_DAC_SWITCH(1),
+	AD1836_DAC_VOLUME(2),
+	AD1836_DAC_SWITCH(2),
+	AD1836_DAC_VOLUME(3),
+	AD1836_DAC_SWITCH(3),
+	AD1836_DAC_VOLUME(4),
+	AD1836_DAC_SWITCH(4),
+};
+
+static const struct snd_soc_dapm_widget ad183x_dac_dapm_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("DAC1OUT"),
+	SND_SOC_DAPM_OUTPUT("DAC2OUT"),
+	SND_SOC_DAPM_OUTPUT("DAC3OUT"),
+	SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+};
+
+static const struct snd_soc_dapm_route ad183x_dac_routes[] = {
+	{ "DAC1OUT", NULL, "DAC" },
+	{ "DAC2OUT", NULL, "DAC" },
+	{ "DAC3OUT", NULL, "DAC" },
+	{ "DAC4OUT", NULL, "DAC" },
+};
+
+static const struct snd_kcontrol_new ad183x_adc_controls[] = {
+	AD1836_ADC_SWITCH(1),
+	AD1836_ADC_SWITCH(2),
+	AD1836_ADC_SWITCH(3),
+};
+
+static const struct snd_soc_dapm_widget ad183x_adc_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("ADC1IN"),
+	SND_SOC_DAPM_INPUT("ADC2IN"),
+};
+
+static const struct snd_soc_dapm_route ad183x_adc_routes[] = {
+	{ "ADC", NULL, "ADC1IN" },
+	{ "ADC", NULL, "ADC2IN" },
+};
+
+static const struct snd_kcontrol_new ad183x_controls[] = {
 	/* ADC high-pass filter */
 	SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1,
 			AD1836_ADC_HIGHPASS_FILTER, 1, 0),
@@ -75,27 +102,24 @@
 	SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum),
 };
 
-static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget ad183x_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1,
 				AD1836_DAC_POWERDOWN, 1),
 	SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1,
 				AD1836_ADC_POWERDOWN, 1, NULL, 0),
-	SND_SOC_DAPM_OUTPUT("DAC1OUT"),
-	SND_SOC_DAPM_OUTPUT("DAC2OUT"),
-	SND_SOC_DAPM_OUTPUT("DAC3OUT"),
-	SND_SOC_DAPM_INPUT("ADC1IN"),
-	SND_SOC_DAPM_INPUT("ADC2IN"),
 };
 
-static const struct snd_soc_dapm_route audio_paths[] = {
+static const struct snd_soc_dapm_route ad183x_dapm_routes[] = {
 	{ "DAC", NULL, "ADC_PWR" },
 	{ "ADC", NULL, "ADC_PWR" },
-	{ "DAC1OUT", "DAC1 Switch", "DAC" },
-	{ "DAC2OUT", "DAC2 Switch", "DAC" },
-	{ "DAC3OUT", "DAC3 Switch", "DAC" },
-	{ "ADC", "ADC1 Switch", "ADC1IN" },
-	{ "ADC", "ADC2 Switch", "ADC2IN" },
+};
+
+static const DECLARE_TLV_DB_SCALE(ad1836_in_tlv, 0, 300, 0);
+
+static const struct snd_kcontrol_new ad1836_controls[] = {
+	SOC_DOUBLE_TLV("ADC2 Capture Volume", AD1836_ADC_CTRL1, 3, 0, 4, 0,
+	    ad1836_in_tlv),
 };
 
 /*
@@ -165,64 +189,69 @@
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int ad1836_soc_suspend(struct snd_soc_codec *codec,
-		pm_message_t state)
-{
-	/* reset clock control mode */
-	u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
-	adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
-
-	return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
-}
-
-static int ad1836_soc_resume(struct snd_soc_codec *codec)
-{
-	/* restore clock control mode */
-	u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
-	adc_ctrl2 |= AD1836_ADC_AUX;
-
-	return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
-}
-#else
-#define ad1836_soc_suspend NULL
-#define ad1836_soc_resume  NULL
-#endif
-
 static struct snd_soc_dai_ops ad1836_dai_ops = {
 	.hw_params = ad1836_hw_params,
 	.set_fmt = ad1836_set_dai_fmt,
 };
 
-/* codec DAI instance */
-static struct snd_soc_dai_driver ad1836_dai = {
-	.name = "ad1836-hifi",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 2,
-		.channels_max = 6,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 2,
-		.channels_max = 4,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.ops = &ad1836_dai_ops,
+#define AD183X_DAI(_name, num_dacs, num_adcs) \
+{ \
+	.name = _name "-hifi", \
+	.playback = { \
+		.stream_name = "Playback", \
+		.channels_min = 2, \
+		.channels_max = (num_dacs) * 2, \
+		.rates = SNDRV_PCM_RATE_48000,  \
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
+	}, \
+	.capture = { \
+		.stream_name = "Capture", \
+		.channels_min = 2, \
+		.channels_max = (num_adcs) * 2, \
+		.rates = SNDRV_PCM_RATE_48000, \
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
+	}, \
+	.ops = &ad1836_dai_ops, \
+}
+
+static struct snd_soc_dai_driver ad183x_dais[] = {
+	[AD1835] = AD183X_DAI("ad1835", 4, 1),
+	[AD1836] = AD183X_DAI("ad1836", 3, 2),
+	[AD1838] = AD183X_DAI("ad1838", 3, 1),
 };
 
+#ifdef CONFIG_PM
+static int ad1836_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	/* reset clock control mode */
+	return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+		AD1836_ADC_SERFMT_MASK, 0);
+}
+
+static int ad1836_resume(struct snd_soc_codec *codec)
+{
+	/* restore clock control mode */
+	return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+		AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX);
+}
+#else
+#define ad1836_suspend NULL
+#define ad1836_resume  NULL
+#endif
+
 static int ad1836_probe(struct snd_soc_codec *codec)
 {
 	struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int num_dacs, num_adcs;
 	int ret = 0;
+	int i;
 
-	codec->control_data = ad1836->control_data;
+	num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2;
+	num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2;
+
 	ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI);
 	if (ret < 0) {
 		dev_err(codec->dev, "failed to set cache I/O: %d\n",
@@ -239,21 +268,46 @@
 	snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100);
 	/* unmute adc channles, adc aux mode */
 	snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180);
-	/* left/right diff:PGA/MUX */
-	snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
 	/* volume */
-	snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF);
-	snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF);
-	snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF);
-	snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF);
-	snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF);
-	snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF);
+	for (i = 1; i <= num_dacs; ++i) {
+		snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF);
+		snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF);
+	}
 
-	snd_soc_add_controls(codec, ad1836_snd_controls,
-			     ARRAY_SIZE(ad1836_snd_controls));
-	snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets,
-				  ARRAY_SIZE(ad1836_dapm_widgets));
-	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
+	if (ad1836->type == AD1836) {
+		/* left/right diff:PGA/MUX */
+		snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
+		ret = snd_soc_add_controls(codec, ad1836_controls,
+				ARRAY_SIZE(ad1836_controls));
+		if (ret)
+			return ret;
+	} else {
+		snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00);
+	}
+
+	ret = snd_soc_add_controls(codec, ad183x_dac_controls, num_dacs * 2);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_add_controls(codec, ad183x_adc_controls, num_adcs);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, ad183x_dac_dapm_widgets, num_dacs);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, ad183x_adc_dapm_widgets, num_adcs);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_add_routes(dapm, ad183x_dac_routes, num_dacs);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_add_routes(dapm, ad183x_adc_routes, num_adcs);
+	if (ret)
+		return ret;
 
 	return ret;
 }
@@ -262,19 +316,24 @@
 static int ad1836_remove(struct snd_soc_codec *codec)
 {
 	/* reset clock control mode */
-	u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
-	adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
-
-	return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
+	return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+		AD1836_ADC_SERFMT_MASK, 0);
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_ad1836 = {
-	.probe = 	ad1836_probe,
-	.remove = 	ad1836_remove,
-	.suspend =      ad1836_soc_suspend,
-	.resume =       ad1836_soc_resume,
+	.probe = ad1836_probe,
+	.remove = ad1836_remove,
+	.suspend = ad1836_suspend,
+	.resume = ad1836_resume,
 	.reg_cache_size = AD1836_NUM_REGS,
 	.reg_word_size = sizeof(u16),
+
+	.controls = ad183x_controls,
+	.num_controls = ARRAY_SIZE(ad183x_controls),
+	.dapm_widgets = ad183x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ad183x_dapm_widgets),
+	.dapm_routes = ad183x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes),
 };
 
 static int __devinit ad1836_spi_probe(struct spi_device *spi)
@@ -286,12 +345,12 @@
 	if (ad1836 == NULL)
 		return -ENOMEM;
 
+	ad1836->type = spi_get_device_id(spi)->driver_data;
+
 	spi_set_drvdata(spi, ad1836);
-	ad1836->control_data = spi;
-	ad1836->control_type = SND_SOC_SPI;
 
 	ret = snd_soc_register_codec(&spi->dev,
-			&soc_codec_dev_ad1836, &ad1836_dai, 1);
+			&soc_codec_dev_ad1836, &ad183x_dais[ad1836->type], 1);
 	if (ret < 0)
 		kfree(ad1836);
 	return ret;
@@ -303,27 +362,29 @@
 	kfree(spi_get_drvdata(spi));
 	return 0;
 }
+static const struct spi_device_id ad1836_ids[] = {
+	{ "ad1835", AD1835 },
+	{ "ad1836", AD1836 },
+	{ "ad1837", AD1835 },
+	{ "ad1838", AD1838 },
+	{ "ad1839", AD1838 },
+	{ },
+};
+MODULE_DEVICE_TABLE(spi, ad1836_ids);
 
 static struct spi_driver ad1836_spi_driver = {
 	.driver = {
-		.name	= "ad1836-codec",
+		.name	= "ad1836",
 		.owner	= THIS_MODULE,
 	},
 	.probe		= ad1836_spi_probe,
 	.remove		= __devexit_p(ad1836_spi_remove),
+	.id_table	= ad1836_ids,
 };
 
 static int __init ad1836_init(void)
 {
-	int ret;
-
-	ret = spi_register_driver(&ad1836_spi_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n",
-				ret);
-	}
-
-	return ret;
+	return spi_register_driver(&ad1836_spi_driver);
 }
 module_init(ad1836_init);
 
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 9d6a3f8..444747f 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -1,19 +1,10 @@
 /*
- * File:         sound/soc/codecs/ad1836.h
- * Based on:
- * Author:       Barry Song <Barry.Song@analog.com>
+ * Audio Codec driver supporting:
+ *  AD1835A, AD1836, AD1837A, AD1838A, AD1839A
  *
- * Created:      Aug 04, 2009
- * Description:  definitions for AD1836 registers
+ * Copyright 2009-2011 Analog Devices Inc.
  *
- * Modified:
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * Licensed under the GPL-2 or later.
  */
 
 #ifndef __AD1836_H__
@@ -21,39 +12,30 @@
 
 #define AD1836_DAC_CTRL1               0
 #define AD1836_DAC_POWERDOWN           2
-#define AD1836_DAC_SERFMT_MASK	       0xE0
+#define AD1836_DAC_SERFMT_MASK         0xE0
 #define AD1836_DAC_SERFMT_PCK256       (0x4 << 5)
 #define AD1836_DAC_SERFMT_PCK128       (0x5 << 5)
 #define AD1836_DAC_WORD_LEN_MASK       0x18
 #define AD1836_DAC_WORD_LEN_OFFSET     3
 
 #define AD1836_DAC_CTRL2               1
-#define AD1836_DACL1_MUTE              0
-#define AD1836_DACR1_MUTE              1
-#define AD1836_DACL2_MUTE              2
-#define AD1836_DACR2_MUTE              3
-#define AD1836_DACL3_MUTE              4
-#define AD1836_DACR3_MUTE              5
 
-#define AD1836_DAC_L1_VOL              2
-#define AD1836_DAC_R1_VOL              3
-#define AD1836_DAC_L2_VOL              4
-#define AD1836_DAC_R2_VOL              5
-#define AD1836_DAC_L3_VOL              6
-#define AD1836_DAC_R3_VOL              7
+/* These macros are one-based. So AD183X_MUTE_LEFT(1) will return the mute bit
+ * for the first ADC/DAC */
+#define AD1836_MUTE_LEFT(x) (((x) * 2) - 2)
+#define AD1836_MUTE_RIGHT(x) (((x) * 2) - 1)
+
+#define AD1836_DAC_L_VOL(x) ((x) * 2)
+#define AD1836_DAC_R_VOL(x) (1 + ((x) * 2))
 
 #define AD1836_ADC_CTRL1               12
 #define AD1836_ADC_POWERDOWN           7
 #define AD1836_ADC_HIGHPASS_FILTER     8
 
 #define AD1836_ADC_CTRL2               13
-#define AD1836_ADCL1_MUTE 		0
-#define AD1836_ADCR1_MUTE 		1
-#define AD1836_ADCL2_MUTE 		2
-#define AD1836_ADCR2_MUTE 		3
 #define AD1836_ADC_WORD_LEN_MASK       0x30
 #define AD1836_ADC_WORD_OFFSET         5
-#define AD1836_ADC_SERFMT_MASK	       (7 << 6)
+#define AD1836_ADC_SERFMT_MASK         (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
 #define AD1836_ADC_AUX                 (0x6 << 6)
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
new file mode 100644
index 0000000..2758d5f
--- /dev/null
+++ b/sound/soc/codecs/adau1701.c
@@ -0,0 +1,549 @@
+/*
+ * Driver for ADAU1701 SigmaDSP processor
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *	based on an inital version by Cliff Cai <cliff.cai@analog.com>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/sigma.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "adau1701.h"
+
+#define ADAU1701_DSPCTRL	0x1c
+#define ADAU1701_SEROCTL	0x1e
+#define ADAU1701_SERICTL	0x1f
+
+#define ADAU1701_AUXNPOW	0x22
+
+#define ADAU1701_OSCIPOW	0x26
+#define ADAU1701_DACSET		0x27
+
+#define ADAU1701_NUM_REGS	0x28
+
+#define ADAU1701_DSPCTRL_CR		(1 << 2)
+#define ADAU1701_DSPCTRL_DAM		(1 << 3)
+#define ADAU1701_DSPCTRL_ADM		(1 << 4)
+#define ADAU1701_DSPCTRL_SR_48		0x00
+#define ADAU1701_DSPCTRL_SR_96		0x01
+#define ADAU1701_DSPCTRL_SR_192		0x02
+#define ADAU1701_DSPCTRL_SR_MASK	0x03
+
+#define ADAU1701_SEROCTL_INV_LRCLK	0x2000
+#define ADAU1701_SEROCTL_INV_BCLK	0x1000
+#define ADAU1701_SEROCTL_MASTER		0x0800
+
+#define ADAU1701_SEROCTL_OBF16		0x0000
+#define ADAU1701_SEROCTL_OBF8		0x0200
+#define ADAU1701_SEROCTL_OBF4		0x0400
+#define ADAU1701_SEROCTL_OBF2		0x0600
+#define ADAU1701_SEROCTL_OBF_MASK	0x0600
+
+#define ADAU1701_SEROCTL_OLF1024	0x0000
+#define ADAU1701_SEROCTL_OLF512		0x0080
+#define ADAU1701_SEROCTL_OLF256		0x0100
+#define ADAU1701_SEROCTL_OLF_MASK	0x0180
+
+#define ADAU1701_SEROCTL_MSB_DEALY1	0x0000
+#define ADAU1701_SEROCTL_MSB_DEALY0	0x0004
+#define ADAU1701_SEROCTL_MSB_DEALY8	0x0008
+#define ADAU1701_SEROCTL_MSB_DEALY12	0x000c
+#define ADAU1701_SEROCTL_MSB_DEALY16	0x0010
+#define ADAU1701_SEROCTL_MSB_DEALY_MASK	0x001c
+
+#define ADAU1701_SEROCTL_WORD_LEN_24	0x0000
+#define ADAU1701_SEROCTL_WORD_LEN_20	0x0001
+#define ADAU1701_SEROCTL_WORD_LEN_16	0x0010
+#define ADAU1701_SEROCTL_WORD_LEN_MASK	0x0003
+
+#define ADAU1701_AUXNPOW_VBPD		0x40
+#define ADAU1701_AUXNPOW_VRPD		0x20
+
+#define ADAU1701_SERICTL_I2S		0
+#define ADAU1701_SERICTL_LEFTJ		1
+#define ADAU1701_SERICTL_TDM		2
+#define ADAU1701_SERICTL_RIGHTJ_24	3
+#define ADAU1701_SERICTL_RIGHTJ_20	4
+#define ADAU1701_SERICTL_RIGHTJ_18	5
+#define ADAU1701_SERICTL_RIGHTJ_16	6
+#define ADAU1701_SERICTL_MODE_MASK	7
+#define ADAU1701_SERICTL_INV_BCLK	BIT(3)
+#define ADAU1701_SERICTL_INV_LRCLK	BIT(4)
+
+#define ADAU1701_OSCIPOW_OPD		0x04
+#define ADAU1701_DACSET_DACINIT		1
+
+#define ADAU1701_FIRMWARE "adau1701.bin"
+
+struct adau1701 {
+	unsigned int dai_fmt;
+};
+
+static const struct snd_kcontrol_new adau1701_controls[] = {
+	SOC_SINGLE("Master Capture Switch", ADAU1701_DSPCTRL, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget adau1701_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC0", "Playback", ADAU1701_AUXNPOW, 3, 1),
+	SND_SOC_DAPM_DAC("DAC1", "Playback", ADAU1701_AUXNPOW, 2, 1),
+	SND_SOC_DAPM_DAC("DAC2", "Playback", ADAU1701_AUXNPOW, 1, 1),
+	SND_SOC_DAPM_DAC("DAC3", "Playback", ADAU1701_AUXNPOW, 0, 1),
+	SND_SOC_DAPM_ADC("ADC", "Capture", ADAU1701_AUXNPOW, 7, 1),
+
+	SND_SOC_DAPM_OUTPUT("OUT0"),
+	SND_SOC_DAPM_OUTPUT("OUT1"),
+	SND_SOC_DAPM_OUTPUT("OUT2"),
+	SND_SOC_DAPM_OUTPUT("OUT3"),
+	SND_SOC_DAPM_INPUT("IN0"),
+	SND_SOC_DAPM_INPUT("IN1"),
+};
+
+static const struct snd_soc_dapm_route adau1701_dapm_routes[] = {
+	{ "OUT0", NULL, "DAC0" },
+	{ "OUT1", NULL, "DAC1" },
+	{ "OUT2", NULL, "DAC2" },
+	{ "OUT3", NULL, "DAC3" },
+
+	{ "ADC", NULL, "IN0" },
+	{ "ADC", NULL, "IN1" },
+};
+
+static unsigned int adau1701_register_size(struct snd_soc_codec *codec,
+		unsigned int reg)
+{
+	switch (reg) {
+	case ADAU1701_DSPCTRL:
+	case ADAU1701_SEROCTL:
+	case ADAU1701_AUXNPOW:
+	case ADAU1701_OSCIPOW:
+	case ADAU1701_DACSET:
+		return 2;
+	case ADAU1701_SERICTL:
+		return 1;
+	}
+
+	dev_err(codec->dev, "Unsupported register address: %d\n", reg);
+	return 0;
+}
+
+static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg,
+		unsigned int value)
+{
+	unsigned int i;
+	unsigned int size;
+	uint8_t buf[4];
+	int ret;
+
+	size = adau1701_register_size(codec, reg);
+	if (size == 0)
+		return -EINVAL;
+
+	snd_soc_cache_write(codec, reg, value);
+
+	buf[0] = 0x08;
+	buf[1] = reg;
+
+	for (i = size + 1; i >= 2; --i) {
+		buf[i] = value;
+		value >>= 8;
+	}
+
+	ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2);
+	if (ret == size + 2)
+		return 0;
+	else if (ret < 0)
+		return ret;
+	else
+		return -EIO;
+}
+
+static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	unsigned int value;
+	unsigned int ret;
+
+	ret = snd_soc_cache_read(codec, reg, &value);
+	if (ret)
+		return ret;
+
+	return value;
+}
+
+static int adau1701_load_firmware(struct snd_soc_codec *codec)
+{
+	return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE);
+}
+
+static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
+		snd_pcm_format_t format)
+{
+	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+	unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK;
+	unsigned int val;
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		val = ADAU1701_SEROCTL_WORD_LEN_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val = ADAU1701_SEROCTL_WORD_LEN_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val = ADAU1701_SEROCTL_WORD_LEN_24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) {
+		switch (format) {
+		case SNDRV_PCM_FORMAT_S16_LE:
+			val |= ADAU1701_SEROCTL_MSB_DEALY16;
+			break;
+		case SNDRV_PCM_FORMAT_S20_3LE:
+			val |= ADAU1701_SEROCTL_MSB_DEALY12;
+			break;
+		case SNDRV_PCM_FORMAT_S24_LE:
+			val |= ADAU1701_SEROCTL_MSB_DEALY8;
+			break;
+		}
+		mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK;
+	}
+
+	snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val);
+
+	return 0;
+}
+
+static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
+			snd_pcm_format_t format)
+{
+	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+
+	if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
+		return 0;
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		val = ADAU1701_SERICTL_RIGHTJ_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val = ADAU1701_SERICTL_RIGHTJ_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val = ADAU1701_SERICTL_RIGHTJ_24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ADAU1701_SERICTL,
+		ADAU1701_SERICTL_MODE_MASK, val);
+
+	return 0;
+}
+
+static int adau1701_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	snd_pcm_format_t format;
+	unsigned int val;
+
+	switch (params_rate(params)) {
+	case 192000:
+		val = ADAU1701_DSPCTRL_SR_192;
+		break;
+	case 96000:
+		val = ADAU1701_DSPCTRL_SR_96;
+		break;
+	case 48000:
+		val = ADAU1701_DSPCTRL_SR_48;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ADAU1701_DSPCTRL,
+		ADAU1701_DSPCTRL_SR_MASK, val);
+
+	format = params_format(params);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return adau1701_set_playback_pcm_format(codec, format);
+	else
+		return adau1701_set_capture_pcm_format(codec, format);
+}
+
+static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+	unsigned int serictl = 0x00, seroctl = 0x00;
+	bool invert_lrclk;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		/* master, 64-bits per sample, 1 frame per sample */
+		seroctl |= ADAU1701_SEROCTL_MASTER | ADAU1701_SEROCTL_OBF16
+				| ADAU1701_SEROCTL_OLF1024;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		invert_lrclk = false;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		invert_lrclk = true;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		invert_lrclk = false;
+		serictl |= ADAU1701_SERICTL_INV_BCLK;
+		seroctl |= ADAU1701_SEROCTL_INV_BCLK;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		invert_lrclk = true;
+		serictl |= ADAU1701_SERICTL_INV_BCLK;
+		seroctl |= ADAU1701_SEROCTL_INV_BCLK;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		serictl |= ADAU1701_SERICTL_LEFTJ;
+		seroctl |= ADAU1701_SEROCTL_MSB_DEALY0;
+		invert_lrclk = !invert_lrclk;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		serictl |= ADAU1701_SERICTL_RIGHTJ_24;
+		seroctl |= ADAU1701_SEROCTL_MSB_DEALY8;
+		invert_lrclk = !invert_lrclk;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (invert_lrclk) {
+		seroctl |= ADAU1701_SEROCTL_INV_LRCLK;
+		serictl |= ADAU1701_SERICTL_INV_LRCLK;
+	}
+
+	adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+	snd_soc_write(codec, ADAU1701_SERICTL, serictl);
+	snd_soc_update_bits(codec, ADAU1701_SEROCTL,
+		~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl);
+
+	return 0;
+}
+
+static int adau1701_set_bias_level(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* Enable VREF and VREF buffer */
+		snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* Disable VREF and VREF buffer */
+		snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int mask = ADAU1701_DSPCTRL_DAM;
+	unsigned int val;
+
+	if (mute)
+		val = 0;
+	else
+		val = mask;
+
+	snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val);
+
+	return 0;
+}
+
+static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+	unsigned int freq, int dir)
+{
+	unsigned int val;
+
+	switch (clk_id) {
+	case ADAU1701_CLK_SRC_OSC:
+		val = 0x0;
+		break;
+	case ADAU1701_CLK_SRC_MCLK:
+		val = ADAU1701_OSCIPOW_OPD;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val);
+
+	return 0;
+}
+
+#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+	SNDRV_PCM_RATE_192000)
+
+#define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops adau1701_dai_ops = {
+	.set_fmt	= adau1701_set_dai_fmt,
+	.hw_params	= adau1701_hw_params,
+	.digital_mute	= adau1701_digital_mute,
+};
+
+static struct snd_soc_dai_driver adau1701_dai = {
+	.name = "adau1701",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = ADAU1701_RATES,
+		.formats = ADAU1701_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = ADAU1701_RATES,
+		.formats = ADAU1701_FORMATS,
+	},
+	.ops = &adau1701_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static int adau1701_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	codec->dapm.idle_bias_off = 1;
+
+	ret = adau1701_load_firmware(codec);
+	if (ret)
+		dev_warn(codec->dev, "Failed to load firmware\n");
+
+	snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT);
+	snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver adau1701_codec_drv = {
+	.probe			= adau1701_probe,
+	.set_bias_level		= adau1701_set_bias_level,
+
+	.reg_cache_size		= ADAU1701_NUM_REGS,
+	.reg_word_size		= sizeof(u16),
+
+	.controls		= adau1701_controls,
+	.num_controls		= ARRAY_SIZE(adau1701_controls),
+	.dapm_widgets		= adau1701_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(adau1701_dapm_widgets),
+	.dapm_routes		= adau1701_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(adau1701_dapm_routes),
+
+	.write			= adau1701_write,
+	.read			= adau1701_read,
+
+	.set_sysclk		= adau1701_set_sysclk,
+};
+
+static __devinit int adau1701_i2c_probe(struct i2c_client *client,
+		const struct i2c_device_id *id)
+{
+	struct adau1701 *adau1701;
+	int ret;
+
+	adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL);
+	if (!adau1701)
+		return -ENOMEM;
+
+	i2c_set_clientdata(client, adau1701);
+	ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv,
+			&adau1701_dai, 1);
+	if (ret < 0)
+		kfree(adau1701);
+
+	return ret;
+}
+
+static __devexit int adau1701_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(i2c_get_clientdata(client));
+	return 0;
+}
+
+static const struct i2c_device_id adau1701_i2c_id[] = {
+	{ "adau1701", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id);
+
+static struct i2c_driver adau1701_i2c_driver = {
+	.driver = {
+		.name	= "adau1701",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= adau1701_i2c_probe,
+	.remove		= __devexit_p(adau1701_i2c_remove),
+	.id_table	= adau1701_i2c_id,
+};
+
+static int __init adau1701_init(void)
+{
+	return i2c_add_driver(&adau1701_i2c_driver);
+}
+module_init(adau1701_init);
+
+static void __exit adau1701_exit(void)
+{
+	i2c_del_driver(&adau1701_i2c_driver);
+}
+module_exit(adau1701_exit);
+
+MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver");
+MODULE_AUTHOR("Cliff Cai <cliff.cai@analog.com>");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1701.h b/sound/soc/codecs/adau1701.h
new file mode 100644
index 0000000..8d0949a
--- /dev/null
+++ b/sound/soc/codecs/adau1701.h
@@ -0,0 +1,17 @@
+/*
+ * header file for ADAU1701 SigmaDSP processor
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAU1701_H
+#define _ADAU1701_H
+
+enum adau1701_clk_src {
+	ADAU1701_CLK_SRC_OSC,
+	ADAU1701_CLK_SRC_MCLK,
+};
+
+#endif
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
new file mode 100644
index 0000000..e30fba6
--- /dev/null
+++ b/sound/soc/codecs/adav80x.c
@@ -0,0 +1,951 @@
+/*
+ * ADAV80X Audio Codec driver supporting ADAV801, ADAV803
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Yi Li <yi.li@analog.com>
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+#define ADAV80X_PLAYBACK_CTRL	0x04
+#define ADAV80X_AUX_IN_CTRL	0x05
+#define ADAV80X_REC_CTRL	0x06
+#define ADAV80X_AUX_OUT_CTRL	0x07
+#define ADAV80X_DPATH_CTRL1	0x62
+#define ADAV80X_DPATH_CTRL2	0x63
+#define ADAV80X_DAC_CTRL1	0x64
+#define ADAV80X_DAC_CTRL2	0x65
+#define ADAV80X_DAC_CTRL3	0x66
+#define ADAV80X_DAC_L_VOL	0x68
+#define ADAV80X_DAC_R_VOL	0x69
+#define ADAV80X_PGA_L_VOL	0x6c
+#define ADAV80X_PGA_R_VOL	0x6d
+#define ADAV80X_ADC_CTRL1	0x6e
+#define ADAV80X_ADC_CTRL2	0x6f
+#define ADAV80X_ADC_L_VOL	0x70
+#define ADAV80X_ADC_R_VOL	0x71
+#define ADAV80X_PLL_CTRL1	0x74
+#define ADAV80X_PLL_CTRL2	0x75
+#define ADAV80X_ICLK_CTRL1	0x76
+#define ADAV80X_ICLK_CTRL2	0x77
+#define ADAV80X_PLL_CLK_SRC	0x78
+#define ADAV80X_PLL_OUTE	0x7a
+
+#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll)	0x00
+#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll)	(0x40 << (pll))
+#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll)	(0x40 << (pll))
+
+#define ADAV80X_ICLK_CTRL1_DAC_SRC(src)		((src) << 5)
+#define ADAV80X_ICLK_CTRL1_ADC_SRC(src)		((src) << 2)
+#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src)	(src)
+#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src)	((src) << 3)
+
+#define ADAV80X_PLL_CTRL1_PLLDIV		0x10
+#define ADAV80X_PLL_CTRL1_PLLPD(pll)		(0x04 << (pll))
+#define ADAV80X_PLL_CTRL1_XTLPD			0x02
+
+#define ADAV80X_PLL_CTRL2_FIELD(pll, x)		((x) << ((pll) * 4))
+
+#define ADAV80X_PLL_CTRL2_FS_48(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x00)
+#define ADAV80X_PLL_CTRL2_FS_32(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x08)
+#define ADAV80X_PLL_CTRL2_FS_44(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c)
+
+#define ADAV80X_PLL_CTRL2_SEL(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x02)
+#define ADAV80X_PLL_CTRL2_DOUB(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x01)
+#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f)
+
+#define ADAV80X_ADC_CTRL1_MODULATOR_MASK	0x80
+#define ADAV80X_ADC_CTRL1_MODULATOR_128FS	0x00
+#define ADAV80X_ADC_CTRL1_MODULATOR_64FS	0x80
+
+#define ADAV80X_DAC_CTRL1_PD			0x80
+
+#define ADAV80X_DAC_CTRL2_DIV1			0x00
+#define ADAV80X_DAC_CTRL2_DIV1_5		0x10
+#define ADAV80X_DAC_CTRL2_DIV2			0x20
+#define ADAV80X_DAC_CTRL2_DIV3			0x30
+#define ADAV80X_DAC_CTRL2_DIV_MASK		0x30
+
+#define ADAV80X_DAC_CTRL2_INTERPOL_256FS	0x00
+#define ADAV80X_DAC_CTRL2_INTERPOL_128FS	0x40
+#define ADAV80X_DAC_CTRL2_INTERPOL_64FS		0x80
+#define ADAV80X_DAC_CTRL2_INTERPOL_MASK		0xc0
+
+#define ADAV80X_DAC_CTRL2_DEEMPH_NONE		0x00
+#define ADAV80X_DAC_CTRL2_DEEMPH_44		0x01
+#define ADAV80X_DAC_CTRL2_DEEMPH_32		0x02
+#define ADAV80X_DAC_CTRL2_DEEMPH_48		0x03
+#define ADAV80X_DAC_CTRL2_DEEMPH_MASK		0x01
+
+#define ADAV80X_CAPTURE_MODE_MASTER		0x20
+#define ADAV80X_CAPTURE_WORD_LEN24		0x00
+#define ADAV80X_CAPTURE_WORD_LEN20		0x04
+#define ADAV80X_CAPTRUE_WORD_LEN18		0x08
+#define ADAV80X_CAPTURE_WORD_LEN16		0x0c
+#define ADAV80X_CAPTURE_WORD_LEN_MASK		0x0c
+
+#define ADAV80X_CAPTURE_MODE_LEFT_J		0x00
+#define ADAV80X_CAPTURE_MODE_I2S		0x01
+#define ADAV80X_CAPTURE_MODE_RIGHT_J		0x03
+#define ADAV80X_CAPTURE_MODE_MASK		0x03
+
+#define ADAV80X_PLAYBACK_MODE_MASTER		0x10
+#define ADAV80X_PLAYBACK_MODE_LEFT_J		0x00
+#define ADAV80X_PLAYBACK_MODE_I2S		0x01
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24	0x04
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20	0x05
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18	0x06
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16	0x07
+#define ADAV80X_PLAYBACK_MODE_MASK		0x07
+
+#define ADAV80X_PLL_OUTE_SYSCLKPD(x)		BIT(2 - (x))
+
+static u8 adav80x_default_regs[] = {
+	0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00,
+	0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37,
+	0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b,
+	0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00,
+	0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee,
+	0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f,
+	0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x52, 0x00,
+};
+
+struct adav80x {
+	enum snd_soc_control_type control_type;
+
+	enum adav80x_clk_src clk_src;
+	unsigned int sysclk;
+	enum adav80x_pll_src pll_src;
+
+	unsigned int dai_fmt[2];
+	unsigned int rate;
+	bool deemph;
+	bool sysclk_pd[3];
+};
+
+static const char *adav80x_mux_text[] = {
+	"ADC",
+	"Playback",
+	"Aux Playback",
+};
+
+static const unsigned int adav80x_mux_values[] = {
+	0, 2, 3,
+};
+
+#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \
+	SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \
+		ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \
+		adav80x_mux_values)
+
+static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0);
+static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3);
+static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3);
+
+static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl =
+	SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum);
+static const struct snd_kcontrol_new adav80x_capture_mux_ctrl =
+	SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum);
+static const struct snd_kcontrol_new adav80x_dac_mux_ctrl =
+	SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum);
+
+#define ADAV80X_MUX(name, ctrl) \
+	SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1),
+	SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1),
+
+	SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0),
+
+	SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl),
+	ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl),
+	ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl),
+
+	SND_SOC_DAPM_INPUT("VINR"),
+	SND_SOC_DAPM_INPUT("VINL"),
+	SND_SOC_DAPM_OUTPUT("VOUTR"),
+	SND_SOC_DAPM_OUTPUT("VOUTL"),
+
+	SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0),
+};
+
+static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_codec *codec = source->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	const char *clk;
+
+	switch (adav80x->clk_src) {
+	case ADAV80X_CLK_PLL1:
+		clk = "PLL1";
+		break;
+	case ADAV80X_CLK_PLL2:
+		clk = "PLL2";
+		break;
+	case ADAV80X_CLK_XTAL:
+		clk = "OSC";
+		break;
+	default:
+		return 0;
+	}
+
+	return strcmp(source->name, clk) == 0;
+}
+
+static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_codec *codec = source->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL;
+}
+
+
+static const struct snd_soc_dapm_route adav80x_dapm_routes[] = {
+	{ "DAC Select", "ADC", "ADC" },
+	{ "DAC Select", "Playback", "AIFIN" },
+	{ "DAC Select", "Aux Playback", "AIFAUXIN" },
+	{ "DAC", NULL,  "DAC Select" },
+
+	{ "Capture Select", "ADC", "ADC" },
+	{ "Capture Select", "Playback", "AIFIN" },
+	{ "Capture Select", "Aux Playback", "AIFAUXIN" },
+	{ "AIFOUT", NULL,  "Capture Select" },
+
+	{ "Aux Capture Select", "ADC", "ADC" },
+	{ "Aux Capture Select", "Playback", "AIFIN" },
+	{ "Aux Capture Select", "Aux Playback", "AIFAUXIN" },
+	{ "AIFAUXOUT", NULL,  "Aux Capture Select" },
+
+	{ "VOUTR",  NULL, "DAC" },
+	{ "VOUTL",  NULL, "DAC" },
+
+	{ "Left PGA", NULL, "VINL" },
+	{ "Right PGA", NULL, "VINR" },
+	{ "ADC", NULL, "Left PGA" },
+	{ "ADC", NULL, "Right PGA" },
+
+	{ "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check },
+	{ "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check },
+	{ "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check },
+	{ "PLL1", NULL, "OSC", adav80x_dapm_pll_check },
+	{ "PLL2", NULL, "OSC", adav80x_dapm_pll_check },
+
+	{ "ADC", NULL, "SYSCLK" },
+	{ "DAC", NULL, "SYSCLK" },
+	{ "AIFOUT", NULL, "SYSCLK" },
+	{ "AIFAUXOUT", NULL, "SYSCLK" },
+	{ "AIFIN", NULL, "SYSCLK" },
+	{ "AIFAUXIN", NULL, "SYSCLK" },
+};
+
+static int adav80x_set_deemph(struct snd_soc_codec *codec)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+
+	if (adav80x->deemph) {
+		switch (adav80x->rate) {
+		case 32000:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_32;
+			break;
+		case 44100:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_44;
+			break;
+		case 48000:
+		case 64000:
+		case 88200:
+		case 96000:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_48;
+			break;
+		default:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+			break;
+		}
+	} else {
+		val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+	}
+
+	return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+		ADAV80X_DAC_CTRL2_DEEMPH_MASK, val);
+}
+
+static int adav80x_put_deemph(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int deemph = ucontrol->value.enumerated.item[0];
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	adav80x->deemph = deemph;
+
+	return adav80x_set_deemph(codec);
+}
+
+static int adav80x_get_deemph(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = adav80x->deemph;
+	return 0;
+};
+
+static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0);
+
+static const struct snd_kcontrol_new adav80x_controls[] = {
+	SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL,
+		ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+	SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL,
+			ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+
+	SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL,
+			ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv),
+
+	SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0),
+	SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1),
+
+	SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0),
+
+	SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0,
+			adav80x_get_deemph, adav80x_put_deemph),
+};
+
+static unsigned int adav80x_port_ctrl_regs[2][2] = {
+	{ ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, },
+	{ ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL },
+};
+
+static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int capture = 0x00;
+	unsigned int playback = 0x00;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		capture |= ADAV80X_CAPTURE_MODE_MASTER;
+		playback |= ADAV80X_PLAYBACK_MODE_MASTER;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		capture |= ADAV80X_CAPTURE_MODE_I2S;
+		playback |= ADAV80X_PLAYBACK_MODE_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		capture |= ADAV80X_CAPTURE_MODE_LEFT_J;
+		playback |= ADAV80X_PLAYBACK_MODE_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		capture |= ADAV80X_CAPTURE_MODE_RIGHT_J;
+		playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+		ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER,
+		capture);
+	snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback);
+
+	adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+	return 0;
+}
+
+static int adav80x_set_adc_clock(struct snd_soc_codec *codec,
+		unsigned int sample_rate)
+{
+	unsigned int val;
+
+	if (sample_rate <= 48000)
+		val = ADAV80X_ADC_CTRL1_MODULATOR_128FS;
+	else
+		val = ADAV80X_ADC_CTRL1_MODULATOR_64FS;
+
+	snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1,
+		ADAV80X_ADC_CTRL1_MODULATOR_MASK, val);
+
+	return 0;
+}
+
+static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
+		unsigned int sample_rate)
+{
+	unsigned int val;
+
+	if (sample_rate <= 48000)
+		val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS;
+	else
+		val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS;
+
+	snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+		ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK,
+		val);
+
+	return 0;
+}
+
+static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
+		struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+	unsigned int val;
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		val = ADAV80X_CAPTURE_WORD_LEN16;
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+		val = ADAV80X_CAPTRUE_WORD_LEN18;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val = ADAV80X_CAPTURE_WORD_LEN20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val = ADAV80X_CAPTURE_WORD_LEN24;
+		break;
+	default:
+		break;
+	}
+
+	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+		ADAV80X_CAPTURE_WORD_LEN_MASK, val);
+
+	return 0;
+}
+
+static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
+		struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+
+	if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J)
+		return 0;
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16;
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+		break;
+	default:
+		break;
+	}
+
+	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1],
+		ADAV80X_PLAYBACK_MODE_MASK, val);
+
+	return 0;
+}
+
+static int adav80x_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int rate = params_rate(params);
+
+	if (rate * 256 != adav80x->sysclk)
+		return -EINVAL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		adav80x_set_playback_pcm_format(codec, dai,
+			params_format(params));
+		adav80x_set_dac_clock(codec, rate);
+	} else {
+		adav80x_set_capture_pcm_format(codec, dai,
+			params_format(params));
+		adav80x_set_adc_clock(codec, rate);
+	}
+	adav80x->rate = rate;
+	adav80x_set_deemph(codec);
+
+	return 0;
+}
+
+static int adav80x_set_sysclk(struct snd_soc_codec *codec,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	if (dir == SND_SOC_CLOCK_IN) {
+		switch (clk_id) {
+		case ADAV80X_CLK_XIN:
+		case ADAV80X_CLK_XTAL:
+		case ADAV80X_CLK_MCLKI:
+		case ADAV80X_CLK_PLL1:
+		case ADAV80X_CLK_PLL2:
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		adav80x->sysclk = freq;
+
+		if (adav80x->clk_src != clk_id) {
+			unsigned int iclk_ctrl1, iclk_ctrl2;
+
+			adav80x->clk_src = clk_id;
+			if (clk_id == ADAV80X_CLK_XTAL)
+				clk_id = ADAV80X_CLK_XIN;
+
+			iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) |
+					ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) |
+					ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id);
+			iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id);
+
+			snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1);
+			snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2);
+
+			snd_soc_dapm_sync(&codec->dapm);
+		}
+	} else {
+		unsigned int mask;
+
+		switch (clk_id) {
+		case ADAV80X_CLK_SYSCLK1:
+		case ADAV80X_CLK_SYSCLK2:
+		case ADAV80X_CLK_SYSCLK3:
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		clk_id -= ADAV80X_CLK_SYSCLK1;
+		mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id);
+
+		if (freq == 0) {
+			snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask);
+			adav80x->sysclk_pd[clk_id] = true;
+		} else {
+			snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0);
+			adav80x->sysclk_pd[clk_id] = false;
+		}
+
+		if (adav80x->sysclk_pd[0])
+			snd_soc_dapm_disable_pin(&codec->dapm, "PLL1");
+		else
+			snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+
+		if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2])
+			snd_soc_dapm_disable_pin(&codec->dapm, "PLL2");
+		else
+			snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+		snd_soc_dapm_sync(&codec->dapm);
+	}
+
+	return 0;
+}
+
+static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int pll_ctrl1 = 0;
+	unsigned int pll_ctrl2 = 0;
+	unsigned int pll_src;
+
+	switch (source) {
+	case ADAV80X_PLL_SRC_XTAL:
+	case ADAV80X_PLL_SRC_XIN:
+	case ADAV80X_PLL_SRC_MCLKI:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (!freq_out)
+		return 0;
+
+	switch (freq_in) {
+	case 27000000:
+		break;
+	case 54000000:
+		if (source == ADAV80X_PLL_SRC_XIN) {
+			pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
+			break;
+		}
+	default:
+		return -EINVAL;
+	}
+
+	if (freq_out > 12288000) {
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id);
+		freq_out /= 2;
+	}
+
+	/* freq_out = sample_rate * 256 */
+	switch (freq_out) {
+	case 8192000:
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id);
+		break;
+	case 11289600:
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id);
+		break;
+	case 12288000:
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV,
+		pll_ctrl1);
+	snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2,
+			ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2);
+
+	if (source != adav80x->pll_src) {
+		if (source == ADAV80X_PLL_SRC_MCLKI)
+			pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id);
+		else
+			pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id);
+
+		snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC,
+				ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src);
+
+		adav80x->pll_src = source;
+
+		snd_soc_dapm_sync(&codec->dapm);
+	}
+
+	return 0;
+}
+
+static int adav80x_set_bias_level(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	unsigned int mask = ADAV80X_DAC_CTRL1_PD;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+/* Enforce the same sample rate on all audio interfaces */
+static int adav80x_dai_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	if (!codec->active || !adav80x->rate)
+		return 0;
+
+	return snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+}
+
+static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	if (!codec->active)
+		adav80x->rate = 0;
+}
+
+static const struct snd_soc_dai_ops adav80x_dai_ops = {
+	.set_fmt = adav80x_set_dai_fmt,
+	.hw_params = adav80x_hw_params,
+	.startup = adav80x_dai_startup,
+	.shutdown = adav80x_dai_shutdown,
+};
+
+#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \
+	SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
+	SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver adav80x_dais[] = {
+	{
+		.name = "adav80x-hifi",
+		.id = 0,
+		.playback = {
+			.stream_name = "HiFi Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_PLAYBACK_RATES,
+			.formats = ADAV80X_FORMATS,
+	},
+		.capture = {
+			.stream_name = "HiFi Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_CAPTURE_RATES,
+			.formats = ADAV80X_FORMATS,
+		},
+		.ops = &adav80x_dai_ops,
+	},
+	{
+		.name = "adav80x-aux",
+		.id = 1,
+		.playback = {
+			.stream_name = "Aux Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_PLAYBACK_RATES,
+			.formats = ADAV80X_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Aux Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_CAPTURE_RATES,
+			.formats = ADAV80X_FORMATS,
+		},
+		.ops = &adav80x_dai_ops,
+	},
+};
+
+static int adav80x_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type);
+	if (ret) {
+		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	/* Force PLLs on for SYSCLK output */
+	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+	/* Power down S/PDIF receiver, since it is currently not supported */
+	snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20);
+	/* Disable DAC zero flag */
+	snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6);
+
+	return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int adav80x_resume(struct snd_soc_codec *codec)
+{
+	adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	codec->cache_sync = 1;
+	snd_soc_cache_sync(codec);
+
+	return 0;
+}
+
+static int adav80x_remove(struct snd_soc_codec *codec)
+{
+	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static struct snd_soc_codec_driver adav80x_codec_driver = {
+	.probe = adav80x_probe,
+	.remove = adav80x_remove,
+	.suspend = adav80x_suspend,
+	.resume = adav80x_resume,
+	.set_bias_level = adav80x_set_bias_level,
+
+	.set_pll = adav80x_set_pll,
+	.set_sysclk = adav80x_set_sysclk,
+
+	.reg_word_size = sizeof(u8),
+	.reg_cache_size = ARRAY_SIZE(adav80x_default_regs),
+	.reg_cache_default = adav80x_default_regs,
+
+	.controls = adav80x_controls,
+	.num_controls = ARRAY_SIZE(adav80x_controls),
+	.dapm_widgets = adav80x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets),
+	.dapm_routes = adav80x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
+};
+
+static int __devinit adav80x_bus_probe(struct device *dev,
+		enum snd_soc_control_type control_type)
+{
+	struct adav80x *adav80x;
+	int ret;
+
+	adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
+	if (!adav80x)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, adav80x);
+	adav80x->control_type = control_type;
+
+	ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
+		adav80x_dais, ARRAY_SIZE(adav80x_dais));
+	if (ret)
+		kfree(adav80x);
+
+	return ret;
+}
+
+static int __devexit adav80x_bus_remove(struct device *dev)
+{
+	snd_soc_unregister_codec(dev);
+	kfree(dev_get_drvdata(dev));
+	return 0;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit adav80x_spi_probe(struct spi_device *spi)
+{
+	return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
+}
+
+static int __devexit adav80x_spi_remove(struct spi_device *spi)
+{
+	return adav80x_bus_remove(&spi->dev);
+}
+
+static struct spi_driver adav80x_spi_driver = {
+	.driver = {
+		.name	= "adav801",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= adav80x_spi_probe,
+	.remove		= __devexit_p(adav80x_spi_remove),
+};
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct i2c_device_id adav80x_id[] = {
+	{ "adav803", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, adav80x_id);
+
+static int __devinit adav80x_i2c_probe(struct i2c_client *client,
+		const struct i2c_device_id *id)
+{
+	return adav80x_bus_probe(&client->dev, SND_SOC_I2C);
+}
+
+static int __devexit adav80x_i2c_remove(struct i2c_client *client)
+{
+	return adav80x_bus_remove(&client->dev);
+}
+
+static struct i2c_driver adav80x_i2c_driver = {
+	.driver = {
+		.name = "adav803",
+		.owner = THIS_MODULE,
+	},
+	.probe = adav80x_i2c_probe,
+	.remove = __devexit_p(adav80x_i2c_remove),
+	.id_table = adav80x_id,
+};
+#endif
+
+static int __init adav80x_init(void)
+{
+	int ret = 0;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&adav80x_i2c_driver);
+	if (ret)
+		return ret;
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&adav80x_spi_driver);
+#endif
+
+	return ret;
+}
+module_init(adav80x_init);
+
+static void __exit adav80x_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&adav80x_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&adav80x_spi_driver);
+#endif
+}
+module_exit(adav80x_exit);
+
+MODULE_DESCRIPTION("ASoC ADAV80x driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h
new file mode 100644
index 0000000..adb0fc7
--- /dev/null
+++ b/sound/soc/codecs/adav80x.h
@@ -0,0 +1,35 @@
+/*
+ * header file for ADAV80X parts
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAV80X_H
+#define _ADAV80X_H
+
+enum adav80x_pll_src {
+	ADAV80X_PLL_SRC_XIN,
+	ADAV80X_PLL_SRC_XTAL,
+	ADAV80X_PLL_SRC_MCLKI,
+};
+
+enum adav80x_pll {
+	ADAV80X_PLL1 = 0,
+	ADAV80X_PLL2 = 1,
+};
+
+enum adav80x_clk_src {
+	ADAV80X_CLK_XIN = 0,
+	ADAV80X_CLK_MCLKI = 1,
+	ADAV80X_CLK_PLL1 = 2,
+	ADAV80X_CLK_PLL2 = 3,
+	ADAV80X_CLK_XTAL = 6,
+
+	ADAV80X_CLK_SYSCLK1 = 6,
+	ADAV80X_CLK_SYSCLK2 = 7,
+	ADAV80X_CLK_SYSCLK3 = 8,
+};
+
+#endif
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index ed96f247c..7a64e58 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -457,7 +457,7 @@
 	.set_sysclk   = ak4641_set_dai_sysclk,
 };
 
-struct snd_soc_dai_driver ak4641_dai[] = {
+static struct snd_soc_dai_driver ak4641_dai[] = {
 {
 	.name = "ak4641-hifi",
 	.id = 1,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0206a17..6cc8678 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -636,10 +636,7 @@
 #endif /* CONFIG_PM */
 
 /*
- * ASoC codec device structure
- *
- * Assign this variable to the codec_dev field of the machine driver's
- * snd_soc_device structure.
+ * ASoC codec driver structure
  */
 static const struct snd_soc_codec_driver soc_codec_device_cs4270 = {
 	.probe =		cs4270_probe,
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 4173b67..ac65a2d 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1397,8 +1397,6 @@
        if (freq == max98088->sysclk)
                return 0;
 
-       max98088->sysclk = freq; /* remember current sysclk */
-
        /* Setup clocks for slave mode, and using the PLL
         * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
         *         0x02 (when master clk is 20MHz to 30MHz)..
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index e1d282d..668434d 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1517,8 +1517,6 @@
 	if (freq == max98095->sysclk)
 		return 0;
 
-	max98095->sysclk = freq; /* remember current sysclk */
-
 	/* Setup clocks for slave mode, and using the PLL
 	 * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
 	 *         0x02 (when master clk is 20MHz to 40MHz)..
@@ -2261,11 +2259,11 @@
 
 	ret = snd_soc_read(codec, M98095_0FF_REV_ID);
 	if (ret < 0) {
-		dev_err(codec->dev, "Failed to read device revision: %d\n",
+		dev_err(codec->dev, "Failure reading hardware revision: %d\n",
 			ret);
 		goto err_access;
 	}
-	dev_info(codec->dev, "revision %c\n", ret + 'A');
+	dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A');
 
 	snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV);
 
@@ -2342,8 +2340,8 @@
 	max98095->control_data = i2c;
 	max98095->pdata = i2c->dev.platform_data;
 
-	ret = snd_soc_register_codec(&i2c->dev,
-			&soc_codec_dev_max98095, &max98095_dai[0], 3);
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095,
+				     max98095_dai, ARRAY_SIZE(max98095_dai));
 	if (ret < 0)
 		kfree(max98095);
 	return ret;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
new file mode 100644
index 0000000..486628a
--- /dev/null
+++ b/sound/soc/codecs/sta32x.c
@@ -0,0 +1,777 @@
+/*
+ * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on code from:
+ *	Wolfson Microelectronics PLC.
+ *	  Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *	Freescale Semiconductor, Inc.
+ *	  Timur Tabi <timur@freescale.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.
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "sta32x.h"
+
+#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \
+		      SNDRV_PCM_RATE_44100 | \
+		      SNDRV_PCM_RATE_48000 | \
+		      SNDRV_PCM_RATE_88200 | \
+		      SNDRV_PCM_RATE_96000 | \
+		      SNDRV_PCM_RATE_176400 | \
+		      SNDRV_PCM_RATE_192000)
+
+#define STA32X_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE  | SNDRV_PCM_FMTBIT_S16_BE  | \
+	 SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
+	 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
+	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \
+	 SNDRV_PCM_FMTBIT_S24_LE  | SNDRV_PCM_FMTBIT_S24_BE  | \
+	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_S32_BE)
+
+/* Power-up register defaults */
+static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = {
+	0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60,
+	0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69,
+	0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d,
+	0xc0, 0xf3, 0x33, 0x00, 0x0c,
+};
+
+/* regulator power supply names */
+static const char *sta32x_supply_names[] = {
+	"Vdda",	/* analog supply, 3.3VV */
+	"Vdd3",	/* digital supply, 3.3V */
+	"Vcc"	/* power amp spply, 10V - 36V */
+};
+
+/* codec private data */
+struct sta32x_priv {
+	struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
+	struct snd_soc_codec *codec;
+
+	unsigned int mclk;
+	unsigned int format;
+};
+
+static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1);
+static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0);
+
+static const char *sta32x_drc_ac[] = {
+	"Anti-Clipping", "Dynamic Range Compression" };
+static const char *sta32x_auto_eq_mode[] = {
+	"User", "Preset", "Loudness" };
+static const char *sta32x_auto_gc_mode[] = {
+	"User", "AC no clipping", "AC limited clipping (10%)",
+	"DRC nighttime listening mode" };
+static const char *sta32x_auto_xo_mode[] = {
+	"User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz",
+	"220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" };
+static const char *sta32x_preset_eq_mode[] = {
+	"Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft",
+	"Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1",
+	"Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2",
+	"Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7",
+	"Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12",
+	"Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" };
+static const char *sta32x_limiter_select[] = {
+	"Limiter Disabled", "Limiter #1", "Limiter #2" };
+static const char *sta32x_limiter_attack_rate[] = {
+	"3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024",
+	"0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752",
+	"0.0645", "0.0564", "0.0501", "0.0451" };
+static const char *sta32x_limiter_release_rate[] = {
+	"0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299",
+	"0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137",
+	"0.0134", "0.0117", "0.0110", "0.0104" };
+
+static const unsigned int sta32x_limiter_ac_attack_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0),
+	8, 16, TLV_DB_SCALE_ITEM(300, 100, 0),
+};
+
+static const unsigned int sta32x_limiter_ac_release_tlv[] = {
+	TLV_DB_RANGE_HEAD(5),
+	0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
+	1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0),
+	2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0),
+	3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0),
+	8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0),
+};
+
+static const unsigned int sta32x_limiter_drc_attack_tlv[] = {
+	TLV_DB_RANGE_HEAD(3),
+	0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0),
+	8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0),
+	14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0),
+};
+
+static const unsigned int sta32x_limiter_drc_release_tlv[] = {
+	TLV_DB_RANGE_HEAD(5),
+	0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
+	1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0),
+	3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0),
+	5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0),
+	13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+};
+
+static const struct soc_enum sta32x_drc_ac_enum =
+	SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
+			2, sta32x_drc_ac);
+static const struct soc_enum sta32x_auto_eq_enum =
+	SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
+			3, sta32x_auto_eq_mode);
+static const struct soc_enum sta32x_auto_gc_enum =
+	SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
+			4, sta32x_auto_gc_mode);
+static const struct soc_enum sta32x_auto_xo_enum =
+	SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
+			16, sta32x_auto_xo_mode);
+static const struct soc_enum sta32x_preset_eq_enum =
+	SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
+			32, sta32x_preset_eq_mode);
+static const struct soc_enum sta32x_limiter_ch1_enum =
+	SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
+			3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter_ch2_enum =
+	SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
+			3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter_ch3_enum =
+	SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
+			3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter1_attack_rate_enum =
+	SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT,
+			16, sta32x_limiter_attack_rate);
+static const struct soc_enum sta32x_limiter2_attack_rate_enum =
+	SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT,
+			16, sta32x_limiter_attack_rate);
+static const struct soc_enum sta32x_limiter1_release_rate_enum =
+	SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT,
+			16, sta32x_limiter_release_rate);
+static const struct soc_enum sta32x_limiter2_release_rate_enum =
+	SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT,
+			16, sta32x_limiter_release_rate);
+static const struct snd_kcontrol_new sta32x_snd_controls[] = {
+SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv),
+SOC_SINGLE("Master Switch", STA32X_MMUTE, 0, 1, 1),
+SOC_SINGLE("Ch1 Switch", STA32X_MMUTE, 1, 1, 1),
+SOC_SINGLE("Ch2 Switch", STA32X_MMUTE, 2, 1, 1),
+SOC_SINGLE("Ch3 Switch", STA32X_MMUTE, 3, 1, 1),
+SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0),
+SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum),
+SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0),
+SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0),
+SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0),
+SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0),
+SOC_ENUM("Automode EQ", sta32x_auto_eq_enum),
+SOC_ENUM("Automode GC", sta32x_auto_gc_enum),
+SOC_ENUM("Automode XO", sta32x_auto_xo_enum),
+SOC_ENUM("Preset EQ", sta32x_preset_eq_enum),
+SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0),
+SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum),
+SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum),
+SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum),
+SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv),
+SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv),
+SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum),
+SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum),
+SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+
+/* depending on mode, the attack/release thresholds have
+ * two different enum definitions; provide both
+ */
+SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT,
+	       16, 0, sta32x_limiter_ac_attack_tlv),
+SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT,
+	       16, 0, sta32x_limiter_ac_attack_tlv),
+SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT,
+	       16, 0, sta32x_limiter_ac_release_tlv),
+SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT,
+	       16, 0, sta32x_limiter_ac_release_tlv),
+SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT,
+	       16, 0, sta32x_limiter_drc_attack_tlv),
+SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT,
+	       16, 0, sta32x_limiter_drc_attack_tlv),
+SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT,
+	       16, 0, sta32x_limiter_drc_release_tlv),
+SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT,
+	       16, 0, sta32x_limiter_drc_release_tlv),
+};
+
+static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("LEFT"),
+SND_SOC_DAPM_OUTPUT("RIGHT"),
+SND_SOC_DAPM_OUTPUT("SUB"),
+};
+
+static const struct snd_soc_dapm_route sta32x_dapm_routes[] = {
+	{ "LEFT", NULL, "DAC" },
+	{ "RIGHT", NULL, "DAC" },
+	{ "SUB", NULL, "DAC" },
+};
+
+/* MCLK interpolation ratio per fs */
+static struct {
+	int fs;
+	int ir;
+} interpolation_ratios[] = {
+	{ 32000, 0 },
+	{ 44100, 0 },
+	{ 48000, 0 },
+	{ 88200, 1 },
+	{ 96000, 1 },
+	{ 176400, 2 },
+	{ 192000, 2 },
+};
+
+/* MCLK to fs clock ratios */
+static struct {
+	int ratio;
+	int mcs;
+} mclk_ratios[3][7] = {
+	{ { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 },
+	  { 128, 4 }, { 576, 5 }, { 0, 0 } },
+	{ { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } },
+	{ { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } },
+};
+
+
+/**
+ * sta32x_set_dai_sysclk - configure MCLK
+ * @codec_dai: the codec DAI
+ * @clk_id: the clock ID (ignored)
+ * @freq: the MCLK input frequency
+ * @dir: the clock direction (ignored)
+ *
+ * The value of MCLK is used to determine which sample rates are supported
+ * by the STA32X, based on the mclk_ratios table.
+ *
+ * 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.
+ *
+ * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause
+ * theoretically possible sample rates to be enabled. Call it again with a
+ * proper value set one the external clock is set (most probably you would do
+ * that from a machine's driver 'hw_param' hook.
+ */
+static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+	int i, j, ir, fs;
+	unsigned int rates = 0;
+	unsigned int rate_min = -1;
+	unsigned int rate_max = 0;
+
+	pr_debug("mclk=%u\n", freq);
+	sta32x->mclk = freq;
+
+	if (sta32x->mclk) {
+		for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) {
+			ir = interpolation_ratios[i].ir;
+			fs = interpolation_ratios[i].fs;
+			for (j = 0; mclk_ratios[ir][j].ratio; j++) {
+				if (mclk_ratios[ir][j].ratio * fs == freq) {
+					rates |= snd_pcm_rate_to_rate_bit(fs);
+					if (fs < rate_min)
+						rate_min = fs;
+					if (fs > rate_max)
+						rate_max = fs;
+				}
+			}
+		}
+		/* FIXME: soc should support a rate list */
+		rates &= ~SNDRV_PCM_RATE_KNOT;
+
+		if (!rates) {
+			dev_err(codec->dev, "could not find a valid sample rate\n");
+			return -EINVAL;
+		}
+	} else {
+		/* enable all possible rates */
+		rates = STA32X_RATES;
+		rate_min = 32000;
+		rate_max = 192000;
+	}
+
+	codec_dai->driver->playback.rates = rates;
+	codec_dai->driver->playback.rate_min = rate_min;
+	codec_dai->driver->playback.rate_max = rate_max;
+	return 0;
+}
+
+/**
+ * sta32x_set_dai_fmt - configure the codec for the selected audio format
+ * @codec_dai: the codec DAI
+ * @fmt: 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.
+ */
+static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+	u8 confb = snd_soc_read(codec, STA32X_CONFB);
+
+	pr_debug("\n");
+	confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_LEFT_J:
+		sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		confb |= STA32X_CONFB_C2IM;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		confb |= STA32X_CONFB_C1IM;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, STA32X_CONFB, confb);
+	return 0;
+}
+
+/**
+ * sta32x_hw_params - program the STA32X 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.
+ */
+static int sta32x_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+	unsigned int rate;
+	int i, mcs = -1, ir = -1;
+	u8 confa, confb;
+
+	rate = params_rate(params);
+	pr_debug("rate: %u\n", rate);
+	for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++)
+		if (interpolation_ratios[i].fs == rate)
+			ir = interpolation_ratios[i].ir;
+	if (ir < 0)
+		return -EINVAL;
+	for (i = 0; mclk_ratios[ir][i].ratio; i++)
+		if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk)
+			mcs = mclk_ratios[ir][i].mcs;
+	if (mcs < 0)
+		return -EINVAL;
+
+	confa = snd_soc_read(codec, STA32X_CONFA);
+	confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK);
+	confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT);
+
+	confb = snd_soc_read(codec, STA32X_CONFB);
+	confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB);
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S24_LE:
+	case SNDRV_PCM_FORMAT_S24_BE:
+	case SNDRV_PCM_FORMAT_S24_3LE:
+	case SNDRV_PCM_FORMAT_S24_3BE:
+		pr_debug("24bit\n");
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S32_LE:
+	case SNDRV_PCM_FORMAT_S32_BE:
+		pr_debug("24bit or 32bit\n");
+		switch (sta32x->format) {
+		case SND_SOC_DAIFMT_I2S:
+			confb |= 0x0;
+			break;
+		case SND_SOC_DAIFMT_LEFT_J:
+			confb |= 0x1;
+			break;
+		case SND_SOC_DAIFMT_RIGHT_J:
+			confb |= 0x2;
+			break;
+		}
+
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+	case SNDRV_PCM_FORMAT_S20_3BE:
+		pr_debug("20bit\n");
+		switch (sta32x->format) {
+		case SND_SOC_DAIFMT_I2S:
+			confb |= 0x4;
+			break;
+		case SND_SOC_DAIFMT_LEFT_J:
+			confb |= 0x5;
+			break;
+		case SND_SOC_DAIFMT_RIGHT_J:
+			confb |= 0x6;
+			break;
+		}
+
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+	case SNDRV_PCM_FORMAT_S18_3BE:
+		pr_debug("18bit\n");
+		switch (sta32x->format) {
+		case SND_SOC_DAIFMT_I2S:
+			confb |= 0x8;
+			break;
+		case SND_SOC_DAIFMT_LEFT_J:
+			confb |= 0x9;
+			break;
+		case SND_SOC_DAIFMT_RIGHT_J:
+			confb |= 0xa;
+			break;
+		}
+
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+	case SNDRV_PCM_FORMAT_S16_BE:
+		pr_debug("16bit\n");
+		switch (sta32x->format) {
+		case SND_SOC_DAIFMT_I2S:
+			confb |= 0x0;
+			break;
+		case SND_SOC_DAIFMT_LEFT_J:
+			confb |= 0xd;
+			break;
+		case SND_SOC_DAIFMT_RIGHT_J:
+			confb |= 0xe;
+			break;
+		}
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, STA32X_CONFA, confa);
+	snd_soc_write(codec, STA32X_CONFB, confb);
+	return 0;
+}
+
+/**
+ * sta32x_set_bias_level - DAPM callback
+ * @codec: the codec device
+ * @level: DAPM power level
+ *
+ * This is called by ALSA to put the codec into low power mode
+ * or to wake it up.  If the codec is powered off completely
+ * all registers must be restored after power on.
+ */
+static int sta32x_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	int ret;
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+
+	pr_debug("level = %d\n", level);
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* Full power on */
+		snd_soc_update_bits(codec, STA32X_CONFF,
+				    STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+				    STA32X_CONFF_PWDN | STA32X_CONFF_EAPD);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
+						    sta32x->supplies);
+			if (ret != 0) {
+				dev_err(codec->dev,
+					"Failed to enable supplies: %d\n", ret);
+				return ret;
+			}
+
+			snd_soc_cache_sync(codec);
+		}
+
+		/* Power up to mute */
+		/* FIXME */
+		snd_soc_update_bits(codec, STA32X_CONFF,
+				    STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+				    STA32X_CONFF_PWDN | STA32X_CONFF_EAPD);
+
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* The chip runs through the power down sequence for us. */
+		snd_soc_update_bits(codec, STA32X_CONFF,
+				    STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+				    STA32X_CONFF_PWDN);
+		msleep(300);
+
+		regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies),
+				       sta32x->supplies);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static struct snd_soc_dai_ops sta32x_dai_ops = {
+	.hw_params	= sta32x_hw_params,
+	.set_sysclk	= sta32x_set_dai_sysclk,
+	.set_fmt	= sta32x_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver sta32x_dai = {
+	.name = "STA32X",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = STA32X_RATES,
+		.formats = STA32X_FORMATS,
+	},
+	.ops = &sta32x_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int sta32x_resume(struct snd_soc_codec *codec)
+{
+	sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+#else
+#define sta32x_suspend NULL
+#define sta32x_resume NULL
+#endif
+
+static int sta32x_probe(struct snd_soc_codec *codec)
+{
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+	int i, ret = 0;
+
+	sta32x->codec = codec;
+
+	/* regulators */
+	for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++)
+		sta32x->supplies[i].supply = sta32x_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies),
+				 sta32x->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		goto err;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
+				    sta32x->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_get;
+	}
+
+	/* Tell ASoC what kind of I/O to use to read the registers.  ASoC will
+	 * then do the I2C transactions itself.
+	 */
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret);
+		return ret;
+	}
+
+	/* read reg reset values into cache */
+	for (i = 0; i < STA32X_REGISTER_COUNT; i++)
+		snd_soc_cache_write(codec, i, sta32x_regs[i]);
+
+	/* FIXME enable thermal warning adjustment and recovery  */
+	snd_soc_update_bits(codec, STA32X_CONFA,
+			    STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0);
+
+	/* FIXME select 2.1 mode  */
+	snd_soc_update_bits(codec, STA32X_CONFF,
+			    STA32X_CONFF_OCFG_MASK,
+			    1 << STA32X_CONFF_OCFG_SHIFT);
+
+	/* FIXME channel to output mapping */
+	snd_soc_update_bits(codec, STA32X_C1CFG,
+			    STA32X_CxCFG_OM_MASK,
+			    0 << STA32X_CxCFG_OM_SHIFT);
+	snd_soc_update_bits(codec, STA32X_C2CFG,
+			    STA32X_CxCFG_OM_MASK,
+			    1 << STA32X_CxCFG_OM_SHIFT);
+	snd_soc_update_bits(codec, STA32X_C3CFG,
+			    STA32X_CxCFG_OM_MASK,
+			    2 << STA32X_CxCFG_OM_SHIFT);
+
+	sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	/* Bias level configuration will have done an extra enable */
+	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+	return 0;
+
+err_get:
+	regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+err:
+	return ret;
+}
+
+static int sta32x_remove(struct snd_soc_codec *codec)
+{
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+
+	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+	regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+	return 0;
+}
+
+static const struct snd_soc_codec_driver sta32x_codec = {
+	.probe =		sta32x_probe,
+	.remove =		sta32x_remove,
+	.suspend =		sta32x_suspend,
+	.resume =		sta32x_resume,
+	.reg_cache_size =	STA32X_REGISTER_COUNT,
+	.reg_word_size =	sizeof(u8),
+	.set_bias_level =	sta32x_set_bias_level,
+	.controls =		sta32x_snd_controls,
+	.num_controls =		ARRAY_SIZE(sta32x_snd_controls),
+	.dapm_widgets =		sta32x_dapm_widgets,
+	.num_dapm_widgets =	ARRAY_SIZE(sta32x_dapm_widgets),
+	.dapm_routes =		sta32x_dapm_routes,
+	.num_dapm_routes =	ARRAY_SIZE(sta32x_dapm_routes),
+};
+
+static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct sta32x_priv *sta32x;
+	int ret;
+
+	sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL);
+	if (!sta32x)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, sta32x);
+
+	ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static __devexit int sta32x_i2c_remove(struct i2c_client *client)
+{
+	struct sta32x_priv *sta32x = i2c_get_clientdata(client);
+	struct snd_soc_codec *codec = sta32x->codec;
+
+	if (codec)
+		sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+	if (codec) {
+		snd_soc_unregister_codec(&client->dev);
+		snd_soc_codec_set_drvdata(codec, NULL);
+	}
+
+	kfree(sta32x);
+	return 0;
+}
+
+static const struct i2c_device_id sta32x_i2c_id[] = {
+	{ "sta326", 0 },
+	{ "sta328", 0 },
+	{ "sta329", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id);
+
+static struct i2c_driver sta32x_i2c_driver = {
+	.driver = {
+		.name = "sta32x",
+		.owner = THIS_MODULE,
+	},
+	.probe =    sta32x_i2c_probe,
+	.remove =   __devexit_p(sta32x_i2c_remove),
+	.id_table = sta32x_i2c_id,
+};
+
+static int __init sta32x_init(void)
+{
+	return i2c_add_driver(&sta32x_i2c_driver);
+}
+module_init(sta32x_init);
+
+static void __exit sta32x_exit(void)
+{
+	i2c_del_driver(&sta32x_i2c_driver);
+}
+module_exit(sta32x_exit);
+
+MODULE_DESCRIPTION("ASoC STA32X driver");
+MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
new file mode 100644
index 0000000..b97ee5a
--- /dev/null
+++ b/sound/soc/codecs/sta32x.h
@@ -0,0 +1,210 @@
+/*
+ * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on code from:
+ *	Wolfson Microelectronics PLC.
+ *	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 _ASOC_STA_32X_H
+#define _ASOC_STA_32X_H
+
+/* STA326 register addresses */
+
+#define STA32X_REGISTER_COUNT	0x2d
+
+#define STA32X_CONFA	0x00
+#define STA32X_CONFB    0x01
+#define STA32X_CONFC    0x02
+#define STA32X_CONFD    0x03
+#define STA32X_CONFE    0x04
+#define STA32X_CONFF    0x05
+#define STA32X_MMUTE    0x06
+#define STA32X_MVOL     0x07
+#define STA32X_C1VOL    0x08
+#define STA32X_C2VOL    0x09
+#define STA32X_C3VOL    0x0a
+#define STA32X_AUTO1    0x0b
+#define STA32X_AUTO2    0x0c
+#define STA32X_AUTO3    0x0d
+#define STA32X_C1CFG    0x0e
+#define STA32X_C2CFG    0x0f
+#define STA32X_C3CFG    0x10
+#define STA32X_TONE     0x11
+#define STA32X_L1AR     0x12
+#define STA32X_L1ATRT   0x13
+#define STA32X_L2AR     0x14
+#define STA32X_L2ATRT   0x15
+#define STA32X_CFADDR2  0x16
+#define STA32X_B1CF1    0x17
+#define STA32X_B1CF2    0x18
+#define STA32X_B1CF3    0x19
+#define STA32X_B2CF1    0x1a
+#define STA32X_B2CF2    0x1b
+#define STA32X_B2CF3    0x1c
+#define STA32X_A1CF1    0x1d
+#define STA32X_A1CF2    0x1e
+#define STA32X_A1CF3    0x1f
+#define STA32X_A2CF1    0x20
+#define STA32X_A2CF2    0x21
+#define STA32X_A2CF3    0x22
+#define STA32X_B0CF1    0x23
+#define STA32X_B0CF2    0x24
+#define STA32X_B0CF3    0x25
+#define STA32X_CFUD     0x26
+#define STA32X_MPCC1    0x27
+#define STA32X_MPCC2    0x28
+/* Reserved 0x29 */
+/* Reserved 0x2a */
+#define STA32X_Reserved 0x2a
+#define STA32X_FDRC1    0x2b
+#define STA32X_FDRC2    0x2c
+/* Reserved 0x2d */
+
+
+/* STA326 register field definitions */
+
+/* 0x00 CONFA */
+#define STA32X_CONFA_MCS_MASK	0x03
+#define STA32X_CONFA_MCS_SHIFT	0
+#define STA32X_CONFA_IR_MASK	0x18
+#define STA32X_CONFA_IR_SHIFT	3
+#define STA32X_CONFA_TWRB	0x20
+#define STA32X_CONFA_TWAB	0x40
+#define STA32X_CONFA_FDRB	0x80
+
+/* 0x01 CONFB */
+#define STA32X_CONFB_SAI_MASK	0x0f
+#define STA32X_CONFB_SAI_SHIFT	0
+#define STA32X_CONFB_SAIFB	0x10
+#define STA32X_CONFB_DSCKE	0x20
+#define STA32X_CONFB_C1IM	0x40
+#define STA32X_CONFB_C2IM	0x80
+
+/* 0x02 CONFC */
+#define STA32X_CONFC_OM_MASK	0x03
+#define STA32X_CONFC_OM_SHIFT	0
+#define STA32X_CONFC_CSZ_MASK	0x7c
+#define STA32X_CONFC_CSZ_SHIFT	2
+
+/* 0x03 CONFD */
+#define STA32X_CONFD_HPB	0x01
+#define STA32X_CONFD_HPB_SHIFT	0
+#define STA32X_CONFD_DEMP	0x02
+#define STA32X_CONFD_DEMP_SHIFT	1
+#define STA32X_CONFD_DSPB	0x04
+#define STA32X_CONFD_DSPB_SHIFT	2
+#define STA32X_CONFD_PSL	0x08
+#define STA32X_CONFD_PSL_SHIFT	3
+#define STA32X_CONFD_BQL	0x10
+#define STA32X_CONFD_BQL_SHIFT	4
+#define STA32X_CONFD_DRC	0x20
+#define STA32X_CONFD_DRC_SHIFT	5
+#define STA32X_CONFD_ZDE	0x40
+#define STA32X_CONFD_ZDE_SHIFT	6
+#define STA32X_CONFD_MME	0x80
+#define STA32X_CONFD_MME_SHIFT	7
+
+/* 0x04 CONFE */
+#define STA32X_CONFE_MPCV	0x01
+#define STA32X_CONFE_MPCV_SHIFT	0
+#define STA32X_CONFE_MPC	0x02
+#define STA32X_CONFE_MPC_SHIFT	1
+#define STA32X_CONFE_AME	0x08
+#define STA32X_CONFE_AME_SHIFT	3
+#define STA32X_CONFE_PWMS	0x10
+#define STA32X_CONFE_PWMS_SHIFT	4
+#define STA32X_CONFE_ZCE	0x40
+#define STA32X_CONFE_ZCE_SHIFT	6
+#define STA32X_CONFE_SVE	0x80
+#define STA32X_CONFE_SVE_SHIFT	7
+
+/* 0x05 CONFF */
+#define STA32X_CONFF_OCFG_MASK	0x03
+#define STA32X_CONFF_OCFG_SHIFT	0
+#define STA32X_CONFF_IDE	0x04
+#define STA32X_CONFF_IDE_SHIFT	3
+#define STA32X_CONFF_BCLE	0x08
+#define STA32X_CONFF_ECLE	0x20
+#define STA32X_CONFF_PWDN	0x40
+#define STA32X_CONFF_EAPD	0x80
+
+/* 0x06 MMUTE */
+#define STA32X_MMUTE_MMUTE	0x01
+
+/* 0x0b AUTO1 */
+#define STA32X_AUTO1_AMEQ_MASK	0x03
+#define STA32X_AUTO1_AMEQ_SHIFT	0
+#define STA32X_AUTO1_AMV_MASK	0xc0
+#define STA32X_AUTO1_AMV_SHIFT	2
+#define STA32X_AUTO1_AMGC_MASK	0x30
+#define STA32X_AUTO1_AMGC_SHIFT	4
+#define STA32X_AUTO1_AMPS	0x80
+
+/* 0x0c AUTO2 */
+#define STA32X_AUTO2_AMAME	0x01
+#define STA32X_AUTO2_AMAM_MASK	0x0e
+#define STA32X_AUTO2_AMAM_SHIFT	1
+#define STA32X_AUTO2_XO_MASK	0xf0
+#define STA32X_AUTO2_XO_SHIFT	4
+
+/* 0x0d AUTO3 */
+#define STA32X_AUTO3_PEQ_MASK	0x1f
+#define STA32X_AUTO3_PEQ_SHIFT	0
+
+/* 0x0e 0x0f 0x10 CxCFG */
+#define STA32X_CxCFG_TCB	0x01	/* only C1 and C2 */
+#define STA32X_CxCFG_TCB_SHIFT	0
+#define STA32X_CxCFG_EQBP	0x02	/* only C1 and C2 */
+#define STA32X_CxCFG_EQBP_SHIFT	1
+#define STA32X_CxCFG_VBP	0x03
+#define STA32X_CxCFG_VBP_SHIFT	2
+#define STA32X_CxCFG_BO		0x04
+#define STA32X_CxCFG_LS_MASK	0x30
+#define STA32X_CxCFG_LS_SHIFT	4
+#define STA32X_CxCFG_OM_MASK	0xc0
+#define STA32X_CxCFG_OM_SHIFT	6
+
+/* 0x11 TONE */
+#define STA32X_TONE_BTC_SHIFT	0
+#define STA32X_TONE_TTC_SHIFT	4
+
+/* 0x12 0x13 0x14 0x15 limiter attack/release */
+#define STA32X_LxA_SHIFT	0
+#define STA32X_LxR_SHIFT	4
+
+/* 0x26 CFUD */
+#define STA32X_CFUD_W1		0x01
+#define STA32X_CFUD_WA		0x02
+#define STA32X_CFUD_R1		0x04
+#define STA32X_CFUD_RA		0x08
+
+
+/* biquad filter coefficient table offsets */
+#define STA32X_C1_BQ_BASE	0
+#define STA32X_C2_BQ_BASE	20
+#define STA32X_CH_BQ_NUM	4
+#define STA32X_BQ_NUM_COEF	5
+#define STA32X_XO_HP_BQ_BASE	40
+#define STA32X_XO_LP_BQ_BASE	45
+#define STA32X_C1_PRESCALE	50
+#define STA32X_C2_PRESCALE	51
+#define STA32X_C1_POSTSCALE	52
+#define STA32X_C2_POSTSCALE	53
+#define STA32X_C3_POSTSCALE	54
+#define STA32X_TW_POSTSCALE	55
+#define STA32X_C1_MIX1		56
+#define STA32X_C1_MIX2		57
+#define STA32X_C2_MIX1		58
+#define STA32X_C2_MIX2		59
+#define STA32X_C3_MIX1		60
+#define STA32X_C3_MIX2		61
+
+#endif /* _ASOC_STA_32X_H */
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
new file mode 100644
index 0000000..a2a09f8
--- /dev/null
+++ b/sound/soc/codecs/wm8782.c
@@ -0,0 +1,80 @@
+/*
+ * sound/soc/codecs/wm8782.c
+ * simple, strap-pin configured 24bit 2ch ADC
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on ad73311.c
+ * Copyright:	Analog Device Inc.
+ * Author:	Cliff Cai <cliff.cai@analog.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/init.h>
+#include <linux/slab.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>
+
+static struct snd_soc_dai_driver wm8782_dai = {
+	.name = "wm8782",
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		/* For configurations with FSAMPEN=0 */
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			   SNDRV_PCM_FMTBIT_S20_3LE |
+			   SNDRV_PCM_FMTBIT_S24_LE,
+	},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8782;
+
+static __devinit int wm8782_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev,
+			&soc_codec_dev_wm8782, &wm8782_dai, 1);
+}
+
+static int __devexit wm8782_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver wm8782_codec_driver = {
+	.driver = {
+		.name = "wm8782",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8782_probe,
+	.remove = wm8782_remove,
+};
+
+static int __init wm8782_init(void)
+{
+	return platform_driver_register(&wm8782_codec_driver);
+}
+module_init(wm8782_init);
+
+static void __exit wm8782_exit(void)
+{
+	platform_driver_unregister(&wm8782_codec_driver);
+}
+module_exit(wm8782_exit);
+
+MODULE_DESCRIPTION("ASoC WM8782 driver");
+MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9b3bba4..b085575 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2560,6 +2560,7 @@
 static const struct i2c_device_id wm8904_i2c_id[] = {
 	{ "wm8904", WM8904 },
 	{ "wm8912", WM8912 },
+	{ "wm8918", WM8904 },   /* Actually a subset, updates to follow */
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id);
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index e2ab4fa..423baa9 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -41,14 +41,12 @@
 #define HPOUT2L 4
 #define HPOUT2R 8
 
-#define WM8915_NUM_SUPPLIES 6
+#define WM8915_NUM_SUPPLIES 4
 static const char *wm8915_supply_names[WM8915_NUM_SUPPLIES] = {
-	"DCVDD",
 	"DBVDD",
 	"AVDD1",
 	"AVDD2",
 	"CPVDD",
-	"MICVDD",
 };
 
 struct wm8915_priv {
@@ -57,6 +55,7 @@
 	int ldo1ena;
 
 	int sysclk;
+	int sysclk_src;
 
 	int fll_src;
 	int fll_fref;
@@ -76,6 +75,7 @@
 	struct wm8915_pdata pdata;
 
 	int rx_rate[WM8915_AIFS];
+	int bclk_rate[WM8915_AIFS];
 
 	/* Platform dependant ReTune mobile configuration */
 	int num_retune_mobile_texts;
@@ -113,8 +113,6 @@
 WM8915_REGULATOR_EVENT(1)
 WM8915_REGULATOR_EVENT(2)
 WM8915_REGULATOR_EVENT(3)
-WM8915_REGULATOR_EVENT(4)
-WM8915_REGULATOR_EVENT(5)
 
 static const u16 wm8915_reg[WM8915_MAX_REGISTER] = {
 	[WM8915_SOFTWARE_RESET] = 0x8915,
@@ -1565,6 +1563,50 @@
 	return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915);
 }
 
+static const int bclk_divs[] = {
+	1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
+};
+
+static void wm8915_update_bclk(struct snd_soc_codec *codec)
+{
+	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+	int aif, best, cur_val, bclk_rate, bclk_reg, i;
+
+	/* Don't bother if we're in a low frequency idle mode that
+	 * can't support audio.
+	 */
+	if (wm8915->sysclk < 64000)
+		return;
+
+	for (aif = 0; aif < WM8915_AIFS; aif++) {
+		switch (aif) {
+		case 0:
+			bclk_reg = WM8915_AIF1_BCLK;
+			break;
+		case 1:
+			bclk_reg = WM8915_AIF2_BCLK;
+			break;
+		}
+
+		bclk_rate = wm8915->bclk_rate[aif];
+
+		/* Pick a divisor for BCLK as close as we can get to ideal */
+		best = 0;
+		for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+			cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
+			if (cur_val < 0) /* BCLK table is sorted */
+				break;
+			best = i;
+		}
+		bclk_rate = wm8915->sysclk / bclk_divs[best];
+		dev_dbg(codec->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
+			bclk_divs[best], bclk_rate);
+
+		snd_soc_update_bits(codec, bclk_reg,
+				    WM8915_AIF1_BCLK_DIV_MASK, best);
+	}
+}
+
 static int wm8915_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
@@ -1717,10 +1759,6 @@
 	return 0;
 }
 
-static const int bclk_divs[] = {
-	1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
-};
-
 static const int dsp_divs[] = {
 	48000, 32000, 16000, 8000
 };
@@ -1731,17 +1769,11 @@
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
-	int bits, i, bclk_rate, best, cur_val;
+	int bits, i, bclk_rate;
 	int aifdata = 0;
-	int bclk = 0;
 	int lrclk = 0;
 	int dsp = 0;
-	int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift;
-
-	if (!wm8915->sysclk) {
-		dev_err(codec->dev, "SYSCLK not configured\n");
-		return -EINVAL;
-	}
+	int aifdata_reg, lrclk_reg, dsp_shift;
 
 	switch (dai->id) {
 	case 0:
@@ -1753,7 +1785,6 @@
 			aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1;
 			lrclk_reg = WM8915_AIF1_TX_LRCLK_1;
 		}
-		bclk_reg = WM8915_AIF1_BCLK;
 		dsp_shift = 0;
 		break;
 	case 1:
@@ -1765,7 +1796,6 @@
 			aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1;
 			lrclk_reg = WM8915_AIF2_TX_LRCLK_1;
 		}
-		bclk_reg = WM8915_AIF2_BCLK;
 		dsp_shift = WM8915_DSP2_DIV_SHIFT;
 		break;
 	default:
@@ -1779,6 +1809,9 @@
 		return bclk_rate;
 	}
 
+	wm8915->bclk_rate[dai->id] = bclk_rate;
+	wm8915->rx_rate[dai->id] = params_rate(params);
+
 	/* Needs looking at for TDM */
 	bits = snd_pcm_format_width(params_format(params));
 	if (bits < 0)
@@ -1796,18 +1829,7 @@
 	}
 	dsp |= i << dsp_shift;
 
-	/* Pick a divisor for BCLK as close as we can get to ideal */
-	best = 0;
-	for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
-		cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
-		if (cur_val < 0) /* BCLK table is sorted */
-			break;
-		best = i;
-	}
-	bclk_rate = wm8915->sysclk / bclk_divs[best];
-	dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
-		bclk_divs[best], bclk_rate);
-	bclk |= best;
+	wm8915_update_bclk(codec);
 
 	lrclk = bclk_rate / params_rate(params);
 	dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
@@ -1817,14 +1839,11 @@
 			    WM8915_AIF1TX_WL_MASK |
 			    WM8915_AIF1TX_SLOT_LEN_MASK,
 			    aifdata);
-	snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk);
 	snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK,
 			    lrclk);
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2,
 			    WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp);
 
-	wm8915->rx_rate[dai->id] = params_rate(params);
-
 	return 0;
 }
 
@@ -1838,6 +1857,9 @@
 	int src;
 	int old;
 
+	if (freq == wm8915->sysclk && clk_id == wm8915->sysclk_src)
+		return 0;
+
 	/* Disable SYSCLK while we reconfigure */
 	old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
@@ -1882,6 +1904,8 @@
 		return -EINVAL;
 	}
 
+	wm8915_update_bclk(codec);
+
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
 			    WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK,
 			    src << WM8915_SYSCLK_SRC_SHIFT | ratediv);
@@ -1889,6 +1913,8 @@
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
 			    WM8915_SYSCLK_ENA, old);
 
+	wm8915->sysclk_src = clk_id;
+
 	return 0;
 }
 
@@ -2007,6 +2033,7 @@
 			  unsigned int Fref, unsigned int Fout)
 {
 	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+	struct i2c_client *i2c = to_i2c_client(codec->dev);
 	struct _fll_div fll_div;
 	unsigned long timeout;
 	int ret, reg;
@@ -2093,7 +2120,18 @@
 	else
 		timeout = msecs_to_jiffies(2);
 
-	wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+	/* Allow substantially longer if we've actually got the IRQ */
+	if (i2c->irq)
+		timeout *= 1000;
+
+	ret = wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+
+	if (ret == 0 && i2c->irq) {
+		dev_err(codec->dev, "Timed out waiting for FLL\n");
+		ret = -ETIMEDOUT;
+	} else {
+		ret = 0;
+	}
 
 	dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
 
@@ -2101,7 +2139,7 @@
 	wm8915->fll_fout = Fout;
 	wm8915->fll_src = source;
 
-	return 0;
+	return ret;
 }
 
 #ifdef CONFIG_GPIOLIB
@@ -2293,6 +2331,12 @@
 				    SND_JACK_HEADSET | SND_JACK_BTN_0);
 		wm8915->jack_mic = true;
 		wm8915->detecting = false;
+
+		/* Increase poll rate to give better responsiveness
+		 * for buttons */
+		snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+				    WM8915_MICD_RATE_MASK,
+				    5 << WM8915_MICD_RATE_SHIFT);
 	}
 
 	/* If we detected a lower impedence during initial startup
@@ -2333,15 +2377,17 @@
 					    SND_JACK_HEADPHONE,
 					    SND_JACK_HEADSET |
 					    SND_JACK_BTN_0);
+
+			/* Increase the detection rate a bit for
+			 * responsiveness.
+			 */
+			snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+					    WM8915_MICD_RATE_MASK,
+					    7 << WM8915_MICD_RATE_SHIFT);
+
 			wm8915->detecting = false;
 		}
 	}
-
-	/* Increase poll rate to give better responsiveness for buttons */
-	if (!wm8915->detecting)
-		snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
-				    WM8915_MICD_RATE_MASK,
-				    5 << WM8915_MICD_RATE_SHIFT);
 }
 
 static irqreturn_t wm8915_irq(int irq, void *data)
@@ -2383,6 +2429,20 @@
 	}
 }
 
+static irqreturn_t wm8915_edge_irq(int irq, void *data)
+{
+	irqreturn_t ret = IRQ_NONE;
+	irqreturn_t val;
+
+	do {
+		val = wm8915_irq(irq, data);
+		if (val != IRQ_NONE)
+			ret = val;
+	} while (val != IRQ_NONE);
+
+	return ret;
+}
+
 static void wm8915_retune_mobile_pdata(struct snd_soc_codec *codec)
 {
 	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
@@ -2482,8 +2542,6 @@
 	wm8915->disable_nb[1].notifier_call = wm8915_regulator_event_1;
 	wm8915->disable_nb[2].notifier_call = wm8915_regulator_event_2;
 	wm8915->disable_nb[3].notifier_call = wm8915_regulator_event_3;
-	wm8915->disable_nb[4].notifier_call = wm8915_regulator_event_4;
-	wm8915->disable_nb[5].notifier_call = wm8915_regulator_event_5;
 
 	/* This should really be moved into the regulator core */
 	for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) {
@@ -2709,8 +2767,14 @@
 
 		irq_flags |= IRQF_ONESHOT;
 
-		ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
-					   irq_flags, "wm8915", codec);
+		if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING))
+			ret = request_threaded_irq(i2c->irq, NULL,
+						   wm8915_edge_irq,
+						   irq_flags, "wm8915", codec);
+		else
+			ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
+						   irq_flags, "wm8915", codec);
+
 		if (ret == 0) {
 			/* Unmask the interrupt */
 			snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 25580e3..056daa0 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -297,8 +297,6 @@
 	if (ret)
 		goto error_ret;
 	ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-	if (ret)
-		goto error_ret;
 
 error_ret:
 	return ret;
@@ -683,8 +681,6 @@
 		}
 	}
 	ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	if (ret)
-		goto error_ret;
 
 error_ret:
 	return ret;
@@ -730,9 +726,6 @@
 	if (ret)
 		return ret;
 	ret = wm8940_add_widgets(codec);
-	if (ret)
-		return ret;
-
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 5e05eed..8499c56 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -78,6 +78,8 @@
 #ifdef CONFIG_GPIOLIB
 	struct gpio_chip gpio_chip;
 #endif
+
+	int irq;
 };
 
 /* We can't use the same notifier block for more than one supply and
@@ -1982,6 +1984,7 @@
 	0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
 	7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
 };
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 
 /* The VU bits for the headphones are in a different register to the mute
  * bits and only take effect on the PGA if it is actually powered.
@@ -2119,6 +2122,18 @@
 
 SOC_SINGLE_TLV("Speaker Boost Volume", WM8962_CLASS_D_CONTROL_2, 0, 7, 0,
 	       classd_tlv),
+
+SOC_SINGLE("EQ Switch", WM8962_EQ1, WM8962_EQ_ENA_SHIFT, 1, 0),
+SOC_DOUBLE_R_TLV("EQ1 Volume", WM8962_EQ2, WM8962_EQ22,
+		 WM8962_EQL_B1_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ2 Volume", WM8962_EQ2, WM8962_EQ22,
+		 WM8962_EQL_B2_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ3 Volume", WM8962_EQ2, WM8962_EQ22,
+		 WM8962_EQL_B3_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
+		 WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
+		 WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
 };
 
 static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2184,6 +2199,8 @@
 			struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+	unsigned long timeout;
 	int src;
 	int fll;
 
@@ -2203,9 +2220,19 @@
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		if (fll)
+		if (fll) {
 			snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 					    WM8962_FLL_ENA, WM8962_FLL_ENA);
+			if (wm8962->irq) {
+				timeout = msecs_to_jiffies(5);
+				timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+								      timeout);
+
+				if (timeout == 0)
+					dev_err(codec->dev,
+						"Timed out starting FLL\n");
+			}
+		}
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
@@ -2763,18 +2790,44 @@
 	1, -1, 2, 3, 4, -1, 6, 8, -1, 12, 16, 24, -1, 32, 32, 32
 };
 
+static const int sysclk_rates[] = {
+	64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
+};
+
 static void wm8962_configure_bclk(struct snd_soc_codec *codec)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	int dspclk, i;
 	int clocking2 = 0;
+	int clocking4 = 0;
 	int aif2 = 0;
 
-	if (!wm8962->bclk) {
-		dev_dbg(codec->dev, "No BCLK rate configured\n");
+	if (!wm8962->sysclk_rate) {
+		dev_dbg(codec->dev, "No SYSCLK configured\n");
 		return;
 	}
 
+	if (!wm8962->bclk || !wm8962->lrclk) {
+		dev_dbg(codec->dev, "No audio clocks configured\n");
+		return;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
+		if (sysclk_rates[i] == wm8962->sysclk_rate / wm8962->lrclk) {
+			clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
+			break;
+		}
+	}
+
+	if (i == ARRAY_SIZE(sysclk_rates)) {
+		dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
+			wm8962->sysclk_rate / wm8962->lrclk);
+		return;
+	}
+
+	snd_soc_update_bits(codec, WM8962_CLOCKING_4,
+			    WM8962_SYSCLK_RATE_MASK, clocking4);
+
 	dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
 	if (dspclk < 0) {
 		dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
@@ -2844,6 +2897,8 @@
 		/* VMID 2*50k */
 		snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
 				    WM8962_VMID_SEL_MASK, 0x80);
+
+		wm8962_configure_bclk(codec);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -2876,8 +2931,6 @@
 			snd_soc_update_bits(codec, WM8962_CLOCKING2,
 					    WM8962_CLKREG_OVD,
 					    WM8962_CLKREG_OVD);
-
-			wm8962_configure_bclk(codec);
 		}
 
 		/* VMID 2*250k */
@@ -2918,10 +2971,6 @@
 	{ 96000, 6 },
 };
 
-static const int sysclk_rates[] = {
-	64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
-};
-
 static int wm8962_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_pcm_hw_params *params,
 			    struct snd_soc_dai *dai)
@@ -2929,41 +2978,27 @@
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	int rate = params_rate(params);
 	int i;
 	int aif0 = 0;
 	int adctl3 = 0;
-	int clocking4 = 0;
 
 	wm8962->bclk = snd_soc_params_to_bclk(params);
 	wm8962->lrclk = params_rate(params);
 
 	for (i = 0; i < ARRAY_SIZE(sr_vals); i++) {
-		if (sr_vals[i].rate == rate) {
+		if (sr_vals[i].rate == wm8962->lrclk) {
 			adctl3 |= sr_vals[i].reg;
 			break;
 		}
 	}
 	if (i == ARRAY_SIZE(sr_vals)) {
-		dev_err(codec->dev, "Unsupported rate %dHz\n", rate);
+		dev_err(codec->dev, "Unsupported rate %dHz\n", wm8962->lrclk);
 		return -EINVAL;
 	}
 
-	if (rate % 8000 == 0)
+	if (wm8962->lrclk % 8000 == 0)
 		adctl3 |= WM8962_SAMPLE_RATE_INT_MODE;
 
-	for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
-		if (sysclk_rates[i] == wm8962->sysclk_rate / rate) {
-			clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
-			break;
-		}
-	}
-	if (i == ARRAY_SIZE(sysclk_rates)) {
-		dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
-			wm8962->sysclk_rate / rate);
-		return -EINVAL;
-	}
-
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
@@ -2985,8 +3020,6 @@
 	snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_3,
 			    WM8962_SAMPLE_RATE_INT_MODE |
 			    WM8962_SAMPLE_RATE_MASK, adctl3);
-	snd_soc_update_bits(codec, WM8962_CLOCKING_4,
-			    WM8962_SYSCLK_RATE_MASK, clocking4);
 
 	wm8962_configure_bclk(codec);
 
@@ -3261,16 +3294,31 @@
 
 	dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
 
-	/* This should be a massive overestimate */
-	timeout = msecs_to_jiffies(1);
+	ret = 0;
 
-	wait_for_completion_timeout(&wm8962->fll_lock, timeout);
+	if (fll1 & WM8962_FLL_ENA) {
+		/* This should be a massive overestimate but go even
+		 * higher if we'll error out
+		 */
+		if (wm8962->irq)
+			timeout = msecs_to_jiffies(5);
+		else
+			timeout = msecs_to_jiffies(1);
+
+		timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+						      timeout);
+
+		if (timeout == 0 && wm8962->irq) {
+			dev_err(codec->dev, "FLL lock timed out");
+			ret = -ETIMEDOUT;
+		}
+	}
 
 	wm8962->fll_fref = Fref;
 	wm8962->fll_fout = Fout;
 	wm8962->fll_src = source;
 
-	return 0;
+	return ret;
 }
 
 static int wm8962_mute(struct snd_soc_dai *dai, int mute)
@@ -3731,8 +3779,6 @@
 	int ret;
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
-	struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
-					      dev);
 	u16 *reg_cache = codec->reg_cache;
 	int i, trigger, irq_pol;
 	bool dmicclk, dmicdat;
@@ -3871,6 +3917,9 @@
 	snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME,
 			    WM8962_HPOUT_VU, WM8962_HPOUT_VU);
 
+	/* Stereo control for EQ */
+	snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0);
+
 	wm8962_add_widgets(codec);
 
 	/* Save boards having to disable DMIC when not in use */
@@ -3899,7 +3948,7 @@
 	wm8962_init_beep(codec);
 	wm8962_init_gpio(codec);
 
-	if (i2c->irq) {
+	if (wm8962->irq) {
 		if (pdata && pdata->irq_active_low) {
 			trigger = IRQF_TRIGGER_LOW;
 			irq_pol = WM8962_IRQ_POL;
@@ -3911,12 +3960,13 @@
 		snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL,
 				    WM8962_IRQ_POL, irq_pol);
 
-		ret = request_threaded_irq(i2c->irq, NULL, wm8962_irq,
+		ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq,
 					   trigger | IRQF_ONESHOT,
 					   "wm8962", codec);
 		if (ret != 0) {
 			dev_err(codec->dev, "Failed to request IRQ %d: %d\n",
-				i2c->irq, ret);
+				wm8962->irq, ret);
+			wm8962->irq = 0;
 			/* Non-fatal */
 		} else {
 			/* Enable some IRQs by default */
@@ -3941,12 +3991,10 @@
 static int wm8962_remove(struct snd_soc_codec *codec)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
-					      dev);
 	int i;
 
-	if (i2c->irq)
-		free_irq(i2c->irq, codec);
+	if (wm8962->irq)
+		free_irq(wm8962->irq, codec);
 
 	cancel_delayed_work_sync(&wm8962->mic_work);
 
@@ -3986,6 +4034,8 @@
 
 	i2c_set_clientdata(i2c, wm8962);
 
+	wm8962->irq = i2c->irq;
+
 	ret = snd_soc_register_codec(&i2c->dev,
 				     &soc_codec_dev_wm8962, &wm8962_dai, 1);
 	if (ret < 0)
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 9e5ff78..03af7ad 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1434,6 +1434,7 @@
 
 	wm8993->hubs_data.hp_startup_mode = 1;
 	wm8993->hubs_data.dcs_codes = -2;
+	wm8993->hubs_data.series_startup = 1;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
 	if (ret != 0) {
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index c2fc035..70a68fd 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2905,6 +2905,7 @@
 			wm8994->hubs.dcs_codes = -5;
 			wm8994->hubs.hp_startup_mode = 1;
 			wm8994->hubs.dcs_readback_mode = 1;
+			wm8994->hubs.series_startup = 1;
 			break;
 		default:
 			wm8994->hubs.dcs_readback_mode = 1;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 91c6b39..a469132 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -727,7 +727,7 @@
 SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
 
 SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
 
 SND_SOC_DAPM_OUTPUT("LINEOUT"),
 SND_SOC_DAPM_OUTPUT("SPKN"),
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 9e370d1..2d6c88b 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -107,8 +107,7 @@
 		return;
 	}
 
-	/* Devices not using a DCS code correction have startup mode */
-	if (hubs->dcs_codes) {
+	if (hubs->series_startup) {
 		/* Set for 32 series updates */
 		snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
 				    WM8993_DCS_SERIES_NO_01_MASK,
@@ -134,9 +133,9 @@
 		break;
 	case 1:
 		reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
-		reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+		reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
 			>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
-		reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+		reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
 		break;
 	default:
 		WARN(1, "Unknown DCS readback method\n");
@@ -150,13 +149,13 @@
 		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
 			hubs->dcs_codes);
 
-		/* HPOUT1L */
-		offset = reg_l;
+		/* HPOUT1R */
+		offset = reg_r;
 		offset += hubs->dcs_codes;
 		dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
-		/* HPOUT1R */
-		offset = reg_r;
+		/* HPOUT1L */
+		offset = reg_l;
 		offset += hubs->dcs_codes;
 		dcs_cfg |= (u8)offset;
 
@@ -168,8 +167,8 @@
 				  WM8993_DCS_TRIG_DAC_WR_0 |
 				  WM8993_DCS_TRIG_DAC_WR_1);
 	} else {
-		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
-		dcs_cfg |= reg_r;
+		dcs_cfg = reg_r << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		dcs_cfg |= reg_l;
 	}
 
 	/* Save the callibrated offset if we're in class W mode and
@@ -195,7 +194,7 @@
 
 	/* If we're applying an offset correction then updating the
 	 * callibration would be likely to introduce further offsets. */
-	if (hubs->dcs_codes)
+	if (hubs->dcs_codes || hubs->no_series_update)
 		return ret;
 
 	/* Only need to do this if the outputs are active */
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index f8a5e97..0d290d2 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -23,6 +23,8 @@
 	int dcs_codes;
 	int dcs_readback_mode;
 	int hp_startup_mode;
+	int series_startup;
+	int no_series_update;
 
 	bool class_w;
 	u16 class_w_dcs;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 9d35b8c..a49e667 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -46,11 +46,28 @@
 }
 #endif
 
+#define DAVINCI_PCM_FMTBITS	(\
+				SNDRV_PCM_FMTBIT_S8	|\
+				SNDRV_PCM_FMTBIT_U8	|\
+				SNDRV_PCM_FMTBIT_S16_LE	|\
+				SNDRV_PCM_FMTBIT_S16_BE	|\
+				SNDRV_PCM_FMTBIT_U16_LE	|\
+				SNDRV_PCM_FMTBIT_U16_BE	|\
+				SNDRV_PCM_FMTBIT_S24_LE	|\
+				SNDRV_PCM_FMTBIT_S24_BE	|\
+				SNDRV_PCM_FMTBIT_U24_LE	|\
+				SNDRV_PCM_FMTBIT_U24_BE	|\
+				SNDRV_PCM_FMTBIT_S32_LE	|\
+				SNDRV_PCM_FMTBIT_S32_BE	|\
+				SNDRV_PCM_FMTBIT_U32_LE	|\
+				SNDRV_PCM_FMTBIT_U32_BE)
+
 static struct snd_pcm_hardware pcm_hardware_playback = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
-	.formats = (SNDRV_PCM_FMTBIT_S16_LE),
+		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME|
+		 SNDRV_PCM_INFO_BATCH),
+	.formats = DAVINCI_PCM_FMTBITS,
 	.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
 		  SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
 		  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
@@ -59,7 +76,7 @@
 	.rate_min = 8000,
 	.rate_max = 96000,
 	.channels_min = 2,
-	.channels_max = 2,
+	.channels_max = 384,
 	.buffer_bytes_max = 128 * 1024,
 	.period_bytes_min = 32,
 	.period_bytes_max = 8 * 1024,
@@ -71,8 +88,9 @@
 static struct snd_pcm_hardware pcm_hardware_capture = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_PAUSE),
-	.formats = (SNDRV_PCM_FMTBIT_S16_LE),
+		 SNDRV_PCM_INFO_PAUSE |
+		 SNDRV_PCM_INFO_BATCH),
+	.formats = DAVINCI_PCM_FMTBITS,
 	.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
 		  SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
 		  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
@@ -81,7 +99,7 @@
 	.rate_min = 8000,
 	.rate_max = 96000,
 	.channels_min = 2,
-	.channels_max = 2,
+	.channels_max = 384,
 	.buffer_bytes_max = 128 * 1024,
 	.period_bytes_min = 32,
 	.period_bytes_max = 8 * 1024,
@@ -139,6 +157,22 @@
 	struct edmacc_param ram_params;
 };
 
+static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream)
+{
+	struct davinci_runtime_data *prtd = substream->runtime->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	prtd->period++;
+	if (unlikely(prtd->period >= runtime->periods))
+		prtd->period = 0;
+}
+
+static void davinci_pcm_period_reset(struct snd_pcm_substream *substream)
+{
+	struct davinci_runtime_data *prtd = substream->runtime->private_data;
+
+	prtd->period = 0;
+}
 /*
  * Not used with ping/pong
  */
@@ -199,10 +233,6 @@
 	else
 		edma_set_transfer_params(link, acnt, fifo_level, count,
 							fifo_level, ABSYNC);
-
-	prtd->period++;
-	if (unlikely(prtd->period >= runtime->periods))
-		prtd->period = 0;
 }
 
 static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
@@ -217,12 +247,13 @@
 		return;
 
 	if (snd_pcm_running(substream)) {
+		spin_lock(&prtd->lock);
 		if (prtd->ram_channel < 0) {
 			/* No ping/pong must fix up link dma data*/
-			spin_lock(&prtd->lock);
 			davinci_pcm_enqueue_dma(substream);
-			spin_unlock(&prtd->lock);
 		}
+		davinci_pcm_period_elapsed(substream);
+		spin_unlock(&prtd->lock);
 		snd_pcm_period_elapsed(substream);
 	}
 }
@@ -425,7 +456,8 @@
 
 	edma_read_slot(link, &prtd->asp_params);
 	prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN);
-	prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f);
+	prtd->asp_params.opt |= TCCHEN |
+		EDMA_TCC(prtd->ram_channel & 0x3f);
 	edma_write_slot(link, &prtd->asp_params);
 
 	/* pong */
@@ -439,7 +471,7 @@
 	prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f));
 	/* interrupt after every pong completion */
 	prtd->asp_params.opt |= TCINTEN | TCCHEN |
-		EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel));
+		EDMA_TCC(prtd->ram_channel & 0x3f);
 	edma_write_slot(link, &prtd->asp_params);
 
 	/* ram */
@@ -527,6 +559,13 @@
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
+		edma_start(prtd->asp_channel);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+		    prtd->ram_channel >= 0) {
+			/* copy 1st iram buffer */
+			edma_start(prtd->ram_channel);
+		}
+		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		edma_resume(prtd->asp_channel);
@@ -550,6 +589,7 @@
 {
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 
+	davinci_pcm_period_reset(substream);
 	if (prtd->ram_channel >= 0) {
 		int ret = ping_pong_dma_setup(substream);
 		if (ret < 0)
@@ -565,21 +605,31 @@
 		print_buf_info(prtd->asp_link[0], "asp_link[0]");
 		print_buf_info(prtd->asp_link[1], "asp_link[1]");
 
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			/* copy 1st iram buffer */
-			edma_start(prtd->ram_channel);
-		}
-		edma_start(prtd->asp_channel);
+		/*
+		 * There is a phase offset of 2 periods between the position
+		 * used by dma setup and the position reported in the pointer
+		 * function.
+		 *
+		 * The phase offset, when not using ping-pong buffers, is due to
+		 * the two consecutive calls to davinci_pcm_enqueue_dma() below.
+		 *
+		 * Whereas here, with ping-pong buffers, the phase is due to
+		 * there being an entire buffer transfer complete before the
+		 * first dma completion event triggers davinci_pcm_dma_irq().
+		 */
+		davinci_pcm_period_elapsed(substream);
+		davinci_pcm_period_elapsed(substream);
+
 		return 0;
 	}
-	prtd->period = 0;
 	davinci_pcm_enqueue_dma(substream);
+	davinci_pcm_period_elapsed(substream);
 
 	/* Copy self-linked parameter RAM entry into master channel */
 	edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
 	edma_write_slot(prtd->asp_channel, &prtd->asp_params);
 	davinci_pcm_enqueue_dma(substream);
-	edma_start(prtd->asp_channel);
+	davinci_pcm_period_elapsed(substream);
 
 	return 0;
 }
@@ -591,51 +641,23 @@
 	struct davinci_runtime_data *prtd = runtime->private_data;
 	unsigned int offset;
 	int asp_count;
-	dma_addr_t asp_src, asp_dst;
+	unsigned int period_size = snd_pcm_lib_period_bytes(substream);
 
+	/*
+	 * There is a phase offset of 2 periods between the position used by dma
+	 * setup and the position reported in the pointer function. Either +2 in
+	 * the dma setup or -2 here in the pointer function (with wrapping,
+	 * both) accounts for this offset -- choose the latter since it makes
+	 * the first-time setup clearer.
+	 */
 	spin_lock(&prtd->lock);
-	if (prtd->ram_channel >= 0) {
-		int ram_count;
-		int mod_ram;
-		dma_addr_t ram_src, ram_dst;
-		unsigned int period_size = snd_pcm_lib_period_bytes(substream);
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			/* reading ram before asp should be safe
-			 * as long as the asp transfers less than a ping size
-			 * of bytes between the 2 reads
-			 */
-			edma_get_position(prtd->ram_channel,
-					&ram_src, &ram_dst);
-			edma_get_position(prtd->asp_channel,
-					&asp_src, &asp_dst);
-			asp_count = asp_src - prtd->asp_params.src;
-			ram_count = ram_src - prtd->ram_params.src;
-			mod_ram = ram_count % period_size;
-			mod_ram -= asp_count;
-			if (mod_ram < 0)
-				mod_ram += period_size;
-			else if (mod_ram == 0) {
-				if (snd_pcm_running(substream))
-					mod_ram += period_size;
-			}
-			ram_count -= mod_ram;
-			if (ram_count < 0)
-				ram_count += period_size * runtime->periods;
-		} else {
-			edma_get_position(prtd->ram_channel,
-					&ram_src, &ram_dst);
-			ram_count = ram_dst - prtd->ram_params.dst;
-		}
-		asp_count = ram_count;
-	} else {
-		edma_get_position(prtd->asp_channel, &asp_src, &asp_dst);
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			asp_count = asp_src - runtime->dma_addr;
-		else
-			asp_count = asp_dst - runtime->dma_addr;
-	}
+	asp_count = prtd->period - 2;
 	spin_unlock(&prtd->lock);
 
+	if (asp_count < 0)
+		asp_count += runtime->periods;
+	asp_count *= period_size;
+
 	offset = bytes_to_frames(runtime, asp_count);
 	if (offset >= runtime->buffer_size)
 		offset = 0;
@@ -811,9 +833,11 @@
 
 static u64 davinci_pcm_dmamask = 0xffffffff;
 
-static int davinci_pcm_new(struct snd_card *card,
-			   struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index a456e49..e27c417 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -266,9 +266,11 @@
 
 static u64 ep93xx_pcm_dmamask = 0xffffffff;
 
-static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-			  struct snd_pcm *pcm)
+static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 6680c0b..732208c 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -294,9 +294,11 @@
  * Regardless of where the memory is actually allocated, since the device can
  * technically DMA to any 36-bit address, we do need to set the DMA mask to 36.
  */
-static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	static u64 fsl_dma_dmamask = DMA_BIT_MASK(36);
 	int ret;
 
@@ -939,7 +941,7 @@
 
 	iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
 	if (iprop)
-		dma->ssi_fifo_depth = *iprop;
+		dma->ssi_fifo_depth = be32_to_cpup(iprop);
 	else
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		dma->ssi_fifo_depth = 8;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 313e0cc..d48afea 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -678,7 +678,12 @@
 		kfree(ssi_private);
 		return ret;
 	}
-	ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start);
+	ssi_private->ssi = of_iomap(np, 0);
+	if (!ssi_private->ssi) {
+		dev_err(&pdev->dev, "could not map device resources\n");
+		kfree(ssi_private);
+		return -ENOMEM;
+	}
 	ssi_private->ssi_phys = res.start;
 	ssi_private->irq = irq_of_parse_and_map(np, 0);
 
@@ -691,7 +696,7 @@
 	/* Determine the FIFO depth. */
 	iprop = of_get_property(np, "fsl,fifo-depth", NULL);
 	if (iprop)
-		ssi_private->fifo_depth = *iprop;
+		ssi_private->fifo_depth = be32_to_cpup(iprop);
 	else
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		ssi_private->fifo_depth = 8;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index fff695c..19ad0c1 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -299,10 +299,11 @@
 };
 
 static u64 psc_dma_dmamask = 0xffffffff;
-static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
-			   struct snd_pcm *pcm)
+static int psc_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
 	size_t size = psc_dma_hardware.buffer_bytes_max;
 	int rc = 0;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index c16c6b2..a192979 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -233,7 +233,7 @@
 	if (!iprop)
 		return -1;
 
-	return *iprop;
+	return be32_to_cpup(iprop);
 }
 
 /**
@@ -258,7 +258,7 @@
 	if (!iprop)
 		return -EINVAL;
 
-	addr = *iprop;
+	addr = be32_to_cpup(iprop);
 
 	bus = get_parent_cell_index(np);
 	if (bus < 0)
@@ -305,7 +305,7 @@
 		return -EINVAL;
 	}
 
-	*dma_channel_id = *iprop;
+	*dma_channel_id = be32_to_cpup(iprop);
 	*dma_id = get_parent_cell_index(dma_channel_np);
 	of_node_put(dma_channel_np);
 
@@ -379,7 +379,7 @@
 		ret = -EINVAL;
 		goto error;
 	}
-	machine_data->ssi_id = *iprop;
+	machine_data->ssi_id = be32_to_cpup(iprop);
 
 	/* Get the serial format and clock direction. */
 	sprop = of_get_property(np, "fsl,mode", NULL);
@@ -405,7 +405,7 @@
 			ret = -EINVAL;
 			goto error;
 		}
-		machine_data->clk_frequency = *iprop;
+		machine_data->clk_frequency = be32_to_cpup(iprop);
 	} else if (strcasecmp(sprop, "i2s-master") == 0) {
 		machine_data->dai_format = SND_SOC_DAIFMT_I2S;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 66e0b68..8fa4d5f 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -232,7 +232,7 @@
 
 	iprop = of_get_property(parent, "cell-index", NULL);
 	if (iprop)
-		ret = *iprop;
+		ret = be32_to_cpup(iprop);
 
 	of_node_put(parent);
 
@@ -261,7 +261,7 @@
 	if (!iprop)
 		return -EINVAL;
 
-	addr = *iprop;
+	addr = be32_to_cpup(iprop);
 
 	bus = get_parent_cell_index(np);
 	if (bus < 0)
@@ -308,7 +308,7 @@
 		return -EINVAL;
 	}
 
-	*dma_channel_id = *iprop;
+	*dma_channel_id = be32_to_cpup(iprop);
 	*dma_id = get_parent_cell_index(dma_channel_np);
 	of_node_put(dma_channel_np);
 
@@ -379,7 +379,7 @@
 		ret = -EINVAL;
 		goto error;
 	}
-	mdata->ssi_id = *iprop;
+	mdata->ssi_id = be32_to_cpup(iprop);
 
 	/* Get the serial format and clock direction. */
 	sprop = of_get_property(np, "fsl,mode", NULL);
@@ -405,7 +405,7 @@
 			ret = -EINVAL;
 			goto error;
 		}
-		mdata->clk_frequency = *iprop;
+		mdata->clk_frequency = be32_to_cpup(iprop);
 	} else if (strcasecmp(sprop, "i2s-master") == 0) {
 		mdata->dai_format = SND_SOC_DAIFMT_I2S;
 		mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 413b78d..309c59e 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -238,12 +238,14 @@
 
 static int ssi_irq = 0;
 
-static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
-	ret = imx_pcm_new(card, dai, pcm);
+	ret = imx_pcm_new(rtd);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 61fceb0..10a8e27 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -388,10 +388,11 @@
 
 static u64 imx_pcm_dmamask = DMA_BIT_MASK(32);
 
-int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
-
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index dc8a875..0a84cec 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -225,8 +225,7 @@
 		struct imx_ssi *ssi);
 
 int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma);
-int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm);
+int imx_pcm_new(struct snd_soc_pcm_runtime *rtd);
 void imx_pcm_free(struct snd_pcm *pcm);
 
 /*
diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c
index fb1483f..a7c9578 100644
--- a/sound/soc/jz4740/jz4740-pcm.c
+++ b/sound/soc/jz4740/jz4740-pcm.c
@@ -299,9 +299,11 @@
 
 static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32);
 
-int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index e13c6ce..cd33de1 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -312,9 +312,11 @@
 	return 0;
 }
 
-static int kirkwood_dma_new(struct snd_card *card,
-		struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index 5a946b4..3e78260 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -402,9 +402,10 @@
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-			struct snd_pcm *pcm)
+int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int retval = 0;
 
 	pr_debug("sst_pcm_new called\n");
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index dac6732..9c0edad 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -356,7 +356,7 @@
 	nuc900_audio->irq_num = platform_get_irq(pdev, 0);
 	if (!nuc900_audio->irq_num) {
 		ret = -EBUSY;
-		goto out2;
+		goto out3;
 	}
 
 	nuc900_ac97_data = nuc900_audio;
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 8263f56..d589ef1 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -315,9 +315,12 @@
 }
 
 static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32);
-static int nuc900_dma_new(struct snd_card *card,
-	struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
+
 	if (!card->dev->dma_mask)
 		card->dev->dma_mask = &nuc900_pcm_dmamask;
 	if (!card->dev->coherent_dma_mask)
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 462cbcb..b40095a 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -427,7 +427,8 @@
 
 /* Board specific codec bias level control */
 static int ams_delta_set_bias_level(struct snd_soc_card *card,
-					enum snd_soc_bias_level level)
+				    struct snd_soc_dapm_context *dapm,
+				    enum snd_soc_bias_level level)
 {
 	struct snd_soc_codec *codec = card->rtd->codec;
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e6a6b99..b2f5751 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -366,9 +366,11 @@
 	}
 }
 
-static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-		 struct snd_pcm *pcm)
+static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index fab20a5..c430600 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -85,9 +85,10 @@
 
 static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index ab3ccae..80c85fd6 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -443,10 +443,11 @@
 
 static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32);
 
-static int s6000_pcm_new(struct snd_card *card,
-			 struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime)
 {
-	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+	struct snd_card *card = runtime->card->snd_card;
+	struct snd_soc_dai *dai = runtime->cpu_dai;
+	struct snd_pcm *pcm = runtime->pcm;
 	struct s6000_pcm_dma_params *params;
 	int res;
 
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index d155cbb..54b0e4b 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -158,7 +158,7 @@
 
 config SND_SOC_SAMSUNG_SMDK_SPDIF
 	tristate "SoC S/PDIF Audio support for SMDK"
-	depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210)
+	depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310)
 	select SND_SAMSUNG_SPDIF
 	help
 	  Say Y if you want to add support for SoC S/PDIF audio on the SMDK.
@@ -171,9 +171,23 @@
 	help
 	  Say Y if you want to add support for SoC audio on the SMDK.
 
+config SND_SOC_SMDK_WM8994_PCM
+	tristate "SoC PCM Audio support for WM8994 on SMDK"
+	depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310)
+	select SND_SOC_WM8994
+	select SND_SAMSUNG_PCM
+	help
+	  Say Y if you want to add support for SoC audio on the SMDK
+
 config SND_SOC_SPEYSIDE
 	tristate "Audio support for Wolfson Speyside"
 	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM8915
 	select SND_SOC_WM9081
+
+config SND_SOC_SPEYSIDE_WM8962
+	tristate "Audio support for Wolfson Speyside with WM8962"
+	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+	select SND_SAMSUNG_I2S
+	select SND_SOC_WM8962
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 683843a..9eb3b12 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -35,7 +35,9 @@
 snd-soc-goni-wm8994-objs := goni_wm8994.o
 snd-soc-smdk-spdif-objs := smdk_spdif.o
 snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o
+snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o
 snd-soc-speyside-objs := speyside.o
+snd-soc-speyside-wm8962-objs := speyside_wm8962.o
 
 obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -54,4 +56,6 @@
 obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o
 obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o
 obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o
+obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o
 obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o
+obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 5cb3b88..9465588 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -425,9 +425,11 @@
 
 static u64 dma_mask = DMA_BIT_MASK(32);
 
-static int dma_new(struct snd_card *card,
-	struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	pr_debug("Entered %s\n", __func__);
diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h
new file mode 100644
index 0000000..c0e6d9a
--- /dev/null
+++ b/sound/soc/samsung/i2s-regs.h
@@ -0,0 +1,143 @@
+/*
+ * linux/sound/soc/samsung/i2s-regs.h
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd.
+ *		http://www.samsung.com
+ *
+ * Samsung I2S driver's register header
+ *
+ * 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 __SND_SOC_SAMSUNG_I2S_REGS_H
+#define __SND_SOC_SAMSUNG_I2S_REGS_H
+
+#define I2SCON		0x0
+#define I2SMOD		0x4
+#define I2SFIC		0x8
+#define I2SPSR		0xc
+#define I2STXD		0x10
+#define I2SRXD		0x14
+#define I2SFICS		0x18
+#define I2STXDS		0x1c
+#define I2SAHB		0x20
+#define I2SSTR0		0x24
+#define I2SSIZE		0x28
+#define I2STRNCNT	0x2c
+#define I2SLVL0ADDR	0x30
+#define I2SLVL1ADDR	0x34
+#define I2SLVL2ADDR	0x38
+#define I2SLVL3ADDR	0x3c
+
+#define CON_RSTCLR		(1 << 31)
+#define CON_FRXOFSTATUS		(1 << 26)
+#define CON_FRXORINTEN		(1 << 25)
+#define CON_FTXSURSTAT		(1 << 24)
+#define CON_FTXSURINTEN		(1 << 23)
+#define CON_TXSDMA_PAUSE	(1 << 20)
+#define CON_TXSDMA_ACTIVE	(1 << 18)
+
+#define CON_FTXURSTATUS		(1 << 17)
+#define CON_FTXURINTEN		(1 << 16)
+#define CON_TXFIFO2_EMPTY	(1 << 15)
+#define CON_TXFIFO1_EMPTY	(1 << 14)
+#define CON_TXFIFO2_FULL	(1 << 13)
+#define CON_TXFIFO1_FULL	(1 << 12)
+
+#define CON_LRINDEX		(1 << 11)
+#define CON_TXFIFO_EMPTY	(1 << 10)
+#define CON_RXFIFO_EMPTY	(1 << 9)
+#define CON_TXFIFO_FULL		(1 << 8)
+#define CON_RXFIFO_FULL		(1 << 7)
+#define CON_TXDMA_PAUSE		(1 << 6)
+#define CON_RXDMA_PAUSE		(1 << 5)
+#define CON_TXCH_PAUSE		(1 << 4)
+#define CON_RXCH_PAUSE		(1 << 3)
+#define CON_TXDMA_ACTIVE	(1 << 2)
+#define CON_RXDMA_ACTIVE	(1 << 1)
+#define CON_ACTIVE		(1 << 0)
+
+#define MOD_OPCLK_CDCLK_OUT	(0 << 30)
+#define MOD_OPCLK_CDCLK_IN	(1 << 30)
+#define MOD_OPCLK_BCLK_OUT	(2 << 30)
+#define MOD_OPCLK_PCLK		(3 << 30)
+#define MOD_OPCLK_MASK		(3 << 30)
+#define MOD_TXS_IDMA		(1 << 28) /* Sec_TXFIFO use I-DMA */
+
+#define MOD_BLCS_SHIFT		26
+#define MOD_BLCS_16BIT		(0 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_8BIT		(1 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_24BIT		(2 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_MASK		(3 << MOD_BLCS_SHIFT)
+#define MOD_BLCP_SHIFT		24
+#define MOD_BLCP_16BIT		(0 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_8BIT		(1 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_24BIT		(2 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_MASK		(3 << MOD_BLCP_SHIFT)
+
+#define MOD_C2DD_HHALF		(1 << 21) /* Discard Higher-half */
+#define MOD_C2DD_LHALF		(1 << 20) /* Discard Lower-half */
+#define MOD_C1DD_HHALF		(1 << 19)
+#define MOD_C1DD_LHALF		(1 << 18)
+#define MOD_DC2_EN		(1 << 17)
+#define MOD_DC1_EN		(1 << 16)
+#define MOD_BLC_16BIT		(0 << 13)
+#define MOD_BLC_8BIT		(1 << 13)
+#define MOD_BLC_24BIT		(2 << 13)
+#define MOD_BLC_MASK		(3 << 13)
+
+#define MOD_IMS_SYSMUX		(1 << 10)
+#define MOD_SLAVE		(1 << 11)
+#define MOD_TXONLY		(0 << 8)
+#define MOD_RXONLY		(1 << 8)
+#define MOD_TXRX		(2 << 8)
+#define MOD_MASK		(3 << 8)
+#define MOD_LR_LLOW		(0 << 7)
+#define MOD_LR_RLOW		(1 << 7)
+#define MOD_SDF_IIS		(0 << 5)
+#define MOD_SDF_MSB		(1 << 5)
+#define MOD_SDF_LSB		(2 << 5)
+#define MOD_SDF_MASK		(3 << 5)
+#define MOD_RCLK_256FS		(0 << 3)
+#define MOD_RCLK_512FS		(1 << 3)
+#define MOD_RCLK_384FS		(2 << 3)
+#define MOD_RCLK_768FS		(3 << 3)
+#define MOD_RCLK_MASK		(3 << 3)
+#define MOD_BCLK_32FS		(0 << 1)
+#define MOD_BCLK_48FS		(1 << 1)
+#define MOD_BCLK_16FS		(2 << 1)
+#define MOD_BCLK_24FS		(3 << 1)
+#define MOD_BCLK_MASK		(3 << 1)
+#define MOD_8BIT		(1 << 0)
+
+#define MOD_CDCLKCON		(1 << 12)
+
+#define PSR_PSREN		(1 << 15)
+
+#define FIC_TX2COUNT(x)		(((x) >>  24) & 0xf)
+#define FIC_TX1COUNT(x)		(((x) >>  16) & 0xf)
+
+#define FIC_TXFLUSH		(1 << 15)
+#define FIC_RXFLUSH		(1 << 7)
+
+#define FIC_TXCOUNT(x)		(((x) >>  8) & 0xf)
+#define FIC_RXCOUNT(x)		(((x) >>  0) & 0xf)
+#define FICS_TXCOUNT(x)		(((x) >>  8) & 0x7f)
+
+#define AHB_INTENLVL0		(1 << 24)
+#define AHB_LVL0INT		(1 << 20)
+#define AHB_CLRLVL0INT		(1 << 16)
+#define AHB_DMARLD		(1 << 5)
+#define AHB_INTMASK		(1 << 3)
+#define AHB_DMAEN		(1 << 0)
+#define AHB_LVLINTMASK		(0xf << 20)
+
+#define I2SSIZE_TRNMSK		(0xffff)
+#define I2SSIZE_SHIFT		(16)
+
+#endif /* __SND_SOC_SAMSUNG_I2S_REGS_H */
+
+
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 992a732..1568eea 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -22,109 +22,7 @@
 
 #include "dma.h"
 #include "i2s.h"
-
-#define I2SCON		0x0
-#define I2SMOD		0x4
-#define I2SFIC		0x8
-#define I2SPSR		0xc
-#define I2STXD		0x10
-#define I2SRXD		0x14
-#define I2SFICS		0x18
-#define I2STXDS		0x1c
-
-#define CON_RSTCLR		(1 << 31)
-#define CON_FRXOFSTATUS		(1 << 26)
-#define CON_FRXORINTEN		(1 << 25)
-#define CON_FTXSURSTAT		(1 << 24)
-#define CON_FTXSURINTEN		(1 << 23)
-#define CON_TXSDMA_PAUSE	(1 << 20)
-#define CON_TXSDMA_ACTIVE	(1 << 18)
-
-#define CON_FTXURSTATUS		(1 << 17)
-#define CON_FTXURINTEN		(1 << 16)
-#define CON_TXFIFO2_EMPTY	(1 << 15)
-#define CON_TXFIFO1_EMPTY	(1 << 14)
-#define CON_TXFIFO2_FULL	(1 << 13)
-#define CON_TXFIFO1_FULL	(1 << 12)
-
-#define CON_LRINDEX		(1 << 11)
-#define CON_TXFIFO_EMPTY	(1 << 10)
-#define CON_RXFIFO_EMPTY	(1 << 9)
-#define CON_TXFIFO_FULL		(1 << 8)
-#define CON_RXFIFO_FULL		(1 << 7)
-#define CON_TXDMA_PAUSE		(1 << 6)
-#define CON_RXDMA_PAUSE		(1 << 5)
-#define CON_TXCH_PAUSE		(1 << 4)
-#define CON_RXCH_PAUSE		(1 << 3)
-#define CON_TXDMA_ACTIVE	(1 << 2)
-#define CON_RXDMA_ACTIVE	(1 << 1)
-#define CON_ACTIVE		(1 << 0)
-
-#define MOD_OPCLK_CDCLK_OUT	(0 << 30)
-#define MOD_OPCLK_CDCLK_IN	(1 << 30)
-#define MOD_OPCLK_BCLK_OUT	(2 << 30)
-#define MOD_OPCLK_PCLK		(3 << 30)
-#define MOD_OPCLK_MASK		(3 << 30)
-#define MOD_TXS_IDMA		(1 << 28) /* Sec_TXFIFO use I-DMA */
-
-#define MOD_BLCS_SHIFT	26
-#define MOD_BLCS_16BIT	(0 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_8BIT	(1 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_24BIT	(2 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_MASK	(3 << MOD_BLCS_SHIFT)
-#define MOD_BLCP_SHIFT	24
-#define MOD_BLCP_16BIT	(0 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_8BIT	(1 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_24BIT	(2 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_MASK	(3 << MOD_BLCP_SHIFT)
-
-#define MOD_C2DD_HHALF		(1 << 21) /* Discard Higher-half */
-#define MOD_C2DD_LHALF		(1 << 20) /* Discard Lower-half */
-#define MOD_C1DD_HHALF		(1 << 19)
-#define MOD_C1DD_LHALF		(1 << 18)
-#define MOD_DC2_EN		(1 << 17)
-#define MOD_DC1_EN		(1 << 16)
-#define MOD_BLC_16BIT		(0 << 13)
-#define MOD_BLC_8BIT		(1 << 13)
-#define MOD_BLC_24BIT		(2 << 13)
-#define MOD_BLC_MASK		(3 << 13)
-
-#define MOD_IMS_SYSMUX		(1 << 10)
-#define MOD_SLAVE		(1 << 11)
-#define MOD_TXONLY		(0 << 8)
-#define MOD_RXONLY		(1 << 8)
-#define MOD_TXRX		(2 << 8)
-#define MOD_MASK		(3 << 8)
-#define MOD_LR_LLOW		(0 << 7)
-#define MOD_LR_RLOW		(1 << 7)
-#define MOD_SDF_IIS		(0 << 5)
-#define MOD_SDF_MSB		(1 << 5)
-#define MOD_SDF_LSB		(2 << 5)
-#define MOD_SDF_MASK		(3 << 5)
-#define MOD_RCLK_256FS		(0 << 3)
-#define MOD_RCLK_512FS		(1 << 3)
-#define MOD_RCLK_384FS		(2 << 3)
-#define MOD_RCLK_768FS		(3 << 3)
-#define MOD_RCLK_MASK		(3 << 3)
-#define MOD_BCLK_32FS		(0 << 1)
-#define MOD_BCLK_48FS		(1 << 1)
-#define MOD_BCLK_16FS		(2 << 1)
-#define MOD_BCLK_24FS		(3 << 1)
-#define MOD_BCLK_MASK		(3 << 1)
-#define MOD_8BIT		(1 << 0)
-
-#define MOD_CDCLKCON		(1 << 12)
-
-#define PSR_PSREN		(1 << 15)
-
-#define FIC_TX2COUNT(x)		(((x) >>  24) & 0xf)
-#define FIC_TX1COUNT(x)		(((x) >>  16) & 0xf)
-
-#define FIC_TXFLUSH		(1 << 15)
-#define FIC_RXFLUSH		(1 << 7)
-#define FIC_TXCOUNT(x)		(((x) >>  8) & 0xf)
-#define FIC_RXCOUNT(x)		(((x) >>  0) & 0xf)
-#define FICS_TXCOUNT(x)		(((x) >>  8) & 0x7f)
+#include "i2s-regs.h"
 
 #define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
 
diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c
new file mode 100644
index 0000000..5f21116
--- /dev/null
+++ b/sound/soc/samsung/smdk_wm8994pcm.c
@@ -0,0 +1,176 @@
+/*
+ *  sound/soc/samsung/smdk_wm8994pcm.c
+ *
+ *  Copyright (c) 2011 Samsung Electronics Co., Ltd
+ *		http://www.samsung.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/soc.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/wm8994.h"
+#include "dma.h"
+#include "pcm.h"
+
+/*
+ * Board Settings:
+ *  o '1' means 'ON'
+ *  o '0' means 'OFF'
+ *  o 'X' means 'Don't care'
+ *
+ * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111
+ */
+
+/*
+ * Configure audio route as :-
+ * $ amixer sset 'DAC1' on,on
+ * $ amixer sset 'Right Headphone Mux' 'DAC'
+ * $ amixer sset 'Left Headphone Mux' 'DAC'
+ * $ amixer sset 'DAC1R Mixer AIF1.1' on
+ * $ amixer sset 'DAC1L Mixer AIF1.1' on
+ * $ amixer sset 'IN2L' on
+ * $ amixer sset 'IN2L PGA IN2LN' on
+ * $ amixer sset 'MIXINL IN2L' on
+ * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on
+ * $ amixer sset 'IN2R' on
+ * $ amixer sset 'IN2R PGA IN2RN' on
+ * $ amixer sset 'MIXINR IN2R' on
+ * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on
+ */
+
+/* SMDK has a 16.9344MHZ crystal attached to WM8994 */
+#define SMDK_WM8994_FREQ 16934400
+
+static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	unsigned long mclk_freq;
+	int rfs, ret;
+
+	switch(params_rate(params)) {
+	case 8000:
+		rfs = 512;
+		break;
+	default:
+		dev_err(cpu_dai->dev, "%s:%d Sampling Rate %u not supported!\n",
+		__func__, __LINE__, params_rate(params));
+		return -EINVAL;
+	}
+
+	mclk_freq = params_rate(params) * rfs;
+
+	/* Set the codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B
+				| SND_SOC_DAIFMT_IB_NF
+				| SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* Set the cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B
+				| SND_SOC_DAIFMT_IB_NF
+				| SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
+					mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
+					SMDK_WM8994_FREQ, mclk_freq);
+	if (ret < 0)
+		return ret;
+
+	/* Set PCM source clock on CPU */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX,
+					mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* Set SCLK_DIV for making bclk */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops smdk_wm8994_pcm_ops = {
+	.hw_params = smdk_wm8994_pcm_hw_params,
+};
+
+static struct snd_soc_dai_link smdk_dai[] = {
+	{
+		.name = "WM8994 PAIF PCM",
+		.stream_name = "Primary PCM",
+		.cpu_dai_name = "samsung-pcm.0",
+		.codec_dai_name = "wm8994-aif1",
+		.platform_name = "samsung-audio",
+		.codec_name = "wm8994-codec",
+		.ops = &smdk_wm8994_pcm_ops,
+	},
+};
+
+static struct snd_soc_card smdk_pcm = {
+	.name = "SMDK-PCM",
+	.dai_link = smdk_dai,
+	.num_links = 1,
+};
+
+static int __devinit snd_smdk_probe(struct platform_device *pdev)
+{
+	int ret = 0;
+
+	smdk_pcm.dev = &pdev->dev;
+	ret = snd_soc_register_card(&smdk_pcm);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit snd_smdk_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&smdk_pcm);
+	platform_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+static struct platform_driver snd_smdk_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "samsung-smdk-pcm",
+	},
+	.probe = snd_smdk_probe,
+	.remove = __devexit_p(snd_smdk_remove),
+};
+
+static int __init smdk_audio_init(void)
+{
+	return platform_driver_register(&snd_smdk_driver);
+}
+
+module_init(smdk_audio_init);
+
+static void __exit smdk_audio_exit(void)
+{
+	platform_driver_unregister(&snd_smdk_driver);
+}
+
+module_exit(smdk_audio_exit);
+
+MODULE_AUTHOR("Sangbeom Kim, <sbkim73@samsung.com>");
+MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 360a333..d6dee4d 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -20,24 +20,29 @@
 #define WM8915_HPSEL_GPIO 214
 
 static int speyside_set_bias_level(struct snd_soc_card *card,
+				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level)
 {
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	int ret;
 
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
-		ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK1,
+		ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK2,
 					     32768, SND_SOC_CLOCK_IN);
 		if (ret < 0)
 			return ret;
 
-		ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK1,
+		ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK2,
 					  0, 0, 0);
 		if (ret < 0) {
 			pr_err("Failed to stop FLL\n");
 			return ret;
 		}
+		break;
 
 	default:
 		break;
@@ -46,6 +51,45 @@
 	return 0;
 }
 
+static int speyside_set_bias_level_post(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
+					enum snd_soc_bias_level level)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+			ret = snd_soc_dai_set_pll(codec_dai, 0,
+						  WM8915_FLL_MCLK2,
+						  32768, 48000 * 256);
+			if (ret < 0) {
+				pr_err("Failed to start FLL\n");
+				return ret;
+			}
+
+			ret = snd_soc_dai_set_sysclk(codec_dai,
+						     WM8915_SYSCLK_FLL,
+						     48000 * 256,
+						     SND_SOC_CLOCK_IN);
+			if (ret < 0)
+				return ret;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	card->dapm.bias_level = level;
+
+	return 0;
+}
+
 static int speyside_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params)
 {
@@ -66,16 +110,6 @@
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK1,
-				  32768, 256 * 48000);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_FLL,
-				     256 * 48000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
@@ -127,7 +161,7 @@
 	struct snd_soc_codec *codec = rtd->codec;
 	int ret;
 
-	ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0);
+	ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK2, 32768, 0);
 	if (ret < 0)
 		return ret;
 
@@ -267,6 +301,7 @@
 	.num_configs = ARRAY_SIZE(speyside_codec_conf),
 
 	.set_bias_level = speyside_set_bias_level,
+	.set_bias_level_post = speyside_set_bias_level_post,
 
 	.controls = controls,
 	.num_controls = ARRAY_SIZE(controls),
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
new file mode 100644
index 0000000..8ac42bf
--- /dev/null
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -0,0 +1,264 @@
+/*
+ * Speyside with WM8962 audio support
+ *
+ * Copyright 2011 Wolfson Microelectronics
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <linux/gpio.h>
+
+#include "../codecs/wm8962.h"
+
+static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
+					  struct snd_soc_dapm_context *dapm,
+					  enum snd_soc_bias_level level)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+			ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+						  WM8962_FLL_MCLK, 32768,
+						  44100 * 256);
+			if (ret < 0)
+				pr_err("Failed to start FLL: %d\n", ret);
+
+			ret = snd_soc_dai_set_sysclk(codec_dai,
+						     WM8962_SYSCLK_FLL,
+						     44100 * 256,
+						     SND_SOC_CLOCK_IN);
+			if (ret < 0) {
+				pr_err("Failed to set SYSCLK: %d\n");
+				return ret;
+			}
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
+					       struct snd_soc_dapm_context *dapm,
+					       enum snd_soc_bias_level level)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_STANDBY:
+		ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+					     32768, SND_SOC_CLOCK_IN);
+		if (ret < 0) {
+			pr_err("Failed to switch away from FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+					  0, 0, 0);
+		if (ret < 0) {
+			pr_err("Failed to stop FLL: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	dapm->bias_level = level;
+
+	return 0;
+}
+
+static int speyside_wm8962_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 *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+					 | SND_SOC_DAIFMT_NB_NF
+					 | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+					 | SND_SOC_DAIFMT_NB_NF
+					 | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops speyside_wm8962_ops = {
+	.hw_params = speyside_wm8962_hw_params,
+};
+
+static struct snd_soc_dai_link speyside_wm8962_dai[] = {
+	{
+		.name = "CPU",
+		.stream_name = "CPU",
+		.cpu_dai_name = "samsung-i2s.0",
+		.codec_dai_name = "wm8962",
+		.platform_name = "samsung-audio",
+		.codec_name = "wm8962.1-001a",
+		.ops = &speyside_wm8962_ops,
+	},
+};
+
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("Main Speaker"),
+};
+
+static struct snd_soc_dapm_widget widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
+	SND_SOC_DAPM_MIC("DMIC", NULL),
+
+	SND_SOC_DAPM_SPK("Main Speaker", NULL),
+};
+
+static struct snd_soc_dapm_route audio_paths[] = {
+	{ "Headphone", NULL, "HPOUTL" },
+	{ "Headphone", NULL, "HPOUTR" },
+
+	{ "Main Speaker", NULL, "SPKOUTL" },
+	{ "Main Speaker", NULL, "SPKOUTR" },
+
+	{ "MICBIAS", NULL, "Headset Mic" },
+	{ "IN4L", NULL, "MICBIAS" },
+	{ "IN4R", NULL, "MICBIAS" },
+
+	{ "MICBIAS", NULL, "DMIC" },
+	{ "DMICDAT", NULL, "MICBIAS" },
+};
+
+static struct snd_soc_jack speyside_wm8962_headset;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin speyside_wm8962_headset_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headphone",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+static int speyside_wm8962_late_probe(struct snd_soc_card *card)
+{
+	struct snd_soc_codec *codec = card->rtd[0].codec;
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+				     32768, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_jack_new(codec, "Headset",
+			       SND_JACK_HEADSET | SND_JACK_BTN_0,
+			       &speyside_wm8962_headset);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_pins(&speyside_wm8962_headset,
+				    ARRAY_SIZE(speyside_wm8962_headset_pins),
+				    speyside_wm8962_headset_pins);
+	if (ret)
+		return ret;
+
+	wm8962_mic_detect(codec, &speyside_wm8962_headset);
+
+	return 0;
+}
+
+static struct snd_soc_card speyside_wm8962 = {
+	.name = "Speyside WM8962",
+	.dai_link = speyside_wm8962_dai,
+	.num_links = ARRAY_SIZE(speyside_wm8962_dai),
+
+	.set_bias_level = speyside_wm8962_set_bias_level,
+	.set_bias_level_post = speyside_wm8962_set_bias_level_post,
+
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
+	.dapm_widgets = widgets,
+	.num_dapm_widgets = ARRAY_SIZE(widgets),
+	.dapm_routes = audio_paths,
+	.num_dapm_routes = ARRAY_SIZE(audio_paths),
+
+	.late_probe = speyside_wm8962_late_probe,
+};
+
+static __devinit int speyside_wm8962_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &speyside_wm8962;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit speyside_wm8962_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver speyside_wm8962_driver = {
+	.driver = {
+		.name = "speyside-wm8962",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = speyside_wm8962_probe,
+	.remove = __devexit_p(speyside_wm8962_remove),
+};
+
+static int __init speyside_wm8962_audio_init(void)
+{
+	return platform_driver_register(&speyside_wm8962_driver);
+}
+module_init(speyside_wm8962_audio_init);
+
+static void __exit speyside_wm8962_audio_exit(void)
+{
+	platform_driver_unregister(&speyside_wm8962_driver);
+}
+module_exit(speyside_wm8962_audio_exit);
+
+MODULE_DESCRIPTION("Speyside WM8962 audio support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:speyside-wm8962");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index c326d29..db74005 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -327,10 +327,10 @@
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-static int camelot_pcm_new(struct snd_card *card,
-			   struct snd_soc_dai *dai,
-			   struct snd_pcm *pcm)
+static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_pcm *pcm = rtd->pcm;
+
 	/* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
 	 * in MMAP mode (i.e. aplay -M)
 	 */
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 4a9da6b..8e112cc 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -118,10 +118,38 @@
 /*
  * FSI driver use below type name for variable
  *
- * xxx_len	: data length
- * xxx_width	: data width
- * xxx_offset	: data offset
  * xxx_num	: number of data
+ * xxx_pos	: position of data
+ * xxx_capa	: capacity of data
+ */
+
+/*
+ *	period/frame/sample image
+ *
+ * ex) PCM (2ch)
+ *
+ * period pos					   period pos
+ *   [n]					     [n + 1]
+ *   |<-------------------- period--------------------->|
+ * ==|============================================ ... =|==
+ *   |							|
+ *   ||<-----  frame ----->|<------ frame ----->|  ...	|
+ *   |+--------------------+--------------------+- ...	|
+ *   ||[ sample ][ sample ]|[ sample ][ sample ]|  ...	|
+ *   |+--------------------+--------------------+- ...	|
+ * ==|============================================ ... =|==
+ */
+
+/*
+ *	FSI FIFO image
+ *
+ *	|	     |
+ *	|	     |
+ *	| [ sample ] |
+ *	| [ sample ] |
+ *	| [ sample ] |
+ *	| [ sample ] |
+ *		--> go to codecs
  */
 
 /*
@@ -131,12 +159,11 @@
 struct fsi_stream {
 	struct snd_pcm_substream *substream;
 
-	int fifo_max_num;
-
-	int buff_offset;
-	int buff_len;
-	int period_len;
-	int period_num;
+	int fifo_sample_capa;	/* sample capacity of FSI FIFO */
+	int buff_sample_capa;	/* sample capacity of ALSA buffer */
+	int buff_sample_pos;	/* sample position of ALSA buffer */
+	int period_samples;	/* sample number / 1 period */
+	int period_pos;		/* current period position */
 
 	int uerr_num;
 	int oerr_num;
@@ -149,17 +176,14 @@
 	struct fsi_stream playback;
 	struct fsi_stream capture;
 
+	u32 do_fmt;
+	u32 di_fmt;
+
 	int chan_num:16;
 	int clk_master:1;
+	int spdif:1;
 
 	long rate;
-
-	/* for suspend/resume */
-	u32 saved_do_fmt;
-	u32 saved_di_fmt;
-	u32 saved_ckg1;
-	u32 saved_ckg2;
-	u32 saved_out_sel;
 };
 
 struct fsi_core {
@@ -180,14 +204,6 @@
 	struct fsi_core *core;
 	struct sh_fsi_platform_info *info;
 	spinlock_t lock;
-
-	/* for suspend/resume */
-	u32 saved_a_mclk;
-	u32 saved_b_mclk;
-	u32 saved_iemsk;
-	u32 saved_imsk;
-	u32 saved_clk_rst;
-	u32 saved_soft_rst;
 };
 
 /*
@@ -271,6 +287,11 @@
 	return fsi->master->base == fsi->base;
 }
 
+static int fsi_is_spdif(struct fsi_priv *fsi)
+{
+	return fsi->spdif;
+}
+
 static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -342,28 +363,59 @@
 	return shift;
 }
 
-static void fsi_stream_push(struct fsi_priv *fsi,
-			    int is_play,
-			    struct snd_pcm_substream *substream,
-			    u32 buffer_len,
-			    u32 period_len)
+static int fsi_frame2sample(struct fsi_priv *fsi, int frames)
+{
+	return frames * fsi->chan_num;
+}
+
+static int fsi_sample2frame(struct fsi_priv *fsi, int samples)
+{
+	return samples / fsi->chan_num;
+}
+
+static int fsi_stream_is_working(struct fsi_priv *fsi,
+				  int is_play)
 {
 	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+	struct fsi_master *master = fsi_get_master(fsi);
+	unsigned long flags;
+	int ret;
 
+	spin_lock_irqsave(&master->lock, flags);
+	ret = !!io->substream;
+	spin_unlock_irqrestore(&master->lock, flags);
+
+	return ret;
+}
+
+static void fsi_stream_push(struct fsi_priv *fsi,
+			    int is_play,
+			    struct snd_pcm_substream *substream)
+{
+	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct fsi_master *master = fsi_get_master(fsi);
+	unsigned long flags;
+
+	spin_lock_irqsave(&master->lock, flags);
 	io->substream	= substream;
-	io->buff_len	= buffer_len;
-	io->buff_offset	= 0;
-	io->period_len	= period_len;
-	io->period_num	= 0;
+	io->buff_sample_capa	= fsi_frame2sample(fsi, runtime->buffer_size);
+	io->buff_sample_pos	= 0;
+	io->period_samples	= fsi_frame2sample(fsi, runtime->period_size);
+	io->period_pos		= 0;
 	io->oerr_num	= -1; /* ignore 1st err */
 	io->uerr_num	= -1; /* ignore 1st err */
+	spin_unlock_irqrestore(&master->lock, flags);
 }
 
 static void fsi_stream_pop(struct fsi_priv *fsi, int is_play)
 {
 	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
 	struct snd_soc_dai *dai = fsi_get_dai(io->substream);
+	struct fsi_master *master = fsi_get_master(fsi);
+	unsigned long flags;
 
+	spin_lock_irqsave(&master->lock, flags);
 
 	if (io->oerr_num > 0)
 		dev_err(dai->dev, "over_run = %d\n", io->oerr_num);
@@ -372,47 +424,27 @@
 		dev_err(dai->dev, "under_run = %d\n", io->uerr_num);
 
 	io->substream	= NULL;
-	io->buff_len	= 0;
-	io->buff_offset	= 0;
-	io->period_len	= 0;
-	io->period_num	= 0;
+	io->buff_sample_capa	= 0;
+	io->buff_sample_pos	= 0;
+	io->period_samples	= 0;
+	io->period_pos		= 0;
 	io->oerr_num	= 0;
 	io->uerr_num	= 0;
+	spin_unlock_irqrestore(&master->lock, flags);
 }
 
-static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
+static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, int is_play)
 {
 	u32 status;
-	int data_num;
+	int frames;
 
 	status = is_play ?
 		fsi_reg_read(fsi, DOFF_ST) :
 		fsi_reg_read(fsi, DIFF_ST);
 
-	data_num = 0x1ff & (status >> 8);
-	data_num *= fsi->chan_num;
+	frames = 0x1ff & (status >> 8);
 
-	return data_num;
-}
-
-static int fsi_len2num(int len, int width)
-{
-	return len / width;
-}
-
-#define fsi_num2offset(a, b) fsi_num2len(a, b)
-static int fsi_num2len(int num, int width)
-{
-	return num * width;
-}
-
-static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play)
-{
-	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
-	struct snd_pcm_substream *substream = io->substream;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	return frames_to_bytes(runtime, 1) / fsi->chan_num;
+	return fsi_frame2sample(fsi, frames);
 }
 
 static void fsi_count_fifo_err(struct fsi_priv *fsi)
@@ -444,8 +476,10 @@
 {
 	int is_play = fsi_stream_is_play(stream);
 	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+	struct snd_pcm_runtime *runtime = io->substream->runtime;
 
-	return io->substream->runtime->dma_area + io->buff_offset;
+	return runtime->dma_area +
+		samples_to_bytes(runtime, io->buff_sample_pos);
 }
 
 static void fsi_dma_soft_push16(struct fsi_priv *fsi, int num)
@@ -559,37 +593,94 @@
 /*
  *		clock function
  */
-#define fsi_module_init(m, d)	__fsi_module_clk_ctrl(m, d, 1)
-#define fsi_module_kill(m, d)	__fsi_module_clk_ctrl(m, d, 0)
-static void __fsi_module_clk_ctrl(struct fsi_master *master,
-				  struct device *dev,
-				  int enable)
-{
-	pm_runtime_get_sync(dev);
-
-	if (enable) {
-		/* enable only SR */
-		fsi_master_mask_set(master, SOFT_RST, FSISR, FSISR);
-		fsi_master_mask_set(master, SOFT_RST, PASR | PBSR, 0);
-	} else {
-		/* clear all registers */
-		fsi_master_mask_set(master, SOFT_RST, FSISR, 0);
-	}
-
-	pm_runtime_put_sync(dev);
-}
-
-#define fsi_port_start(f)	__fsi_port_clk_ctrl(f, 1)
-#define fsi_port_stop(f)	__fsi_port_clk_ctrl(f, 0)
-static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int enable)
+static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
+			      long rate, int enable)
 {
 	struct fsi_master *master = fsi_get_master(fsi);
-	u32 soft = fsi_is_port_a(fsi) ? PASR : PBSR;
-	u32 clk  = fsi_is_port_a(fsi) ? CRA  : CRB;
-	int is_master = fsi_is_clk_master(fsi);
+	set_rate_func set_rate = fsi_get_info_set_rate(master);
+	int fsi_ver = master->core->ver;
+	int ret;
 
-	fsi_master_mask_set(master, SOFT_RST, soft, (enable) ? soft : 0);
-	if (is_master)
+	ret = set_rate(dev, fsi_is_port_a(fsi), rate, enable);
+	if (ret < 0) /* error */
+		return ret;
+
+	if (!enable)
+		return 0;
+
+	if (ret > 0) {
+		u32 data = 0;
+
+		switch (ret & SH_FSI_ACKMD_MASK) {
+		default:
+			/* FALL THROUGH */
+		case SH_FSI_ACKMD_512:
+			data |= (0x0 << 12);
+			break;
+		case SH_FSI_ACKMD_256:
+			data |= (0x1 << 12);
+			break;
+		case SH_FSI_ACKMD_128:
+			data |= (0x2 << 12);
+			break;
+		case SH_FSI_ACKMD_64:
+			data |= (0x3 << 12);
+			break;
+		case SH_FSI_ACKMD_32:
+			if (fsi_ver < 2)
+				dev_err(dev, "unsupported ACKMD\n");
+			else
+				data |= (0x4 << 12);
+			break;
+		}
+
+		switch (ret & SH_FSI_BPFMD_MASK) {
+		default:
+			/* FALL THROUGH */
+		case SH_FSI_BPFMD_32:
+			data |= (0x0 << 8);
+			break;
+		case SH_FSI_BPFMD_64:
+			data |= (0x1 << 8);
+			break;
+		case SH_FSI_BPFMD_128:
+			data |= (0x2 << 8);
+			break;
+		case SH_FSI_BPFMD_256:
+			data |= (0x3 << 8);
+			break;
+		case SH_FSI_BPFMD_512:
+			data |= (0x4 << 8);
+			break;
+		case SH_FSI_BPFMD_16:
+			if (fsi_ver < 2)
+				dev_err(dev, "unsupported ACKMD\n");
+			else
+				data |= (0x7 << 8);
+			break;
+		}
+
+		fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
+		udelay(10);
+		ret = 0;
+	}
+
+	return ret;
+}
+
+#define fsi_port_start(f, i)	__fsi_port_clk_ctrl(f, i, 1)
+#define fsi_port_stop(f, i)	__fsi_port_clk_ctrl(f, i, 0)
+static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable)
+{
+	struct fsi_master *master = fsi_get_master(fsi);
+	u32 clk  = fsi_is_port_a(fsi) ? CRA  : CRB;
+
+	if (enable)
+		fsi_irq_enable(fsi, is_play);
+	else
+		fsi_irq_disable(fsi, is_play);
+
+	if (fsi_is_clk_master(fsi))
 		fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
 }
 
@@ -598,18 +689,19 @@
  */
 static void fsi_fifo_init(struct fsi_priv *fsi,
 			  int is_play,
-			  struct snd_soc_dai *dai)
+			  struct device *dev)
 {
 	struct fsi_master *master = fsi_get_master(fsi);
 	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
 	u32 shift, i;
+	int frame_capa;
 
 	/* get on-chip RAM capacity */
 	shift = fsi_master_read(master, FIFO_SZ);
 	shift >>= fsi_get_port_shift(fsi, is_play);
 	shift &= FIFO_SZ_MASK;
-	io->fifo_max_num = 256 << shift;
-	dev_dbg(dai->dev, "fifo = %d words\n", io->fifo_max_num);
+	frame_capa = 256 << shift;
+	dev_dbg(dev, "fifo = %d words\n", frame_capa);
 
 	/*
 	 * The maximum number of sample data varies depending
@@ -631,9 +723,11 @@
 	 * 8 channels:  32 ( 32 x 8 = 256)
 	 */
 	for (i = 1; i < fsi->chan_num; i <<= 1)
-		io->fifo_max_num >>= 1;
-	dev_dbg(dai->dev, "%d channel %d store\n",
-		fsi->chan_num, io->fifo_max_num);
+		frame_capa >>= 1;
+	dev_dbg(dev, "%d channel %d store\n",
+		fsi->chan_num, frame_capa);
+
+	io->fifo_sample_capa = fsi_frame2sample(fsi, frame_capa);
 
 	/*
 	 * set interrupt generation factor
@@ -654,10 +748,10 @@
 	struct snd_pcm_substream *substream = NULL;
 	int is_play = fsi_stream_is_play(stream);
 	struct fsi_stream *io = fsi_get_stream(fsi, is_play);
-	int data_residue_num;
-	int data_num;
-	int data_num_max;
-	int ch_width;
+	int sample_residues;
+	int sample_width;
+	int samples;
+	int samples_max;
 	int over_period;
 	void (*fn)(struct fsi_priv *fsi, int size);
 
@@ -673,36 +767,35 @@
 	/* FSI FIFO has limit.
 	 * So, this driver can not send periods data at a time
 	 */
-	if (io->buff_offset >=
-	    fsi_num2offset(io->period_num + 1, io->period_len)) {
+	if (io->buff_sample_pos >=
+	    io->period_samples * (io->period_pos + 1)) {
 
 		over_period = 1;
-		io->period_num = (io->period_num + 1) % runtime->periods;
+		io->period_pos = (io->period_pos + 1) % runtime->periods;
 
-		if (0 == io->period_num)
-			io->buff_offset = 0;
+		if (0 == io->period_pos)
+			io->buff_sample_pos = 0;
 	}
 
-	/* get 1 channel data width */
-	ch_width = fsi_get_frame_width(fsi, is_play);
+	/* get 1 sample data width */
+	sample_width = samples_to_bytes(runtime, 1);
 
-	/* get residue data number of alsa */
-	data_residue_num = fsi_len2num(io->buff_len - io->buff_offset,
-				       ch_width);
+	/* get number of residue samples */
+	sample_residues = io->buff_sample_capa - io->buff_sample_pos;
 
 	if (is_play) {
 		/*
 		 * for play-back
 		 *
-		 * data_num_max	: number of FSI fifo free space
-		 * data_num	: number of ALSA residue data
+		 * samples_max	: number of FSI fifo free samples space
+		 * samples	: number of ALSA residue samples
 		 */
-		data_num_max  = io->fifo_max_num * fsi->chan_num;
-		data_num_max -= fsi_get_fifo_data_num(fsi, is_play);
+		samples_max  = io->fifo_sample_capa;
+		samples_max -= fsi_get_current_fifo_samples(fsi, is_play);
 
-		data_num = data_residue_num;
+		samples = sample_residues;
 
-		switch (ch_width) {
+		switch (sample_width) {
 		case 2:
 			fn = fsi_dma_soft_push16;
 			break;
@@ -716,13 +809,13 @@
 		/*
 		 * for capture
 		 *
-		 * data_num_max	: number of ALSA free space
-		 * data_num	: number of data in FSI fifo
+		 * samples_max	: number of ALSA free samples space
+		 * samples	: number of samples in FSI fifo
 		 */
-		data_num_max = data_residue_num;
-		data_num     = fsi_get_fifo_data_num(fsi, is_play);
+		samples_max = sample_residues;
+		samples     = fsi_get_current_fifo_samples(fsi, is_play);
 
-		switch (ch_width) {
+		switch (sample_width) {
 		case 2:
 			fn = fsi_dma_soft_pop16;
 			break;
@@ -734,12 +827,12 @@
 		}
 	}
 
-	data_num = min(data_num, data_num_max);
+	samples = min(samples, samples_max);
 
-	fn(fsi, data_num);
+	fn(fsi, samples);
 
-	/* update buff_offset */
-	io->buff_offset += fsi_num2offset(data_num, ch_width);
+	/* update buff_sample_pos */
+	io->buff_sample_pos += samples;
 
 	if (over_period)
 		snd_pcm_period_elapsed(substream);
@@ -788,16 +881,20 @@
  *		dai ops
  */
 
-static int fsi_dai_startup(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
+static int fsi_hw_startup(struct fsi_priv *fsi,
+			  int is_play,
+			  struct device *dev)
 {
-	struct fsi_priv *fsi = fsi_get_priv(substream);
 	u32 flags = fsi_get_info_flags(fsi);
-	u32 data;
-	int is_play = fsi_is_play(substream);
+	u32 data = 0;
 
-	pm_runtime_get_sync(dai->dev);
+	pm_runtime_get_sync(dev);
 
+	/* clock setting */
+	if (fsi_is_clk_master(fsi))
+		data = DIMD | DOMD;
+
+	fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
 
 	/* clock inversion (CKG2) */
 	data = 0;
@@ -812,54 +909,70 @@
 
 	fsi_reg_write(fsi, CKG2, data);
 
+	/* set format */
+	fsi_reg_write(fsi, DO_FMT, fsi->do_fmt);
+	fsi_reg_write(fsi, DI_FMT, fsi->di_fmt);
+
+	/* spdif ? */
+	if (fsi_is_spdif(fsi)) {
+		fsi_spdif_clk_ctrl(fsi, 1);
+		fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+	}
+
 	/* irq clear */
 	fsi_irq_disable(fsi, is_play);
 	fsi_irq_clear_status(fsi);
 
 	/* fifo init */
-	fsi_fifo_init(fsi, is_play, dai);
+	fsi_fifo_init(fsi, is_play, dev);
 
 	return 0;
 }
 
+static void fsi_hw_shutdown(struct fsi_priv *fsi,
+			    int is_play,
+			    struct device *dev)
+{
+	if (fsi_is_clk_master(fsi))
+		fsi_set_master_clk(dev, fsi, fsi->rate, 0);
+
+	pm_runtime_put_sync(dev);
+}
+
+static int fsi_dai_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct fsi_priv *fsi = fsi_get_priv(substream);
+	int is_play = fsi_is_play(substream);
+
+	return fsi_hw_startup(fsi, is_play, dai->dev);
+}
+
 static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	int is_play = fsi_is_play(substream);
-	struct fsi_master *master = fsi_get_master(fsi);
-	set_rate_func set_rate = fsi_get_info_set_rate(master);
 
-	fsi_irq_disable(fsi, is_play);
-
-	if (fsi_is_clk_master(fsi))
-		set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
-
+	fsi_hw_shutdown(fsi, is_play, dai->dev);
 	fsi->rate = 0;
-
-	pm_runtime_put_sync(dai->dev);
 }
 
 static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 			   struct snd_soc_dai *dai)
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int is_play = fsi_is_play(substream);
 	int ret = 0;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		fsi_stream_push(fsi, is_play, substream,
-				frames_to_bytes(runtime, runtime->buffer_size),
-				frames_to_bytes(runtime, runtime->period_size));
+		fsi_stream_push(fsi, is_play, substream);
 		ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi);
-		fsi_irq_enable(fsi, is_play);
-		fsi_port_start(fsi);
+		fsi_port_start(fsi, is_play);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
-		fsi_port_stop(fsi);
-		fsi_irq_disable(fsi, is_play);
+		fsi_port_stop(fsi, is_play);
 		fsi_stream_pop(fsi, is_play);
 		break;
 	}
@@ -884,8 +997,8 @@
 		return -EINVAL;
 	}
 
-	fsi_reg_write(fsi, DO_FMT, data);
-	fsi_reg_write(fsi, DI_FMT, data);
+	fsi->do_fmt = data;
+	fsi->di_fmt = data;
 
 	return 0;
 }
@@ -900,11 +1013,10 @@
 
 	data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
 	fsi->chan_num = 2;
-	fsi_spdif_clk_ctrl(fsi, 1);
-	fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+	fsi->spdif = 1;
 
-	fsi_reg_write(fsi, DO_FMT, data);
-	fsi_reg_write(fsi, DI_FMT, data);
+	fsi->do_fmt = data;
+	fsi->di_fmt = data;
 
 	return 0;
 }
@@ -915,32 +1027,24 @@
 	struct fsi_master *master = fsi_get_master(fsi);
 	set_rate_func set_rate = fsi_get_info_set_rate(master);
 	u32 flags = fsi_get_info_flags(fsi);
-	u32 data = 0;
 	int ret;
 
-	pm_runtime_get_sync(dai->dev);
-
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
-		data = DIMD | DOMD;
 		fsi->clk_master = 1;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
 		break;
 	default:
-		ret = -EINVAL;
-		goto set_fmt_exit;
+		return -EINVAL;
 	}
 
 	if (fsi_is_clk_master(fsi) && !set_rate) {
 		dev_err(dai->dev, "platform doesn't have set_rate\n");
-		ret = -EINVAL;
-		goto set_fmt_exit;
+		return -EINVAL;
 	}
 
-	fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
-
 	/* set format */
 	switch (flags & SH_FSI_FMT_MASK) {
 	case SH_FSI_FMT_DAI:
@@ -953,9 +1057,6 @@
 		ret = -EINVAL;
 	}
 
-set_fmt_exit:
-	pm_runtime_put_sync(dai->dev);
-
 	return ret;
 }
 
@@ -964,79 +1065,19 @@
 			     struct snd_soc_dai *dai)
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
-	struct fsi_master *master = fsi_get_master(fsi);
-	set_rate_func set_rate = fsi_get_info_set_rate(master);
-	int fsi_ver = master->core->ver;
 	long rate = params_rate(params);
 	int ret;
 
 	if (!fsi_is_clk_master(fsi))
 		return 0;
 
-	ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
-	if (ret < 0) /* error */
+	ret = fsi_set_master_clk(dai->dev, fsi, rate, 1);
+	if (ret < 0)
 		return ret;
 
 	fsi->rate = rate;
-	if (ret > 0) {
-		u32 data = 0;
-
-		switch (ret & SH_FSI_ACKMD_MASK) {
-		default:
-			/* FALL THROUGH */
-		case SH_FSI_ACKMD_512:
-			data |= (0x0 << 12);
-			break;
-		case SH_FSI_ACKMD_256:
-			data |= (0x1 << 12);
-			break;
-		case SH_FSI_ACKMD_128:
-			data |= (0x2 << 12);
-			break;
-		case SH_FSI_ACKMD_64:
-			data |= (0x3 << 12);
-			break;
-		case SH_FSI_ACKMD_32:
-			if (fsi_ver < 2)
-				dev_err(dai->dev, "unsupported ACKMD\n");
-			else
-				data |= (0x4 << 12);
-			break;
-		}
-
-		switch (ret & SH_FSI_BPFMD_MASK) {
-		default:
-			/* FALL THROUGH */
-		case SH_FSI_BPFMD_32:
-			data |= (0x0 << 8);
-			break;
-		case SH_FSI_BPFMD_64:
-			data |= (0x1 << 8);
-			break;
-		case SH_FSI_BPFMD_128:
-			data |= (0x2 << 8);
-			break;
-		case SH_FSI_BPFMD_256:
-			data |= (0x3 << 8);
-			break;
-		case SH_FSI_BPFMD_512:
-			data |= (0x4 << 8);
-			break;
-		case SH_FSI_BPFMD_16:
-			if (fsi_ver < 2)
-				dev_err(dai->dev, "unsupported ACKMD\n");
-			else
-				data |= (0x7 << 8);
-			break;
-		}
-
-		fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
-		udelay(10);
-		ret = 0;
-	}
 
 	return ret;
-
 }
 
 static struct snd_soc_dai_ops fsi_dai_ops = {
@@ -1097,16 +1138,14 @@
 
 static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
-	long location;
+	int samples_pos = io->buff_sample_pos - 1;
 
-	location = (io->buff_offset - 1);
-	if (location < 0)
-		location = 0;
+	if (samples_pos < 0)
+		samples_pos = 0;
 
-	return bytes_to_frames(runtime, location);
+	return fsi_sample2frame(fsi, samples_pos);
 }
 
 static struct snd_pcm_ops fsi_pcm_ops = {
@@ -1129,10 +1168,10 @@
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-static int fsi_pcm_new(struct snd_card *card,
-		       struct snd_soc_dai *dai,
-		       struct snd_pcm *pcm)
+static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_pcm *pcm = rtd->pcm;
+
 	/*
 	 * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
 	 * in MMAP mode (i.e. aplay -M)
@@ -1246,8 +1285,6 @@
 	pm_runtime_enable(&pdev->dev);
 	dev_set_drvdata(&pdev->dev, master);
 
-	fsi_module_init(master, &pdev->dev);
-
 	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
 			  id_entry->name, master);
 	if (ret) {
@@ -1290,8 +1327,6 @@
 
 	master = dev_get_drvdata(&pdev->dev);
 
-	fsi_module_kill(master, &pdev->dev);
-
 	free_irq(master->irq, master);
 	pm_runtime_disable(&pdev->dev);
 
@@ -1305,53 +1340,43 @@
 }
 
 static void __fsi_suspend(struct fsi_priv *fsi,
-			  struct device *dev,
-			  set_rate_func set_rate)
+			  int is_play,
+			  struct device *dev)
 {
-	fsi->saved_do_fmt	= fsi_reg_read(fsi, DO_FMT);
-	fsi->saved_di_fmt	= fsi_reg_read(fsi, DI_FMT);
-	fsi->saved_ckg1		= fsi_reg_read(fsi, CKG1);
-	fsi->saved_ckg2		= fsi_reg_read(fsi, CKG2);
-	fsi->saved_out_sel	= fsi_reg_read(fsi, OUT_SEL);
+	if (!fsi_stream_is_working(fsi, is_play))
+		return;
 
-	if (fsi_is_clk_master(fsi))
-		set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 0);
+	fsi_port_stop(fsi, is_play);
+	fsi_hw_shutdown(fsi, is_play, dev);
 }
 
 static void __fsi_resume(struct fsi_priv *fsi,
-			 struct device *dev,
-			 set_rate_func set_rate)
+			 int is_play,
+			 struct device *dev)
 {
-	fsi_reg_write(fsi, DO_FMT,	fsi->saved_do_fmt);
-	fsi_reg_write(fsi, DI_FMT,	fsi->saved_di_fmt);
-	fsi_reg_write(fsi, CKG1,	fsi->saved_ckg1);
-	fsi_reg_write(fsi, CKG2,	fsi->saved_ckg2);
-	fsi_reg_write(fsi, OUT_SEL,	fsi->saved_out_sel);
+	if (!fsi_stream_is_working(fsi, is_play))
+		return;
 
-	if (fsi_is_clk_master(fsi))
-		set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 1);
+	fsi_hw_startup(fsi, is_play, dev);
+
+	if (fsi_is_clk_master(fsi) && fsi->rate)
+		fsi_set_master_clk(dev, fsi, fsi->rate, 1);
+
+	fsi_port_start(fsi, is_play);
+
 }
 
 static int fsi_suspend(struct device *dev)
 {
 	struct fsi_master *master = dev_get_drvdata(dev);
-	set_rate_func set_rate = fsi_get_info_set_rate(master);
+	struct fsi_priv *fsia = &master->fsia;
+	struct fsi_priv *fsib = &master->fsib;
 
-	pm_runtime_get_sync(dev);
+	__fsi_suspend(fsia, 1, dev);
+	__fsi_suspend(fsia, 0, dev);
 
-	__fsi_suspend(&master->fsia, dev, set_rate);
-	__fsi_suspend(&master->fsib, dev, set_rate);
-
-	master->saved_a_mclk	= fsi_core_read(master, a_mclk);
-	master->saved_b_mclk	= fsi_core_read(master, b_mclk);
-	master->saved_iemsk	= fsi_core_read(master, iemsk);
-	master->saved_imsk	= fsi_core_read(master, imsk);
-	master->saved_clk_rst	= fsi_master_read(master, CLK_RST);
-	master->saved_soft_rst	= fsi_master_read(master, SOFT_RST);
-
-	fsi_module_kill(master, dev);
-
-	pm_runtime_put_sync(dev);
+	__fsi_suspend(fsib, 1, dev);
+	__fsi_suspend(fsib, 0, dev);
 
 	return 0;
 }
@@ -1359,23 +1384,14 @@
 static int fsi_resume(struct device *dev)
 {
 	struct fsi_master *master = dev_get_drvdata(dev);
-	set_rate_func set_rate = fsi_get_info_set_rate(master);
+	struct fsi_priv *fsia = &master->fsia;
+	struct fsi_priv *fsib = &master->fsib;
 
-	pm_runtime_get_sync(dev);
+	__fsi_resume(fsia, 1, dev);
+	__fsi_resume(fsia, 0, dev);
 
-	fsi_module_init(master, dev);
-
-	fsi_master_mask_set(master, SOFT_RST, 0xffff, master->saved_soft_rst);
-	fsi_master_mask_set(master, CLK_RST, 0xffff, master->saved_clk_rst);
-	fsi_core_mask_set(master, a_mclk, 0xffff, master->saved_a_mclk);
-	fsi_core_mask_set(master, b_mclk, 0xffff, master->saved_b_mclk);
-	fsi_core_mask_set(master, iemsk, 0xffff, master->saved_iemsk);
-	fsi_core_mask_set(master, imsk, 0xffff, master->saved_imsk);
-
-	__fsi_resume(&master->fsia, dev, set_rate);
-	__fsi_resume(&master->fsib, dev, set_rate);
-
-	pm_runtime_put_sync(dev);
+	__fsi_resume(fsib, 1, dev);
+	__fsi_resume(fsib, 0, dev);
 
 	return 0;
 }
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index a423bab..f8f6816 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -527,10 +527,11 @@
 	return bytes_to_frames(ss->runtime, ptr);
 }
 
-static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-		       struct snd_pcm *pcm)
+static int siu_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	/* card->dev == socdev->dev, see snd_soc_new_pcms() */
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
 	struct siu_info *info = siu_i2s_data;
 	struct platform_device *pdev = to_platform_device(card->dev);
 	int ret;
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 039b953..d9f8ade 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -20,422 +20,6 @@
 
 #include <trace/events/asoc.h>
 
-#ifdef CONFIG_SPI_MASTER
-static int do_spi_write(void *control, const char *data, int len)
-{
-	struct spi_device *spi = control;
-	int ret;
-
-	ret = spi_write(spi, data, len);
-	if (ret < 0)
-		return ret;
-
-	return len;
-}
-#endif
-
-static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
-		       unsigned int value, const void *data, int len)
-{
-	int ret;
-
-	if (!snd_soc_codec_volatile_register(codec, reg) &&
-	    reg < codec->driver->reg_cache_size &&
-	    !codec->cache_bypass) {
-		ret = snd_soc_cache_write(codec, reg, value);
-		if (ret < 0)
-			return -1;
-	}
-
-	if (codec->cache_only) {
-		codec->cache_sync = 1;
-		return 0;
-	}
-
-	ret = codec->hw_write(codec->control_data, data, len);
-	if (ret == len)
-		return 0;
-	if (ret < 0)
-		return ret;
-	else
-		return -EIO;
-}
-
-static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg)
-{
-	int ret;
-	unsigned int val;
-
-	if (reg >= codec->driver->reg_cache_size ||
-	    snd_soc_codec_volatile_register(codec, reg) ||
-	    codec->cache_bypass) {
-		if (codec->cache_only)
-			return -1;
-
-		BUG_ON(!codec->hw_read);
-		return codec->hw_read(codec, reg);
-	}
-
-	ret = snd_soc_cache_read(codec, reg, &val);
-	if (ret < 0)
-		return -1;
-	return val;
-}
-
-static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	return do_hw_read(codec, reg);
-}
-
-static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	u16 data;
-
-	data = cpu_to_be16((reg << 12) | (value & 0xffffff));
-
-	return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
-				     unsigned int reg)
-{
-	return do_hw_read(codec, reg);
-}
-
-static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
-			     unsigned int value)
-{
-	u8 data[2];
-
-	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-	data[1] = value & 0x00ff;
-
-	return do_hw_write(codec, reg, value, data, 2);
-}
-
-static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
-			     unsigned int value)
-{
-	u8 data[2];
-
-	reg &= 0xff;
-	data[0] = reg;
-	data[1] = value & 0xff;
-
-	return do_hw_write(codec, reg, value, data, 2);
-}
-
-static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
-				     unsigned int reg)
-{
-	return do_hw_read(codec, reg);
-}
-
-static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	u8 data[3];
-
-	data[0] = reg;
-	data[1] = (value >> 8) & 0xff;
-	data[2] = value & 0xff;
-
-	return do_hw_write(codec, reg, value, data, 3);
-}
-
-static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	return do_hw_read(codec, reg);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int do_i2c_read(struct snd_soc_codec *codec,
-				void *reg, int reglen,
-				void *data, int datalen)
-{
-	struct i2c_msg xfer[2];
-	int ret;
-	struct i2c_client *client = codec->control_data;
-
-	/* Write register */
-	xfer[0].addr = client->addr;
-	xfer[0].flags = 0;
-	xfer[0].len = reglen;
-	xfer[0].buf = reg;
-
-	/* Read data */
-	xfer[1].addr = client->addr;
-	xfer[1].flags = I2C_M_RD;
-	xfer[1].len = datalen;
-	xfer[1].buf = data;
-
-	ret = i2c_transfer(client->adapter, xfer, 2);
-	if (ret == 2)
-		return 0;
-	else if (ret < 0)
-		return ret;
-	else
-		return -EIO;
-}
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
-					 unsigned int r)
-{
-	u8 reg = r;
-	u8 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 1, &data, 1);
-	if (ret < 0)
-		return 0;
-	return data;
-}
-#else
-#define snd_soc_8_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
-					  unsigned int r)
-{
-	u8 reg = r;
-	u16 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 1, &data, 2);
-	if (ret < 0)
-		return 0;
-	return (data >> 8) | ((data & 0xff) << 8);
-}
-#else
-#define snd_soc_8_16_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
-					  unsigned int r)
-{
-	u16 reg = r;
-	u8 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 2, &data, 1);
-	if (ret < 0)
-		return 0;
-	return data;
-}
-#else
-#define snd_soc_16_8_read_i2c NULL
-#endif
-
-static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	return do_hw_read(codec, reg);
-}
-
-static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	u8 data[3];
-
-	data[0] = (reg >> 8) & 0xff;
-	data[1] = reg & 0xff;
-	data[2] = value;
-
-	return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
-					   unsigned int r)
-{
-	u16 reg = cpu_to_be16(r);
-	u16 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 2, &data, 2);
-	if (ret < 0)
-		return 0;
-	return be16_to_cpu(data);
-}
-#else
-#define snd_soc_16_16_read_i2c NULL
-#endif
-
-static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec,
-				       unsigned int reg)
-{
-	return do_hw_read(codec, reg);
-}
-
-static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
-			       unsigned int value)
-{
-	u8 data[4];
-
-	data[0] = (reg >> 8) & 0xff;
-	data[1] = reg & 0xff;
-	data[2] = (value >> 8) & 0xff;
-	data[3] = value & 0xff;
-
-	return do_hw_write(codec, reg, value, data, 4);
-}
-
-/* Primitive bulk write support for soc-cache.  The data pointed to by
- * `data' needs to already be in the form the hardware expects
- * including any leading register specific data.  Any data written
- * through this function will not go through the cache as it only
- * handles writing to volatile or out of bounds registers.
- */
-static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
-				     const void *data, size_t len)
-{
-	int ret;
-
-	/* To ensure that we don't get out of sync with the cache, check
-	 * whether the base register is volatile or if we've directly asked
-	 * to bypass the cache.  Out of bounds registers are considered
-	 * volatile.
-	 */
-	if (!codec->cache_bypass
-	    && !snd_soc_codec_volatile_register(codec, reg)
-	    && reg < codec->driver->reg_cache_size)
-		return -EINVAL;
-
-	switch (codec->control_type) {
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-	case SND_SOC_I2C:
-		ret = i2c_master_send(codec->control_data, data, len);
-		break;
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	case SND_SOC_SPI:
-		ret = spi_write(codec->control_data, data, len);
-		break;
-#endif
-	default:
-		BUG();
-	}
-
-	if (ret == len)
-		return 0;
-	if (ret < 0)
-		return ret;
-	else
-		return -EIO;
-}
-
-static struct {
-	int addr_bits;
-	int data_bits;
-	int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
-	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
-	unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
-} io_types[] = {
-	{
-		.addr_bits = 4, .data_bits = 12,
-		.write = snd_soc_4_12_write, .read = snd_soc_4_12_read,
-	},
-	{
-		.addr_bits = 7, .data_bits = 9,
-		.write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
-	},
-	{
-		.addr_bits = 8, .data_bits = 8,
-		.write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
-		.i2c_read = snd_soc_8_8_read_i2c,
-	},
-	{
-		.addr_bits = 8, .data_bits = 16,
-		.write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
-		.i2c_read = snd_soc_8_16_read_i2c,
-	},
-	{
-		.addr_bits = 16, .data_bits = 8,
-		.write = snd_soc_16_8_write, .read = snd_soc_16_8_read,
-		.i2c_read = snd_soc_16_8_read_i2c,
-	},
-	{
-		.addr_bits = 16, .data_bits = 16,
-		.write = snd_soc_16_16_write, .read = snd_soc_16_16_read,
-		.i2c_read = snd_soc_16_16_read_i2c,
-	},
-};
-
-/**
- * snd_soc_codec_set_cache_io: Set up standard I/O functions.
- *
- * @codec: CODEC to configure.
- * @addr_bits: Number of bits of register address data.
- * @data_bits: Number of bits of data per register.
- * @control: Control bus used.
- *
- * Register formats are frequently shared between many I2C and SPI
- * devices.  In order to promote code reuse the ASoC core provides
- * some standard implementations of CODEC read and write operations
- * which can be set up using this function.
- *
- * The caller is responsible for allocating and initialising the
- * actual cache.
- *
- * Note that at present this code cannot be used by CODECs with
- * volatile registers.
- */
-int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
-			       int addr_bits, int data_bits,
-			       enum snd_soc_control_type control)
-{
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(io_types); i++)
-		if (io_types[i].addr_bits == addr_bits &&
-		    io_types[i].data_bits == data_bits)
-			break;
-	if (i == ARRAY_SIZE(io_types)) {
-		printk(KERN_ERR
-		       "No I/O functions for %d bit address %d bit data\n",
-		       addr_bits, data_bits);
-		return -EINVAL;
-	}
-
-	codec->write = io_types[i].write;
-	codec->read = io_types[i].read;
-	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
-
-	switch (control) {
-	case SND_SOC_I2C:
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-		codec->hw_write = (hw_write_t)i2c_master_send;
-#endif
-		if (io_types[i].i2c_read)
-			codec->hw_read = io_types[i].i2c_read;
-
-		codec->control_data = container_of(codec->dev,
-						   struct i2c_client,
-						   dev);
-		break;
-
-	case SND_SOC_SPI:
-#ifdef CONFIG_SPI_MASTER
-		codec->hw_write = do_spi_write;
-#endif
-
-		codec->control_data = container_of(codec->dev,
-						   struct spi_device,
-						   dev);
-		break;
-	}
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
-
 static bool snd_soc_set_cache_val(void *base, unsigned int idx,
 				  unsigned int val, unsigned int word_size)
 {
@@ -483,31 +67,86 @@
 }
 
 struct snd_soc_rbtree_node {
-	struct rb_node node;
-	unsigned int reg;
-	unsigned int value;
-	unsigned int defval;
+	struct rb_node node; /* the actual rbtree node holding this block */
+	unsigned int base_reg; /* base register handled by this block */
+	unsigned int word_size; /* number of bytes needed to represent the register index */
+	void *block; /* block of adjacent registers */
+	unsigned int blklen; /* number of registers available in the block */
 } __attribute__ ((packed));
 
 struct snd_soc_rbtree_ctx {
 	struct rb_root root;
+	struct snd_soc_rbtree_node *cached_rbnode;
 };
 
+static inline void snd_soc_rbtree_get_base_top_reg(
+	struct snd_soc_rbtree_node *rbnode,
+	unsigned int *base, unsigned int *top)
+{
+	*base = rbnode->base_reg;
+	*top = rbnode->base_reg + rbnode->blklen - 1;
+}
+
+static unsigned int snd_soc_rbtree_get_register(
+	struct snd_soc_rbtree_node *rbnode, unsigned int idx)
+{
+	unsigned int val;
+
+	switch (rbnode->word_size) {
+	case 1: {
+		u8 *p = rbnode->block;
+		val = p[idx];
+		return val;
+	}
+	case 2: {
+		u16 *p = rbnode->block;
+		val = p[idx];
+		return val;
+	}
+	default:
+		BUG();
+		break;
+	}
+	return -1;
+}
+
+static void snd_soc_rbtree_set_register(struct snd_soc_rbtree_node *rbnode,
+					unsigned int idx, unsigned int val)
+{
+	switch (rbnode->word_size) {
+	case 1: {
+		u8 *p = rbnode->block;
+		p[idx] = val;
+		break;
+	}
+	case 2: {
+		u16 *p = rbnode->block;
+		p[idx] = val;
+		break;
+	}
+	default:
+		BUG();
+		break;
+	}
+}
+
 static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup(
 	struct rb_root *root, unsigned int reg)
 {
 	struct rb_node *node;
 	struct snd_soc_rbtree_node *rbnode;
+	unsigned int base_reg, top_reg;
 
 	node = root->rb_node;
 	while (node) {
 		rbnode = container_of(node, struct snd_soc_rbtree_node, node);
-		if (rbnode->reg < reg)
-			node = node->rb_left;
-		else if (rbnode->reg > reg)
-			node = node->rb_right;
-		else
+		snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+		if (reg >= base_reg && reg <= top_reg)
 			return rbnode;
+		else if (reg > top_reg)
+			node = node->rb_right;
+		else if (reg < base_reg)
+			node = node->rb_left;
 	}
 
 	return NULL;
@@ -518,19 +157,28 @@
 {
 	struct rb_node **new, *parent;
 	struct snd_soc_rbtree_node *rbnode_tmp;
+	unsigned int base_reg_tmp, top_reg_tmp;
+	unsigned int base_reg;
 
 	parent = NULL;
 	new = &root->rb_node;
 	while (*new) {
 		rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node,
 					  node);
+		/* base and top registers of the current rbnode */
+		snd_soc_rbtree_get_base_top_reg(rbnode_tmp, &base_reg_tmp,
+						&top_reg_tmp);
+		/* base register of the rbnode to be added */
+		base_reg = rbnode->base_reg;
 		parent = *new;
-		if (rbnode_tmp->reg < rbnode->reg)
-			new = &((*new)->rb_left);
-		else if (rbnode_tmp->reg > rbnode->reg)
-			new = &((*new)->rb_right);
-		else
+		/* if this register has already been inserted, just return */
+		if (base_reg >= base_reg_tmp &&
+		    base_reg <= top_reg_tmp)
 			return 0;
+		else if (base_reg > top_reg_tmp)
+			new = &((*new)->rb_right);
+		else if (base_reg < base_reg_tmp)
+			new = &((*new)->rb_left);
 	}
 
 	/* insert the node into the rbtree */
@@ -545,58 +193,146 @@
 	struct snd_soc_rbtree_ctx *rbtree_ctx;
 	struct rb_node *node;
 	struct snd_soc_rbtree_node *rbnode;
-	unsigned int val;
+	unsigned int regtmp;
+	unsigned int val, def;
 	int ret;
+	int i;
 
 	rbtree_ctx = codec->reg_cache;
 	for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) {
 		rbnode = rb_entry(node, struct snd_soc_rbtree_node, node);
-		if (rbnode->value == rbnode->defval)
-			continue;
-		WARN_ON(codec->writable_register &&
-			codec->writable_register(codec, rbnode->reg));
-		ret = snd_soc_cache_read(codec, rbnode->reg, &val);
-		if (ret)
-			return ret;
-		codec->cache_bypass = 1;
-		ret = snd_soc_write(codec, rbnode->reg, val);
-		codec->cache_bypass = 0;
-		if (ret)
-			return ret;
-		dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
-			rbnode->reg, val);
+		for (i = 0; i < rbnode->blklen; ++i) {
+			regtmp = rbnode->base_reg + i;
+			WARN_ON(codec->writable_register &&
+				codec->writable_register(codec, regtmp));
+			val = snd_soc_rbtree_get_register(rbnode, i);
+			def = snd_soc_get_cache_val(codec->reg_def_copy, i,
+						    rbnode->word_size);
+			if (val == def)
+				continue;
+
+			codec->cache_bypass = 1;
+			ret = snd_soc_write(codec, regtmp, val);
+			codec->cache_bypass = 0;
+			if (ret)
+				return ret;
+			dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
+				regtmp, val);
+		}
 	}
 
 	return 0;
 }
 
+static int snd_soc_rbtree_insert_to_block(struct snd_soc_rbtree_node *rbnode,
+					  unsigned int pos, unsigned int reg,
+					  unsigned int value)
+{
+	u8 *blk;
+
+	blk = krealloc(rbnode->block,
+		       (rbnode->blklen + 1) * rbnode->word_size, GFP_KERNEL);
+	if (!blk)
+		return -ENOMEM;
+
+	/* insert the register value in the correct place in the rbnode block */
+	memmove(blk + (pos + 1) * rbnode->word_size,
+		blk + pos * rbnode->word_size,
+		(rbnode->blklen - pos) * rbnode->word_size);
+
+	/* update the rbnode block, its size and the base register */
+	rbnode->block = blk;
+	rbnode->blklen++;
+	if (!pos)
+		rbnode->base_reg = reg;
+
+	snd_soc_rbtree_set_register(rbnode, pos, value);
+	return 0;
+}
+
 static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec,
 				      unsigned int reg, unsigned int value)
 {
 	struct snd_soc_rbtree_ctx *rbtree_ctx;
-	struct snd_soc_rbtree_node *rbnode;
+	struct snd_soc_rbtree_node *rbnode, *rbnode_tmp;
+	struct rb_node *node;
+	unsigned int val;
+	unsigned int reg_tmp;
+	unsigned int base_reg, top_reg;
+	unsigned int pos;
+	int i;
+	int ret;
 
 	rbtree_ctx = codec->reg_cache;
+	/* look up the required register in the cached rbnode */
+	rbnode = rbtree_ctx->cached_rbnode;
+	if (rbnode) {
+		snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+		if (reg >= base_reg && reg <= top_reg) {
+			reg_tmp = reg - base_reg;
+			val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+			if (val == value)
+				return 0;
+			snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
+			return 0;
+		}
+	}
+	/* if we can't locate it in the cached rbnode we'll have
+	 * to traverse the rbtree looking for it.
+	 */
 	rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
 	if (rbnode) {
-		if (rbnode->value == value)
+		reg_tmp = reg - rbnode->base_reg;
+		val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+		if (val == value)
 			return 0;
-		rbnode->value = value;
+		snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
+		rbtree_ctx->cached_rbnode = rbnode;
 	} else {
 		/* bail out early, no need to create the rbnode yet */
 		if (!value)
 			return 0;
-		/*
-		 * for uninitialized registers whose value is changed
-		 * from the default zero, create an rbnode and insert
-		 * it into the tree.
+		/* look for an adjacent register to the one we are about to add */
+		for (node = rb_first(&rbtree_ctx->root); node;
+		     node = rb_next(node)) {
+			rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node);
+			for (i = 0; i < rbnode_tmp->blklen; ++i) {
+				reg_tmp = rbnode_tmp->base_reg + i;
+				if (abs(reg_tmp - reg) != 1)
+					continue;
+				/* decide where in the block to place our register */
+				if (reg_tmp + 1 == reg)
+					pos = i + 1;
+				else
+					pos = i;
+				ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos,
+								     reg, value);
+				if (ret)
+					return ret;
+				rbtree_ctx->cached_rbnode = rbnode_tmp;
+				return 0;
+			}
+		}
+		/* we did not manage to find a place to insert it in an existing
+		 * block so create a new rbnode with a single register in its block.
+		 * This block will get populated further if any other adjacent
+		 * registers get modified in the future.
 		 */
 		rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL);
 		if (!rbnode)
 			return -ENOMEM;
-		rbnode->reg = reg;
-		rbnode->value = value;
+		rbnode->blklen = 1;
+		rbnode->base_reg = reg;
+		rbnode->word_size = codec->driver->reg_word_size;
+		rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size,
+					GFP_KERNEL);
+		if (!rbnode->block) {
+			kfree(rbnode);
+			return -ENOMEM;
+		}
+		snd_soc_rbtree_set_register(rbnode, 0, value);
 		snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode);
+		rbtree_ctx->cached_rbnode = rbnode;
 	}
 
 	return 0;
@@ -607,11 +343,28 @@
 {
 	struct snd_soc_rbtree_ctx *rbtree_ctx;
 	struct snd_soc_rbtree_node *rbnode;
+	unsigned int base_reg, top_reg;
+	unsigned int reg_tmp;
 
 	rbtree_ctx = codec->reg_cache;
+	/* look up the required register in the cached rbnode */
+	rbnode = rbtree_ctx->cached_rbnode;
+	if (rbnode) {
+		snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+		if (reg >= base_reg && reg <= top_reg) {
+			reg_tmp = reg - base_reg;
+			*value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+			return 0;
+		}
+	}
+	/* if we can't locate it in the cached rbnode we'll have
+	 * to traverse the rbtree looking for it.
+	 */
 	rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
 	if (rbnode) {
-		*value = rbnode->value;
+		reg_tmp = reg - rbnode->base_reg;
+		*value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+		rbtree_ctx->cached_rbnode = rbnode;
 	} else {
 		/* uninitialized registers default to 0 */
 		*value = 0;
@@ -637,6 +390,7 @@
 		rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node);
 		next = rb_next(&rbtree_node->node);
 		rb_erase(&rbtree_node->node, &rbtree_ctx->root);
+		kfree(rbtree_node->block);
 		kfree(rbtree_node);
 	}
 
@@ -649,10 +403,9 @@
 
 static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
 {
-	struct snd_soc_rbtree_node *rbtree_node;
 	struct snd_soc_rbtree_ctx *rbtree_ctx;
-	unsigned int val;
 	unsigned int word_size;
+	unsigned int val;
 	int i;
 	int ret;
 
@@ -662,32 +415,27 @@
 
 	rbtree_ctx = codec->reg_cache;
 	rbtree_ctx->root = RB_ROOT;
+	rbtree_ctx->cached_rbnode = NULL;
 
 	if (!codec->reg_def_copy)
 		return 0;
 
-	/*
-	 * populate the rbtree with the initialized registers.  All other
-	 * registers will be inserted when they are first modified.
-	 */
 	word_size = codec->driver->reg_word_size;
 	for (i = 0; i < codec->driver->reg_cache_size; ++i) {
-		val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size);
+		val = snd_soc_get_cache_val(codec->reg_def_copy, i,
+					    word_size);
 		if (!val)
 			continue;
-		rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL);
-		if (!rbtree_node) {
-			ret = -ENOMEM;
-			snd_soc_cache_exit(codec);
-			break;
-		}
-		rbtree_node->reg = i;
-		rbtree_node->value = val;
-		rbtree_node->defval = val;
-		snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node);
+		ret = snd_soc_rbtree_cache_write(codec, i, val);
+		if (ret)
+			goto err;
 	}
 
 	return 0;
+
+err:
+	snd_soc_cache_exit(codec);
+	return ret;
 }
 
 #ifdef CONFIG_SND_SOC_CACHE_LZO
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043e..70f4779 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -44,7 +44,6 @@
 
 #define NAME_SIZE	32
 
-static DEFINE_MUTEX(pcm_mutex);
 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
 
 #ifdef CONFIG_DEBUG_FS
@@ -58,7 +57,7 @@
 static LIST_HEAD(platform_list);
 static LIST_HEAD(codec_list);
 
-static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
 
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -485,552 +484,6 @@
 }
 #endif
 
-static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int ret;
-
-	if (!codec_dai->driver->symmetric_rates &&
-	    !cpu_dai->driver->symmetric_rates &&
-	    !rtd->dai_link->symmetric_rates)
-		return 0;
-
-	/* This can happen if multiple streams are starting simultaneously -
-	 * the second can need to get its constraints before the first has
-	 * picked a rate.  Complain and allow the application to carry on.
-	 */
-	if (!rtd->rate) {
-		dev_warn(&rtd->dev,
-			 "Not enforcing symmetric_rates due to race\n");
-		return 0;
-	}
-
-	dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
-
-	ret = snd_pcm_hw_constraint_minmax(substream->runtime,
-					   SNDRV_PCM_HW_PARAM_RATE,
-					   rtd->rate, rtd->rate);
-	if (ret < 0) {
-		dev_err(&rtd->dev,
-			"Unable to apply rate symmetry constraint: %d\n", ret);
-		return ret;
-	}
-
-	return 0;
-}
-
-/*
- * Called by ALSA when a PCM substream is opened, the runtime->hw record is
- * then initialized and any private data can be allocated. This also calls
- * startup for the cpu DAI, platform, machine and codec DAI.
- */
-static int soc_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
-	struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver;
-	int ret = 0;
-
-	mutex_lock(&pcm_mutex);
-
-	/* startup the audio subsystem */
-	if (cpu_dai->driver->ops->startup) {
-		ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: can't open interface %s\n",
-				cpu_dai->name);
-			goto out;
-		}
-	}
-
-	if (platform->driver->ops && platform->driver->ops->open) {
-		ret = platform->driver->ops->open(substream);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
-			goto platform_err;
-		}
-	}
-
-	if (codec_dai->driver->ops->startup) {
-		ret = codec_dai->driver->ops->startup(substream, codec_dai);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: can't open codec %s\n",
-				codec_dai->name);
-			goto codec_dai_err;
-		}
-	}
-
-	if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
-		ret = rtd->dai_link->ops->startup(substream);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name);
-			goto machine_err;
-		}
-	}
-
-	/* Check that the codec and cpu DAIs are compatible */
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		runtime->hw.rate_min =
-			max(codec_dai_drv->playback.rate_min,
-			    cpu_dai_drv->playback.rate_min);
-		runtime->hw.rate_max =
-			min(codec_dai_drv->playback.rate_max,
-			    cpu_dai_drv->playback.rate_max);
-		runtime->hw.channels_min =
-			max(codec_dai_drv->playback.channels_min,
-				cpu_dai_drv->playback.channels_min);
-		runtime->hw.channels_max =
-			min(codec_dai_drv->playback.channels_max,
-				cpu_dai_drv->playback.channels_max);
-		runtime->hw.formats =
-			codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats;
-		runtime->hw.rates =
-			codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates;
-		if (codec_dai_drv->playback.rates
-			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
-			runtime->hw.rates |= cpu_dai_drv->playback.rates;
-		if (cpu_dai_drv->playback.rates
-			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
-			runtime->hw.rates |= codec_dai_drv->playback.rates;
-	} else {
-		runtime->hw.rate_min =
-			max(codec_dai_drv->capture.rate_min,
-			    cpu_dai_drv->capture.rate_min);
-		runtime->hw.rate_max =
-			min(codec_dai_drv->capture.rate_max,
-			    cpu_dai_drv->capture.rate_max);
-		runtime->hw.channels_min =
-			max(codec_dai_drv->capture.channels_min,
-				cpu_dai_drv->capture.channels_min);
-		runtime->hw.channels_max =
-			min(codec_dai_drv->capture.channels_max,
-				cpu_dai_drv->capture.channels_max);
-		runtime->hw.formats =
-			codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats;
-		runtime->hw.rates =
-			codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates;
-		if (codec_dai_drv->capture.rates
-			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
-			runtime->hw.rates |= cpu_dai_drv->capture.rates;
-		if (cpu_dai_drv->capture.rates
-			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
-			runtime->hw.rates |= codec_dai_drv->capture.rates;
-	}
-
-	ret = -EINVAL;
-	snd_pcm_limit_hw_rates(runtime);
-	if (!runtime->hw.rates) {
-		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
-			codec_dai->name, cpu_dai->name);
-		goto config_err;
-	}
-	if (!runtime->hw.formats) {
-		printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
-			codec_dai->name, cpu_dai->name);
-		goto config_err;
-	}
-	if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
-	    runtime->hw.channels_min > runtime->hw.channels_max) {
-		printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
-				codec_dai->name, cpu_dai->name);
-		goto config_err;
-	}
-
-	/* Symmetry only applies if we've already got an active stream. */
-	if (cpu_dai->active || codec_dai->active) {
-		ret = soc_pcm_apply_symmetry(substream);
-		if (ret != 0)
-			goto config_err;
-	}
-
-	pr_debug("asoc: %s <-> %s info:\n",
-			codec_dai->name, cpu_dai->name);
-	pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
-	pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
-		 runtime->hw.channels_max);
-	pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
-		 runtime->hw.rate_max);
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		cpu_dai->playback_active++;
-		codec_dai->playback_active++;
-	} else {
-		cpu_dai->capture_active++;
-		codec_dai->capture_active++;
-	}
-	cpu_dai->active++;
-	codec_dai->active++;
-	rtd->codec->active++;
-	mutex_unlock(&pcm_mutex);
-	return 0;
-
-config_err:
-	if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
-		rtd->dai_link->ops->shutdown(substream);
-
-machine_err:
-	if (codec_dai->driver->ops->shutdown)
-		codec_dai->driver->ops->shutdown(substream, codec_dai);
-
-codec_dai_err:
-	if (platform->driver->ops && platform->driver->ops->close)
-		platform->driver->ops->close(substream);
-
-platform_err:
-	if (cpu_dai->driver->ops->shutdown)
-		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
-out:
-	mutex_unlock(&pcm_mutex);
-	return ret;
-}
-
-/*
- * Power down the audio subsystem pmdown_time msecs after close is called.
- * This is to ensure there are no pops or clicks in between any music tracks
- * due to DAPM power cycling.
- */
-static void close_delayed_work(struct work_struct *work)
-{
-	struct snd_soc_pcm_runtime *rtd =
-			container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
-	mutex_lock(&pcm_mutex);
-
-	pr_debug("pop wq checking: %s status: %s waiting: %s\n",
-		 codec_dai->driver->playback.stream_name,
-		 codec_dai->playback_active ? "active" : "inactive",
-		 codec_dai->pop_wait ? "yes" : "no");
-
-	/* are we waiting on this codec DAI stream */
-	if (codec_dai->pop_wait == 1) {
-		codec_dai->pop_wait = 0;
-		snd_soc_dapm_stream_event(rtd,
-			codec_dai->driver->playback.stream_name,
-			SND_SOC_DAPM_STREAM_STOP);
-	}
-
-	mutex_unlock(&pcm_mutex);
-}
-
-/*
- * Called by ALSA when a PCM substream is closed. Private data can be
- * freed here. The cpu DAI, codec DAI, machine and platform are also
- * shutdown.
- */
-static int soc_codec_close(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
-
-	mutex_lock(&pcm_mutex);
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		cpu_dai->playback_active--;
-		codec_dai->playback_active--;
-	} else {
-		cpu_dai->capture_active--;
-		codec_dai->capture_active--;
-	}
-
-	cpu_dai->active--;
-	codec_dai->active--;
-	codec->active--;
-
-	/* Muting the DAC suppresses artifacts caused during digital
-	 * shutdown, for example from stopping clocks.
-	 */
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		snd_soc_dai_digital_mute(codec_dai, 1);
-
-	if (cpu_dai->driver->ops->shutdown)
-		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
-
-	if (codec_dai->driver->ops->shutdown)
-		codec_dai->driver->ops->shutdown(substream, codec_dai);
-
-	if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
-		rtd->dai_link->ops->shutdown(substream);
-
-	if (platform->driver->ops && platform->driver->ops->close)
-		platform->driver->ops->close(substream);
-	cpu_dai->runtime = NULL;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		/* start delayed pop wq here for playback streams */
-		codec_dai->pop_wait = 1;
-		schedule_delayed_work(&rtd->delayed_work,
-			msecs_to_jiffies(rtd->pmdown_time));
-	} else {
-		/* capture streams can be powered down now */
-		snd_soc_dapm_stream_event(rtd,
-			codec_dai->driver->capture.stream_name,
-			SND_SOC_DAPM_STREAM_STOP);
-	}
-
-	mutex_unlock(&pcm_mutex);
-	return 0;
-}
-
-/*
- * Called by ALSA when the PCM substream is prepared, can set format, sample
- * rate, etc.  This function is non atomic and can be called multiple times,
- * it can refer to the runtime info.
- */
-static int soc_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int ret = 0;
-
-	mutex_lock(&pcm_mutex);
-
-	if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
-		ret = rtd->dai_link->ops->prepare(substream);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: machine prepare error\n");
-			goto out;
-		}
-	}
-
-	if (platform->driver->ops && platform->driver->ops->prepare) {
-		ret = platform->driver->ops->prepare(substream);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: platform prepare error\n");
-			goto out;
-		}
-	}
-
-	if (codec_dai->driver->ops->prepare) {
-		ret = codec_dai->driver->ops->prepare(substream, codec_dai);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: codec DAI prepare error\n");
-			goto out;
-		}
-	}
-
-	if (cpu_dai->driver->ops->prepare) {
-		ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
-			goto out;
-		}
-	}
-
-	/* cancel any delayed stream shutdown that is pending */
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
-	    codec_dai->pop_wait) {
-		codec_dai->pop_wait = 0;
-		cancel_delayed_work(&rtd->delayed_work);
-	}
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		snd_soc_dapm_stream_event(rtd,
-					  codec_dai->driver->playback.stream_name,
-					  SND_SOC_DAPM_STREAM_START);
-	else
-		snd_soc_dapm_stream_event(rtd,
-					  codec_dai->driver->capture.stream_name,
-					  SND_SOC_DAPM_STREAM_START);
-
-	snd_soc_dai_digital_mute(codec_dai, 0);
-
-out:
-	mutex_unlock(&pcm_mutex);
-	return ret;
-}
-
-/*
- * Called by ALSA when the hardware params are set by application. This
- * function can also be called multiple times and can allocate buffers
- * (using snd_pcm_lib_* ). It's non-atomic.
- */
-static int soc_pcm_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_platform *platform = rtd->platform;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int ret = 0;
-
-	mutex_lock(&pcm_mutex);
-
-	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
-		ret = rtd->dai_link->ops->hw_params(substream, params);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: machine hw_params failed\n");
-			goto out;
-		}
-	}
-
-	if (codec_dai->driver->ops->hw_params) {
-		ret = codec_dai->driver->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);
-			goto codec_err;
-		}
-	}
-
-	if (cpu_dai->driver->ops->hw_params) {
-		ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: interface %s hw params failed\n",
-				cpu_dai->name);
-			goto interface_err;
-		}
-	}
-
-	if (platform->driver->ops && platform->driver->ops->hw_params) {
-		ret = platform->driver->ops->hw_params(substream, params);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: platform %s hw params failed\n",
-				platform->name);
-			goto platform_err;
-		}
-	}
-
-	rtd->rate = params_rate(params);
-
-out:
-	mutex_unlock(&pcm_mutex);
-	return ret;
-
-platform_err:
-	if (cpu_dai->driver->ops->hw_free)
-		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
-
-interface_err:
-	if (codec_dai->driver->ops->hw_free)
-		codec_dai->driver->ops->hw_free(substream, codec_dai);
-
-codec_err:
-	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
-		rtd->dai_link->ops->hw_free(substream);
-
-	mutex_unlock(&pcm_mutex);
-	return ret;
-}
-
-/*
- * Frees resources allocated by hw_params, can be called multiple times
- */
-static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
-
-	mutex_lock(&pcm_mutex);
-
-	/* apply codec digital mute */
-	if (!codec->active)
-		snd_soc_dai_digital_mute(codec_dai, 1);
-
-	/* free any machine hw params */
-	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
-		rtd->dai_link->ops->hw_free(substream);
-
-	/* free any DMA resources */
-	if (platform->driver->ops && platform->driver->ops->hw_free)
-		platform->driver->ops->hw_free(substream);
-
-	/* now free hw params for the DAIs  */
-	if (codec_dai->driver->ops->hw_free)
-		codec_dai->driver->ops->hw_free(substream, codec_dai);
-
-	if (cpu_dai->driver->ops->hw_free)
-		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
-
-	mutex_unlock(&pcm_mutex);
-	return 0;
-}
-
-static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int ret;
-
-	if (codec_dai->driver->ops->trigger) {
-		ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
-		if (ret < 0)
-			return ret;
-	}
-
-	if (platform->driver->ops && platform->driver->ops->trigger) {
-		ret = platform->driver->ops->trigger(substream, cmd);
-		if (ret < 0)
-			return ret;
-	}
-
-	if (cpu_dai->driver->ops->trigger) {
-		ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
-		if (ret < 0)
-			return ret;
-	}
-	return 0;
-}
-
-/*
- * soc level wrapper for pointer callback
- * If cpu_dai, codec_dai, platform driver has the delay callback, than
- * the runtime->delay will be updated accordingly.
- */
-static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	snd_pcm_uframes_t offset = 0;
-	snd_pcm_sframes_t delay = 0;
-
-	if (platform->driver->ops && platform->driver->ops->pointer)
-		offset = platform->driver->ops->pointer(substream);
-
-	if (cpu_dai->driver->ops->delay)
-		delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
-
-	if (codec_dai->driver->ops->delay)
-		delay += codec_dai->driver->ops->delay(substream, codec_dai);
-
-	if (platform->driver->delay)
-		delay += platform->driver->delay(substream, codec_dai);
-
-	runtime->delay = delay;
-
-	return offset;
-}
-
-/* ASoC PCM operations */
-static struct snd_pcm_ops soc_pcm_ops = {
-	.open		= soc_pcm_open,
-	.close		= soc_codec_close,
-	.hw_params	= soc_pcm_hw_params,
-	.hw_free	= soc_pcm_hw_free,
-	.prepare	= soc_pcm_prepare,
-	.trigger	= soc_pcm_trigger,
-	.pointer	= soc_pcm_pointer,
-};
-
 #ifdef CONFIG_PM_SLEEP
 /* powers down audio subsystem for suspend */
 int snd_soc_suspend(struct device *dev)
@@ -1256,7 +709,7 @@
 int snd_soc_resume(struct device *dev)
 {
 	struct snd_soc_card *card = dev_get_drvdata(dev);
-	int i;
+	int i, ac97_control = 0;
 
 	/* AC97 devices might have other drivers hanging off them so
 	 * need to resume immediately.  Other drivers don't have that
@@ -1265,14 +718,15 @@
 	 */
 	for (i = 0; i < card->num_rtd; i++) {
 		struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
-		if (cpu_dai->driver->ac97_control) {
-			dev_dbg(dev, "Resuming AC97 immediately\n");
-			soc_resume_deferred(&card->deferred_resume_work);
-		} else {
-			dev_dbg(dev, "Scheduling resume work\n");
-			if (!schedule_work(&card->deferred_resume_work))
-				dev_err(dev, "resume work item may be lost\n");
-		}
+		ac97_control |= cpu_dai->driver->ac97_control;
+	}
+	if (ac97_control) {
+		dev_dbg(dev, "Resuming AC97 immediately\n");
+		soc_resume_deferred(&card->deferred_resume_work);
+	} else {
+		dev_dbg(dev, "Scheduling resume work\n");
+		if (!schedule_work(&card->deferred_resume_work))
+			dev_err(dev, "resume work item may be lost\n");
 	}
 
 	return 0;
@@ -1393,7 +847,7 @@
 	module_put(codec->dev->driver->owner);
 }
 
-static void soc_remove_dai_link(struct snd_soc_card *card, int num)
+static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 {
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
 	struct snd_soc_codec *codec = rtd->codec;
@@ -1410,7 +864,8 @@
 	}
 
 	/* remove the CODEC DAI */
-	if (codec_dai && codec_dai->probed) {
+	if (codec_dai && codec_dai->probed &&
+			codec_dai->driver->remove_order == order) {
 		if (codec_dai->driver->remove) {
 			err = codec_dai->driver->remove(codec_dai);
 			if (err < 0)
@@ -1421,7 +876,8 @@
 	}
 
 	/* remove the platform */
-	if (platform && platform->probed) {
+	if (platform && platform->probed &&
+			platform->driver->remove_order == order) {
 		if (platform->driver->remove) {
 			err = platform->driver->remove(platform);
 			if (err < 0)
@@ -1433,11 +889,13 @@
 	}
 
 	/* remove the CODEC */
-	if (codec && codec->probed)
+	if (codec && codec->probed &&
+			codec->driver->remove_order == order)
 		soc_remove_codec(codec);
 
 	/* remove the cpu_dai */
-	if (cpu_dai && cpu_dai->probed) {
+	if (cpu_dai && cpu_dai->probed &&
+			cpu_dai->driver->remove_order == order) {
 		if (cpu_dai->driver->remove) {
 			err = cpu_dai->driver->remove(cpu_dai);
 			if (err < 0)
@@ -1451,11 +909,13 @@
 
 static void soc_remove_dai_links(struct snd_soc_card *card)
 {
-	int i;
+	int dai, order;
 
-	for (i = 0; i < card->num_rtd; i++)
-		soc_remove_dai_link(card, i);
-
+	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+			order++) {
+		for (dai = 0; dai < card->num_rtd; dai++)
+			soc_remove_dai_link(card, dai, order);
+	}
 	card->num_rtd = 0;
 }
 
@@ -1526,6 +986,52 @@
 	return ret;
 }
 
+static int soc_probe_platform(struct snd_soc_card *card,
+			   struct snd_soc_platform *platform)
+{
+	int ret = 0;
+	const struct snd_soc_platform_driver *driver = platform->driver;
+
+	platform->card = card;
+	platform->dapm.card = card;
+
+	if (!try_module_get(platform->dev->driver->owner))
+		return -ENODEV;
+
+	if (driver->dapm_widgets)
+		snd_soc_dapm_new_controls(&platform->dapm,
+			driver->dapm_widgets, driver->num_dapm_widgets);
+
+	if (driver->probe) {
+		ret = driver->probe(platform);
+		if (ret < 0) {
+			dev_err(platform->dev,
+				"asoc: failed to probe platform %s: %d\n",
+				platform->name, ret);
+			goto err_probe;
+		}
+	}
+
+	if (driver->controls)
+		snd_soc_add_platform_controls(platform, driver->controls,
+				     driver->num_controls);
+	if (driver->dapm_routes)
+		snd_soc_dapm_add_routes(&platform->dapm, driver->dapm_routes,
+					driver->num_dapm_routes);
+
+	/* mark platform as probed and add to card platform list */
+	platform->probed = 1;
+	list_add(&platform->card_list, &card->platform_dev_list);
+	list_add(&platform->dapm.list, &card->dapm_list);
+
+	return 0;
+
+err_probe:
+	module_put(platform->dev->driver->owner);
+
+	return ret;
+}
+
 static void rtd_release(struct device *dev) {}
 
 static int soc_post_component_init(struct snd_soc_card *card,
@@ -1572,6 +1078,7 @@
 	rtd->dev.parent = card->dev;
 	rtd->dev.release = rtd_release;
 	rtd->dev.init_name = name;
+	mutex_init(&rtd->pcm_mutex);
 	ret = device_register(&rtd->dev);
 	if (ret < 0) {
 		dev_err(card->dev,
@@ -1596,7 +1103,7 @@
 	return 0;
 }
 
-static int soc_probe_dai_link(struct snd_soc_card *card, int num)
+static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 {
 	struct snd_soc_dai_link *dai_link = &card->dai_link[num];
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
@@ -1605,7 +1112,8 @@
 	struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
 	int ret;
 
-	dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num);
+	dev_dbg(card->dev, "probe %s dai link %d late %d\n",
+			card->name, num, order);
 
 	/* config components */
 	codec_dai->codec = codec;
@@ -1617,7 +1125,8 @@
 	rtd->pmdown_time = pmdown_time;
 
 	/* probe the cpu_dai */
-	if (!cpu_dai->probed) {
+	if (!cpu_dai->probed &&
+			cpu_dai->driver->probe_order == order) {
 		if (!try_module_get(cpu_dai->dev->driver->owner))
 			return -ENODEV;
 
@@ -1636,33 +1145,23 @@
 	}
 
 	/* probe the CODEC */
-	if (!codec->probed) {
+	if (!codec->probed &&
+			codec->driver->probe_order == order) {
 		ret = soc_probe_codec(card, codec);
 		if (ret < 0)
 			return ret;
 	}
 
 	/* probe the platform */
-	if (!platform->probed) {
-		if (!try_module_get(platform->dev->driver->owner))
-			return -ENODEV;
-
-		if (platform->driver->probe) {
-			ret = platform->driver->probe(platform);
-			if (ret < 0) {
-				printk(KERN_ERR "asoc: failed to probe platform %s\n",
-						platform->name);
-				module_put(platform->dev->driver->owner);
-				return ret;
-			}
-		}
-		/* mark platform as probed and add to card platform list */
-		platform->probed = 1;
-		list_add(&platform->card_list, &card->platform_dev_list);
+	if (!platform->probed &&
+			platform->driver->probe_order == order) {
+		ret = soc_probe_platform(card, platform);
+		if (ret < 0)
+			return ret;
 	}
 
 	/* probe the CODEC DAI */
-	if (!codec_dai->probed) {
+	if (!codec_dai->probed && codec_dai->driver->probe_order == order) {
 		if (codec_dai->driver->probe) {
 			ret = codec_dai->driver->probe(codec_dai);
 			if (ret < 0) {
@@ -1677,8 +1176,9 @@
 		list_add(&codec_dai->card_list, &card->dai_dev_list);
 	}
 
-	/* DAPM dai link stream work */
-	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+	/* complete DAI probe during last probe */
+	if (order != SND_SOC_COMP_ORDER_LAST)
+		return 0;
 
 	ret = soc_post_component_init(card, codec, num, 0);
 	if (ret)
@@ -1817,7 +1317,7 @@
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_conf *codec_conf;
 	enum snd_soc_compress_type compress_type;
-	int ret, i;
+	int ret, i, order;
 
 	mutex_lock(&card->mutex);
 
@@ -1895,12 +1395,16 @@
 			goto card_probe_error;
 	}
 
-	for (i = 0; i < card->num_links; i++) {
-		ret = soc_probe_dai_link(card, i);
-		if (ret < 0) {
-			pr_err("asoc: failed to instantiate card %s: %d\n",
+	/* early DAI link probe */
+	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+			order++) {
+		for (i = 0; i < card->num_links; i++) {
+			ret = soc_probe_dai_link(card, i, order);
+			if (ret < 0) {
+				pr_err("asoc: failed to instantiate card %s: %d\n",
 			       card->name, ret);
-			goto probe_dai_err;
+				goto probe_dai_err;
+			}
 		}
 	}
 
@@ -2095,67 +1599,6 @@
 	.remove		= soc_remove,
 };
 
-/* create a new pcm */
-static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_pcm *pcm;
-	char new_name[64];
-	int ret = 0, playback = 0, capture = 0;
-
-	/* check client and interface hw capabilities */
-	snprintf(new_name, sizeof(new_name), "%s %s-%d",
-			rtd->dai_link->stream_name, codec_dai->name, num);
-
-	if (codec_dai->driver->playback.channels_min)
-		playback = 1;
-	if (codec_dai->driver->capture.channels_min)
-		capture = 1;
-
-	dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
-	ret = snd_pcm_new(rtd->card->snd_card, new_name,
-			num, playback, capture, &pcm);
-	if (ret < 0) {
-		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
-		return ret;
-	}
-
-	rtd->pcm = pcm;
-	pcm->private_data = rtd;
-	if (platform->driver->ops) {
-		soc_pcm_ops.mmap = platform->driver->ops->mmap;
-		soc_pcm_ops.pointer = platform->driver->ops->pointer;
-		soc_pcm_ops.ioctl = platform->driver->ops->ioctl;
-		soc_pcm_ops.copy = platform->driver->ops->copy;
-		soc_pcm_ops.silence = platform->driver->ops->silence;
-		soc_pcm_ops.ack = platform->driver->ops->ack;
-		soc_pcm_ops.page = platform->driver->ops->page;
-	}
-
-	if (playback)
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
-
-	if (capture)
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
-
-	if (platform->driver->pcm_new) {
-		ret = platform->driver->pcm_new(rtd->card->snd_card,
-						codec_dai, pcm);
-		if (ret < 0) {
-			pr_err("asoc: platform pcm constructor failed\n");
-			return ret;
-		}
-	}
-
-	pcm->private_free = platform->driver->pcm_free;
-	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
-		cpu_dai->name);
-	return ret;
-}
-
 /**
  * snd_soc_codec_volatile_register: Report if a register is volatile.
  *
@@ -2210,6 +1653,38 @@
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register);
 
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+					unsigned int reg)
+{
+	unsigned int ret;
+
+	if (!platform->driver->read) {
+		dev_err(platform->dev, "platform has no read back\n");
+		return -1;
+	}
+
+	ret = platform->driver->read(platform, reg);
+	dev_dbg(platform->dev, "read %x => %x\n", reg, ret);
+	trace_snd_soc_preg_read(platform, reg, ret);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_platform_read);
+
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+					 unsigned int reg, unsigned int val)
+{
+	if (!platform->driver->write) {
+		dev_err(platform->dev, "platform has no write back\n");
+		return -1;
+	}
+
+	dev_dbg(platform->dev, "write %x = %x\n", reg, val);
+	trace_snd_soc_preg_write(platform, reg, val);
+	return platform->driver->write(platform, reg, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_platform_write);
+
 /**
  * snd_soc_new_ac97_codec - initailise AC97 device
  * @codec: audio codec
@@ -2322,7 +1797,7 @@
 		return ret;
 
 	old = ret;
-	new = (old & ~mask) | value;
+	new = (old & ~mask) | (value & mask);
 	change = old != new;
 	if (change) {
 		ret = snd_soc_write(codec, reg, new);
@@ -2489,6 +1964,36 @@
 EXPORT_SYMBOL_GPL(snd_soc_add_controls);
 
 /**
+ * snd_soc_add_platform_controls - add an array of controls to a platform.
+ * Convienience function to add a list of controls.
+ *
+ * @platform: platform 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_platform_controls(struct snd_soc_platform *platform,
+	const struct snd_kcontrol_new *controls, int num_controls)
+{
+	struct snd_card *card = platform->card->snd_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, platform,
+				control->name, NULL));
+		if (err < 0) {
+			dev_err(platform->dev, "Failed to add %s %d\n",control->name, err);
+			return err;
+		}
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls);
+
+/**
  * snd_soc_info_enum_double - enumerated double mixer info callback
  * @kcontrol: mixer control
  * @uinfo: control element information
@@ -3632,6 +3137,8 @@
 
 	platform->dev = dev;
 	platform->driver = platform_drv;
+	platform->dapm.dev = dev;
+	platform->dapm.platform = platform;
 
 	mutex_lock(&client_mutex);
 	list_add(&platform->list, &platform_list);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 32ab7fc..54fa2e5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -124,6 +124,51 @@
 	return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
 }
 
+static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg)
+{
+	if (w->codec)
+		return snd_soc_read(w->codec, reg);
+	else if (w->platform)
+		return snd_soc_platform_read(w->platform, reg);
+
+	dev_err(w->dapm->dev, "no valid widget read method\n");
+	return -1;
+}
+
+static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val)
+{
+	if (w->codec)
+		return snd_soc_write(w->codec, reg, val);
+	else if (w->platform)
+		return snd_soc_platform_write(w->platform, reg, val);
+
+	dev_err(w->dapm->dev, "no valid widget write method\n");
+	return -1;
+}
+
+static int soc_widget_update_bits(struct snd_soc_dapm_widget *w,
+	unsigned short reg, unsigned int mask, unsigned int value)
+{
+	int change;
+	unsigned int old, new;
+	int ret;
+
+	ret = soc_widget_read(w, reg);
+	if (ret < 0)
+		return ret;
+
+	old = ret;
+	new = (old & ~mask) | (value & mask);
+	change = old != new;
+	if (change) {
+		ret = soc_widget_write(w, reg, new);
+		if (ret < 0)
+			return ret;
+	}
+
+	return change;
+}
+
 /**
  * snd_soc_dapm_set_bias_level - set the bias level for the system
  * @dapm: DAPM context
@@ -139,39 +184,26 @@
 	struct snd_soc_card *card = dapm->card;
 	int ret = 0;
 
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		dev_dbg(dapm->dev, "Setting full bias\n");
-		break;
-	case SND_SOC_BIAS_PREPARE:
-		dev_dbg(dapm->dev, "Setting bias prepare\n");
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		dev_dbg(dapm->dev, "Setting standby bias\n");
-		break;
-	case SND_SOC_BIAS_OFF:
-		dev_dbg(dapm->dev, "Setting bias off\n");
-		break;
-	default:
-		dev_err(dapm->dev, "Setting invalid bias %d\n", level);
-		return -EINVAL;
-	}
-
 	trace_snd_soc_bias_level_start(card, level);
 
 	if (card && card->set_bias_level)
-		ret = card->set_bias_level(card, level);
-	if (ret == 0) {
-		if (dapm->codec && dapm->codec->driver->set_bias_level)
-			ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
+		ret = card->set_bias_level(card, dapm, level);
+	if (ret != 0)
+		goto out;
+
+	if (dapm->codec) {
+		if (dapm->codec->driver->set_bias_level)
+			ret = dapm->codec->driver->set_bias_level(dapm->codec,
+								  level);
 		else
 			dapm->bias_level = level;
 	}
-	if (ret == 0) {
-		if (card && card->set_bias_level_post)
-			ret = card->set_bias_level_post(card, level);
-	}
+	if (ret != 0)
+		goto out;
 
+	if (card && card->set_bias_level_post)
+		ret = card->set_bias_level_post(card, dapm, level);
+out:
 	trace_snd_soc_bias_level_done(card, level);
 
 	return ret;
@@ -194,7 +226,7 @@
 		unsigned int mask = (1 << fls(max)) - 1;
 		unsigned int invert = mc->invert;
 
-		val = snd_soc_read(w->codec, reg);
+		val = soc_widget_read(w, reg);
 		val = (val >> shift) & mask;
 
 		if ((invert && !val) || (!invert && val))
@@ -209,8 +241,8 @@
 		int val, item, bitmask;
 
 		for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
-		;
-		val = snd_soc_read(w->codec, e->reg);
+			;
+		val = soc_widget_read(w, e->reg);
 		item = (val >> e->shift_l) & (bitmask - 1);
 
 		p->connect = 0;
@@ -240,7 +272,7 @@
 			w->kcontrol_news[i].private_value;
 		int val, item;
 
-		val = snd_soc_read(w->codec, e->reg);
+		val = soc_widget_read(w, e->reg);
 		val = (val >> e->shift_l) & e->mask;
 		for (item = 0; item < e->max; item++) {
 			if (val == e->values[item])
@@ -606,6 +638,9 @@
 	}
 
 	list_for_each_entry(path, &widget->sinks, list_source) {
+		if (path->weak)
+			continue;
+
 		if (path->walked)
 			continue;
 
@@ -656,6 +691,9 @@
 	}
 
 	list_for_each_entry(path, &widget->sources, list_sink) {
+		if (path->weak)
+			continue;
+
 		if (path->walked)
 			continue;
 
@@ -681,7 +719,7 @@
 	else
 		val = w->off_val;
 
-	snd_soc_update_bits(w->codec, -(w->reg + 1),
+	soc_widget_update_bits(w, -(w->reg + 1),
 			    w->mask << w->shift, val << w->shift);
 
 	return 0;
@@ -737,6 +775,9 @@
 
 	/* Check if one of our outputs is connected */
 	list_for_each_entry(path, &w->sinks, list_source) {
+		if (path->weak)
+			continue;
+
 		if (path->connected &&
 		    !path->connected(path->source, path->sink))
 			continue;
@@ -885,11 +926,17 @@
 	}
 
 	if (reg >= 0) {
+		/* Any widget will do, they should all be updating the
+		 * same register.
+		 */
+		w = list_first_entry(pending, struct snd_soc_dapm_widget,
+				     power_list);
+
 		pop_dbg(dapm->dev, card->pop_time,
 			"pop test : Applying 0x%x/0x%x to %x in %dms\n",
 			value, mask, reg, card->pop_time);
 		pop_wait(card->pop_time);
-		snd_soc_update_bits(dapm->codec, reg, mask, value);
+		soc_widget_update_bits(w, reg, mask, value);
 	}
 
 	list_for_each_entry(w, pending, power_list) {
@@ -1041,16 +1088,17 @@
 	struct snd_soc_dapm_context *d = data;
 	int ret;
 
-	if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
+	/* If we're off and we're not supposed to be go into STANDBY */
+	if (d->bias_level == SND_SOC_BIAS_OFF &&
+	    d->target_bias_level != SND_SOC_BIAS_OFF) {
 		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
 		if (ret != 0)
 			dev_err(d->dev,
 				"Failed to turn on bias: %d\n", ret);
 	}
 
-	/* If we're changing to all on or all off then prepare */
-	if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
-	    (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
+	/* Prepare for a STADDBY->ON or ON->STANDBY transition */
+	if (d->bias_level != d->target_bias_level) {
 		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
 		if (ret != 0)
 			dev_err(d->dev,
@@ -1067,7 +1115,9 @@
 	int ret;
 
 	/* If we just powered the last thing off drop to standby bias */
-	if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
+	if (d->bias_level == SND_SOC_BIAS_PREPARE &&
+	    (d->target_bias_level == SND_SOC_BIAS_STANDBY ||
+	     d->target_bias_level == SND_SOC_BIAS_OFF)) {
 		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
 		if (ret != 0)
 			dev_err(d->dev, "Failed to apply standby bias: %d\n",
@@ -1075,14 +1125,16 @@
 	}
 
 	/* If we're in standby and can support bias off then do that */
-	if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) {
+	if (d->bias_level == SND_SOC_BIAS_STANDBY &&
+	    d->target_bias_level == SND_SOC_BIAS_OFF) {
 		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF);
 		if (ret != 0)
 			dev_err(d->dev, "Failed to turn off bias: %d\n", ret);
 	}
 
 	/* If we just powered up then move to active bias */
-	if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
+	if (d->bias_level == SND_SOC_BIAS_PREPARE &&
+	    d->target_bias_level == SND_SOC_BIAS_ON) {
 		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON);
 		if (ret != 0)
 			dev_err(d->dev, "Failed to apply active bias: %d\n",
@@ -1107,13 +1159,19 @@
 	LIST_HEAD(up_list);
 	LIST_HEAD(down_list);
 	LIST_HEAD(async_domain);
+	enum snd_soc_bias_level bias;
 	int power;
 
 	trace_snd_soc_dapm_start(card);
 
-	list_for_each_entry(d, &card->dapm_list, list)
-		if (d->n_widgets || d->codec == NULL)
-			d->dev_power = 0;
+	list_for_each_entry(d, &card->dapm_list, list) {
+		if (d->n_widgets || d->codec == NULL) {
+			if (d->idle_bias_off)
+				d->target_bias_level = SND_SOC_BIAS_OFF;
+			else
+				d->target_bias_level = SND_SOC_BIAS_STANDBY;
+		}
+	}
 
 	/* Check which widgets we need to power and store them in
 	 * lists indicating if they should be powered up or down.
@@ -1135,8 +1193,27 @@
 				power = w->power_check(w);
 			else
 				power = 1;
-			if (power)
-				w->dapm->dev_power = 1;
+
+			if (power) {
+				d = w->dapm;
+
+				/* Supplies and micbiases only bring
+				 * the context up to STANDBY as unless
+				 * something else is active and
+				 * passing audio they generally don't
+				 * require full power.
+				 */
+				switch (w->id) {
+				case snd_soc_dapm_supply:
+				case snd_soc_dapm_micbias:
+					if (d->target_bias_level < SND_SOC_BIAS_STANDBY)
+						d->target_bias_level = SND_SOC_BIAS_STANDBY;
+					break;
+				default:
+					d->target_bias_level = SND_SOC_BIAS_ON;
+					break;
+				}
+			}
 
 			if (w->power == power)
 				continue;
@@ -1160,24 +1237,19 @@
 		switch (event) {
 		case SND_SOC_DAPM_STREAM_START:
 		case SND_SOC_DAPM_STREAM_RESUME:
-			dapm->dev_power = 1;
+			dapm->target_bias_level = SND_SOC_BIAS_ON;
 			break;
 		case SND_SOC_DAPM_STREAM_STOP:
-			dapm->dev_power = !!dapm->codec->active;
+			if (dapm->codec->active)
+				dapm->target_bias_level = SND_SOC_BIAS_ON;
+			else
+				dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
 			break;
 		case SND_SOC_DAPM_STREAM_SUSPEND:
-			dapm->dev_power = 0;
+			dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
 			break;
 		case SND_SOC_DAPM_STREAM_NOP:
-			switch (dapm->bias_level) {
-				case SND_SOC_BIAS_STANDBY:
-				case SND_SOC_BIAS_OFF:
-					dapm->dev_power = 0;
-					break;
-				default:
-					dapm->dev_power = 1;
-					break;
-			}
+			dapm->target_bias_level = dapm->bias_level;
 			break;
 		default:
 			break;
@@ -1185,12 +1257,12 @@
 	}
 
 	/* Force all contexts in the card to the same bias state */
-	power = 0;
+	bias = SND_SOC_BIAS_OFF;
 	list_for_each_entry(d, &card->dapm_list, list)
-		if (d->dev_power)
-			power = 1;
+		if (d->target_bias_level > bias)
+			bias = d->target_bias_level;
 	list_for_each_entry(d, &card->dapm_list, list)
-		d->dev_power = power;
+		d->target_bias_level = bias;
 
 
 	/* Run all the bias changes in parallel */
@@ -1794,6 +1866,84 @@
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
 
+static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm,
+				   const struct snd_soc_dapm_route *route)
+{
+	struct snd_soc_dapm_widget *source = dapm_find_widget(dapm,
+							      route->source,
+							      true);
+	struct snd_soc_dapm_widget *sink = dapm_find_widget(dapm,
+							    route->sink,
+							    true);
+	struct snd_soc_dapm_path *path;
+	int count = 0;
+
+	if (!source) {
+		dev_err(dapm->dev, "Unable to find source %s for weak route\n",
+			route->source);
+		return -ENODEV;
+	}
+
+	if (!sink) {
+		dev_err(dapm->dev, "Unable to find sink %s for weak route\n",
+			route->sink);
+		return -ENODEV;
+	}
+
+	if (route->control || route->connected)
+		dev_warn(dapm->dev, "Ignoring control for weak route %s->%s\n",
+			 route->source, route->sink);
+
+	list_for_each_entry(path, &source->sinks, list_source) {
+		if (path->sink == sink) {
+			path->weak = 1;
+			count++;
+		}
+	}
+
+	if (count == 0)
+		dev_err(dapm->dev, "No path found for weak route %s->%s\n",
+			route->source, route->sink);
+	if (count > 1)
+		dev_warn(dapm->dev, "%d paths found for weak route %s->%s\n",
+			 count, route->source, route->sink);
+
+	return 0;
+}
+
+/**
+ * snd_soc_dapm_weak_routes - Mark routes between DAPM widgets as weak
+ * @dapm: DAPM context
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * Mark existing routes matching those specified in the passed array
+ * as being weak, meaning that they are ignored for the purpose of
+ * power decisions.  The main intended use case is for sidetone paths
+ * which couple audio between other independent paths if they are both
+ * active in order to make the combination work better at the user
+ * level but which aren't intended to be "used".
+ *
+ * Note that CODEC drivers should not use this as sidetone type paths
+ * can frequently also be used as bypass paths.
+ */
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+			     const struct snd_soc_dapm_route *route, int num)
+{
+	int i, err;
+	int ret = 0;
+
+	for (i = 0; i < num; i++) {
+		err = snd_soc_dapm_weak_route(dapm, route);
+		if (err)
+			ret = err;
+		route++;
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
+
 /**
  * snd_soc_dapm_new_widgets - add new dapm widgets
  * @dapm: DAPM context
@@ -1865,7 +2015,7 @@
 
 		/* Read the initial power state from the device */
 		if (w->reg >= 0) {
-			val = snd_soc_read(w->codec, w->reg);
+			val = soc_widget_read(w, w->reg);
 			val &= 1 << w->shift;
 			if (w->invert)
 				val = !val;
@@ -2353,6 +2503,7 @@
 	dapm->n_widgets++;
 	w->dapm = dapm;
 	w->codec = dapm->codec;
+	w->platform = dapm->platform;
 	INIT_LIST_HEAD(&w->sources);
 	INIT_LIST_HEAD(&w->sinks);
 	INIT_LIST_HEAD(&w->list);
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
new file mode 100644
index 0000000..cca490c
--- /dev/null
+++ b/sound/soc/soc-io.c
@@ -0,0 +1,396 @@
+/*
+ * soc-io.c  --  ASoC register I/O helpers
+ *
+ * Copyright 2009-2011 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/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include <trace/events/asoc.h>
+
+#ifdef CONFIG_SPI_MASTER
+static int do_spi_write(void *control, const char *data, int len)
+{
+	struct spi_device *spi = control;
+	int ret;
+
+	ret = spi_write(spi, data, len);
+	if (ret < 0)
+		return ret;
+
+	return len;
+}
+#endif
+
+static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
+		       unsigned int value, const void *data, int len)
+{
+	int ret;
+
+	if (!snd_soc_codec_volatile_register(codec, reg) &&
+	    reg < codec->driver->reg_cache_size &&
+	    !codec->cache_bypass) {
+		ret = snd_soc_cache_write(codec, reg, value);
+		if (ret < 0)
+			return -1;
+	}
+
+	if (codec->cache_only) {
+		codec->cache_sync = 1;
+		return 0;
+	}
+
+	ret = codec->hw_write(codec->control_data, data, len);
+	if (ret == len)
+		return 0;
+	if (ret < 0)
+		return ret;
+	else
+		return -EIO;
+}
+
+static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	int ret;
+	unsigned int val;
+
+	if (reg >= codec->driver->reg_cache_size ||
+	    snd_soc_codec_volatile_register(codec, reg) ||
+	    codec->cache_bypass) {
+		if (codec->cache_only)
+			return -1;
+
+		BUG_ON(!codec->hw_read);
+		return codec->hw_read(codec, reg);
+	}
+
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret < 0)
+		return -1;
+	return val;
+}
+
+static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
+			      unsigned int value)
+{
+	u16 data;
+
+	data = cpu_to_be16((reg << 12) | (value & 0xffffff));
+
+	return do_hw_write(codec, reg, value, &data, 2);
+}
+
+static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
+			     unsigned int value)
+{
+	u16 data;
+
+	data = cpu_to_be16((reg << 9) | (value & 0x1ff));
+
+	return do_hw_write(codec, reg, value, &data, 2);
+}
+
+static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
+			     unsigned int value)
+{
+	u8 data[2];
+
+	reg &= 0xff;
+	data[0] = reg;
+	data[1] = value & 0xff;
+
+	return do_hw_write(codec, reg, value, data, 2);
+}
+
+static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
+			      unsigned int value)
+{
+	u8 data[3];
+	u16 val = cpu_to_be16(value);
+
+	data[0] = reg;
+	memcpy(&data[1], &val, sizeof(val));
+
+	return do_hw_write(codec, reg, value, data, 3);
+}
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int do_i2c_read(struct snd_soc_codec *codec,
+				void *reg, int reglen,
+				void *data, int datalen)
+{
+	struct i2c_msg xfer[2];
+	int ret;
+	struct i2c_client *client = codec->control_data;
+
+	/* Write register */
+	xfer[0].addr = client->addr;
+	xfer[0].flags = 0;
+	xfer[0].len = reglen;
+	xfer[0].buf = reg;
+
+	/* Read data */
+	xfer[1].addr = client->addr;
+	xfer[1].flags = I2C_M_RD;
+	xfer[1].len = datalen;
+	xfer[1].buf = data;
+
+	ret = i2c_transfer(client->adapter, xfer, 2);
+	if (ret == 2)
+		return 0;
+	else if (ret < 0)
+		return ret;
+	else
+		return -EIO;
+}
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
+					 unsigned int r)
+{
+	u8 reg = r;
+	u8 data;
+	int ret;
+
+	ret = do_i2c_read(codec, &reg, 1, &data, 1);
+	if (ret < 0)
+		return 0;
+	return data;
+}
+#else
+#define snd_soc_8_8_read_i2c NULL
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
+					  unsigned int r)
+{
+	u8 reg = r;
+	u16 data;
+	int ret;
+
+	ret = do_i2c_read(codec, &reg, 1, &data, 2);
+	if (ret < 0)
+		return 0;
+	return (data >> 8) | ((data & 0xff) << 8);
+}
+#else
+#define snd_soc_8_16_read_i2c NULL
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
+					  unsigned int r)
+{
+	u16 reg = r;
+	u8 data;
+	int ret;
+
+	ret = do_i2c_read(codec, &reg, 2, &data, 1);
+	if (ret < 0)
+		return 0;
+	return data;
+}
+#else
+#define snd_soc_16_8_read_i2c NULL
+#endif
+
+static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
+			      unsigned int value)
+{
+	u8 data[3];
+	u16 rval = cpu_to_be16(reg);
+
+	memcpy(data, &rval, sizeof(rval));
+	data[2] = value;
+
+	return do_hw_write(codec, reg, value, data, 3);
+}
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
+					   unsigned int r)
+{
+	u16 reg = cpu_to_be16(r);
+	u16 data;
+	int ret;
+
+	ret = do_i2c_read(codec, &reg, 2, &data, 2);
+	if (ret < 0)
+		return 0;
+	return be16_to_cpu(data);
+}
+#else
+#define snd_soc_16_16_read_i2c NULL
+#endif
+
+static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
+			       unsigned int value)
+{
+	u16 data[2];
+
+	data[0] = cpu_to_be16(reg);
+	data[1] = cpu_to_be16(value);
+
+	return do_hw_write(codec, reg, value, data, sizeof(data));
+}
+
+/* Primitive bulk write support for soc-cache.  The data pointed to by
+ * `data' needs to already be in the form the hardware expects
+ * including any leading register specific data.  Any data written
+ * through this function will not go through the cache as it only
+ * handles writing to volatile or out of bounds registers.
+ */
+static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
+				     const void *data, size_t len)
+{
+	int ret;
+
+	/* To ensure that we don't get out of sync with the cache, check
+	 * whether the base register is volatile or if we've directly asked
+	 * to bypass the cache.  Out of bounds registers are considered
+	 * volatile.
+	 */
+	if (!codec->cache_bypass
+	    && !snd_soc_codec_volatile_register(codec, reg)
+	    && reg < codec->driver->reg_cache_size)
+		return -EINVAL;
+
+	switch (codec->control_type) {
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+	case SND_SOC_I2C:
+		ret = i2c_master_send(to_i2c_client(codec->dev), data, len);
+		break;
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	case SND_SOC_SPI:
+		ret = spi_write(to_spi_device(codec->dev), data, len);
+		break;
+#endif
+	default:
+		BUG();
+	}
+
+	if (ret == len)
+		return 0;
+	if (ret < 0)
+		return ret;
+	else
+		return -EIO;
+}
+
+static struct {
+	int addr_bits;
+	int data_bits;
+	int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
+	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+	unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+} io_types[] = {
+	{
+		.addr_bits = 4, .data_bits = 12,
+		.write = snd_soc_4_12_write,
+	},
+	{
+		.addr_bits = 7, .data_bits = 9,
+		.write = snd_soc_7_9_write,
+	},
+	{
+		.addr_bits = 8, .data_bits = 8,
+		.write = snd_soc_8_8_write,
+		.i2c_read = snd_soc_8_8_read_i2c,
+	},
+	{
+		.addr_bits = 8, .data_bits = 16,
+		.write = snd_soc_8_16_write,
+		.i2c_read = snd_soc_8_16_read_i2c,
+	},
+	{
+		.addr_bits = 16, .data_bits = 8,
+		.write = snd_soc_16_8_write,
+		.i2c_read = snd_soc_16_8_read_i2c,
+	},
+	{
+		.addr_bits = 16, .data_bits = 16,
+		.write = snd_soc_16_16_write,
+		.i2c_read = snd_soc_16_16_read_i2c,
+	},
+};
+
+/**
+ * snd_soc_codec_set_cache_io: Set up standard I/O functions.
+ *
+ * @codec: CODEC to configure.
+ * @addr_bits: Number of bits of register address data.
+ * @data_bits: Number of bits of data per register.
+ * @control: Control bus used.
+ *
+ * Register formats are frequently shared between many I2C and SPI
+ * devices.  In order to promote code reuse the ASoC core provides
+ * some standard implementations of CODEC read and write operations
+ * which can be set up using this function.
+ *
+ * The caller is responsible for allocating and initialising the
+ * actual cache.
+ *
+ * Note that at present this code cannot be used by CODECs with
+ * volatile registers.
+ */
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+			       int addr_bits, int data_bits,
+			       enum snd_soc_control_type control)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(io_types); i++)
+		if (io_types[i].addr_bits == addr_bits &&
+		    io_types[i].data_bits == data_bits)
+			break;
+	if (i == ARRAY_SIZE(io_types)) {
+		printk(KERN_ERR
+		       "No I/O functions for %d bit address %d bit data\n",
+		       addr_bits, data_bits);
+		return -EINVAL;
+	}
+
+	codec->write = io_types[i].write;
+	codec->read = hw_read;
+	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
+
+	switch (control) {
+	case SND_SOC_I2C:
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+		codec->hw_write = (hw_write_t)i2c_master_send;
+#endif
+		if (io_types[i].i2c_read)
+			codec->hw_read = io_types[i].i2c_read;
+
+		codec->control_data = container_of(codec->dev,
+						   struct i2c_client,
+						   dev);
+		break;
+
+	case SND_SOC_SPI:
+#ifdef CONFIG_SPI_MASTER
+		codec->hw_write = do_spi_write;
+#endif
+
+		codec->control_data = container_of(codec->dev,
+						   struct spi_device,
+						   dev);
+		break;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
+
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
new file mode 100644
index 0000000..b575939
--- /dev/null
+++ b/sound/soc/soc-pcm.c
@@ -0,0 +1,639 @@
+/*
+ * soc-pcm.c  --  ALSA SoC PCM
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ * Copyright (C) 2010 Slimlogic Ltd.
+ * Copyright (C) 2010 Texas Instruments Inc.
+ *
+ * Authors: Liam Girdwood <lrg@ti.com>
+ *          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/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+static DEFINE_MUTEX(pcm_mutex);
+
+static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	if (!codec_dai->driver->symmetric_rates &&
+	    !cpu_dai->driver->symmetric_rates &&
+	    !rtd->dai_link->symmetric_rates)
+		return 0;
+
+	/* This can happen if multiple streams are starting simultaneously -
+	 * the second can need to get its constraints before the first has
+	 * picked a rate.  Complain and allow the application to carry on.
+	 */
+	if (!rtd->rate) {
+		dev_warn(&rtd->dev,
+			 "Not enforcing symmetric_rates due to race\n");
+		return 0;
+	}
+
+	dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
+
+	ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+					   SNDRV_PCM_HW_PARAM_RATE,
+					   rtd->rate, rtd->rate);
+	if (ret < 0) {
+		dev_err(&rtd->dev,
+			"Unable to apply rate symmetry constraint: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+/*
+ * Called by ALSA when a PCM substream is opened, the runtime->hw record is
+ * then initialized and any private data can be allocated. This also calls
+ * startup for the cpu DAI, platform, machine and codec DAI.
+ */
+static int soc_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+	struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver;
+	int ret = 0;
+
+	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+	/* startup the audio subsystem */
+	if (cpu_dai->driver->ops->startup) {
+		ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't open interface %s\n",
+				cpu_dai->name);
+			goto out;
+		}
+	}
+
+	if (platform->driver->ops && platform->driver->ops->open) {
+		ret = platform->driver->ops->open(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
+			goto platform_err;
+		}
+	}
+
+	if (codec_dai->driver->ops->startup) {
+		ret = codec_dai->driver->ops->startup(substream, codec_dai);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't open codec %s\n",
+				codec_dai->name);
+			goto codec_dai_err;
+		}
+	}
+
+	if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
+		ret = rtd->dai_link->ops->startup(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name);
+			goto machine_err;
+		}
+	}
+
+	/* Check that the codec and cpu DAIs are compatible */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		runtime->hw.rate_min =
+			max(codec_dai_drv->playback.rate_min,
+			    cpu_dai_drv->playback.rate_min);
+		runtime->hw.rate_max =
+			min(codec_dai_drv->playback.rate_max,
+			    cpu_dai_drv->playback.rate_max);
+		runtime->hw.channels_min =
+			max(codec_dai_drv->playback.channels_min,
+				cpu_dai_drv->playback.channels_min);
+		runtime->hw.channels_max =
+			min(codec_dai_drv->playback.channels_max,
+				cpu_dai_drv->playback.channels_max);
+		runtime->hw.formats =
+			codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats;
+		runtime->hw.rates =
+			codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates;
+		if (codec_dai_drv->playback.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= cpu_dai_drv->playback.rates;
+		if (cpu_dai_drv->playback.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= codec_dai_drv->playback.rates;
+	} else {
+		runtime->hw.rate_min =
+			max(codec_dai_drv->capture.rate_min,
+			    cpu_dai_drv->capture.rate_min);
+		runtime->hw.rate_max =
+			min(codec_dai_drv->capture.rate_max,
+			    cpu_dai_drv->capture.rate_max);
+		runtime->hw.channels_min =
+			max(codec_dai_drv->capture.channels_min,
+				cpu_dai_drv->capture.channels_min);
+		runtime->hw.channels_max =
+			min(codec_dai_drv->capture.channels_max,
+				cpu_dai_drv->capture.channels_max);
+		runtime->hw.formats =
+			codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats;
+		runtime->hw.rates =
+			codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates;
+		if (codec_dai_drv->capture.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= cpu_dai_drv->capture.rates;
+		if (cpu_dai_drv->capture.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= codec_dai_drv->capture.rates;
+	}
+
+	ret = -EINVAL;
+	snd_pcm_limit_hw_rates(runtime);
+	if (!runtime->hw.rates) {
+		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
+			codec_dai->name, cpu_dai->name);
+		goto config_err;
+	}
+	if (!runtime->hw.formats) {
+		printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
+			codec_dai->name, cpu_dai->name);
+		goto config_err;
+	}
+	if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+	    runtime->hw.channels_min > runtime->hw.channels_max) {
+		printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
+				codec_dai->name, cpu_dai->name);
+		goto config_err;
+	}
+
+	/* Symmetry only applies if we've already got an active stream. */
+	if (cpu_dai->active || codec_dai->active) {
+		ret = soc_pcm_apply_symmetry(substream);
+		if (ret != 0)
+			goto config_err;
+	}
+
+	pr_debug("asoc: %s <-> %s info:\n",
+			codec_dai->name, cpu_dai->name);
+	pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
+	pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+		 runtime->hw.channels_max);
+	pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+		 runtime->hw.rate_max);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		cpu_dai->playback_active++;
+		codec_dai->playback_active++;
+	} else {
+		cpu_dai->capture_active++;
+		codec_dai->capture_active++;
+	}
+	cpu_dai->active++;
+	codec_dai->active++;
+	rtd->codec->active++;
+	mutex_unlock(&rtd->pcm_mutex);
+	return 0;
+
+config_err:
+	if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
+		rtd->dai_link->ops->shutdown(substream);
+
+machine_err:
+	if (codec_dai->driver->ops->shutdown)
+		codec_dai->driver->ops->shutdown(substream, codec_dai);
+
+codec_dai_err:
+	if (platform->driver->ops && platform->driver->ops->close)
+		platform->driver->ops->close(substream);
+
+platform_err:
+	if (cpu_dai->driver->ops->shutdown)
+		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+out:
+	mutex_unlock(&rtd->pcm_mutex);
+	return ret;
+}
+
+/*
+ * Power down the audio subsystem pmdown_time msecs after close is called.
+ * This is to ensure there are no pops or clicks in between any music tracks
+ * due to DAPM power cycling.
+ */
+static void close_delayed_work(struct work_struct *work)
+{
+	struct snd_soc_pcm_runtime *rtd =
+			container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+	pr_debug("pop wq checking: %s status: %s waiting: %s\n",
+		 codec_dai->driver->playback.stream_name,
+		 codec_dai->playback_active ? "active" : "inactive",
+		 codec_dai->pop_wait ? "yes" : "no");
+
+	/* are we waiting on this codec DAI stream */
+	if (codec_dai->pop_wait == 1) {
+		codec_dai->pop_wait = 0;
+		snd_soc_dapm_stream_event(rtd,
+			codec_dai->driver->playback.stream_name,
+			SND_SOC_DAPM_STREAM_STOP);
+	}
+
+	mutex_unlock(&rtd->pcm_mutex);
+}
+
+/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and platform are also
+ * shutdown.
+ */
+static int soc_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = rtd->codec;
+
+	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		cpu_dai->playback_active--;
+		codec_dai->playback_active--;
+	} else {
+		cpu_dai->capture_active--;
+		codec_dai->capture_active--;
+	}
+
+	cpu_dai->active--;
+	codec_dai->active--;
+	codec->active--;
+
+	/* Muting the DAC suppresses artifacts caused during digital
+	 * shutdown, for example from stopping clocks.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_dai_digital_mute(codec_dai, 1);
+
+	if (cpu_dai->driver->ops->shutdown)
+		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+
+	if (codec_dai->driver->ops->shutdown)
+		codec_dai->driver->ops->shutdown(substream, codec_dai);
+
+	if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
+		rtd->dai_link->ops->shutdown(substream);
+
+	if (platform->driver->ops && platform->driver->ops->close)
+		platform->driver->ops->close(substream);
+	cpu_dai->runtime = NULL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* start delayed pop wq here for playback streams */
+		codec_dai->pop_wait = 1;
+		schedule_delayed_work(&rtd->delayed_work,
+			msecs_to_jiffies(rtd->pmdown_time));
+	} else {
+		/* capture streams can be powered down now */
+		snd_soc_dapm_stream_event(rtd,
+			codec_dai->driver->capture.stream_name,
+			SND_SOC_DAPM_STREAM_STOP);
+	}
+
+	mutex_unlock(&rtd->pcm_mutex);
+	return 0;
+}
+
+/*
+ * Called by ALSA when the PCM substream is prepared, can set format, sample
+ * rate, etc.  This function is non atomic and can be called multiple times,
+ * it can refer to the runtime info.
+ */
+static int soc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret = 0;
+
+	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+	if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
+		ret = rtd->dai_link->ops->prepare(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: machine prepare error\n");
+			goto out;
+		}
+	}
+
+	if (platform->driver->ops && platform->driver->ops->prepare) {
+		ret = platform->driver->ops->prepare(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: platform prepare error\n");
+			goto out;
+		}
+	}
+
+	if (codec_dai->driver->ops->prepare) {
+		ret = codec_dai->driver->ops->prepare(substream, codec_dai);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: codec DAI prepare error\n");
+			goto out;
+		}
+	}
+
+	if (cpu_dai->driver->ops->prepare) {
+		ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
+			goto out;
+		}
+	}
+
+	/* cancel any delayed stream shutdown that is pending */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+	    codec_dai->pop_wait) {
+		codec_dai->pop_wait = 0;
+		cancel_delayed_work(&rtd->delayed_work);
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_dapm_stream_event(rtd,
+					  codec_dai->driver->playback.stream_name,
+					  SND_SOC_DAPM_STREAM_START);
+	else
+		snd_soc_dapm_stream_event(rtd,
+					  codec_dai->driver->capture.stream_name,
+					  SND_SOC_DAPM_STREAM_START);
+
+	snd_soc_dai_digital_mute(codec_dai, 0);
+
+out:
+	mutex_unlock(&rtd->pcm_mutex);
+	return ret;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+static int soc_pcm_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_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret = 0;
+
+	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
+		ret = rtd->dai_link->ops->hw_params(substream, params);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: machine hw_params failed\n");
+			goto out;
+		}
+	}
+
+	if (codec_dai->driver->ops->hw_params) {
+		ret = codec_dai->driver->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);
+			goto codec_err;
+		}
+	}
+
+	if (cpu_dai->driver->ops->hw_params) {
+		ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: interface %s hw params failed\n",
+				cpu_dai->name);
+			goto interface_err;
+		}
+	}
+
+	if (platform->driver->ops && platform->driver->ops->hw_params) {
+		ret = platform->driver->ops->hw_params(substream, params);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: platform %s hw params failed\n",
+				platform->name);
+			goto platform_err;
+		}
+	}
+
+	rtd->rate = params_rate(params);
+
+out:
+	mutex_unlock(&rtd->pcm_mutex);
+	return ret;
+
+platform_err:
+	if (cpu_dai->driver->ops->hw_free)
+		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+
+interface_err:
+	if (codec_dai->driver->ops->hw_free)
+		codec_dai->driver->ops->hw_free(substream, codec_dai);
+
+codec_err:
+	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
+		rtd->dai_link->ops->hw_free(substream);
+
+	mutex_unlock(&rtd->pcm_mutex);
+	return ret;
+}
+
+/*
+ * Frees resources allocated by hw_params, can be called multiple times
+ */
+static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = rtd->codec;
+
+	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+	/* apply codec digital mute */
+	if (!codec->active)
+		snd_soc_dai_digital_mute(codec_dai, 1);
+
+	/* free any machine hw params */
+	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
+		rtd->dai_link->ops->hw_free(substream);
+
+	/* free any DMA resources */
+	if (platform->driver->ops && platform->driver->ops->hw_free)
+		platform->driver->ops->hw_free(substream);
+
+	/* now free hw params for the DAIs  */
+	if (codec_dai->driver->ops->hw_free)
+		codec_dai->driver->ops->hw_free(substream, codec_dai);
+
+	if (cpu_dai->driver->ops->hw_free)
+		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+
+	mutex_unlock(&rtd->pcm_mutex);
+	return 0;
+}
+
+static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	if (codec_dai->driver->ops->trigger) {
+		ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (platform->driver->ops && platform->driver->ops->trigger) {
+		ret = platform->driver->ops->trigger(substream, cmd);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (cpu_dai->driver->ops->trigger) {
+		ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
+		if (ret < 0)
+			return ret;
+	}
+	return 0;
+}
+
+/*
+ * soc level wrapper for pointer callback
+ * If cpu_dai, codec_dai, platform driver has the delay callback, than
+ * the runtime->delay will be updated accordingly.
+ */
+static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	snd_pcm_uframes_t offset = 0;
+	snd_pcm_sframes_t delay = 0;
+
+	if (platform->driver->ops && platform->driver->ops->pointer)
+		offset = platform->driver->ops->pointer(substream);
+
+	if (cpu_dai->driver->ops->delay)
+		delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
+
+	if (codec_dai->driver->ops->delay)
+		delay += codec_dai->driver->ops->delay(substream, codec_dai);
+
+	if (platform->driver->delay)
+		delay += platform->driver->delay(substream, codec_dai);
+
+	runtime->delay = delay;
+
+	return offset;
+}
+
+/* ASoC PCM operations */
+static struct snd_pcm_ops soc_pcm_ops = {
+	.open		= soc_pcm_open,
+	.close		= soc_pcm_close,
+	.hw_params	= soc_pcm_hw_params,
+	.hw_free	= soc_pcm_hw_free,
+	.prepare	= soc_pcm_prepare,
+	.trigger	= soc_pcm_trigger,
+	.pointer	= soc_pcm_pointer,
+};
+
+/* create a new pcm */
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_pcm *pcm;
+	char new_name[64];
+	int ret = 0, playback = 0, capture = 0;
+
+	/* check client and interface hw capabilities */
+	snprintf(new_name, sizeof(new_name), "%s %s-%d",
+			rtd->dai_link->stream_name, codec_dai->name, num);
+
+	if (codec_dai->driver->playback.channels_min)
+		playback = 1;
+	if (codec_dai->driver->capture.channels_min)
+		capture = 1;
+
+	dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
+	ret = snd_pcm_new(rtd->card->snd_card, new_name,
+			num, playback, capture, &pcm);
+	if (ret < 0) {
+		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+		return ret;
+	}
+
+	/* DAPM dai link stream work */
+	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+
+	rtd->pcm = pcm;
+	pcm->private_data = rtd;
+	if (platform->driver->ops) {
+		soc_pcm_ops.mmap = platform->driver->ops->mmap;
+		soc_pcm_ops.pointer = platform->driver->ops->pointer;
+		soc_pcm_ops.ioctl = platform->driver->ops->ioctl;
+		soc_pcm_ops.copy = platform->driver->ops->copy;
+		soc_pcm_ops.silence = platform->driver->ops->silence;
+		soc_pcm_ops.ack = platform->driver->ops->ack;
+		soc_pcm_ops.page = platform->driver->ops->page;
+	}
+
+	if (playback)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
+
+	if (capture)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
+
+	if (platform->driver->pcm_new) {
+		ret = platform->driver->pcm_new(rtd);
+		if (ret < 0) {
+			pr_err("asoc: platform pcm constructor failed\n");
+			return ret;
+		}
+	}
+
+	pcm->private_free = platform->driver->pcm_free;
+	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+		cpu_dai->name);
+	return ret;
+}
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 035d39a..c6af1fd 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -12,6 +12,15 @@
 	  Tegra I2S interface. You will also need to select the individual
 	  machine drivers to support below.
 
+config SND_SOC_TEGRA_SPDIF
+	tristate
+	depends on SND_SOC_TEGRA
+	default m
+	help
+	  Say Y or M if you want to add support for the SPDIF interface.
+	  You will also need to select the individual machine drivers to support
+	  below.
+
 config MACH_HAS_SND_SOC_TEGRA_WM8903
 	bool
 	help
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index fa6574d..4d943b3 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -2,12 +2,14 @@
 snd-soc-tegra-das-objs := tegra_das.o
 snd-soc-tegra-pcm-objs := tegra_pcm.o
 snd-soc-tegra-i2s-objs := tegra_i2s.o
+snd-soc-tegra-spdif-objs := tegra_spdif.o
 snd-soc-tegra-utils-objs += tegra_asoc_utils.o
 
 obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o
 obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o
 obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o
 obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o
+obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o
 
 # Tegra machine Support
 snd-soc-tegra-wm8903-objs := tegra_wm8903.o
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 95f03c1..f36b996 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -354,7 +354,6 @@
 static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 {
 	struct tegra_i2s * i2s;
-	char clk_name[12]; /* tegra-i2s.0 */
 	struct resource *mem, *memregion, *dmareq;
 	int ret;
 
@@ -389,8 +388,7 @@
 	}
 	dev_set_drvdata(&pdev->dev, i2s);
 
-	snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id);
-	i2s->clk_i2s = clk_get_sys(clk_name, NULL);
+	i2s->clk_i2s = clk_get(&pdev->dev, NULL);
 	if (IS_ERR(i2s->clk_i2s)) {
 		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
 		ret = PTR_ERR(i2s->clk_i2s);
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 3c271f9..ff86e5e 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -322,9 +322,11 @@
 
 static u64 tegra_dma_mask = DMA_BIT_MASK(32);
 
-static int tegra_pcm_new(struct snd_card *card,
-				struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
new file mode 100644
index 0000000..abe606b
--- /dev/null
+++ b/sound/soc/tegra/tegra_spdif.c
@@ -0,0 +1,371 @@
+/*
+ * tegra_spdif.c - Tegra SPDIF driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2011 - NVIDIA, Inc.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/debugfs.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/seq_file.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <mach/iomap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "tegra_spdif.h"
+
+#define DRV_NAME "tegra-spdif"
+
+static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg,
+					u32 val)
+{
+	__raw_writel(val, spdif->regs + reg);
+}
+
+static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg)
+{
+	return __raw_readl(spdif->regs + reg);
+}
+
+#ifdef CONFIG_DEBUG_FS
+static int tegra_spdif_show(struct seq_file *s, void *unused)
+{
+#define REG(r) { r, #r }
+	static const struct {
+		int offset;
+		const char *name;
+	} regs[] = {
+		REG(TEGRA_SPDIF_CTRL),
+		REG(TEGRA_SPDIF_STATUS),
+		REG(TEGRA_SPDIF_STROBE_CTRL),
+		REG(TEGRA_SPDIF_DATA_FIFO_CSR),
+		REG(TEGRA_SPDIF_CH_STA_RX_A),
+		REG(TEGRA_SPDIF_CH_STA_RX_B),
+		REG(TEGRA_SPDIF_CH_STA_RX_C),
+		REG(TEGRA_SPDIF_CH_STA_RX_D),
+		REG(TEGRA_SPDIF_CH_STA_RX_E),
+		REG(TEGRA_SPDIF_CH_STA_RX_F),
+		REG(TEGRA_SPDIF_CH_STA_TX_A),
+		REG(TEGRA_SPDIF_CH_STA_TX_B),
+		REG(TEGRA_SPDIF_CH_STA_TX_C),
+		REG(TEGRA_SPDIF_CH_STA_TX_D),
+		REG(TEGRA_SPDIF_CH_STA_TX_E),
+		REG(TEGRA_SPDIF_CH_STA_TX_F),
+	};
+#undef REG
+
+	struct tegra_spdif *spdif = s->private;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(regs); i++) {
+		u32 val = tegra_spdif_read(spdif, regs[i].offset);
+		seq_printf(s, "%s = %08x\n", regs[i].name, val);
+	}
+
+	return 0;
+}
+
+static int tegra_spdif_debug_open(struct inode *inode, struct file *file)
+{
+	return single_open(file, tegra_spdif_show, inode->i_private);
+}
+
+static const struct file_operations tegra_spdif_debug_fops = {
+	.open    = tegra_spdif_debug_open,
+	.read    = seq_read,
+	.llseek  = seq_lseek,
+	.release = single_release,
+};
+
+static void tegra_spdif_debug_add(struct tegra_spdif *spdif)
+{
+	spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO,
+						snd_soc_debugfs_root, spdif,
+						&tegra_spdif_debug_fops);
+}
+
+static void tegra_spdif_debug_remove(struct tegra_spdif *spdif)
+{
+	if (spdif->debug)
+		debugfs_remove(spdif->debug);
+}
+#else
+static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif)
+{
+}
+
+static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif)
+{
+}
+#endif
+
+static int tegra_spdif_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct device *dev = substream->pcm->card->dev;
+	struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+	int ret, srate, spdifclock;
+
+	spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK;
+	spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK;
+		spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	srate = params_rate(params);
+	switch (params_rate(params)) {
+	case 32000:
+		spdifclock = 4096000;
+		break;
+	case 44100:
+		spdifclock = 5644800;
+		break;
+	case 48000:
+		spdifclock = 6144000;
+		break;
+	case 88200:
+		spdifclock = 11289600;
+		break;
+	case 96000:
+		spdifclock = 12288000;
+		break;
+	case 176400:
+		spdifclock = 22579200;
+		break;
+	case 192000:
+		spdifclock = 24576000;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ret = clk_set_rate(spdif->clk_spdif_out, spdifclock);
+	if (ret) {
+		dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static void tegra_spdif_start_playback(struct tegra_spdif *spdif)
+{
+	spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN;
+	tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl);
+}
+
+static void tegra_spdif_stop_playback(struct tegra_spdif *spdif)
+{
+	spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN;
+	tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl);
+}
+
+static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		if (!spdif->clk_refs)
+			clk_enable(spdif->clk_spdif_out);
+		spdif->clk_refs++;
+		tegra_spdif_start_playback(spdif);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		tegra_spdif_stop_playback(spdif);
+		spdif->clk_refs--;
+		if (!spdif->clk_refs)
+			clk_disable(spdif->clk_spdif_out);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int tegra_spdif_probe(struct snd_soc_dai *dai)
+{
+	struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+	dai->capture_dma_data = NULL;
+	dai->playback_dma_data = &spdif->playback_dma_data;
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops tegra_spdif_dai_ops = {
+	.hw_params	= tegra_spdif_hw_params,
+	.trigger	= tegra_spdif_trigger,
+};
+
+struct snd_soc_dai_driver tegra_spdif_dai = {
+	.name = DRV_NAME,
+	.probe = tegra_spdif_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+				SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &tegra_spdif_dai_ops,
+};
+
+static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev)
+{
+	struct tegra_spdif *spdif;
+	struct resource *mem, *memregion, *dmareq;
+	int ret;
+
+	spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL);
+	if (!spdif) {
+		dev_err(&pdev->dev, "Can't allocate tegra_spdif\n");
+		ret = -ENOMEM;
+		goto exit;
+	}
+	dev_set_drvdata(&pdev->dev, spdif);
+
+	spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out");
+	if (IS_ERR(spdif->clk_spdif_out)) {
+		pr_err("Can't retrieve spdif clock\n");
+		ret = PTR_ERR(spdif->clk_spdif_out);
+		goto err_free;
+	}
+
+	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem) {
+		dev_err(&pdev->dev, "No memory resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmareq) {
+		dev_err(&pdev->dev, "No DMA resource\n");
+		ret = -ENODEV;
+		goto err_clk_put;
+	}
+
+	memregion = request_mem_region(mem->start, resource_size(mem),
+					DRV_NAME);
+	if (!memregion) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		ret = -EBUSY;
+		goto err_clk_put;
+	}
+
+	spdif->regs = ioremap(mem->start, resource_size(mem));
+	if (!spdif->regs) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_release;
+	}
+
+	spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT;
+	spdif->playback_dma_data.wrap = 4;
+	spdif->playback_dma_data.width = 32;
+	spdif->playback_dma_data.req_sel = dmareq->start;
+
+	ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
+		ret = -ENOMEM;
+		goto err_unmap;
+	}
+
+	tegra_spdif_debug_add(spdif);
+
+	return 0;
+
+err_unmap:
+	iounmap(spdif->regs);
+err_release:
+	release_mem_region(mem->start, resource_size(mem));
+err_clk_put:
+	clk_put(spdif->clk_spdif_out);
+err_free:
+	kfree(spdif);
+exit:
+	return ret;
+}
+
+static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev)
+{
+	struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev);
+	struct resource *res;
+
+	snd_soc_unregister_dai(&pdev->dev);
+
+	tegra_spdif_debug_remove(spdif);
+
+	iounmap(spdif->regs);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	release_mem_region(res->start, resource_size(res));
+
+	clk_put(spdif->clk_spdif_out);
+
+	kfree(spdif);
+
+	return 0;
+}
+
+static struct platform_driver tegra_spdif_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = tegra_spdif_platform_probe,
+	.remove = __devexit_p(tegra_spdif_platform_remove),
+};
+
+static int __init snd_tegra_spdif_init(void)
+{
+	return platform_driver_register(&tegra_spdif_driver);
+}
+module_init(snd_tegra_spdif_init);
+
+static void __exit snd_tegra_spdif_exit(void)
+{
+	platform_driver_unregister(&tegra_spdif_driver);
+}
+module_exit(snd_tegra_spdif_exit);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra SPDIF ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h
new file mode 100644
index 0000000..2e03db4
--- /dev/null
+++ b/sound/soc/tegra/tegra_spdif.h
@@ -0,0 +1,473 @@
+/*
+ * tegra_spdif.h - Definitions for Tegra SPDIF driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2011 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ * Copyright (c) 2008-2009, NVIDIA Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TEGRA_SPDIF_H__
+#define __TEGRA_SPDIF_H__
+
+#include "tegra_pcm.h"
+
+/* Offsets from TEGRA_SPDIF_BASE */
+
+#define TEGRA_SPDIF_CTRL					0x0
+#define TEGRA_SPDIF_STATUS					0x4
+#define TEGRA_SPDIF_STROBE_CTRL					0x8
+#define TEGRA_SPDIF_DATA_FIFO_CSR				0x0C
+#define TEGRA_SPDIF_DATA_OUT					0x40
+#define TEGRA_SPDIF_DATA_IN					0x80
+#define TEGRA_SPDIF_CH_STA_RX_A					0x100
+#define TEGRA_SPDIF_CH_STA_RX_B					0x104
+#define TEGRA_SPDIF_CH_STA_RX_C					0x108
+#define TEGRA_SPDIF_CH_STA_RX_D					0x10C
+#define TEGRA_SPDIF_CH_STA_RX_E					0x110
+#define TEGRA_SPDIF_CH_STA_RX_F					0x114
+#define TEGRA_SPDIF_CH_STA_TX_A					0x140
+#define TEGRA_SPDIF_CH_STA_TX_B					0x144
+#define TEGRA_SPDIF_CH_STA_TX_C					0x148
+#define TEGRA_SPDIF_CH_STA_TX_D					0x14C
+#define TEGRA_SPDIF_CH_STA_TX_E					0x150
+#define TEGRA_SPDIF_CH_STA_TX_F					0x154
+#define TEGRA_SPDIF_USR_STA_RX_A				0x180
+#define TEGRA_SPDIF_USR_DAT_TX_A				0x1C0
+
+/* Fields in TEGRA_SPDIF_CTRL */
+
+/* Start capturing from 0=right, 1=left channel */
+#define TEGRA_SPDIF_CTRL_CAP_LC					(1 << 30)
+
+/* SPDIF receiver(RX) enable */
+#define TEGRA_SPDIF_CTRL_RX_EN					(1 << 29)
+
+/* SPDIF Transmitter(TX) enable */
+#define TEGRA_SPDIF_CTRL_TX_EN					(1 << 28)
+
+/* Transmit Channel status */
+#define TEGRA_SPDIF_CTRL_TC_EN					(1 << 27)
+
+/* Transmit user Data */
+#define TEGRA_SPDIF_CTRL_TU_EN					(1 << 26)
+
+/* Interrupt on transmit error */
+#define TEGRA_SPDIF_CTRL_IE_TXE					(1 << 25)
+
+/* Interrupt on receive error */
+#define TEGRA_SPDIF_CTRL_IE_RXE					(1 << 24)
+
+/* Interrupt on invalid preamble */
+#define TEGRA_SPDIF_CTRL_IE_P					(1 << 23)
+
+/* Interrupt on "B" preamble */
+#define TEGRA_SPDIF_CTRL_IE_B					(1 << 22)
+
+/* Interrupt when block of channel status received */
+#define TEGRA_SPDIF_CTRL_IE_C					(1 << 21)
+
+/* Interrupt when a valid information unit (IU) is received */
+#define TEGRA_SPDIF_CTRL_IE_U					(1 << 20)
+
+/* Interrupt when RX user FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_RU					(1 << 19)
+
+/* Interrupt when TX user FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_TU					(1 << 18)
+
+/* Interrupt when RX data FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_RX					(1 << 17)
+
+/* Interrupt when TX data FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_TX					(1 << 16)
+
+/* Loopback test mode enable */
+#define TEGRA_SPDIF_CTRL_LBK_EN					(1 << 15)
+
+/*
+ * Pack data mode:
+ * 0 = Single data (16 bit needs to be  padded to match the
+ *     interface data bit size).
+ * 1 = Packeted left/right channel data into a single word.
+ */
+#define TEGRA_SPDIF_CTRL_PACK					(1 << 14)
+
+/*
+ * 00 = 16bit data
+ * 01 = 20bit data
+ * 10 = 24bit data
+ * 11 = raw data
+ */
+#define TEGRA_SPDIF_BIT_MODE_16BIT				0
+#define TEGRA_SPDIF_BIT_MODE_20BIT				1
+#define TEGRA_SPDIF_BIT_MODE_24BIT				2
+#define TEGRA_SPDIF_BIT_MODE_RAW				3
+
+#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT				12
+#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK				(3                          << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT				(TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT				(TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT				(TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW				(TEGRA_SPDIF_BIT_MODE_RAW   << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+
+/* Fields in TEGRA_SPDIF_STATUS */
+
+/*
+ * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must
+ * write a 1 to the corresponding bit location to clear the status.
+ */
+
+/*
+ * Receiver(RX) shifter is busy receiving data.
+ * This bit is asserted when the receiver first locked onto the
+ * preamble of the data stream after RX_EN is asserted. This bit is
+ * deasserted when either,
+ * (a) the end of a frame is reached after RX_EN is deeasserted, or
+ * (b) the SPDIF data stream becomes inactive.
+ */
+#define TEGRA_SPDIF_STATUS_RX_BSY				(1 << 29)
+
+/*
+ * Transmitter(TX) shifter is busy transmitting data.
+ * This bit is asserted when TX_EN is asserted.
+ * This bit is deasserted when the end of a frame is reached after
+ * TX_EN is deasserted.
+ */
+#define TEGRA_SPDIF_STATUS_TX_BSY				(1 << 28)
+
+/*
+ * TX is busy shifting out channel status.
+ * This bit is asserted when both TX_EN and TC_EN are asserted and
+ * data from CH_STA_TX_A register is loaded into the internal shifter.
+ * This bit is deasserted when either,
+ * (a) the end of a frame is reached after TX_EN is deasserted, or
+ * (b) CH_STA_TX_F register is loaded into the internal shifter.
+ */
+#define TEGRA_SPDIF_STATUS_TC_BSY				(1 << 27)
+
+/*
+ * TX User data FIFO busy.
+ * This bit is asserted when TX_EN and TXU_EN are asserted and
+ * there's data in the TX user FIFO.  This bit is deassert when either,
+ * (a) the end of a frame is reached after TX_EN is deasserted, or
+ * (b) there's no data left in the TX user FIFO.
+ */
+#define TEGRA_SPDIF_STATUS_TU_BSY				(1 << 26)
+
+/* TX FIFO Underrun error status */
+#define TEGRA_SPDIF_STATUS_TX_ERR				(1 << 25)
+
+/* RX FIFO Overrun error status */
+#define TEGRA_SPDIF_STATUS_RX_ERR				(1 << 24)
+
+/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */
+#define TEGRA_SPDIF_STATUS_IS_P					(1 << 23)
+
+/* B-preamble detection status: 0=not detected, 1=B-preamble detected */
+#define TEGRA_SPDIF_STATUS_IS_B					(1 << 22)
+
+/*
+ * RX channel block data receive status:
+ * 0=entire block not recieved yet.
+ * 1=received entire block of channel status,
+ */
+#define TEGRA_SPDIF_STATUS_IS_C					(1 << 21)
+
+/* RX User Data Valid flag:  1=valid IU detected, 0 = no IU detected. */
+#define TEGRA_SPDIF_STATUS_IS_U					(1 << 20)
+
+/*
+ * RX User FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_RU				(1 << 19)
+
+/*
+ * TX User FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_TU				(1 << 18)
+
+/*
+ * RX Data FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_RX				(1 << 17)
+
+/*
+ * TX Data FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_TX				(1 << 16)
+
+/* Fields in TEGRA_SPDIF_STROBE_CTRL */
+
+/*
+ * Indicates the approximate number of detected SPDIFIN clocks within a
+ * bi-phase period.
+ */
+#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT			16
+#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK			(0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT)
+
+/* Data strobe mode: 0=Auto-locked 1=Manual locked */
+#define TEGRA_SPDIF_STROBE_CTRL_STROBE				(1 << 15)
+
+/*
+ * Manual data strobe time within the bi-phase clock period (in terms of
+ * the number of over-sampling clocks).
+ */
+#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT		8
+#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK		(0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT)
+
+/*
+ * Manual SPDIFIN bi-phase clock period (in terms of the number of
+ * over-sampling clocks).
+ */
+#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT		0
+#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK		(0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT)
+
+/* Fields in SPDIF_DATA_FIFO_CSR */
+
+/* Clear Receiver User FIFO (RX USR.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR			(1 << 31)
+
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT			0
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS			1
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS			2
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS			3
+
+/* RU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT		29
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK		\
+		(0x3                                    << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT    << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+
+/* Number of RX USR.FIFO levels with valid data. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT		24
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK		(0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT)
+
+/* Clear Transmitter User FIFO (TX USR.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR			(1 << 23)
+
+/* TU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT		21
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK		\
+		(0x3                                   << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT    << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+
+/* Number of TX USR.FIFO levels that could be filled. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT		16
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK		(0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT)
+
+/* Clear Receiver Data FIFO (RX DATA.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR			(1 << 15)
+
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT			0
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS			1
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS			2
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS			3
+
+/* RU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT		13
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK		\
+		(0x3                                     << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT     << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+
+/* Number of RX DATA.FIFO levels with valid data. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT		8
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK		(0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT)
+
+/* Clear Transmitter Data FIFO (TX DATA.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR			(1 << 7)
+
+/* TU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT		5
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK		\
+		(0x3                                     << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT     << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS   << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS  << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL	\
+		(TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+
+/* Number of TX DATA.FIFO levels that could be filled. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT		0
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK		(0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT)
+
+/* Fields in TEGRA_SPDIF_DATA_OUT */
+
+/*
+ * This register has 5 different formats:
+ * 16-bit        (BIT_MODE=00, PACK=0)
+ * 20-bit        (BIT_MODE=01, PACK=0)
+ * 24-bit        (BIT_MODE=10, PACK=0)
+ * raw           (BIT_MODE=11, PACK=0)
+ * 16-bit packed (BIT_MODE=00, PACK=1)
+ */
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT			0
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK			(0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT			0
+#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK			(0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT			0
+#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK			(0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P				(1 << 31)
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C				(1 << 30)
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U				(1 << 29)
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V				(1 << 28)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT		8
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK			(0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT			4
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK			(0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT		0
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK		(0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT		16
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK		(0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT		0
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK		(0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT)
+
+/* Fields in TEGRA_SPDIF_DATA_IN */
+
+/*
+ * This register has 5 different formats:
+ * 16-bit        (BIT_MODE=00, PACK=0)
+ * 20-bit        (BIT_MODE=01, PACK=0)
+ * 24-bit        (BIT_MODE=10, PACK=0)
+ * raw           (BIT_MODE=11, PACK=0)
+ * 16-bit packed (BIT_MODE=00, PACK=1)
+ *
+ * Bits 31:24 are common to all modes except 16-bit packed
+ */
+
+#define TEGRA_SPDIF_DATA_IN_DATA_P				(1 << 31)
+#define TEGRA_SPDIF_DATA_IN_DATA_C				(1 << 30)
+#define TEGRA_SPDIF_DATA_IN_DATA_U				(1 << 29)
+#define TEGRA_SPDIF_DATA_IN_DATA_V				(1 << 28)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT			24
+#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK			(0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT			0
+#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK			(0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT			0
+#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK			(0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT			0
+#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK			(0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT			8
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK			(0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT			4
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK			(0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT		0
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK		(0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT		16
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK		(0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT		0
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK		(0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT)
+
+/* Fields in TEGRA_SPDIF_CH_STA_RX_A */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_B */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_C */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_D */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_E */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_F */
+
+/*
+ * The 6-word receive channel data page buffer holds a block (192 frames) of
+ * channel status information. The order of receive is from LSB to MSB
+ * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A.
+ */
+
+/* Fields in TEGRA_SPDIF_CH_STA_TX_A */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_B */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_C */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_D */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_E */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_F */
+
+/*
+ * The 6-word transmit channel data page buffer holds a block (192 frames) of
+ * channel status information. The order of transmission is from LSB to MSB
+ * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A.
+ */
+
+/* Fields in TEGRA_SPDIF_USR_STA_RX_A */
+
+/*
+ * This 4-word deep FIFO receives user FIFO field information. The order of
+ * receive is from LSB to MSB bit.
+ */
+
+/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */
+
+/*
+ * This 4-word deep FIFO transmits user FIFO field information. The order of
+ * transmission is from LSB to MSB bit.
+ */
+
+struct tegra_spdif {
+	struct clk *clk_spdif_out;
+	int clk_refs;
+	struct tegra_pcm_dma_params capture_dma_data;
+	struct tegra_pcm_dma_params playback_dma_data;
+	void __iomem *regs;
+	struct dentry *debug;
+	u32 reg_ctrl;
+};
+
+#endif
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0d6738a..a42e9ac 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -267,7 +267,7 @@
 		}
 		machine->gpio_requested |= GPIO_HP_MUTE;
 
-		gpio_direction_output(pdata->gpio_hp_mute, 0);
+		gpio_direction_output(pdata->gpio_hp_mute, 1);
 	}
 
 	if (gpio_is_valid(pdata->gpio_int_mic_en)) {
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index f4aa4e0..34aa972 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -288,9 +288,10 @@
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-			    struct snd_pcm *pcm)
+static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	struct platform_device *pdev = to_platform_device(dai->platform->dev);
 	struct txx9aclc_soc_device *dev;
 	struct resource *r;