diff --git a/arch/arm/mach-kirkwood/openrd-setup.c b/arch/arm/mach-kirkwood/openrd-setup.c
index c9d77fa..cfcca41 100644
--- a/arch/arm/mach-kirkwood/openrd-setup.c
+++ b/arch/arm/mach-kirkwood/openrd-setup.c
@@ -171,7 +171,7 @@
 
 	kirkwood_i2c_init();
 
-	if (machine_is_openrd_client()) {
+	if (machine_is_openrd_client() || machine_is_openrd_ultimate()) {
 		i2c_register_board_info(0, i2c_board_info,
 			ARRAY_SIZE(i2c_board_info));
 		kirkwood_audio_init();
diff --git a/include/sound/alc5623.h b/include/sound/alc5623.h
new file mode 100644
index 0000000..422c97d
--- /dev/null
+++ b/include/sound/alc5623.h
@@ -0,0 +1,15 @@
+#ifndef _INCLUDE_SOUND_ALC5623_H
+#define _INCLUDE_SOUND_ALC5623_H
+struct alc5623_platform_data {
+	/* configure :                              */
+	/* Lineout/Speaker Amps Vmid ratio control  */
+	/* enable/disable adc/dac high pass filters */
+	unsigned int add_ctrl;
+	/* configure :                              */
+	/* output to enable when jack is low        */
+	/* output to enable when jack is high       */
+	/* jack detect (gpio/nc/jack detect [12]    */
+	unsigned int jack_det_ctrl;
+};
+#endif
+
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 8fd3b41..78d3560 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -297,6 +297,7 @@
 struct snd_soc_dapm_path;
 struct snd_soc_dapm_pin;
 struct snd_soc_dapm_route;
+struct snd_soc_dapm_context;
 
 int dapm_reg_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
@@ -324,16 +325,16 @@
 	struct snd_ctl_elem_value *uncontrol);
 int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *uncontrol);
-int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	const struct snd_soc_dapm_widget *widget);
-int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 	const struct snd_soc_dapm_widget *widget,
 	int num);
 
 /* dapm path setup */
-int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
-void snd_soc_dapm_free(struct snd_soc_codec *codec);
-int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
+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);
 
 /* dapm events */
@@ -343,17 +344,21 @@
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);
-void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec);
+void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm);
 
 /* dapm audio pin control and status */
-int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_sync(struct snd_soc_codec *codec);
-int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec,
+int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm,
+			    const char *pin);
+int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
+			     const char *pin);
+int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin);
+int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm,
+				const char *pin);
+int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
 				  const char *pin);
-int snd_soc_dapm_ignore_suspend(struct snd_soc_codec *codec, const char *pin);
+int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
+				const char *pin);
 
 /* dapm widget types */
 enum snd_soc_dapm_type {
@@ -425,6 +430,7 @@
 	char *sname;	/* stream name */
 	struct snd_soc_codec *codec;
 	struct list_head list;
+	struct snd_soc_dapm_context *dapm;
 
 	/* dapm control */
 	short reg;						/* negative reg = no direct dapm */
@@ -461,4 +467,21 @@
 	struct list_head power_list;
 };
 
+/* DAPM context */
+struct snd_soc_dapm_context {
+	struct list_head widgets;
+	struct list_head paths;
+	enum snd_soc_bias_level bias_level;
+	enum snd_soc_bias_level suspend_bias_level;
+	struct delayed_work delayed_work;
+	unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
+
+	struct device *dev; /* from parent - for debug */
+	struct snd_soc_codec *codec; /* parent codec */
+	struct snd_soc_card *card; /* parent card */
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_dapm;
+#endif
+};
+
 #endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5c3bce8..3eb92ef 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -238,6 +238,8 @@
 struct snd_soc_ac97_ops;
 struct snd_soc_jack;
 struct snd_soc_jack_pin;
+struct snd_soc_cache_ops;
+#include <sound/soc-dapm.h>
 
 #ifdef CONFIG_GPIOLIB
 struct snd_soc_jack_gpio;
@@ -253,6 +255,12 @@
 	SND_SOC_SPI,
 };
 
+enum snd_soc_compress_type {
+	SND_SOC_NO_COMPRESSION,
+	SND_SOC_LZO_COMPRESSION,
+	SND_SOC_RBTREE_COMPRESSION
+};
+
 int snd_soc_register_platform(struct device *dev,
 		struct snd_soc_platform_driver *platform_drv);
 void snd_soc_unregister_platform(struct device *dev);
@@ -264,6 +272,13 @@
 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 snd_soc_cache_sync(struct snd_soc_codec *codec);
+int snd_soc_cache_init(struct snd_soc_codec *codec);
+int snd_soc_cache_exit(struct snd_soc_codec *codec);
+int snd_soc_cache_write(struct snd_soc_codec *codec,
+			unsigned int reg, unsigned int value);
+int snd_soc_cache_read(struct snd_soc_codec *codec,
+		       unsigned int reg, unsigned int *value);
 
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
@@ -420,9 +435,22 @@
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
+/* SoC cache ops */
+struct snd_soc_cache_ops {
+	enum snd_soc_compress_type id;
+	int (*init)(struct snd_soc_codec *codec);
+	int (*exit)(struct snd_soc_codec *codec);
+	int (*read)(struct snd_soc_codec *codec, unsigned int reg,
+		unsigned int *value);
+	int (*write)(struct snd_soc_codec *codec, unsigned int reg,
+		unsigned int value);
+	int (*sync)(struct snd_soc_codec *codec);
+};
+
 /* SoC Audio Codec device */
 struct snd_soc_codec {
 	const char *name;
+	const char *name_prefix;
 	int id;
 	struct device *dev;
 	struct snd_soc_codec_driver *driver;
@@ -436,7 +464,6 @@
 	/* runtime */
 	struct snd_ac97 *ac97;  /* for ad-hoc ac97 devices */
 	unsigned int active;
-	unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
 	unsigned int cache_only:1;  /* Suppress writes to hardware */
 	unsigned int cache_sync:1; /* Cache needs to be synced to hardware */
 	unsigned int suspended:1; /* Codec is in suspend PM state */
@@ -450,19 +477,15 @@
 	hw_write_t hw_write;
 	unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
 	void *reg_cache;
+	const struct snd_soc_cache_ops *cache_ops;
+	struct mutex cache_rw_mutex;
 
 	/* dapm */
-	u32 pop_time;
-	struct list_head dapm_widgets;
-	struct list_head dapm_paths;
-	enum snd_soc_bias_level bias_level;
-	enum snd_soc_bias_level suspend_bias_level;
-	struct delayed_work delayed_work;
+	struct snd_soc_dapm_context dapm;
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_codec_root;
 	struct dentry *debugfs_reg;
-	struct dentry *debugfs_pop_time;
 	struct dentry *debugfs_dapm;
 #endif
 };
@@ -488,6 +511,7 @@
 	short reg_cache_step;
 	short reg_word_size;
 	const void *reg_cache_default;
+	enum snd_soc_compress_type compress_type;
 
 	/* codec bias level */
 	int (*set_bias_level)(struct snd_soc_codec *,
@@ -554,6 +578,11 @@
 	struct snd_soc_ops *ops;
 };
 
+struct snd_soc_prefix_map {
+	const char *dev_name;
+	const char *name_prefix;
+};
+
 /* SoC card */
 struct snd_soc_card {
 	const char *name;
@@ -588,12 +617,25 @@
 	struct snd_soc_pcm_runtime *rtd;
 	int num_rtd;
 
+	/*
+	 * optional map of kcontrol, widget and path name prefixes that are
+	 * associated per device
+	 */
+	struct snd_soc_prefix_map *prefix_map;
+	int num_prefixes;
+
 	struct work_struct deferred_resume_work;
 
 	/* lists of probed devices belonging to this card */
 	struct list_head codec_dev_list;
 	struct list_head platform_dev_list;
 	struct list_head dai_dev_list;
+
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_card_root;
+	struct dentry *debugfs_pop_time;
+#endif
+	u32 pop_time;
 };
 
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
@@ -639,17 +681,9 @@
 };
 
 /* codec IO */
-static inline unsigned int snd_soc_read(struct snd_soc_codec *codec,
-					unsigned int reg)
-{
-	return codec->driver->read(codec, reg);
-}
-
-static inline unsigned int snd_soc_write(struct snd_soc_codec *codec,
-					 unsigned int reg, unsigned int val)
-{
-	return codec->driver->write(codec, reg, val);
-}
+unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg);
+unsigned int snd_soc_write(struct snd_soc_codec *codec,
+			   unsigned int reg, unsigned int val);
 
 /* device driver data */
 
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
new file mode 100644
index 0000000..9978856
--- /dev/null
+++ b/include/trace/events/asoc.h
@@ -0,0 +1,176 @@
+#undef TRACE_SYSTEM
+#define TRACE_SYSTEM asoc
+
+#if !defined(_TRACE_ASOC_H) || defined(TRACE_HEADER_MULTI_READ)
+#define _TRACE_ASOC_H
+
+#include <linux/ktime.h>
+#include <linux/tracepoint.h>
+
+struct snd_soc_codec;
+struct snd_soc_card;
+struct snd_soc_dapm_widget;
+
+/*
+ * Log register events
+ */
+DECLARE_EVENT_CLASS(snd_soc_reg,
+
+	TP_PROTO(struct snd_soc_codec *codec, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(codec, reg, val),
+
+	TP_STRUCT__entry(
+		__string(	name,		codec->name	)
+		__field(	int,		id		)
+		__field(	unsigned int,	reg		)
+		__field(	unsigned int,	val		)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, codec->name);
+		__entry->id = codec->id;
+		__entry->reg = reg;
+		__entry->val = val;
+	),
+
+	TP_printk("codec=%s.%d reg=%x val=%x", __get_str(name),
+		  (int)__entry->id, (unsigned int)__entry->reg,
+		  (unsigned int)__entry->val)
+);
+
+DEFINE_EVENT(snd_soc_reg, snd_soc_reg_write,
+
+	TP_PROTO(struct snd_soc_codec *codec, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(codec, reg, val)
+
+);
+
+DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read,
+
+	TP_PROTO(struct snd_soc_codec *codec, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(codec, reg, val)
+
+);
+
+DECLARE_EVENT_CLASS(snd_soc_card,
+
+	TP_PROTO(struct snd_soc_card *card, int val),
+
+	TP_ARGS(card, val),
+
+	TP_STRUCT__entry(
+		__string(	name,		card->name	)
+		__field(	int,		val		)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, card->name);
+		__entry->val = val;
+	),
+
+	TP_printk("card=%s val=%d", __get_str(name), (int)__entry->val)
+);
+
+DEFINE_EVENT(snd_soc_card, snd_soc_bias_level_start,
+
+	TP_PROTO(struct snd_soc_card *card, int val),
+
+	TP_ARGS(card, val)
+
+);
+
+DEFINE_EVENT(snd_soc_card, snd_soc_bias_level_done,
+
+	TP_PROTO(struct snd_soc_card *card, int val),
+
+	TP_ARGS(card, val)
+
+);
+
+DECLARE_EVENT_CLASS(snd_soc_dapm_basic,
+
+	TP_PROTO(struct snd_soc_card *card),
+
+	TP_ARGS(card),
+
+	TP_STRUCT__entry(
+		__string(	name,	card->name	)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, card->name);
+	),
+
+	TP_printk("card=%s", __get_str(name))
+);
+
+DEFINE_EVENT(snd_soc_dapm_basic, snd_soc_dapm_start,
+
+	TP_PROTO(struct snd_soc_card *card),
+
+	TP_ARGS(card)
+
+);
+
+DEFINE_EVENT(snd_soc_dapm_basic, snd_soc_dapm_done,
+
+	TP_PROTO(struct snd_soc_card *card),
+
+	TP_ARGS(card)
+
+);
+
+DECLARE_EVENT_CLASS(snd_soc_dapm_widget,
+
+	TP_PROTO(struct snd_soc_dapm_widget *w, int val),
+
+	TP_ARGS(w, val),
+
+	TP_STRUCT__entry(
+		__string(	name,	w->name		)
+		__field(	int,	val		)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, w->name);
+		__entry->val = val;
+	),
+
+	TP_printk("widget=%s val=%d", __get_str(name),
+		  (int)__entry->val)
+);
+
+DEFINE_EVENT(snd_soc_dapm_widget, snd_soc_dapm_widget_power,
+
+	TP_PROTO(struct snd_soc_dapm_widget *w, int val),
+
+	TP_ARGS(w, val)
+
+);
+
+DEFINE_EVENT(snd_soc_dapm_widget, snd_soc_dapm_widget_event_start,
+
+	TP_PROTO(struct snd_soc_dapm_widget *w, int val),
+
+	TP_ARGS(w, val)
+
+);
+
+DEFINE_EVENT(snd_soc_dapm_widget, snd_soc_dapm_widget_event_done,
+
+	TP_PROTO(struct snd_soc_dapm_widget *w, int val),
+
+	TP_ARGS(w, val)
+
+);
+
+#endif /* _TRACE_ASOC_H */
+
+/* This part must be outside protection */
+#include <trace/define_trace.h>
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 3e598e7..4562c89 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -4,6 +4,8 @@
 
 menuconfig SND_SOC
 	tristate "ALSA for SoC audio support"
+	select LZO_COMPRESS
+	select LZO_DECOMPRESS
 	select SND_PCM
 	select AC97_BUS if SND_SOC_AC97_BUS
 	select SND_JACK if INPUT=y || INPUT=SND
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 5f4e59f..aede7e7 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -318,27 +318,28 @@
 static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int i;
 
 	/*
 	 * Add DAPM widgets
 	 */
 	for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]);
+		snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
 
 
 
 	/*
 	 * Setup audio path interconnects
 	 */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 
 
 	/* always connected pins */
-	snd_soc_dapm_enable_pin(codec, "Int Mic");
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_enable_pin(dapm, "Int Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_sync(dapm);
 
 
 
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 293569d..da9c303 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -140,6 +140,7 @@
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	printk(KERN_DEBUG
@@ -154,25 +155,25 @@
 	}
 
 	/* Add specific widgets */
-	snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets,
 				  ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
 	/* Set up specific audio path interconnects */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	/* not connected */
-	snd_soc_dapm_nc_pin(codec, "RLINEIN");
-	snd_soc_dapm_nc_pin(codec, "LLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "LLINEIN");
 
 #ifdef ENABLE_MIC_INPUT
-	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(dapm, "Int Mic");
 #else
-	snd_soc_dapm_nc_pin(codec, "Int Mic");
+	snd_soc_dapm_nc_pin(dapm, "Int Mic");
 #endif
 
 	/* always connected */
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index e3d2835..92c709e 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -105,19 +105,20 @@
 static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add afeb9260 specific widgets */
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
 				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
 
 	/* Set up afeb9260 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Line In");
-	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 01d19e9..a15a3e9 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1172,7 +1172,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable Audio PLL & Audio section */
 			data = AUDIO_PLL | AUDIO_SECTION_RESET
 				| AUDIO_SECTION_ON;
@@ -1185,7 +1185,7 @@
 		pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -1346,6 +1346,7 @@
 static int pm860x_probe(struct snd_soc_codec *codec)
 {
 	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int i, ret;
 
 	pm860x->codec = codec;
@@ -1374,9 +1375,9 @@
 
 	snd_soc_add_controls(codec, pm860x_snd_controls,
 			     ARRAY_SIZE(pm860x_snd_controls));
-	snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets,
 				  ARRAY_SIZE(pm860x_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 	return 0;
 
 out_codec:
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 3b5690d..6ebd3a6 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -22,6 +22,7 @@
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
+	select SND_SOC_ALC5623 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS42L51 if I2C
 	select SND_SOC_CS4270 if I2C
@@ -57,6 +58,7 @@
 	select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8770 if SPI_MASTER
 	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8900 if I2C
@@ -130,6 +132,9 @@
 config SND_SOC_AK4671
 	tristate
 
+config SND_SOC_ALC5623
+       tristate
+
 config SND_SOC_CQ0093VC
 	tristate
 
@@ -240,6 +245,9 @@
 config SND_SOC_WM8753
 	tristate
 
+config SND_SOC_WM8770
+	tristate
+
 config SND_SOC_WM8776
 	tristate
 
@@ -318,3 +326,4 @@
 
 config SND_SOC_WM9090
 	tristate
+
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index f67a2d6..42f185d 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -17,6 +17,7 @@
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-alc5623-objs := alc5623.o
 snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-stac9766-objs := stac9766.o
@@ -41,6 +42,7 @@
 snd-soc-wm8741-objs := wm8741.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
+snd-soc-wm8770-objs := wm8770.o
 snd-soc-wm8776-objs := wm8776.o
 snd-soc-wm8804-objs := wm8804.o
 snd-soc-wm8900-objs := wm8900.o
@@ -92,6 +94,7 @@
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
@@ -116,6 +119,7 @@
 obj-$(CONFIG_SND_SOC_WM8741)	+= snd-soc-wm8741.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
+obj-$(CONFIG_SND_SOC_WM8770)	+= snd-soc-wm8770.o
 obj-$(CONFIG_SND_SOC_WM8776)	+= snd-soc-wm8776.o
 obj-$(CONFIG_SND_SOC_WM8804)	+= snd-soc-wm8804.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index d272534..e5d3db9 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -220,6 +220,7 @@
 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 ret = 0;
 
 	codec->control_data = ad1836->control_data;
@@ -227,7 +228,6 @@
 	if (ret < 0) {
 		dev_err(codec->dev, "failed to set cache I/O: %d\n",
 				ret);
-		kfree(ad1836);
 		return ret;
 	}
 
@@ -252,9 +252,9 @@
 
 	snd_soc_add_controls(codec, ad1836_snd_controls,
 			     ARRAY_SIZE(ad1836_snd_controls));
-	snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets,
 				  ARRAY_SIZE(ad1836_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
 
 	return ret;
 }
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index fa2834c..fd3e659 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -353,6 +353,7 @@
 static int ad193x_probe(struct snd_soc_codec *codec)
 {
 	struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	codec->control_data = ad193x->control_data;
@@ -363,7 +364,6 @@
 	if (ret < 0) {
 		dev_err(codec->dev, "failed to set cache I/O: %d\n",
 				ret);
-		kfree(ad193x);
 		return ret;
 	}
 
@@ -385,9 +385,9 @@
 
 	snd_soc_add_controls(codec, ad193x_snd_controls,
 			     ARRAY_SIZE(ad193x_snd_controls));
-	snd_soc_dapm_new_controls(codec, ad193x_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, ad193x_dapm_widgets,
 				  ARRAY_SIZE(ad193x_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
 
 	return ret;
 }
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index cd88c8f..52abb93 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -290,10 +290,11 @@
 
 static int ak4535_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets,
-				  ARRAY_SIZE(ak4535_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, ak4535_dapm_widgets,
+				  ARRAY_SIZE(ak4535_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -399,7 +400,7 @@
 		ak4535_write(codec, AK4535_PM1, i & (~0x80));
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 90c90b7..f00eba3 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -26,7 +26,7 @@
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
 
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 24f5f49..1d6573c 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -437,10 +437,11 @@
 
 static int ak4671_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets,
-				  ARRAY_SIZE(ak4671_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, ak4671_dapm_widgets,
+				  ARRAY_SIZE(ak4671_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -602,7 +603,7 @@
 		snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
new file mode 100644
index 0000000..5a45067
--- /dev/null
+++ b/sound/soc/codecs/alc5623.c
@@ -0,0 +1,1119 @@
+/*
+ * alc5623.c  --  alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
+ *
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ *
+ * Based on WM8753.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/alc5623.h>
+
+#include "alc5623.h"
+
+static int caps_charge = 2000;
+module_param(caps_charge, int, 0);
+MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
+
+/* codec private data */
+struct alc5623_priv {
+	enum snd_soc_control_type control_type;
+	void *control_data;
+	struct mutex mutex;
+	u8 id;
+	unsigned int sysclk;
+	u16 reg_cache[ALC5623_VENDOR_ID2+2];
+	unsigned int add_ctrl;
+	unsigned int jack_det_ctrl;
+};
+
+static void alc5623_fill_cache(struct snd_soc_codec *codec)
+{
+	int i, step = codec->driver->reg_cache_step;
+	u16 *cache = codec->reg_cache;
+
+	/* not really efficient ... */
+	for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
+		cache[i] = codec->hw_read(codec, i);
+}
+
+static inline int alc5623_reset(struct snd_soc_codec *codec)
+{
+	return snd_soc_write(codec, ALC5623_RESET, 0);
+}
+
+static int amp_mixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	/* to power-on/off class-d amp generators/speaker */
+	/* need to write to 'index-46h' register :        */
+	/* so write index num (here 0x46) to reg 0x6a     */
+	/* and then 0xffff/0 to reg 0x6c                  */
+	snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
+		break;
+	}
+
+	return 0;
+}
+
+/*
+ * ALC5623 Controls
+ */
+
+static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
+static const unsigned int boost_tlv[] = {
+	TLV_DB_RANGE_HEAD(3),
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
+	2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = {
+	SOC_DOUBLE_TLV("Speaker Playback Volume",
+			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Speaker Playback Switch",
+			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Playback Volume",
+			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Headphone Playback Switch",
+			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = {
+	SOC_DOUBLE_TLV("Speaker Playback Volume",
+			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Speaker Playback Switch",
+			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("Line Playback Volume",
+			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Line Playback Switch",
+			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
+	SOC_DOUBLE_TLV("Line Playback Volume",
+			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Line Playback Switch",
+			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Playback Volume",
+			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Headphone Playback Switch",
+			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_snd_controls[] = {
+	SOC_DOUBLE_TLV("Auxout Playback Volume",
+			ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+	SOC_DOUBLE("Auxout Playback Switch",
+			ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
+	SOC_DOUBLE_TLV("PCM Playback Volume",
+			ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
+	SOC_DOUBLE_TLV("AuxI Capture Volume",
+			ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
+	SOC_DOUBLE_TLV("LineIn Capture Volume",
+			ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
+	SOC_SINGLE_TLV("Mic1 Capture Volume",
+			ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
+	SOC_SINGLE_TLV("Mic2 Capture Volume",
+			ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
+	SOC_DOUBLE_TLV("Rec Capture Volume",
+			ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
+	SOC_SINGLE_TLV("Mic 1 Boost Volume",
+			ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
+	SOC_SINGLE_TLV("Mic 2 Boost Volume",
+			ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
+	SOC_SINGLE_TLV("Digital Boost Volume",
+			ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
+SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
+SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
+SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
+SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
+SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
+};
+
+/* Left Record Mixer */
+static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
+SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
+SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
+SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
+};
+
+/* Right Record Mixer */
+static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
+SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
+SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
+SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
+};
+
+static const char *alc5623_spk_n_sour_sel[] = {
+		"RN/-R", "RP/+R", "LN/-R", "Vmid" };
+static const char *alc5623_hpl_out_input_sel[] = {
+		"Vmid", "HP Left Mix"};
+static const char *alc5623_hpr_out_input_sel[] = {
+		"Vmid", "HP Right Mix"};
+static const char *alc5623_spkout_input_sel[] = {
+		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+static const char *alc5623_aux_out_input_sel[] = {
+		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+
+/* auxout output mux */
+static const struct soc_enum alc5623_aux_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
+static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
+
+/* speaker output mux */
+static const struct soc_enum alc5623_spkout_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
+static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
+
+/* headphone left output mux */
+static const struct soc_enum alc5623_hpl_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
+static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
+
+/* headphone right output mux */
+static const struct soc_enum alc5623_hpr_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
+static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
+
+/* speaker output N select */
+static const struct soc_enum alc5623_spk_n_sour_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
+static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
+
+static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
+/* Muxes */
+SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_auxout_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_spkout_mux_controls),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_hpl_out_mux_controls),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_hpr_out_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_spkoutn_mux_controls),
+
+/* output mixers */
+SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
+	&alc5623_hp_mixer_controls[0],
+	ARRAY_SIZE(alc5623_hp_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
+	&alc5623_hpr_mixer_controls[0],
+	ARRAY_SIZE(alc5623_hpr_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
+	&alc5623_hpl_mixer_controls[0],
+	ARRAY_SIZE(alc5623_hpl_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
+	&alc5623_mono_mixer_controls[0],
+	ARRAY_SIZE(alc5623_mono_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
+	&alc5623_speaker_mixer_controls[0],
+	ARRAY_SIZE(alc5623_speaker_mixer_controls)),
+
+/* input mixers */
+SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
+	&alc5623_captureL_mixer_controls[0],
+	ARRAY_SIZE(alc5623_captureL_mixer_controls)),
+SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
+	&alc5623_captureR_mixer_controls[0],
+	ARRAY_SIZE(alc5623_captureR_mixer_controls)),
+
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+	ALC5623_PWR_MANAG_ADD2, 9, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+	ALC5623_PWR_MANAG_ADD2, 8, 0),
+SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
+	ALC5623_PWR_MANAG_ADD2, 7, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
+	ALC5623_PWR_MANAG_ADD2, 6, 0),
+SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
+
+SND_SOC_DAPM_OUTPUT("AUXOUTL"),
+SND_SOC_DAPM_OUTPUT("AUXOUTR"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_OUTPUT("SPKOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+SND_SOC_DAPM_INPUT("LINEINL"),
+SND_SOC_DAPM_INPUT("LINEINR"),
+SND_SOC_DAPM_INPUT("AUXINL"),
+SND_SOC_DAPM_INPUT("AUXINR"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_VMID("Vmid"),
+};
+
+static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
+static const struct soc_enum alc5623_amp_enum =
+	SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
+static const struct snd_kcontrol_new alc5623_amp_mux_controls =
+	SOC_DAPM_ENUM("Route", alc5623_amp_enum);
+
+static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
+SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
+	amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
+SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
+	&alc5623_amp_mux_controls),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* virtual mixer - mixes left & right channels */
+	{"I2S Mix", NULL,				"Left DAC"},
+	{"I2S Mix", NULL,				"Right DAC"},
+	{"Line Mix", NULL,				"Right LineIn"},
+	{"Line Mix", NULL,				"Left LineIn"},
+	{"AuxI Mix", NULL,				"Left AuxI"},
+	{"AuxI Mix", NULL,				"Right AuxI"},
+	{"AUXOUTL", NULL,				"Left AuxOut"},
+	{"AUXOUTR", NULL,				"Right AuxOut"},
+
+	/* HP mixer */
+	{"HPL Mix", "ADC2HP_L Playback Switch",		"Left Capture Mix"},
+	{"HPL Mix", NULL,				"HP Mix"},
+	{"HPR Mix", "ADC2HP_R Playback Switch",		"Right Capture Mix"},
+	{"HPR Mix", NULL,				"HP Mix"},
+	{"HP Mix", "LI2HP Playback Switch",		"Line Mix"},
+	{"HP Mix", "AUXI2HP Playback Switch",		"AuxI Mix"},
+	{"HP Mix", "MIC12HP Playback Switch",		"MIC1 PGA"},
+	{"HP Mix", "MIC22HP Playback Switch",		"MIC2 PGA"},
+	{"HP Mix", "DAC2HP Playback Switch",		"I2S Mix"},
+
+	/* speaker mixer */
+	{"Speaker Mix", "LI2SPK Playback Switch",	"Line Mix"},
+	{"Speaker Mix", "AUXI2SPK Playback Switch",	"AuxI Mix"},
+	{"Speaker Mix", "MIC12SPK Playback Switch",	"MIC1 PGA"},
+	{"Speaker Mix", "MIC22SPK Playback Switch",	"MIC2 PGA"},
+	{"Speaker Mix", "DAC2SPK Playback Switch",	"I2S Mix"},
+
+	/* mono mixer */
+	{"Mono Mix", "ADC2MONO_L Playback Switch",	"Left Capture Mix"},
+	{"Mono Mix", "ADC2MONO_R Playback Switch",	"Right Capture Mix"},
+	{"Mono Mix", "LI2MONO Playback Switch",		"Line Mix"},
+	{"Mono Mix", "AUXI2MONO Playback Switch",	"AuxI Mix"},
+	{"Mono Mix", "MIC12MONO Playback Switch",	"MIC1 PGA"},
+	{"Mono Mix", "MIC22MONO Playback Switch",	"MIC2 PGA"},
+	{"Mono Mix", "DAC2MONO Playback Switch",	"I2S Mix"},
+
+	/* Left record mixer */
+	{"Left Capture Mix", "LineInL Capture Switch",	"LINEINL"},
+	{"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
+	{"Left Capture Mix", "Mic1 Capture Switch",	"MIC1 Pre Amp"},
+	{"Left Capture Mix", "Mic2 Capture Switch",	"MIC2 Pre Amp"},
+	{"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
+	{"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
+	{"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
+
+	/*Right record mixer */
+	{"Right Capture Mix", "LineInR Capture Switch",	"LINEINR"},
+	{"Right Capture Mix", "Right AuxI Capture Switch",	"AUXINR"},
+	{"Right Capture Mix", "Mic1 Capture Switch",	"MIC1 Pre Amp"},
+	{"Right Capture Mix", "Mic2 Capture Switch",	"MIC2 Pre Amp"},
+	{"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
+	{"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
+	{"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
+
+	/* headphone left mux */
+	{"Left Headphone Mux", "HP Left Mix",		"HPL Mix"},
+	{"Left Headphone Mux", "Vmid",			"Vmid"},
+
+	/* headphone right mux */
+	{"Right Headphone Mux", "HP Right Mix",		"HPR Mix"},
+	{"Right Headphone Mux", "Vmid",			"Vmid"},
+
+	/* speaker out mux */
+	{"SpeakerOut Mux", "Vmid",			"Vmid"},
+	{"SpeakerOut Mux", "HPOut Mix",			"HPOut Mix"},
+	{"SpeakerOut Mux", "Speaker Mix",		"Speaker Mix"},
+	{"SpeakerOut Mux", "Mono Mix",			"Mono Mix"},
+
+	/* Mono/Aux Out mux */
+	{"AuxOut Mux", "Vmid",				"Vmid"},
+	{"AuxOut Mux", "HPOut Mix",			"HPOut Mix"},
+	{"AuxOut Mux", "Speaker Mix",			"Speaker Mix"},
+	{"AuxOut Mux", "Mono Mix",			"Mono Mix"},
+
+	/* output pga */
+	{"HPL", NULL,					"Left Headphone"},
+	{"Left Headphone", NULL,			"Left Headphone Mux"},
+	{"HPR", NULL,					"Right Headphone"},
+	{"Right Headphone", NULL,			"Right Headphone Mux"},
+	{"Left AuxOut", NULL,				"AuxOut Mux"},
+	{"Right AuxOut", NULL,				"AuxOut Mux"},
+
+	/* input pga */
+	{"Left LineIn", NULL,				"LINEINL"},
+	{"Right LineIn", NULL,				"LINEINR"},
+	{"Left AuxI", NULL,				"AUXINL"},
+	{"Right AuxI", NULL,				"AUXINR"},
+	{"MIC1 Pre Amp", NULL,				"MIC1"},
+	{"MIC2 Pre Amp", NULL,				"MIC2"},
+	{"MIC1 PGA", NULL,				"MIC1 Pre Amp"},
+	{"MIC2 PGA", NULL,				"MIC2 Pre Amp"},
+
+	/* left ADC */
+	{"Left ADC", NULL,				"Left Capture Mix"},
+
+	/* right ADC */
+	{"Right ADC", NULL,				"Right Capture Mix"},
+
+	{"SpeakerOut N Mux", "RN/-R",			"SpeakerOut"},
+	{"SpeakerOut N Mux", "RP/+R",			"SpeakerOut"},
+	{"SpeakerOut N Mux", "LN/-R",			"SpeakerOut"},
+	{"SpeakerOut N Mux", "Vmid",			"Vmid"},
+
+	{"SPKOUT", NULL,				"SpeakerOut"},
+	{"SPKOUTN", NULL,				"SpeakerOut N Mux"},
+};
+
+static const struct snd_soc_dapm_route intercon_spk[] = {
+	{"SpeakerOut", NULL,				"SpeakerOut Mux"},
+};
+
+static const struct snd_soc_dapm_route intercon_amp_spk[] = {
+	{"AB Amp", NULL,				"SpeakerOut Mux"},
+	{"D Amp", NULL,					"SpeakerOut Mux"},
+	{"AB-D Amp Mux", "AB Amp",			"AB Amp"},
+	{"AB-D Amp Mux", "D Amp",			"D Amp"},
+	{"SpeakerOut", NULL,				"AB-D Amp Mux"},
+};
+
+/* PLL divisors */
+struct _pll_div {
+	u32 pll_in;
+	u32 pll_out;
+	u16 regvalue;
+};
+
+/* Note : pll code from original alc5623 driver. Not sure of how good it is */
+/* usefull only for master mode */
+static const struct _pll_div codec_master_pll_div[] = {
+
+	{  2048000,  8192000,	0x0ea0},
+	{  3686400,  8192000,	0x4e27},
+	{ 12000000,  8192000,	0x456b},
+	{ 13000000,  8192000,	0x495f},
+	{ 13100000,  8192000,	0x0320},
+	{  2048000,  11289600,	0xf637},
+	{  3686400,  11289600,	0x2f22},
+	{ 12000000,  11289600,	0x3e2f},
+	{ 13000000,  11289600,	0x4d5b},
+	{ 13100000,  11289600,	0x363b},
+	{  2048000,  16384000,	0x1ea0},
+	{  3686400,  16384000,	0x9e27},
+	{ 12000000,  16384000,	0x452b},
+	{ 13000000,  16384000,	0x542f},
+	{ 13100000,  16384000,	0x03a0},
+	{  2048000,  16934400,	0xe625},
+	{  3686400,  16934400,	0x9126},
+	{ 12000000,  16934400,	0x4d2c},
+	{ 13000000,  16934400,	0x742f},
+	{ 13100000,  16934400,	0x3c27},
+	{  2048000,  22579200,	0x2aa0},
+	{  3686400,  22579200,	0x2f20},
+	{ 12000000,  22579200,	0x7e2f},
+	{ 13000000,  22579200,	0x742f},
+	{ 13100000,  22579200,	0x3c27},
+	{  2048000,  24576000,	0x2ea0},
+	{  3686400,  24576000,	0xee27},
+	{ 12000000,  24576000,	0x2915},
+	{ 13000000,  24576000,	0x772e},
+	{ 13100000,  24576000,	0x0d20},
+};
+
+static const struct _pll_div codec_slave_pll_div[] = {
+
+	{  1024000,  16384000,  0x3ea0},
+	{  1411200,  22579200,	0x3ea0},
+	{  1536000,  24576000,	0x3ea0},
+	{  2048000,  16384000,  0x1ea0},
+	{  2822400,  22579200,	0x1ea0},
+	{  3072000,  24576000,	0x1ea0},
+
+};
+
+static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
+{
+	int i;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int gbl_clk = 0, pll_div = 0;
+	u16 reg;
+
+	if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
+		return -ENODEV;
+
+	/* Disable PLL power */
+	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
+				ALC5623_PWR_ADD2_PLL,
+				0);
+
+	/* pll is not used in slave mode */
+	reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
+	if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
+		return 0;
+
+	if (!freq_in || !freq_out)
+		return 0;
+
+	switch (pll_id) {
+	case ALC5623_PLL_FR_MCLK:
+		for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
+			if (codec_master_pll_div[i].pll_in == freq_in
+			   && codec_master_pll_div[i].pll_out == freq_out) {
+				/* PLL source from MCLK */
+				pll_div  = codec_master_pll_div[i].regvalue;
+				break;
+			}
+		}
+		break;
+	case ALC5623_PLL_FR_BCK:
+		for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
+			if (codec_slave_pll_div[i].pll_in == freq_in
+			   && codec_slave_pll_div[i].pll_out == freq_out) {
+				/* PLL source from Bitclk */
+				gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
+				pll_div = codec_slave_pll_div[i].regvalue;
+				break;
+			}
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (!pll_div)
+		return -EINVAL;
+
+	snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
+	snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
+	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
+				ALC5623_PWR_ADD2_PLL,
+				ALC5623_PWR_ADD2_PLL);
+	gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
+	snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
+
+	return 0;
+}
+
+struct _coeff_div {
+	u16 fs;
+	u16 regvalue;
+};
+
+/* codec hifi mclk (after PLL) clock divider coefficients */
+/* values inspired from column BCLK=32Fs of Appendix A table */
+static const struct _coeff_div coeff_div[] = {
+	{256*8, 0x3a69},
+	{384*8, 0x3c6b},
+	{256*4, 0x2a69},
+	{384*4, 0x2c6b},
+	{256*2, 0x1a69},
+	{384*2, 0x1c6b},
+	{256*1, 0x0a69},
+	{384*1, 0x0c6b},
+};
+
+static int get_coeff(struct snd_soc_codec *codec, int rate)
+{
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].fs * rate == alc5623->sysclk)
+			return i;
+	}
+	return -EINVAL;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int alc5623_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
+	switch (freq) {
+	case  8192000:
+	case 11289600:
+	case 12288000:
+	case 16384000:
+	case 16934400:
+	case 18432000:
+	case 22579200:
+	case 24576000:
+		alc5623->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface = ALC5623_DAI_SDP_MASTER_MODE;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		iface = ALC5623_DAI_SDP_SLAVE_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= ALC5623_DAI_I2S_DF_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface |= ALC5623_DAI_I2S_DF_RIGHT;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= ALC5623_DAI_I2S_DF_LEFT;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= ALC5623_DAI_I2S_DF_PCM;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
+}
+
+static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+	int coeff, rate;
+	u16 iface;
+
+	iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
+	iface &= ~ALC5623_DAI_I2S_DL_MASK;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		iface |= ALC5623_DAI_I2S_DL_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= ALC5623_DAI_I2S_DL_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= ALC5623_DAI_I2S_DL_24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= ALC5623_DAI_I2S_DL_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set iface & srate */
+	snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
+	rate = params_rate(params);
+	coeff = get_coeff(codec, rate);
+	if (coeff < 0)
+		return -EINVAL;
+
+	coeff = coeff_div[coeff].regvalue;
+	dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
+		__func__, alc5623->sysclk, rate, coeff);
+	snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
+
+	return 0;
+}
+
+static int alc5623_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
+	u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
+
+	if (mute)
+		mute_reg |= hp_mute;
+
+	return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
+}
+
+#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
+	| ALC5623_PWR_ADD2_DAC_REF_CIR)
+
+#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
+	| ALC5623_PWR_ADD3_MIC1_BOOST_AD)
+
+#define ALC5623_ADD1_POWER_EN \
+	(ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
+	| ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
+	| ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
+
+#define ALC5623_ADD1_POWER_EN_5622 \
+	(ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
+	| ALC5623_PWR_ADD1_HP_OUT_AMP)
+
+static void enable_power_depop(struct snd_soc_codec *codec)
+{
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
+	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
+				ALC5623_PWR_ADD1_SOFTGEN_EN,
+				ALC5623_PWR_ADD1_SOFTGEN_EN);
+
+	snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
+
+	snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
+				ALC5623_MISC_HP_DEPOP_MODE2_EN,
+				ALC5623_MISC_HP_DEPOP_MODE2_EN);
+
+	msleep(500);
+
+	snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
+
+	/* avoid writing '1' into 5622 reserved bits */
+	if (alc5623->id == 0x22)
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
+			ALC5623_ADD1_POWER_EN_5622);
+	else
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
+			ALC5623_ADD1_POWER_EN);
+
+	/* disable HP Depop2 */
+	snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
+				ALC5623_MISC_HP_DEPOP_MODE2_EN,
+				0);
+
+}
+
+static int alc5623_set_bias_level(struct snd_soc_codec *codec,
+				      enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		enable_power_depop(codec);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* everything off except vref/vmid, */
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
+				ALC5623_PWR_ADD2_VREF);
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
+				ALC5623_PWR_ADD3_MAIN_BIAS);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
+		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define ALC5623_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE \
+			| SNDRV_PCM_FMTBIT_S24_LE \
+			| SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops alc5623_dai_ops = {
+		.hw_params = alc5623_pcm_hw_params,
+		.digital_mute = alc5623_mute,
+		.set_fmt = alc5623_set_dai_fmt,
+		.set_sysclk = alc5623_set_dai_sysclk,
+		.set_pll = alc5623_set_dai_pll,
+};
+
+static struct snd_soc_dai_driver alc5623_dai = {
+	.name = "alc5623-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rate_min =	8000,
+		.rate_max =	48000,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = ALC5623_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rate_min =	8000,
+		.rate_max =	48000,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = ALC5623_FORMATS,},
+
+	.ops = &alc5623_dai_ops,
+};
+
+static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+{
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int alc5623_resume(struct snd_soc_codec *codec)
+{
+	int i, step = codec->driver->reg_cache_step;
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
+		snd_soc_write(codec, i, cache[i]);
+
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* charge alc5623 caps */
+	if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
+		alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+		codec->dapm.bias_level = SND_SOC_BIAS_ON;
+		alc5623_set_bias_level(codec, codec->dapm.bias_level);
+	}
+
+	return 0;
+}
+
+static int alc5623_probe(struct snd_soc_codec *codec)
+{
+	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	alc5623_reset(codec);
+	alc5623_fill_cache(codec);
+
+	/* power on device */
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	if (alc5623->add_ctrl) {
+		snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
+				alc5623->add_ctrl);
+	}
+
+	if (alc5623->jack_det_ctrl) {
+		snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
+				alc5623->jack_det_ctrl);
+	}
+
+	switch (alc5623->id) {
+	default:
+	case 0x21:
+		snd_soc_add_controls(codec, rt5621_vol_snd_controls,
+			ARRAY_SIZE(rt5621_vol_snd_controls));
+		break;
+	case 0x22:
+		snd_soc_add_controls(codec, rt5622_vol_snd_controls,
+			ARRAY_SIZE(rt5622_vol_snd_controls));
+		break;
+	case 0x23:
+		snd_soc_add_controls(codec, alc5623_vol_snd_controls,
+			ARRAY_SIZE(alc5623_vol_snd_controls));
+		break;
+	}
+
+	snd_soc_add_controls(codec, alc5623_snd_controls,
+			ARRAY_SIZE(alc5623_snd_controls));
+
+	snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
+					ARRAY_SIZE(alc5623_dapm_widgets));
+
+	/* set up audio path interconnects */
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+
+	switch (alc5623->id) {
+	default:
+	case 0x21:
+	case 0x22:
+		snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
+					ARRAY_SIZE(alc5623_dapm_amp_widgets));
+		snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
+					ARRAY_SIZE(intercon_amp_spk));
+		break;
+	case 0x23:
+		snd_soc_dapm_add_routes(dapm, intercon_spk,
+					ARRAY_SIZE(intercon_spk));
+		break;
+	}
+
+	return ret;
+}
+
+/* power down chip */
+static int alc5623_remove(struct snd_soc_codec *codec)
+{
+	alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
+	.probe = alc5623_probe,
+	.remove = alc5623_remove,
+	.suspend = alc5623_suspend,
+	.resume = alc5623_resume,
+	.set_bias_level = alc5623_set_bias_level,
+	.reg_cache_size = ALC5623_VENDOR_ID2+2,
+	.reg_word_size = sizeof(u16),
+	.reg_cache_step = 2,
+};
+
+/*
+ * ALC5623 2 wire address is determined by A1 pin
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+static int alc5623_i2c_probe(struct i2c_client *client,
+				const struct i2c_device_id *id)
+{
+	struct alc5623_platform_data *pdata;
+	struct alc5623_priv *alc5623;
+	int ret, vid1, vid2;
+
+	vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
+	if (vid1 < 0) {
+		dev_err(&client->dev, "failed to read I2C\n");
+		return -EIO;
+	}
+	vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
+
+	vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
+	if (vid2 < 0) {
+		dev_err(&client->dev, "failed to read I2C\n");
+		return -EIO;
+	}
+
+	if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
+		dev_err(&client->dev, "unknown or wrong codec\n");
+		dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
+				0x10ec, id->driver_data,
+				vid1, vid2);
+		return -ENODEV;
+	}
+
+	dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
+
+	alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL);
+	if (alc5623 == NULL) {
+		ret = -ENOMEM;
+		goto err;
+	}
+
+	pdata = client->dev.platform_data;
+	if (pdata) {
+		alc5623->add_ctrl = pdata->add_ctrl;
+		alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
+	}
+
+	alc5623->id = vid2;
+	switch (alc5623->id) {
+	case 0x21:
+		alc5623_dai.name = "alc5621-hifi";
+		break;
+	case 0x22:
+		alc5623_dai.name = "alc5622-hifi";
+		break;
+	default:
+	case 0x23:
+		alc5623_dai.name = "alc5623-hifi";
+		break;
+	}
+
+	i2c_set_clientdata(client, alc5623);
+	alc5623->control_data = client;
+	alc5623->control_type = SND_SOC_I2C;
+	mutex_init(&alc5623->mutex);
+
+	ret =  snd_soc_register_codec(&client->dev,
+		&soc_codec_device_alc5623, &alc5623_dai, 1);
+	if (ret != 0) {
+		dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	return 0;
+
+err:
+	return ret;
+}
+
+static int alc5623_i2c_remove(struct i2c_client *client)
+{
+	struct alc5623_priv *alc5623 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+	kfree(alc5623);
+	return 0;
+}
+
+static const struct i2c_device_id alc5623_i2c_table[] = {
+	{"alc5621", 0x21},
+	{"alc5622", 0x22},
+	{"alc5623", 0x23},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
+
+/*  i2c codec control layer */
+static struct i2c_driver alc5623_i2c_driver = {
+	.driver = {
+		.name = "alc562x-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = alc5623_i2c_probe,
+	.remove =  __devexit_p(alc5623_i2c_remove),
+	.id_table = alc5623_i2c_table,
+};
+
+static int __init alc5623_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&alc5623_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "%s: can't add i2c driver", __func__);
+		return ret;
+	}
+
+	return ret;
+}
+module_init(alc5623_modinit);
+
+static void __exit alc5623_modexit(void)
+{
+	i2c_del_driver(&alc5623_i2c_driver);
+}
+module_exit(alc5623_modexit);
+
+MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
+MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/alc5623.h b/sound/soc/codecs/alc5623.h
new file mode 100644
index 0000000..f3d6826
--- /dev/null
+++ b/sound/soc/codecs/alc5623.h
@@ -0,0 +1,161 @@
+/*
+ * alc5623.h  --  alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * Author: flove <flove@realtek.com>
+ * Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _ALC5623_H
+#define _ALC5623_H
+
+#define ALC5623_RESET				0x00
+/*				5621 5622 5623  */
+/* speaker output vol		   2    2       */
+/* line output vol                      4    2  */
+/* HP output vol		   4    0    4  */
+#define ALC5623_SPK_OUT_VOL			0x02
+#define ALC5623_HP_OUT_VOL			0x04
+#define ALC5623_MONO_AUX_OUT_VOL		0x06
+#define ALC5623_AUXIN_VOL			0x08
+#define ALC5623_LINE_IN_VOL			0x0A
+#define ALC5623_STEREO_DAC_VOL			0x0C
+#define ALC5623_MIC_VOL				0x0E
+#define ALC5623_MIC_ROUTING_CTRL		0x10
+#define ALC5623_ADC_REC_GAIN			0x12
+#define ALC5623_ADC_REC_MIXER			0x14
+#define ALC5623_SOFT_VOL_CTRL_TIME		0x16
+/* ALC5623_OUTPUT_MIXER_CTRL :			*/
+/* same remark as for reg 2 line vs speaker	*/
+#define ALC5623_OUTPUT_MIXER_CTRL		0x1C
+#define ALC5623_MIC_CTRL			0x22
+
+#define	ALC5623_DAI_CONTROL			0x34
+#define ALC5623_DAI_SDP_MASTER_MODE		(0 << 15)
+#define ALC5623_DAI_SDP_SLAVE_MODE		(1 << 15)
+#define ALC5623_DAI_I2S_PCM_MODE		(1 << 14)
+#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL	(1 <<  7)
+#define ALC5623_DAI_ADC_DATA_L_R_SWAP		(1 <<  5)
+#define ALC5623_DAI_DAC_DATA_L_R_SWAP		(1 <<  4)
+#define ALC5623_DAI_I2S_DL_MASK			(3 <<  2)
+#define ALC5623_DAI_I2S_DL_32			(3 <<  2)
+#define	ALC5623_DAI_I2S_DL_24			(2 <<  2)
+#define ALC5623_DAI_I2S_DL_20			(1 <<  2)
+#define ALC5623_DAI_I2S_DL_16			(0 <<  2)
+#define ALC5623_DAI_I2S_DF_PCM			(3 <<  0)
+#define	ALC5623_DAI_I2S_DF_LEFT			(2 <<  0)
+#define ALC5623_DAI_I2S_DF_RIGHT		(1 <<  0)
+#define ALC5623_DAI_I2S_DF_I2S			(0 <<  0)
+
+#define ALC5623_STEREO_AD_DA_CLK_CTRL		0x36
+#define	ALC5623_COMPANDING_CTRL			0x38
+
+#define	ALC5623_PWR_MANAG_ADD1			0x3A
+#define ALC5623_PWR_ADD1_MAIN_I2S_EN		(1 << 15)
+#define ALC5623_PWR_ADD1_ZC_DET_PD_EN		(1 << 14)
+#define ALC5623_PWR_ADD1_MIC1_BIAS_EN		(1 << 11)
+#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN	(1 << 10)
+#define ALC5623_PWR_ADD1_SOFTGEN_EN		(1 <<  8) /* rsvd on 5622 */
+#define	ALC5623_PWR_ADD1_DEPOP_BUF_HP		(1 <<  6) /* rsvd on 5622 */
+#define	ALC5623_PWR_ADD1_HP_OUT_AMP		(1 <<  5)
+#define	ALC5623_PWR_ADD1_HP_OUT_ENH_AMP		(1 <<  4) /* rsvd on 5622 */
+#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX		(1 <<  2)
+#define ALC5623_PWR_ADD1_AUX_OUT_AMP		(1 <<  1)
+#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP	(1 <<  0) /* rsvd on 5622 */
+
+#define ALC5623_PWR_MANAG_ADD2			0x3C
+#define ALC5623_PWR_ADD2_LINEOUT		(1 << 15) /* rt5623 */
+#define ALC5623_PWR_ADD2_CLASS_AB		(1 << 15) /* rt5621 */
+#define ALC5623_PWR_ADD2_CLASS_D		(1 << 14) /* rt5621 */
+#define ALC5623_PWR_ADD2_VREF			(1 << 13)
+#define ALC5623_PWR_ADD2_PLL			(1 << 12)
+#define ALC5623_PWR_ADD2_DAC_REF_CIR		(1 << 10)
+#define ALC5623_PWR_ADD2_L_DAC_CLK		(1 <<  9)
+#define ALC5623_PWR_ADD2_R_DAC_CLK		(1 <<  8)
+#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN		(1 <<  7)
+#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN		(1 <<  6)
+#define ALC5623_PWR_ADD2_L_HP_MIXER		(1 <<  5)
+#define ALC5623_PWR_ADD2_R_HP_MIXER		(1 <<  4)
+#define ALC5623_PWR_ADD2_SPK_MIXER		(1 <<  3)
+#define ALC5623_PWR_ADD2_MONO_MIXER		(1 <<  2)
+#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER	(1 <<  1)
+#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER	(1 <<  0)
+
+#define ALC5623_PWR_MANAG_ADD3			0x3E
+#define ALC5623_PWR_ADD3_MAIN_BIAS		(1 << 15)
+#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP	(1 << 14)
+#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP	(1 << 13)
+#define ALC5623_PWR_ADD3_SPK_OUT		(1 << 12)
+#define ALC5623_PWR_ADD3_HP_L_OUT_VOL		(1 << 10)
+#define ALC5623_PWR_ADD3_HP_R_OUT_VOL		(1 <<  9)
+#define ALC5623_PWR_ADD3_LINEIN_L_VOL		(1 <<  7)
+#define ALC5623_PWR_ADD3_LINEIN_R_VOL		(1 <<  6)
+#define ALC5623_PWR_ADD3_AUXIN_L_VOL		(1 <<  5)
+#define ALC5623_PWR_ADD3_AUXIN_R_VOL		(1 <<  4)
+#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL		(1 <<  3)
+#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL		(1 <<  2)
+#define ALC5623_PWR_ADD3_MIC1_BOOST_AD		(1 <<  1)
+#define ALC5623_PWR_ADD3_MIC2_BOOST_AD		(1 <<  0)
+
+#define ALC5623_ADD_CTRL_REG			0x40
+
+#define	ALC5623_GLOBAL_CLK_CTRL_REG		0x42
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL	(1 << 15)
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK	(0 << 15)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK	(1 << 14)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK	(0 << 14)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8	(3 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4	(2 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2	(1 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1	(0 <<  1)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV2		(1 <<  0)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV1		(0 <<  0)
+
+#define ALC5623_PLL_CTRL			0x44
+#define ALC5623_PLL_CTRL_N_VAL(n)		(((n)&0xff) << 8)
+#define ALC5623_PLL_CTRL_K_VAL(k)		(((k)&0x7)  << 4)
+#define ALC5623_PLL_CTRL_M_VAL(m)		((m)&0xf)
+
+#define ALC5623_GPIO_OUTPUT_PIN_CTRL		0x4A
+#define ALC5623_GPIO_PIN_CONFIG			0x4C
+#define ALC5623_GPIO_PIN_POLARITY		0x4E
+#define ALC5623_GPIO_PIN_STICKY			0x50
+#define ALC5623_GPIO_PIN_WAKEUP			0x52
+#define ALC5623_GPIO_PIN_STATUS			0x54
+#define ALC5623_GPIO_PIN_SHARING		0x56
+#define	ALC5623_OVER_CURR_STATUS		0x58
+#define ALC5623_JACK_DET_CTRL			0x5A
+
+#define ALC5623_MISC_CTRL			0x5E
+#define ALC5623_MISC_DISABLE_FAST_VREG		(1 << 15)
+#define ALC5623_MISC_SPK_CLASS_AB_OC_PD		(1 << 13) /* 5621 */
+#define ALC5623_MISC_SPK_CLASS_AB_OC_DET	(1 << 12) /* 5621 */
+#define ALC5623_MISC_HP_DEPOP_MODE3_EN		(1 << 10)
+#define ALC5623_MISC_HP_DEPOP_MODE2_EN		(1 <<  9)
+#define ALC5623_MISC_HP_DEPOP_MODE1_EN		(1 <<  8)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN	(1 <<  6)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN	(1 <<  5)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN	(1 <<  4)
+#define ALC5623_MISC_M_DAC_L_INPUT		(1 <<  3)
+#define ALC5623_MISC_M_DAC_R_INPUT		(1 <<  2)
+#define ALC5623_MISC_IRQOUT_INV_CTRL		(1 <<  0)
+
+#define	ALC5623_PSEDUEO_SPATIAL_CTRL		0x60
+#define ALC5623_EQ_CTRL				0x62
+#define ALC5623_EQ_MODE_ENABLE			0x66
+#define ALC5623_AVC_CTRL			0x68
+#define ALC5623_HID_CTRL_INDEX			0x6A
+#define ALC5623_HID_CTRL_DATA			0x6C
+#define ALC5623_VENDOR_ID1			0x7C
+#define ALC5623_VENDOR_ID2			0x7E
+
+#define ALC5623_PLL_FR_MCLK			0
+#define ALC5623_PLL_FR_BCK			1
+#endif
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 8236439..98b9e52 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -116,7 +116,7 @@
 			     DAVINCI_VC_REG12_POWER_ALL_OFF);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index cb086ea..a7fdca3 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -519,6 +519,7 @@
 static int cs42l51_probe(struct snd_soc_codec *codec)
 {
 	struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret, reg;
 
 	codec->control_data = cs42l51->control_data;
@@ -550,9 +551,9 @@
 
 	snd_soc_add_controls(codec, cs42l51_snd_controls,
 		ARRAY_SIZE(cs42l51_snd_controls));
-	snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, cs42l51_dapm_widgets,
 		ARRAY_SIZE(cs42l51_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, cs42l51_routes,
+	snd_soc_dapm_add_routes(dapm, cs42l51_routes,
 		ARRAY_SIZE(cs42l51_routes));
 
 	return 0;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index e8d27c8..a9521ac 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -18,7 +18,7 @@
 
 #include <sound/core.h>
 #include <sound/initval.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
 
 #include "cx20442.h"
 
@@ -89,10 +89,11 @@
 
 static int cx20442_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, cx20442_dapm_widgets,
-				  ARRAY_SIZE(cx20442_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, cx20442_audio_map,
+	snd_soc_dapm_new_controls(dapm, cx20442_dapm_widgets,
+				  ARRAY_SIZE(cx20442_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, cx20442_audio_map,
 				ARRAY_SIZE(cx20442_audio_map));
 
 	return 0;
@@ -263,7 +264,7 @@
 	/* Prevent the codec driver from further accessing the modem */
 	codec->hw_write = NULL;
 	cx20442->control_data = NULL;
-	codec->pop_time = 0;
+	codec->card->pop_time = 0;
 }
 
 /* Line discipline .hangup() */
@@ -291,7 +292,7 @@
 		/* Set up codec driver access to modem controls */
 		cx20442->control_data = tty;
 		codec->hw_write = (hw_write_t)tty->ops->write;
-		codec->pop_time = 1;
+		codec->card->pop_time = 1;
 	}
 }
 
@@ -348,7 +349,7 @@
 
 	cx20442->control_data = NULL;
 	codec->hw_write = NULL;
-	codec->pop_time = 0;
+	codec->card->pop_time = 0;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 58bb9b9..92fd9d7 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -21,7 +21,7 @@
 #include <linux/slab.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
 
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 16253ec..8a45562 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -266,7 +266,7 @@
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* The only way to clear the suspend flag is to reset the codec */
-		if (codec->bias_level == SND_SOC_BIAS_OFF)
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
 			jz4740_codec_wakeup(codec);
 
 		mask = JZ4740_CODEC_1_VREF_DISABLE |
@@ -288,23 +288,25 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
 
 static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
 			JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE);
 
 	snd_soc_add_controls(codec, jz4740_codec_controls,
 		ARRAY_SIZE(jz4740_codec_controls));
 
-	snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, jz4740_codec_dapm_widgets,
 		ARRAY_SIZE(jz4740_codec_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes,
+	snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
 		ARRAY_SIZE(jz4740_codec_dapm_routes));
 
 	snd_soc_dapm_new_widgets(codec);
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index bc22ee9..ef06007 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1224,15 +1224,17 @@
 
 static int max98088_add_widgets(struct snd_soc_codec *codec)
 {
-       snd_soc_dapm_new_controls(codec, max98088_dapm_widgets,
+       struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+       snd_soc_dapm_new_controls(dapm, max98088_dapm_widgets,
                                  ARRAY_SIZE(max98088_dapm_widgets));
 
-       snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+       snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
        snd_soc_add_controls(codec, max98088_snd_controls,
                             ARRAY_SIZE(max98088_snd_controls));
 
-       snd_soc_dapm_new_widgets(codec);
+       snd_soc_dapm_new_widgets(dapm);
        return 0;
 }
 
@@ -1617,7 +1619,7 @@
                break;
 
        case SND_SOC_BIAS_STANDBY:
-               if (codec->bias_level == SND_SOC_BIAS_OFF)
+               if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
                        max98088_sync_cache(codec);
 
                snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
@@ -1630,7 +1632,7 @@
                codec->cache_sync = 1;
                break;
        }
-       codec->bias_level = level;
+       codec->dapm.bias_level = level;
        return 0;
 }
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 6f38d61..adbc3e8 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -207,10 +207,11 @@
 
 static int ssm2602_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets,
-				  ARRAY_SIZE(ssm2602_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn));
+	snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets,
+				  ARRAY_SIZE(ssm2602_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_conn, ARRAY_SIZE(audio_conn));
 
 	return 0;
 }
@@ -493,7 +494,7 @@
 		break;
 
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 00d67cc..8aad3a2 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -24,6 +24,7 @@
 #include <sound/initval.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 
 #include "stac9766.h"
@@ -236,7 +237,7 @@
 		stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index e8652b1..d9d8e84 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -391,11 +391,12 @@
 
 static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
-				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
 	/* set up audio path interconnects */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -574,7 +575,7 @@
 		tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index fc68779..df726a5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -183,7 +183,7 @@
 
 	if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) {
 		/* find dapm widget path assoc with kcontrol */
-		list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+		list_for_each_entry(path, &widget->dapm->paths, list) {
 			if (path->kcontrol != kcontrol)
 				continue;
 
@@ -199,7 +199,7 @@
 		}
 
 		if (found)
-			snd_soc_dapm_sync(widget->codec);
+			snd_soc_dapm_sync(widget->dapm);
 	}
 
 	ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
@@ -788,17 +788,19 @@
 static int aic3x_add_widgets(struct snd_soc_codec *codec)
 {
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
 				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* set up audio path interconnects */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	if (aic3x->model == AIC3X_MODEL_3007) {
-		snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets,
+		snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets,
 			ARRAY_SIZE(aic3007_dapm_widgets));
-		snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007));
+		snd_soc_dapm_add_routes(dapm, intercon_3007,
+					ARRAY_SIZE(intercon_3007));
 	}
 
 	return 0;
@@ -1135,7 +1137,7 @@
 	case SND_SOC_BIAS_ON:
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->bias_level == SND_SOC_BIAS_STANDBY &&
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY &&
 		    aic3x->master) {
 			/* enable pll */
 			reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
@@ -1146,7 +1148,7 @@
 	case SND_SOC_BIAS_STANDBY:
 		if (!aic3x->power)
 			aic3x_set_power(codec, 1);
-		if (codec->bias_level == SND_SOC_BIAS_PREPARE &&
+		if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE &&
 		    aic3x->master) {
 			/* disable pll */
 			reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
@@ -1159,7 +1161,7 @@
 			aic3x_set_power(codec, 0);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1351,7 +1353,7 @@
 
 	codec->control_data = aic3x->control_data;
 	aic3x->codec = codec;
-	codec->idle_bias_off = 1;
+	codec->dapm.idle_bias_off = 1;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type);
 	if (ret != 0) {
@@ -1417,7 +1419,6 @@
 	if (aic3x->gpio_reset >= 0)
 		gpio_free(aic3x->gpio_reset);
 err_gpio:
-	kfree(aic3x);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index c5ab8c8..7149c14 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -628,11 +628,12 @@
 
 static int dac33_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, dac33_dapm_widgets,
-				  ARRAY_SIZE(dac33_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
+	snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets,
+				  ARRAY_SIZE(dac33_dapm_widgets));
 	/* set up audio path interconnects */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -649,7 +650,7 @@
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Coming from OFF, switch on the codec */
 			ret = dac33_hard_power(codec, 1);
 			if (ret != 0)
@@ -660,14 +661,14 @@
 		break;
 	case SND_SOC_BIAS_OFF:
 		/* Do not power off, when the codec is already off */
-		if (codec->bias_level == SND_SOC_BIAS_OFF)
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
 			return 0;
 		ret = dac33_hard_power(codec, 0);
 		if (ret != 0)
 			return ret;
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1415,7 +1416,7 @@
 
 	codec->control_data = dac33->control_data;
 	codec->hw_write = (hw_write_t) i2c_master_send;
-	codec->idle_bias_off = 1;
+	codec->dapm.idle_bias_off = 1;
 	dac33->codec = codec;
 
 	/* Read the tlv320dac33 ID registers */
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index ee4fb20..f9a92ea 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -388,16 +388,17 @@
 int tpa6130a2_add_controls(struct snd_soc_codec *codec)
 {
 	struct	tpa6130a2_data *data;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	if (tpa6130a2_client == NULL)
 		return -ENODEV;
 
 	data = i2c_get_clientdata(tpa6130a2_client);
 
-	snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tpa6130a2_dapm_widgets,
 				ARRAY_SIZE(tpa6130a2_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	if (data->id == TPA6140A2)
 		return snd_soc_add_controls(codec, tpa6140a2_controls,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index cbebec6..f4602e8 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1621,10 +1621,11 @@
 
 static int twl4030_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets,
-				 ARRAY_SIZE(twl4030_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, twl4030_dapm_widgets,
+				 ARRAY_SIZE(twl4030_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -1638,14 +1639,14 @@
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF)
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
 			twl4030_codec_enable(codec, 1);
 		break;
 	case SND_SOC_BIAS_OFF:
 		twl4030_codec_enable(codec, 0);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -2245,7 +2246,7 @@
 	snd_soc_codec_set_drvdata(codec, twl4030);
 	/* Set the defaults, and power up the codec */
 	twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
-	codec->idle_bias_off = 1;
+	codec->dapm.idle_bias_off = 1;
 
 	twl4030_init_chip(codec);
 
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 10f6e52..0dd2d53 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -641,12 +641,12 @@
 
 static int twl6040_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets,
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_new_controls(dapm, twl6040_dapm_widgets,
 				 ARRAY_SIZE(twl6040_dapm_widgets));
-
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
-
-	snd_soc_dapm_new_widgets(codec);
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_widgets(dapm);
 
 	return 0;
 }
@@ -739,7 +739,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 7540a50..8ea81d4 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -389,7 +389,7 @@
 			pd->power(0);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 0c6c725..cd6dd19 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -414,10 +414,11 @@
 
 static int uda1380_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
-				  ARRAY_SIZE(uda1380_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
+				  ARRAY_SIZE(uda1380_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -603,7 +604,7 @@
 	int reg;
 	struct uda1380_platform_data *pdata = codec->dev->platform_data;
 
-	if (codec->bias_level == level)
+	if (codec->dapm.bias_level == level)
 		return 0;
 
 	switch (level) {
@@ -613,7 +614,7 @@
 		uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			if (gpio_is_valid(pdata->gpio_power)) {
 				gpio_set_value(pdata->gpio_power, 1);
 				mdelay(1);
@@ -636,7 +637,7 @@
 		for (reg = UDA1380_MVOL; reg < UDA1380_CACHEREGNUM; reg++)
 			set_bit(reg - 0x10, &uda1380_cache_dirty);
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 4bcd168..9277d8d 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -705,6 +705,7 @@
 /* Called from the machine driver */
 int wm2000_add_controls(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	if (!wm2000_i2c) {
@@ -712,12 +713,12 @@
 		return -ENODEV;
 	}
 
-	ret = snd_soc_dapm_new_controls(codec, wm2000_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, wm2000_dapm_widgets,
 					ARRAY_SIZE(wm2000_dapm_widgets));
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 7611add..d5e6e02 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -230,8 +230,9 @@
  */
 static void wm8350_pga_work(struct work_struct *work)
 {
-	struct snd_soc_codec *codec =
-	    container_of(work, struct snd_soc_codec, delayed_work.work);
+	struct snd_soc_dapm_context *dapm =
+	    container_of(work, struct snd_soc_dapm_context, delayed_work.work);
+	struct snd_soc_codec *codec = dapm->codec;
 	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out1 = &wm8350_data->out1,
 	    *out2 = &wm8350_data->out2;
@@ -302,8 +303,8 @@
 		out->ramp = WM8350_RAMP_UP;
 		out->active = 1;
 
-		if (!delayed_work_pending(&codec->delayed_work))
-			schedule_delayed_work(&codec->delayed_work,
+		if (!delayed_work_pending(&codec->dapm.delayed_work))
+			schedule_delayed_work(&codec->dapm.delayed_work,
 					      msecs_to_jiffies(1));
 		break;
 
@@ -311,8 +312,8 @@
 		out->ramp = WM8350_RAMP_DOWN;
 		out->active = 0;
 
-		if (!delayed_work_pending(&codec->delayed_work))
-			schedule_delayed_work(&codec->delayed_work,
+		if (!delayed_work_pending(&codec->dapm.delayed_work))
+			schedule_delayed_work(&codec->dapm.delayed_work,
 					      msecs_to_jiffies(1));
 		break;
 	}
@@ -786,9 +787,10 @@
 
 static int wm8350_add_widgets(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	ret = snd_soc_dapm_new_controls(codec,
+	ret = snd_soc_dapm_new_controls(dapm,
 					wm8350_dapm_widgets,
 					ARRAY_SIZE(wm8350_dapm_widgets));
 	if (ret != 0) {
@@ -797,7 +799,7 @@
 	}
 
 	/* set up audio paths */
-	ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 	if (ret != 0) {
 		dev_err(codec->dev, "DAPM route register failed\n");
 		return ret;
@@ -1184,7 +1186,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
 						    priv->supplies);
 			if (ret != 0)
@@ -1317,7 +1319,7 @@
 				       priv->supplies);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -1550,7 +1552,7 @@
 	/* Put the codec into reset if it wasn't already */
 	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
 
-	INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work);
+	INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8350_pga_work);
 
 	/* Enable the codec */
 	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
@@ -1642,12 +1644,12 @@
 	priv->mic.jack = NULL;
 
 	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(&codec->delayed_work);
+	ret = cancel_delayed_work(&codec->dapm.delayed_work);
 
 	/* if there was any work waiting then we run it now and
 	 * wait for its completion */
 	if (ret) {
-		schedule_delayed_work(&codec->delayed_work, 0);
+		schedule_delayed_work(&codec->dapm.delayed_work, 0);
 		flush_scheduled_work();
 	}
 
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 8502997..96927a4 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -911,10 +911,11 @@
 
 static int wm8400_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets,
-				  ARRAY_SIZE(wm8400_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8400_dapm_widgets,
+				  ARRAY_SIZE(wm8400_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -1219,7 +1220,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(power),
 						    &power[0]);
 			if (ret != 0) {
@@ -1306,7 +1307,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 8f10709..6b3833c 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -216,10 +216,11 @@
 
 static int wm8510_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets,
-				  ARRAY_SIZE(wm8510_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8510_dapm_widgets,
+				  ARRAY_SIZE(wm8510_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -478,7 +479,7 @@
 	case SND_SOC_BIAS_STANDBY:
 		power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
 
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Initial cap charge at VMID 5k */
 			snd_soc_write(codec, WM8510_POWER1, power1 | 0x3);
 			mdelay(100);
@@ -495,7 +496,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 712ef7c..d331888 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -110,10 +110,11 @@
 
 static int wm8523_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8523_dapm_widgets,
-				  ARRAY_SIZE(wm8523_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets,
+				  ARRAY_SIZE(wm8523_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -328,7 +329,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies),
 						    wm8523->supplies);
 			if (ret != 0) {
@@ -367,7 +368,7 @@
 				       wm8523->supplies);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index a2e0ed5..36c035b 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -302,10 +302,11 @@
 
 static int wm8580_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets,
-				  ARRAY_SIZE(wm8580_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets,
+				  ARRAY_SIZE(wm8580_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -767,7 +768,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Power up and get individual control of the DACs */
 			reg = snd_soc_read(codec, WM8580_PWRDN1);
 			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
@@ -785,7 +786,7 @@
 		snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -905,7 +906,7 @@
 	.set_bias_level = wm8580_set_bias_level,
 	.reg_cache_size = ARRAY_SIZE(wm8580_reg),
 	.reg_word_size = sizeof(u16),
-	.reg_cache_default = &wm8580_reg,
+	.reg_cache_default = wm8580_reg,
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 54fbd76..ea2daf4 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -93,10 +93,11 @@
 
 static int wm8711_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets,
-				  ARRAY_SIZE(wm8711_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, wm8711_dapm_widgets,
+				  ARRAY_SIZE(wm8711_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -318,7 +319,7 @@
 		snd_soc_write(codec, WM8711_PWR, 0xffff);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 075f35e..2393997 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -73,10 +73,11 @@
 
 static int wm8728_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets,
-				  ARRAY_SIZE(wm8728_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, wm8728_dapm_widgets,
+				  ARRAY_SIZE(wm8728_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -180,7 +181,7 @@
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Power everything up... */
 			reg = snd_soc_read(codec, WM8728_DACCTL);
 			snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4);
@@ -197,7 +198,7 @@
 		snd_soc_write(codec, WM8728_DACCTL, reg | 0x4);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 6313858..4b70958 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -165,10 +165,11 @@
 
 static int wm8731_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
-				  ARRAY_SIZE(wm8731_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -319,7 +320,7 @@
 		return -EINVAL;
 	}
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(&codec->dapm);
 
 	return 0;
 }
@@ -399,7 +400,7 @@
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies),
 						    wm8731->supplies);
 			if (ret != 0)
@@ -428,7 +429,7 @@
 				       wm8731->supplies);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -542,7 +543,6 @@
 err_regulator_get:
 	regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
 
-	kfree(wm8731);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 90e31e9..2543a26 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -95,10 +95,11 @@
 
 static int wm8741_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets,
-				  ARRAY_SIZE(wm8741_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, wm8741_dapm_widgets,
+				  ARRAY_SIZE(wm8741_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -455,7 +456,7 @@
 	.resume =	wm8741_resume,
 	.reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults),
 	.reg_word_size = sizeof(u16),
-	.reg_cache_default = &wm8741_reg_defaults,
+	.reg_cache_default = wm8741_reg_defaults,
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 6c924cd..178b967 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -399,10 +399,11 @@
 
 static int wm8750_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
-				  ARRAY_SIZE(wm8750_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
+				  ARRAY_SIZE(wm8750_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -615,7 +616,7 @@
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Set VMID to 5k */
 			snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
 
@@ -630,7 +631,7 @@
 		snd_soc_write(codec, WM8750_PWR1, 0x0001);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8f679a1..26096b4 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -670,10 +670,11 @@
 
 static int wm8753_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
-				  ARRAY_SIZE(wm8753_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -1292,7 +1293,7 @@
 		wm8753_write(codec, WM8753_PWR1, 0x0001);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -1482,9 +1483,11 @@
 
 static void wm8753_work(struct work_struct *work)
 {
-	struct snd_soc_codec *codec =
-		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8753_set_bias_level(codec, codec->bias_level);
+	struct snd_soc_dapm_context *dapm =
+		container_of(work, struct snd_soc_dapm_context,
+			     delayed_work.work);
+	struct snd_soc_codec *codec = dapm->codec;
+	wm8753_set_bias_level(codec, dapm->bias_level);
 }
 
 static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state)
@@ -1516,10 +1519,10 @@
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8753 caps */
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+	if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
 		wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
-		codec->bias_level = SND_SOC_BIAS_ON;
-		schedule_delayed_work(&codec->delayed_work,
+		codec->dapm.bias_level = SND_SOC_BIAS_ON;
+		schedule_delayed_work(&codec->dapm.delayed_work,
 			msecs_to_jiffies(caps_charge));
 	}
 
@@ -1550,7 +1553,7 @@
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0, reg;
 
-	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
+	INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work);
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8753->control_type);
 	if (ret < 0) {
@@ -1569,7 +1572,7 @@
 
 	/* charge output caps */
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
-	schedule_delayed_work(&codec->delayed_work,
+	schedule_delayed_work(&codec->dapm.delayed_work,
 			      msecs_to_jiffies(caps_charge));
 
 	/* set the update bits */
@@ -1604,7 +1607,7 @@
 /* power down chip */
 static int wm8753_remove(struct snd_soc_codec *codec)
 {
-	run_delayed_work(&codec->delayed_work);
+	run_delayed_work(&codec->dapm.delayed_work);
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
new file mode 100644
index 0000000..8608b4a
--- /dev/null
+++ b/sound/soc/codecs/wm8770.c
@@ -0,0 +1,750 @@
+/*
+ * wm8770.c  --  WM8770 ALSA SoC Audio driver
+ *
+ * Copyright 2010 Wolfson Microelectronics plc
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.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 "wm8770.h"
+
+#define WM8770_NUM_SUPPLIES 3
+static const char *wm8770_supply_names[WM8770_NUM_SUPPLIES] = {
+	"AVDD1",
+	"AVDD2",
+	"DVDD"
+};
+
+static const u16 wm8770_reg_defs[WM8770_CACHEREGNUM] = {
+	0x7f, 0x7f, 0x7f, 0x7f,
+	0x7f, 0x7f, 0x7f, 0x7f,
+	0x7f, 0xff, 0xff, 0xff,
+	0xff, 0xff, 0xff, 0xff,
+	0xff, 0xff, 0, 0x90, 0,
+	0, 0x22, 0x22, 0x3e,
+	0xc, 0xc, 0x100, 0x189,
+	0x189, 0x8770
+};
+
+struct wm8770_priv {
+	enum snd_soc_control_type control_type;
+	struct regulator_bulk_data supplies[WM8770_NUM_SUPPLIES];
+	struct notifier_block disable_nb[WM8770_NUM_SUPPLIES];
+	struct snd_soc_codec *codec;
+	int sysclk;
+};
+
+static int vout12supply_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event);
+static int vout34supply_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event);
+
+/*
+ * We can't use the same notifier block for more than one supply and
+ * there's no way I can see to get from a callback to the caller
+ * except container_of().
+ */
+#define WM8770_REGULATOR_EVENT(n) \
+static int wm8770_regulator_event_##n(struct notifier_block *nb, \
+				      unsigned long event, void *data)    \
+{ \
+	struct wm8770_priv *wm8770 = container_of(nb, struct wm8770_priv, \
+				     disable_nb[n]); \
+	if (event & REGULATOR_EVENT_DISABLE) { \
+		wm8770->codec->cache_sync = 1; \
+	} \
+	return 0; \
+}
+
+WM8770_REGULATOR_EVENT(0)
+WM8770_REGULATOR_EVENT(1)
+WM8770_REGULATOR_EVENT(2)
+
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(dac_dig_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_alg_tlv, -12700, 100, 1);
+
+static const char *dac_phase_text[][2] = {
+	{ "DAC1 Normal", "DAC1 Inverted" },
+	{ "DAC2 Normal", "DAC2 Inverted" },
+	{ "DAC3 Normal", "DAC3 Inverted" },
+	{ "DAC4 Normal", "DAC4 Inverted" },
+};
+
+static const struct soc_enum dac_phase[] = {
+	SOC_ENUM_DOUBLE(WM8770_DACPHASE, 0, 1, 2, dac_phase_text[0]),
+	SOC_ENUM_DOUBLE(WM8770_DACPHASE, 2, 3, 2, dac_phase_text[1]),
+	SOC_ENUM_DOUBLE(WM8770_DACPHASE, 4, 5, 2, dac_phase_text[2]),
+	SOC_ENUM_DOUBLE(WM8770_DACPHASE, 6, 7, 2, dac_phase_text[3]),
+};
+
+static const struct snd_kcontrol_new wm8770_snd_controls[] = {
+	/* global DAC playback controls */
+	SOC_SINGLE_TLV("DAC Playback Volume", WM8770_MSDIGVOL, 0, 255, 0,
+		dac_dig_tlv),
+	SOC_SINGLE("DAC Playback Switch", WM8770_DACMUTE, 4, 1, 1),
+	SOC_SINGLE("DAC Playback ZC Switch", WM8770_DACCTRL1, 0, 1, 0),
+
+	/* global VOUT playback controls */
+	SOC_SINGLE_TLV("VOUT Playback Volume", WM8770_MSALGVOL, 0, 127, 0,
+		dac_alg_tlv),
+	SOC_SINGLE("VOUT Playback ZC Switch", WM8770_MSALGVOL, 7, 1, 0),
+
+	/* VOUT1/2/3/4 specific controls */
+	SOC_DOUBLE_R_TLV("VOUT1 Playback Volume", WM8770_VOUT1LVOL,
+		WM8770_VOUT1RVOL, 0, 127, 0, dac_alg_tlv),
+	SOC_DOUBLE_R("VOUT1 Playback ZC Switch", WM8770_VOUT1LVOL,
+		WM8770_VOUT1RVOL, 7, 1, 0),
+	SOC_DOUBLE_R_TLV("VOUT2 Playback Volume", WM8770_VOUT2LVOL,
+		WM8770_VOUT2RVOL, 0, 127, 0, dac_alg_tlv),
+	SOC_DOUBLE_R("VOUT2 Playback ZC Switch", WM8770_VOUT2LVOL,
+		WM8770_VOUT2RVOL, 7, 1, 0),
+	SOC_DOUBLE_R_TLV("VOUT3 Playback Volume", WM8770_VOUT3LVOL,
+		WM8770_VOUT3RVOL, 0, 127, 0, dac_alg_tlv),
+	SOC_DOUBLE_R("VOUT3 Playback ZC Switch", WM8770_VOUT3LVOL,
+		WM8770_VOUT3RVOL, 7, 1, 0),
+	SOC_DOUBLE_R_TLV("VOUT4 Playback Volume", WM8770_VOUT4LVOL,
+		WM8770_VOUT4RVOL, 0, 127, 0, dac_alg_tlv),
+	SOC_DOUBLE_R("VOUT4 Playback ZC Switch", WM8770_VOUT4LVOL,
+		WM8770_VOUT4RVOL, 7, 1, 0),
+
+	/* DAC1/2/3/4 specific controls */
+	SOC_DOUBLE_R_TLV("DAC1 Playback Volume", WM8770_DAC1LVOL,
+		WM8770_DAC1RVOL, 0, 255, 0, dac_dig_tlv),
+	SOC_SINGLE("DAC1 Deemphasis Switch", WM8770_DACCTRL2, 0, 1, 0),
+	SOC_ENUM("DAC1 Phase", dac_phase[0]),
+	SOC_DOUBLE_R_TLV("DAC2 Playback Volume", WM8770_DAC2LVOL,
+		WM8770_DAC2RVOL, 0, 255, 0, dac_dig_tlv),
+	SOC_SINGLE("DAC2 Deemphasis Switch", WM8770_DACCTRL2, 1, 1, 0),
+	SOC_ENUM("DAC2 Phase", dac_phase[1]),
+	SOC_DOUBLE_R_TLV("DAC3 Playback Volume", WM8770_DAC3LVOL,
+		WM8770_DAC3RVOL, 0, 255, 0, dac_dig_tlv),
+	SOC_SINGLE("DAC3 Deemphasis Switch", WM8770_DACCTRL2, 2, 1, 0),
+	SOC_ENUM("DAC3 Phase", dac_phase[2]),
+	SOC_DOUBLE_R_TLV("DAC4 Playback Volume", WM8770_DAC4LVOL,
+		WM8770_DAC4RVOL, 0, 255, 0, dac_dig_tlv),
+	SOC_SINGLE("DAC4 Deemphasis Switch", WM8770_DACCTRL2, 3, 1, 0),
+	SOC_ENUM("DAC4 Phase", dac_phase[3]),
+
+	/* ADC specific controls */
+	SOC_DOUBLE_R_TLV("Capture Volume", WM8770_ADCLCTRL, WM8770_ADCRCTRL,
+		0, 31, 0, adc_tlv),
+	SOC_DOUBLE_R("Capture Switch", WM8770_ADCLCTRL, WM8770_ADCRCTRL,
+		5, 1, 1),
+
+	/* other controls */
+	SOC_SINGLE("ADC 128x Oversampling Switch", WM8770_MSTRCTRL, 3, 1, 0),
+	SOC_SINGLE("ADC Highpass Filter Switch", WM8770_IFACECTRL, 8, 1, 1)
+};
+
+static const char *ain_text[] = {
+	"AIN1", "AIN2", "AIN3", "AIN4",
+	"AIN5", "AIN6", "AIN7", "AIN8"
+};
+
+static const struct soc_enum ain_enum =
+	SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text);
+
+static const struct snd_kcontrol_new ain_mux =
+	SOC_DAPM_ENUM("Capture Mux", ain_enum);
+
+static const struct snd_kcontrol_new vout1_mix_controls[] = {
+	SOC_DAPM_SINGLE("DAC1 Switch", WM8770_OUTMUX1, 0, 1, 0),
+	SOC_DAPM_SINGLE("AUX1 Switch", WM8770_OUTMUX1, 1, 1, 0),
+	SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX1, 2, 1, 0)
+};
+
+static const struct snd_kcontrol_new vout2_mix_controls[] = {
+	SOC_DAPM_SINGLE("DAC2 Switch", WM8770_OUTMUX1, 3, 1, 0),
+	SOC_DAPM_SINGLE("AUX2 Switch", WM8770_OUTMUX1, 4, 1, 0),
+	SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX1, 5, 1, 0)
+};
+
+static const struct snd_kcontrol_new vout3_mix_controls[] = {
+	SOC_DAPM_SINGLE("DAC3 Switch", WM8770_OUTMUX2, 0, 1, 0),
+	SOC_DAPM_SINGLE("AUX3 Switch", WM8770_OUTMUX2, 1, 1, 0),
+	SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX2, 2, 1, 0)
+};
+
+static const struct snd_kcontrol_new vout4_mix_controls[] = {
+	SOC_DAPM_SINGLE("DAC4 Switch", WM8770_OUTMUX2, 3, 1, 0),
+	SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX2, 4, 1, 0)
+};
+
+static const struct snd_soc_dapm_widget wm8770_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("AUX1"),
+	SND_SOC_DAPM_INPUT("AUX2"),
+	SND_SOC_DAPM_INPUT("AUX3"),
+
+	SND_SOC_DAPM_INPUT("AIN1"),
+	SND_SOC_DAPM_INPUT("AIN2"),
+	SND_SOC_DAPM_INPUT("AIN3"),
+	SND_SOC_DAPM_INPUT("AIN4"),
+	SND_SOC_DAPM_INPUT("AIN5"),
+	SND_SOC_DAPM_INPUT("AIN6"),
+	SND_SOC_DAPM_INPUT("AIN7"),
+	SND_SOC_DAPM_INPUT("AIN8"),
+
+	SND_SOC_DAPM_MUX("Capture Mux", WM8770_ADCMUX, 8, 1, &ain_mux),
+
+	SND_SOC_DAPM_ADC("ADC", "Capture", WM8770_PWDNCTRL, 1, 1),
+
+	SND_SOC_DAPM_DAC("DAC1", "Playback", WM8770_PWDNCTRL, 2, 1),
+	SND_SOC_DAPM_DAC("DAC2", "Playback", WM8770_PWDNCTRL, 3, 1),
+	SND_SOC_DAPM_DAC("DAC3", "Playback", WM8770_PWDNCTRL, 4, 1),
+	SND_SOC_DAPM_DAC("DAC4", "Playback", WM8770_PWDNCTRL, 5, 1),
+
+	SND_SOC_DAPM_SUPPLY("VOUT12 Supply", SND_SOC_NOPM, 0, 0,
+		vout12supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("VOUT34 Supply", SND_SOC_NOPM, 0, 0,
+		vout34supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	SND_SOC_DAPM_MIXER("VOUT1 Mixer", SND_SOC_NOPM, 0, 0,
+		vout1_mix_controls, ARRAY_SIZE(vout1_mix_controls)),
+	SND_SOC_DAPM_MIXER("VOUT2 Mixer", SND_SOC_NOPM, 0, 0,
+		vout2_mix_controls, ARRAY_SIZE(vout2_mix_controls)),
+	SND_SOC_DAPM_MIXER("VOUT3 Mixer", SND_SOC_NOPM, 0, 0,
+		vout3_mix_controls, ARRAY_SIZE(vout3_mix_controls)),
+	SND_SOC_DAPM_MIXER("VOUT4 Mixer", SND_SOC_NOPM, 0, 0,
+		vout4_mix_controls, ARRAY_SIZE(vout4_mix_controls)),
+
+	SND_SOC_DAPM_OUTPUT("VOUT1"),
+	SND_SOC_DAPM_OUTPUT("VOUT2"),
+	SND_SOC_DAPM_OUTPUT("VOUT3"),
+	SND_SOC_DAPM_OUTPUT("VOUT4")
+};
+
+static const struct snd_soc_dapm_route wm8770_intercon[] = {
+	{ "Capture Mux", "AIN1", "AIN1" },
+	{ "Capture Mux", "AIN2", "AIN2" },
+	{ "Capture Mux", "AIN3", "AIN3" },
+	{ "Capture Mux", "AIN4", "AIN4" },
+	{ "Capture Mux", "AIN5", "AIN5" },
+	{ "Capture Mux", "AIN6", "AIN6" },
+	{ "Capture Mux", "AIN7", "AIN7" },
+	{ "Capture Mux", "AIN8", "AIN8" },
+
+	{ "ADC", NULL, "Capture Mux" },
+
+	{ "VOUT1 Mixer", NULL, "VOUT12 Supply" },
+	{ "VOUT1 Mixer", "DAC1 Switch", "DAC1" },
+	{ "VOUT1 Mixer", "AUX1 Switch", "AUX1" },
+	{ "VOUT1 Mixer", "Bypass Switch", "Capture Mux" },
+
+	{ "VOUT2 Mixer", NULL, "VOUT12 Supply" },
+	{ "VOUT2 Mixer", "DAC2 Switch", "DAC2" },
+	{ "VOUT2 Mixer", "AUX2 Switch", "AUX2" },
+	{ "VOUT2 Mixer", "Bypass Switch", "Capture Mux" },
+
+	{ "VOUT3 Mixer", NULL, "VOUT34 Supply" },
+	{ "VOUT3 Mixer", "DAC3 Switch", "DAC3" },
+	{ "VOUT3 Mixer", "AUX3 Switch", "AUX3" },
+	{ "VOUT3 Mixer", "Bypass Switch", "Capture Mux" },
+
+	{ "VOUT4 Mixer", NULL, "VOUT34 Supply" },
+	{ "VOUT4 Mixer", "DAC4 Switch", "DAC4" },
+	{ "VOUT4 Mixer", "Bypass Switch", "Capture Mux" },
+
+	{ "VOUT1", NULL, "VOUT1 Mixer" },
+	{ "VOUT2", NULL, "VOUT2 Mixer" },
+	{ "VOUT3", NULL, "VOUT3 Mixer" },
+	{ "VOUT4", NULL, "VOUT4 Mixer" }
+};
+
+static int vout12supply_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec;
+
+	codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_update_bits(codec, WM8770_OUTMUX1, 0x180, 0);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_update_bits(codec, WM8770_OUTMUX1, 0x180, 0x180);
+		break;
+	}
+
+	return 0;
+}
+
+static int vout34supply_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec;
+
+	codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_update_bits(codec, WM8770_OUTMUX2, 0x180, 0);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_update_bits(codec, WM8770_OUTMUX2, 0x180, 0x180);
+		break;
+	}
+
+	return 0;
+}
+
+static int wm8770_reset(struct snd_soc_codec *codec)
+{
+	return snd_soc_write(codec, WM8770_RESET, 0);
+}
+
+static int wm8770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec;
+	int iface, master;
+
+	codec = dai->codec;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		master = 0x100;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		master = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	iface = 0;
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x2;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0xc;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= 0x8;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x4;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, WM8770_IFACECTRL, 0xf, iface);
+	snd_soc_update_bits(codec, WM8770_MSTRCTRL, 0x100, master);
+
+	return 0;
+}
+
+static const int mclk_ratios[] = {
+	128,
+	192,
+	256,
+	384,
+	512,
+	768
+};
+
+static int wm8770_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec;
+	struct wm8770_priv *wm8770;
+	int i;
+	int iface;
+	int shift;
+	int ratio;
+
+	codec = dai->codec;
+	wm8770 = snd_soc_codec_get_drvdata(codec);
+
+	iface = 0;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x10;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x20;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x30;
+		break;
+	}
+
+	switch (substream->stream) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		i = 0;
+		shift = 4;
+		break;
+	case SNDRV_PCM_STREAM_CAPTURE:
+		i = 2;
+		shift = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Only need to set MCLK/LRCLK ratio if we're master */
+	if (snd_soc_read(codec, WM8770_MSTRCTRL) & 0x100) {
+		for (; i < ARRAY_SIZE(mclk_ratios); ++i) {
+			ratio = wm8770->sysclk / params_rate(params);
+			if (ratio == mclk_ratios[i])
+				break;
+		}
+
+		if (i == ARRAY_SIZE(mclk_ratios)) {
+			dev_err(codec->dev,
+				"Unable to configure MCLK ratio %d/%d\n",
+				wm8770->sysclk, params_rate(params));
+			return -EINVAL;
+		}
+
+		dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]);
+
+		snd_soc_update_bits(codec, WM8770_MSTRCTRL, 0x7 << shift,
+				    i << shift);
+	}
+
+	snd_soc_update_bits(codec, WM8770_IFACECTRL, 0x30, iface);
+
+	return 0;
+}
+
+static int wm8770_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec;
+
+	codec = dai->codec;
+	return snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10,
+				   !!mute << 4);
+}
+
+static int wm8770_set_sysclk(struct snd_soc_dai *dai,
+			     int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec;
+	struct wm8770_priv *wm8770;
+
+	codec = dai->codec;
+	wm8770 = snd_soc_codec_get_drvdata(codec);
+	wm8770->sysclk = freq;
+	return 0;
+}
+
+static void wm8770_sync_cache(struct snd_soc_codec *codec)
+{
+	int i;
+	u16 *cache;
+
+	if (!codec->cache_sync)
+		return;
+
+	codec->cache_only = 0;
+	cache = codec->reg_cache;
+	for (i = 0; i < codec->driver->reg_cache_size; i++) {
+		if (i == WM8770_RESET || cache[i] == wm8770_reg_defs[i])
+			continue;
+		snd_soc_write(codec, i, cache[i]);
+	}
+	codec->cache_sync = 0;
+}
+
+static int wm8770_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	int ret;
+	struct wm8770_priv *wm8770;
+
+	wm8770 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies),
+						    wm8770->supplies);
+			if (ret) {
+				dev_err(codec->dev,
+					"Failed to enable supplies: %d\n",
+					ret);
+				return ret;
+			}
+			wm8770_sync_cache(codec);
+			/* global powerup */
+			snd_soc_write(codec, WM8770_PWDNCTRL, 0);
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* global powerdown */
+		snd_soc_write(codec, WM8770_PWDNCTRL, 1);
+		regulator_bulk_disable(ARRAY_SIZE(wm8770->supplies),
+				       wm8770->supplies);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define WM8770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8770_dai_ops = {
+	.digital_mute = wm8770_mute,
+	.hw_params = wm8770_hw_params,
+	.set_fmt = wm8770_set_fmt,
+	.set_sysclk = wm8770_set_sysclk,
+};
+
+static struct snd_soc_dai_driver wm8770_dai = {
+	.name = "wm8770-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = WM8770_FORMATS
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = WM8770_FORMATS
+	},
+	.ops = &wm8770_dai_ops,
+	.symmetric_rates = 1
+};
+
+#ifdef CONFIG_PM
+static int wm8770_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8770_resume(struct snd_soc_codec *codec)
+{
+	wm8770_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+#else
+#define wm8770_suspend NULL
+#define wm8770_resume NULL
+#endif
+
+static int wm8770_probe(struct snd_soc_codec *codec)
+{
+	struct wm8770_priv *wm8770;
+	int ret;
+	int i;
+
+	wm8770 = snd_soc_codec_get_drvdata(codec);
+	wm8770->codec = codec;
+
+	codec->dapm.idle_bias_off = 1;
+
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8770->control_type);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++)
+		wm8770->supplies[i].supply = wm8770_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8770->supplies),
+				 wm8770->supplies);
+	if (ret) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	wm8770->disable_nb[0].notifier_call = wm8770_regulator_event_0;
+	wm8770->disable_nb[1].notifier_call = wm8770_regulator_event_1;
+	wm8770->disable_nb[2].notifier_call = wm8770_regulator_event_2;
+
+	/* This should really be moved into the regulator core */
+	for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++) {
+		ret = regulator_register_notifier(wm8770->supplies[i].consumer,
+						  &wm8770->disable_nb[i]);
+		if (ret) {
+			dev_err(codec->dev,
+				"Failed to register regulator notifier: %d\n",
+				ret);
+		}
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies),
+				    wm8770->supplies);
+	if (ret) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_reg_get;
+	}
+
+	ret = wm8770_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
+		goto err_reg_enable;
+	}
+
+	wm8770_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* latch the volume update bits */
+	snd_soc_update_bits(codec, WM8770_MSDIGVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_MSALGVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_VOUT1RVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_VOUT2RVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_VOUT3RVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_VOUT4RVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_DAC1RVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_DAC2RVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_DAC3RVOL, 0x100, 0x100);
+	snd_soc_update_bits(codec, WM8770_DAC4RVOL, 0x100, 0x100);
+
+	/* mute all DACs */
+	snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10, 0x10);
+
+	snd_soc_add_controls(codec, wm8770_snd_controls,
+			     ARRAY_SIZE(wm8770_snd_controls));
+	snd_soc_dapm_new_controls(&codec->dapm, wm8770_dapm_widgets,
+				  ARRAY_SIZE(wm8770_dapm_widgets));
+	snd_soc_dapm_add_routes(&codec->dapm, wm8770_intercon,
+				ARRAY_SIZE(wm8770_intercon));
+	return 0;
+
+err_reg_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8770->supplies), wm8770->supplies);
+err_reg_get:
+	regulator_bulk_free(ARRAY_SIZE(wm8770->supplies), wm8770->supplies);
+	return ret;
+}
+
+static int wm8770_remove(struct snd_soc_codec *codec)
+{
+	struct wm8770_priv *wm8770;
+	int i;
+
+	wm8770 = snd_soc_codec_get_drvdata(codec);
+	wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	for (i = 0; i < ARRAY_SIZE(wm8770->supplies); ++i)
+		regulator_unregister_notifier(wm8770->supplies[i].consumer,
+					      &wm8770->disable_nb[i]);
+	regulator_bulk_free(ARRAY_SIZE(wm8770->supplies), wm8770->supplies);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8770 = {
+	.probe = wm8770_probe,
+	.remove = wm8770_remove,
+	.suspend = wm8770_suspend,
+	.resume = wm8770_resume,
+	.set_bias_level = wm8770_set_bias_level,
+	.reg_cache_size = ARRAY_SIZE(wm8770_reg_defs),
+	.reg_word_size = sizeof (u16),
+	.reg_cache_default = wm8770_reg_defs
+};
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8770_spi_probe(struct spi_device *spi)
+{
+	struct wm8770_priv *wm8770;
+	int ret;
+
+	wm8770 = kzalloc(sizeof(struct wm8770_priv), GFP_KERNEL);
+	if (!wm8770)
+		return -ENOMEM;
+
+	wm8770->control_type = SND_SOC_SPI;
+	spi_set_drvdata(spi, wm8770);
+
+	ret = snd_soc_register_codec(&spi->dev,
+				     &soc_codec_dev_wm8770, &wm8770_dai, 1);
+	if (ret < 0)
+		kfree(wm8770);
+	return ret;
+}
+
+static int __devexit wm8770_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	kfree(spi_get_drvdata(spi));
+	return 0;
+}
+
+static struct spi_driver wm8770_spi_driver = {
+	.driver = {
+		.name = "wm8770",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8770_spi_probe,
+	.remove = __devexit_p(wm8770_spi_remove)
+};
+#endif
+
+static int __init wm8770_modinit(void)
+{
+	int ret = 0;
+
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8770_spi_driver);
+	if (ret) {
+		printk(KERN_ERR "Failed to register wm8770 SPI driver: %d\n",
+		       ret);
+	}
+#endif
+	return ret;
+}
+module_init(wm8770_modinit);
+
+static void __exit wm8770_exit(void)
+{
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8770_spi_driver);
+#endif
+}
+module_exit(wm8770_exit);
+
+MODULE_DESCRIPTION("ASoC WM8770 driver");
+MODULE_AUTHOR("Dimitris Papastamos <dp@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8770.h b/sound/soc/codecs/wm8770.h
new file mode 100644
index 0000000..5f1b3bd
--- /dev/null
+++ b/sound/soc/codecs/wm8770.h
@@ -0,0 +1,51 @@
+/*
+ * wm8770.h  --  WM8770 ASoC driver
+ *
+ * Copyright 2010 Wolfson Microelectronics plc
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8770_H
+#define _WM8770_H
+
+/* Registers */
+#define WM8770_VOUT1LVOL                0
+#define WM8770_VOUT1RVOL                0x1
+#define WM8770_VOUT2LVOL                0x2
+#define WM8770_VOUT2RVOL                0x3
+#define WM8770_VOUT3LVOL                0x4
+#define WM8770_VOUT3RVOL                0x5
+#define WM8770_VOUT4LVOL                0x6
+#define WM8770_VOUT4RVOL                0x7
+#define WM8770_MSALGVOL                 0x8
+#define WM8770_DAC1LVOL                 0x9
+#define WM8770_DAC1RVOL                 0xa
+#define WM8770_DAC2LVOL                 0xb
+#define WM8770_DAC2RVOL                 0xc
+#define WM8770_DAC3LVOL                 0xd
+#define WM8770_DAC3RVOL                 0xe
+#define WM8770_DAC4LVOL                 0xf
+#define WM8770_DAC4RVOL                 0x10
+#define WM8770_MSDIGVOL                 0x11
+#define WM8770_DACPHASE                 0x12
+#define WM8770_DACCTRL1                 0x13
+#define WM8770_DACMUTE                  0x14
+#define WM8770_DACCTRL2                 0x15
+#define WM8770_IFACECTRL                0x16
+#define WM8770_MSTRCTRL                 0x17
+#define WM8770_PWDNCTRL                 0x18
+#define WM8770_ADCLCTRL                 0x19
+#define WM8770_ADCRCTRL                 0x1a
+#define WM8770_ADCMUX                   0x1b
+#define WM8770_OUTMUX1                  0x1c
+#define WM8770_OUTMUX2                  0x1d
+#define WM8770_RESET                    0x31
+
+#define WM8770_CACHEREGNUM 0x20
+
+#endif
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 0132a27..e09ed65 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -306,7 +306,7 @@
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Disable the global powerdown; DAPM does the rest */
 			snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0);
 		}
@@ -317,7 +317,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -404,6 +404,7 @@
 static int wm8776_probe(struct snd_soc_codec *codec)
 {
 	struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret = 0;
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type);
@@ -427,9 +428,9 @@
 
 	snd_soc_add_controls(codec, wm8776_snd_controls,
 			     ARRAY_SIZE(wm8776_snd_controls));
-	snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8776_dapm_widgets,
 				  ARRAY_SIZE(wm8776_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes));
+	snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes));
 
 	return ret;
 }
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 4599e8e..031a0d4 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -515,7 +515,7 @@
 		snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies),
 						    wm8804->supplies);
 			if (ret) {
@@ -537,7 +537,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -581,7 +581,7 @@
 	wm8804 = snd_soc_codec_get_drvdata(codec);
 	wm8804->codec = codec;
 
-	codec->idle_bias_off = 1;
+	codec->dapm.idle_bias_off = 1;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 8, wm8804->control_type);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index aca4b1e..06ea9c0 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -611,10 +611,11 @@
 
 static int wm8900_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8900_dapm_widgets,
-				  ARRAY_SIZE(wm8900_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8900_dapm_widgets,
+				  ARRAY_SIZE(wm8900_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -1051,7 +1052,7 @@
 
 	case SND_SOC_BIAS_STANDBY:
 		/* Charge capacitors if initial power up */
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* STARTUP_BIAS_ENA on */
 			snd_soc_write(codec, WM8900_REG_POWER1,
 				     WM8900_REG_POWER1_STARTUP_BIAS_ENA);
@@ -1119,7 +1120,7 @@
 			     WM8900_REG_POWER2_SYSCLK_ENA);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 622b602..4a6df4b 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -923,10 +923,11 @@
 
 static int wm8903_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8903_dapm_widgets,
-				  ARRAY_SIZE(wm8903_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_controls(dapm, wm8903_dapm_widgets,
+				  ARRAY_SIZE(wm8903_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 }
@@ -946,7 +947,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			snd_soc_write(codec, WM8903_CLOCK_RATES_2,
 				     WM8903_CLK_SYS_ENA);
 
@@ -991,7 +992,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 33be84e..5e57bd2 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1428,10 +1428,11 @@
 static int wm8904_add_widgets(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8904_core_dapm_widgets,
 				  ARRAY_SIZE(wm8904_core_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, core_intercon,
+	snd_soc_dapm_add_routes(dapm, core_intercon,
 				ARRAY_SIZE(core_intercon));
 
 	switch (wm8904->devtype) {
@@ -1443,20 +1444,20 @@
 		snd_soc_add_controls(codec, wm8904_snd_controls,
 				     ARRAY_SIZE(wm8904_snd_controls));
 
-		snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets,
+		snd_soc_dapm_new_controls(dapm, wm8904_adc_dapm_widgets,
 					  ARRAY_SIZE(wm8904_adc_dapm_widgets));
-		snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets,
+		snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets,
 					  ARRAY_SIZE(wm8904_dac_dapm_widgets));
-		snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets,
+		snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets,
 					  ARRAY_SIZE(wm8904_dapm_widgets));
 
-		snd_soc_dapm_add_routes(codec, core_intercon,
+		snd_soc_dapm_add_routes(dapm, core_intercon,
 					ARRAY_SIZE(core_intercon));
-		snd_soc_dapm_add_routes(codec, adc_intercon,
+		snd_soc_dapm_add_routes(dapm, adc_intercon,
 					ARRAY_SIZE(adc_intercon));
-		snd_soc_dapm_add_routes(codec, dac_intercon,
+		snd_soc_dapm_add_routes(dapm, dac_intercon,
 					ARRAY_SIZE(dac_intercon));
-		snd_soc_dapm_add_routes(codec, wm8904_intercon,
+		snd_soc_dapm_add_routes(dapm, wm8904_intercon,
 					ARRAY_SIZE(wm8904_intercon));
 		break;
 
@@ -1464,17 +1465,17 @@
 		snd_soc_add_controls(codec, wm8904_dac_snd_controls,
 				     ARRAY_SIZE(wm8904_dac_snd_controls));
 
-		snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets,
+		snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets,
 					  ARRAY_SIZE(wm8904_dac_dapm_widgets));
 
-		snd_soc_dapm_add_routes(codec, dac_intercon,
+		snd_soc_dapm_add_routes(dapm, dac_intercon,
 					ARRAY_SIZE(dac_intercon));
-		snd_soc_dapm_add_routes(codec, wm8912_intercon,
+		snd_soc_dapm_add_routes(dapm, wm8912_intercon,
 					ARRAY_SIZE(wm8912_intercon));
 		break;
 	}
 
-	snd_soc_dapm_new_widgets(codec);
+	snd_soc_dapm_new_widgets(dapm);
 	return 0;
 }
 
@@ -1590,7 +1591,7 @@
 		       - wm8904->fs);
 	for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
 		cur_val = abs((wm8904->sysclk_rate /
-			       clk_sys_rates[i].ratio) - wm8904->fs);;
+			       clk_sys_rates[i].ratio) - wm8904->fs);
 		if (cur_val < best_val) {
 			best = i;
 			best_val = cur_val;
@@ -2139,7 +2140,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
 						    wm8904->supplies);
 			if (ret != 0) {
@@ -2198,7 +2199,7 @@
 				       wm8904->supplies);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -2373,7 +2374,7 @@
 	int ret, i;
 
 	codec->cache_sync = 1;
-	codec->idle_bias_off = 1;
+	codec->dapm.idle_bias_off = 1;
 
 	switch (wm8904->devtype) {
 	case WM8904:
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 2cb16f8..caed084 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -291,13 +291,14 @@
 
 static int wm8940_add_widgets(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, wm8940_dapm_widgets,
 					ARRAY_SIZE(wm8940_dapm_widgets));
 	if (ret)
 		goto error_ret;
-	ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 	if (ret)
 		goto error_ret;
 
@@ -735,7 +736,6 @@
 		return ret;
 
 	return ret;
-;
 }
 
 static int wm8940_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index f89ad6c..df1940f 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -577,13 +577,14 @@
 
 static int wm8955_add_widgets(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	snd_soc_add_controls(codec, wm8955_snd_controls,
 			     ARRAY_SIZE(wm8955_snd_controls));
 
-	snd_soc_dapm_new_controls(codec, wm8955_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8955_dapm_widgets,
 				  ARRAY_SIZE(wm8955_dapm_widgets));
-
-	snd_soc_dapm_add_routes(codec, wm8955_intercon,
+	snd_soc_dapm_add_routes(dapm, wm8955_intercon,
 				ARRAY_SIZE(wm8955_intercon));
 
 	return 0;
@@ -786,7 +787,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies),
 						    wm8955->supplies);
 			if (ret != 0) {
@@ -850,7 +851,7 @@
 				       wm8955->supplies);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 8d5efb3..0ea5788 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -388,27 +388,28 @@
 {
 	struct wm8960_data *pdata = codec->dev->platform_data;
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_dapm_widget *w;
 
-	snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets,
 				  ARRAY_SIZE(wm8960_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
 
 	/* In capless mode OUT3 is used to provide VMID for the
 	 * headphone outputs, otherwise it is used as a mono mixer.
 	 */
 	if (pdata && pdata->capless) {
-		snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_capless,
+		snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_capless,
 					  ARRAY_SIZE(wm8960_dapm_widgets_capless));
 
-		snd_soc_dapm_add_routes(codec, audio_paths_capless,
+		snd_soc_dapm_add_routes(dapm, audio_paths_capless,
 					ARRAY_SIZE(audio_paths_capless));
 	} else {
-		snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_out3,
+		snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_out3,
 					  ARRAY_SIZE(wm8960_dapm_widgets_out3));
 
-		snd_soc_dapm_add_routes(codec, audio_paths_out3,
+		snd_soc_dapm_add_routes(dapm, audio_paths_out3,
 					ARRAY_SIZE(audio_paths_out3));
 	}
 
@@ -417,7 +418,7 @@
 	 * list each time to find the desired power state do so now
 	 * and save the result.
 	 */
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &codec->dapm.widgets, list) {
 		if (strcmp(w->name, "LOUT1 PGA") == 0)
 			wm8960->lout1 = w;
 		if (strcmp(w->name, "ROUT1 PGA") == 0)
@@ -572,7 +573,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable anti-pop features */
 			snd_soc_write(codec, WM8960_APOP1,
 				      WM8960_POBCTRL | WM8960_SOFT_ST |
@@ -610,7 +611,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -626,7 +627,7 @@
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		switch (codec->bias_level) {
+		switch (codec->dapm.bias_level) {
 		case SND_SOC_BIAS_STANDBY:
 			/* Enable anti pop mode */
 			snd_soc_update_bits(codec, WM8960_APOP1,
@@ -681,7 +682,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		switch (codec->bias_level) {
+		switch (codec->dapm.bias_level) {
 		case SND_SOC_BIAS_PREPARE:
 			/* Disable HP discharge */
 			snd_soc_update_bits(codec, WM8960_APOP2,
@@ -705,7 +706,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 4f326f6..79b6509 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -882,7 +882,7 @@
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
 			/* Enable bias generation */
 			reg = snd_soc_read(codec, WM8961_ANTI_POP);
 			reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN;
@@ -897,7 +897,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_PREPARE) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
 			/* VREF off */
 			reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
 			reg &= ~WM8961_VREF;
@@ -919,7 +919,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -959,6 +959,7 @@
 
 static int wm8961_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret = 0;
 	u16 reg;
 
@@ -1024,9 +1025,9 @@
 
 	snd_soc_add_controls(codec, wm8961_snd_controls,
 				ARRAY_SIZE(wm8961_snd_controls));
-	snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets,
 				  ARRAY_SIZE(wm8961_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index e809274..9f6beca 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1958,7 +1958,7 @@
 
 static int wm8962_reset(struct snd_soc_codec *codec)
 {
-	return snd_soc_write(codec, WM8962_SOFTWARE_RESET, 0);
+	return snd_soc_write(codec, WM8962_SOFTWARE_RESET, 0x6243);
 }
 
 static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0);
@@ -2682,6 +2682,7 @@
 static int wm8962_add_widgets(struct snd_soc_codec *codec)
 {
 	struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	snd_soc_add_controls(codec, wm8962_snd_controls,
 			     ARRAY_SIZE(wm8962_snd_controls));
@@ -2693,26 +2694,26 @@
 				     ARRAY_SIZE(wm8962_spk_stereo_controls));
 
 
-	snd_soc_dapm_new_controls(codec, wm8962_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8962_dapm_widgets,
 				  ARRAY_SIZE(wm8962_dapm_widgets));
 	if (pdata && pdata->spk_mono)
-		snd_soc_dapm_new_controls(codec, wm8962_dapm_spk_mono_widgets,
+		snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_mono_widgets,
 					  ARRAY_SIZE(wm8962_dapm_spk_mono_widgets));
 	else
-		snd_soc_dapm_new_controls(codec, wm8962_dapm_spk_stereo_widgets,
+		snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_stereo_widgets,
 					  ARRAY_SIZE(wm8962_dapm_spk_stereo_widgets));
 
-	snd_soc_dapm_add_routes(codec, wm8962_intercon,
+	snd_soc_dapm_add_routes(dapm, wm8962_intercon,
 				ARRAY_SIZE(wm8962_intercon));
 	if (pdata && pdata->spk_mono)
-		snd_soc_dapm_add_routes(codec, wm8962_spk_mono_intercon,
+		snd_soc_dapm_add_routes(dapm, wm8962_spk_mono_intercon,
 					ARRAY_SIZE(wm8962_spk_mono_intercon));
 	else
-		snd_soc_dapm_add_routes(codec, wm8962_spk_stereo_intercon,
+		snd_soc_dapm_add_routes(dapm, wm8962_spk_stereo_intercon,
 					ARRAY_SIZE(wm8962_spk_stereo_intercon));
 
 
-	snd_soc_dapm_disable_pin(codec, "Beep");
+	snd_soc_dapm_disable_pin(dapm, "Beep");
 
 	return 0;
 }
@@ -2819,7 +2820,7 @@
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	if (level == codec->bias_level)
+	if (level == codec->dapm.bias_level)
 		return 0;
 
 	switch (level) {
@@ -2833,7 +2834,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies),
 						    wm8962->supplies);
 			if (ret != 0) {
@@ -2883,7 +2884,7 @@
 				       wm8962->supplies);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -3353,6 +3354,8 @@
 	if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) {
 		dev_dbg(codec->dev, "Microphone event detected\n");
 
+		pm_wakeup_event(codec->dev, 300);
+
 		schedule_delayed_work(&wm8962->mic_work,
 				      msecs_to_jiffies(250));
 	}
@@ -3439,6 +3442,7 @@
 	struct wm8962_priv *wm8962 =
 		container_of(work, struct wm8962_priv, beep_work);
 	struct snd_soc_codec *codec = wm8962->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int i;
 	int reg = 0;
 	int best = 0;
@@ -3455,16 +3459,16 @@
 
 		reg = WM8962_BEEP_ENA | (best << WM8962_BEEP_RATE_SHIFT);
 
-		snd_soc_dapm_enable_pin(codec, "Beep");
+		snd_soc_dapm_enable_pin(dapm, "Beep");
 	} else {
 		dev_dbg(codec->dev, "Disabling beep\n");
-		snd_soc_dapm_disable_pin(codec, "Beep");
+		snd_soc_dapm_disable_pin(dapm, "Beep");
 	}
 
 	snd_soc_update_bits(codec, WM8962_BEEP_GENERATOR_1,
 			    WM8962_BEEP_ENA | WM8962_BEEP_RATE_MASK, reg);
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 /* For usability define a way of injecting beep events for the device -
@@ -3711,7 +3715,7 @@
 	INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work);
 
 	codec->cache_sync = 1;
-	codec->idle_bias_off = 1;
+	codec->dapm.idle_bias_off = 1;
 
 	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C);
 	if (ret != 0) {
@@ -3870,7 +3874,6 @@
 err_get:
 	regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
 err:
-	kfree(wm8962);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 63f6dbf..84b2dcb 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -333,10 +333,11 @@
 
 static int wm8971_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8971_dapm_widgets,
-				  ARRAY_SIZE(wm8971_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8971_dapm_widgets,
+				  ARRAY_SIZE(wm8971_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -553,7 +554,7 @@
 		snd_soc_write(codec, WM8971_PWR1, 0x0001);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -590,9 +591,11 @@
 
 static void wm8971_work(struct work_struct *work)
 {
-	struct snd_soc_codec *codec =
-		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8971_set_bias_level(codec, codec->bias_level);
+	struct snd_soc_dapm_context *dapm =
+		container_of(work, struct snd_soc_dapm_context,
+			     delayed_work.work);
+	struct snd_soc_codec *codec = dapm->codec;
+	wm8971_set_bias_level(codec, codec->dapm.bias_level);
 }
 
 static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state)
@@ -620,11 +623,11 @@
 	wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8971 caps */
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+	if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
 		reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e;
 		snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0);
-		codec->bias_level = SND_SOC_BIAS_ON;
-		queue_delayed_work(wm8971_workq, &codec->delayed_work,
+		codec->dapm.bias_level = SND_SOC_BIAS_ON;
+		queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work,
 			msecs_to_jiffies(1000));
 	}
 
@@ -643,7 +646,7 @@
 		return ret;
 	}
 
-	INIT_DELAYED_WORK(&codec->delayed_work, wm8971_work);
+	INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work);
 	wm8971_workq = create_workqueue("wm8971");
 	if (wm8971_workq == NULL)
 		return -ENOMEM;
@@ -653,8 +656,8 @@
 	/* charge output caps - set vmid to 5k for quick power up */
 	reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e;
 	snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0);
-	codec->bias_level = SND_SOC_BIAS_STANDBY;
-	queue_delayed_work(wm8971_workq, &codec->delayed_work,
+	codec->dapm.bias_level = SND_SOC_BIAS_STANDBY;
+	queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work,
 		msecs_to_jiffies(1000));
 
 	/* set the update bits */
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index b4363f6..d19bb14 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -274,10 +274,11 @@
 
 static int wm8974_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8974_dapm_widgets,
-				  ARRAY_SIZE(wm8974_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm8974_dapm_widgets,
+				  ARRAY_SIZE(wm8974_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -530,7 +531,7 @@
 	case SND_SOC_BIAS_STANDBY:
 		power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN;
 
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Initial cap charge at VMID 5k */
 			snd_soc_write(codec, WM8974_POWER1, power1 | 0x3);
 			mdelay(100);
@@ -547,7 +548,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 13b979a..ac43b60 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -355,11 +355,12 @@
 
 static int wm8978_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets,
-				  ARRAY_SIZE(wm8978_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
+	snd_soc_dapm_new_controls(dapm, wm8978_dapm_widgets,
+				  ARRAY_SIZE(wm8978_dapm_widgets));
 	/* set up the WM8978 audio map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -837,7 +838,7 @@
 		/* bit 3: enable bias, bit 2: enable I/O tie off buffer */
 		power1 |= 0xc;
 
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Initial cap charge at VMID 5k */
 			snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1,
 				      power1 | 0x3);
@@ -857,7 +858,7 @@
 
 	dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1);
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index fd2e7cc..c3c8fd2 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -533,10 +533,11 @@
 
 static int wm8985_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8985_dapm_widgets,
-				  ARRAY_SIZE(wm8985_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map,
+	snd_soc_dapm_new_controls(dapm, wm8985_dapm_widgets,
+				  ARRAY_SIZE(wm8985_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map,
 				ARRAY_SIZE(audio_map));
 	return 0;
 }
@@ -879,7 +880,7 @@
 				    1 << WM8985_VMIDSEL_SHIFT);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8985->supplies),
 						    wm8985->supplies);
 			if (ret) {
@@ -939,7 +940,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index d7f2597..0bc2eb5 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -677,7 +677,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* VREF, VMID=2x5k */
 			snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
 
@@ -693,7 +693,7 @@
 		snd_soc_write(codec, WM8988_PWR1, 0x0000);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -759,6 +759,7 @@
 static int wm8988_probe(struct snd_soc_codec *codec)
 {
 	struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret = 0;
 	u16 reg;
 
@@ -790,9 +791,9 @@
 
 	snd_soc_add_controls(codec, wm8988_snd_controls,
 				ARRAY_SIZE(wm8988_snd_controls));
-	snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8988_dapm_widgets,
 				  ARRAY_SIZE(wm8988_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 264828e..309664e 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -914,11 +914,12 @@
 
 static int wm8990_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets,
-				  ARRAY_SIZE(wm8990_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
+	snd_soc_dapm_new_controls(dapm, wm8990_dapm_widgets,
+				  ARRAY_SIZE(wm8990_dapm_widgets));
 	/* set up the WM8990 audio map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -1170,7 +1171,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable all output discharge bits */
 			snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
 				WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
@@ -1266,7 +1267,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 589e3fa..991d90c 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -735,6 +735,7 @@
 					    0);
 		}
 		wm8993->class_w_users++;
+		wm8993->hubs_data.class_w = true;
 	}
 
 	/* Implement the change */
@@ -751,6 +752,7 @@
 					    WM8993_CP_DYN_V);
 		}
 		wm8993->class_w_users--;
+		wm8993->hubs_data.class_w = false;
 	}
 
 	dev_dbg(codec->dev, "Indirect DAC use count now %d\n",
@@ -968,7 +970,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies),
 						    wm8993->supplies);
 			if (ret != 0)
@@ -1043,7 +1045,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1225,7 +1227,7 @@
 		       - wm8993->fs);
 	for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
 		cur_val = abs((wm8993->sysclk_rate /
-			       clk_sys_rates[i].ratio) - wm8993->fs);;
+			       clk_sys_rates[i].ratio) - wm8993->fs);
 		if (cur_val < best_val) {
 			best = i;
 			best_val = cur_val;
@@ -1422,6 +1424,7 @@
 static int wm8993_probe(struct snd_soc_codec *codec)
 {
 	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret, i, val;
 
 	wm8993->hubs_data.hp_startup_mode = 1;
@@ -1503,11 +1506,11 @@
 				     ARRAY_SIZE(wm8993_eq_controls));
 	}
 
-	snd_soc_dapm_new_controls(codec, wm8993_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8993_dapm_widgets,
 				  ARRAY_SIZE(wm8993_dapm_widgets));
 	wm_hubs_add_analogue_controls(codec);
 
-	snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes));
+	snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes));
 	wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff,
 				    wm8993->pdata.lineout2_diff);
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 830dfdd66..bfdb7da 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1730,8 +1730,6 @@
 	if (!wm8994_volatile(reg))
 		wm8994->reg_cache[reg] = value;
 
-	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
 	return wm8994_reg_write(codec->control_data, reg, value);
 }
 
@@ -1837,7 +1835,7 @@
 
 	snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new);
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(&codec->dapm);
 
 	return 0;
 }
@@ -2228,6 +2226,7 @@
 
 static void wm8994_update_class_w(struct snd_soc_codec *codec)
 {
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int enable = 1;
 	int source = 0;  /* GCC flow analysis can't track enable */
 	int reg, reg_r;
@@ -2278,11 +2277,13 @@
 				    WM8994_CP_DYN_PWR |
 				    WM8994_CP_DYN_SRC_SEL_MASK,
 				    source | WM8994_CP_DYN_PWR);
+		wm8994->hubs.class_w = true;
 		
 	} else {
 		dev_dbg(codec->dev, "Class W disabled\n");
 		snd_soc_update_bits(codec, WM8994_CLASS_W_1,
 				    WM8994_CP_DYN_PWR, 0);
+		wm8994->hubs.class_w = false;
 	}
 }
 
@@ -3107,7 +3108,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Tweak DC servo and DSP configuration for
 			 * improved performance. */
 			if (wm8994->revision < 4) {
@@ -3151,7 +3152,7 @@
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
 			/* Switch over to startup biases */
 			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
 					    WM8994_BIAS_SRC |
@@ -3186,7 +3187,7 @@
 		}
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -3894,6 +3895,7 @@
 static int wm8994_codec_probe(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret, i;
 
 	codec->control_data = dev_get_drvdata(codec->dev->parent);
@@ -4034,10 +4036,10 @@
 	wm_hubs_add_analogue_controls(codec);
 	snd_soc_add_controls(codec, wm8994_snd_controls,
 			     ARRAY_SIZE(wm8994_snd_controls));
-	snd_soc_dapm_new_controls(codec, wm8994_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets,
 				  ARRAY_SIZE(wm8994_dapm_widgets));
 	wm_hubs_add_analogue_routes(codec, 0, 0);
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	return 0;
 
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index ecc7c37..c03e2c3 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -805,7 +805,7 @@
 
 	case SND_SOC_BIAS_STANDBY:
 		/* Initial cold start */
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Disable LINEOUT discharge */
 			reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL);
 			reg &= ~WM9081_LINEOUT_DISCH;
@@ -865,7 +865,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1228,6 +1228,7 @@
 static int wm9081_probe(struct snd_soc_codec *codec)
 {
 	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 	u16 reg;
 
@@ -1269,9 +1270,9 @@
 				     ARRAY_SIZE(wm9081_eq_controls));
 	}
 
-	snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets,
 				  ARRAY_SIZE(wm9081_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
 
 	return ret;
 }
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 99c046b..b5afa01 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -443,31 +443,32 @@
 static int wm9090_add_controls(struct snd_soc_codec *codec)
 {
 	struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int i;
 
-	snd_soc_dapm_new_controls(codec, wm9090_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm9090_dapm_widgets,
 				  ARRAY_SIZE(wm9090_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_add_controls(codec, wm9090_controls,
 			     ARRAY_SIZE(wm9090_controls));
 
 	if (wm9090->pdata.lin1_diff) {
-		snd_soc_dapm_add_routes(codec, audio_map_in1_diff,
+		snd_soc_dapm_add_routes(dapm, audio_map_in1_diff,
 					ARRAY_SIZE(audio_map_in1_diff));
 	} else {
-		snd_soc_dapm_add_routes(codec, audio_map_in1_se,
+		snd_soc_dapm_add_routes(dapm, audio_map_in1_se,
 					ARRAY_SIZE(audio_map_in1_se));
 		snd_soc_add_controls(codec, wm9090_in1_se_controls,
 				     ARRAY_SIZE(wm9090_in1_se_controls));
 	}
 
 	if (wm9090->pdata.lin2_diff) {
-		snd_soc_dapm_add_routes(codec, audio_map_in2_diff,
+		snd_soc_dapm_add_routes(dapm, audio_map_in2_diff,
 					ARRAY_SIZE(audio_map_in2_diff));
 	} else {
-		snd_soc_dapm_add_routes(codec, audio_map_in2_se,
+		snd_soc_dapm_add_routes(dapm, audio_map_in2_se,
 					ARRAY_SIZE(audio_map_in2_se));
 		snd_soc_add_controls(codec, wm9090_in2_se_controls,
 				     ARRAY_SIZE(wm9090_in2_se_controls));
@@ -514,7 +515,7 @@
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Restore the register cache */
 			for (i = 1; i < codec->driver->reg_cache_size; i++) {
 				if (reg_cache[i] == wm9090_reg_defaults[i])
@@ -544,7 +545,7 @@
 		break;
 	}
 
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index a144acd..58d1208 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -203,9 +203,11 @@
 
 static int wm9705_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_new_controls(dapm, wm9705_dapm_widgets,
 					ARRAY_SIZE(wm9705_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index d2f224d..3ca42a3 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -432,10 +432,11 @@
 
 static int wm9712_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
-				  ARRAY_SIZE(wm9712_dapm_widgets));
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_controls(dapm, wm9712_dapm_widgets,
+				  ARRAY_SIZE(wm9712_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -570,7 +571,7 @@
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 7da13b0..87b236b 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -647,10 +647,12 @@
 
 static int wm9713_add_widgets(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_new_controls(dapm, wm9713_dapm_widgets,
 				  ARRAY_SIZE(wm9713_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
@@ -1147,7 +1149,7 @@
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 19ca782..422c7fb 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -94,6 +94,18 @@
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	u16 reg, reg_l, reg_r, dcs_cfg;
 
+	/* If we're using a digital only path and have a previously
+	 * callibrated DC servo offset stored then use that. */
+	if (hubs->class_w && hubs->class_w_dcs) {
+		dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
+			hubs->class_w_dcs);
+		snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+		wait_for_dc_servo(codec,
+				  WM8993_DCS_TRIG_DAC_WR_0 |
+				  WM8993_DCS_TRIG_DAC_WR_1);
+		return;
+	}
+
 	/* Set for 32 series updates */
 	snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
 			    WM8993_DCS_SERIES_NO_01_MASK,
@@ -101,34 +113,34 @@
 	wait_for_dc_servo(codec,
 			  WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1);
 
+	/* Different chips in the family support different readback
+	 * methods.
+	 */
+	switch (hubs->dcs_readback_mode) {
+	case 0:
+		reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+			& WM8993_DCS_INTEG_CHAN_0_MASK;
+		reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
+			& WM8993_DCS_INTEG_CHAN_1_MASK;
+		break;
+	case 1:
+		reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+		reg_l = (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;
+		break;
+	default:
+		WARN(1, "Unknown DCS readback method\n");
+		break;
+	}
+
+	dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
+
 	/* Apply correction to DC servo result */
 	if (hubs->dcs_codes) {
 		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
 			hubs->dcs_codes);
 
-		/* Different chips in the family support different
-		 * readback methods.
-		 */
-		switch (hubs->dcs_readback_mode) {
-		case 0:
-			reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
-				& WM8993_DCS_INTEG_CHAN_0_MASK;;
-			reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
-				& WM8993_DCS_INTEG_CHAN_1_MASK;
-			break;
-		case 1:
-			reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
-			reg_l = (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;
-			break;
-		default:
-			WARN(1, "Unknown DCS readback method\n");
-			break;
-		}
-
-		dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
-
 		/* HPOUT1L */
 		if (reg_l + hubs->dcs_codes > 0 &&
 		    reg_l + hubs->dcs_codes < 0xff)
@@ -148,7 +160,15 @@
 		wait_for_dc_servo(codec,
 				  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;
 	}
+
+	/* Save the callibrated offset if we're in class W mode and
+	 * therefore don't have any analogue signal mixed in. */
+	if (hubs->class_w)
+		hubs->class_w_dcs = dcs_cfg;
 }
 
 /*
@@ -163,6 +183,9 @@
 
 	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
 
+	/* Updating the analogue gains invalidates the DC servo cache */
+	hubs->class_w_dcs = 0;
+
 	/* If we're applying an offset correction then updating the
 	 * callibration would be likely to introduce further offsets. */
 	if (hubs->dcs_codes)
@@ -791,6 +814,8 @@
 
 int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	/* Latch volume update bits & default ZC on */
 	snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_1_2_VOLUME,
 			    WM8993_IN1_VU, WM8993_IN1_VU);
@@ -819,7 +844,7 @@
 	snd_soc_add_controls(codec, analogue_snd_controls,
 			     ARRAY_SIZE(analogue_snd_controls));
 
-	snd_soc_dapm_new_controls(codec, analogue_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, analogue_dapm_widgets,
 				  ARRAY_SIZE(analogue_dapm_widgets));
 	return 0;
 }
@@ -828,24 +853,26 @@
 int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
 				int lineout1_diff, int lineout2_diff)
 {
-	snd_soc_dapm_add_routes(codec, analogue_routes,
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_add_routes(dapm, analogue_routes,
 				ARRAY_SIZE(analogue_routes));
 
 	if (lineout1_diff)
-		snd_soc_dapm_add_routes(codec,
+		snd_soc_dapm_add_routes(dapm,
 					lineout1_diff_routes,
 					ARRAY_SIZE(lineout1_diff_routes));
 	else
-		snd_soc_dapm_add_routes(codec,
+		snd_soc_dapm_add_routes(dapm,
 					lineout1_se_routes,
 					ARRAY_SIZE(lineout1_se_routes));
 
 	if (lineout2_diff)
-		snd_soc_dapm_add_routes(codec,
+		snd_soc_dapm_add_routes(dapm,
 					lineout2_diff_routes,
 					ARRAY_SIZE(lineout2_diff_routes));
 	else
-		snd_soc_dapm_add_routes(codec,
+		snd_soc_dapm_add_routes(dapm,
 					lineout2_se_routes,
 					ARRAY_SIZE(lineout2_se_routes));
 
@@ -872,7 +899,7 @@
 	 * VMID as an output and can disable it.
 	 */
 	if (lineout1_diff && lineout2_diff)
-		codec->idle_bias_off = 1;
+		codec->dapm.idle_bias_off = 1;
 
 	if (lineout1fb)
 		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index e51c166..f8a5e97 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -23,6 +23,9 @@
 	int dcs_codes;
 	int dcs_readback_mode;
 	int hp_startup_mode;
+
+	bool class_w;
+	u16 class_w_dcs;
 };
 
 extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index bc9e6b0b..fd96905 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -132,26 +132,27 @@
 static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add davinci-evm specific widgets */
-	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
 				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* Set up davinci-evm specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* not connected */
-	snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
-	snd_soc_dapm_disable_pin(codec, "HPLCOM");
-	snd_soc_dapm_disable_pin(codec, "HPRCOM");
+	snd_soc_dapm_disable_pin(dapm, "MONO_LOUT");
+	snd_soc_dapm_disable_pin(dapm, "HPLCOM");
+	snd_soc_dapm_disable_pin(dapm, "HPRCOM");
 
 	/* always connected */
-	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Line Out");
-	snd_soc_dapm_enable_pin(codec, "Mic Jack");
-	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line Out");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c
index 28ab5ff..f1c7851 100644
--- a/sound/soc/ep93xx/snappercl15.c
+++ b/sound/soc/ep93xx/snappercl15.c
@@ -79,11 +79,12 @@
 static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
 				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 	return 0;
 }
 
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index d2d98c7..ad21f81 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -456,13 +456,13 @@
 static struct snd_soc_dai_driver imx_ssi_dai = {
 	.probe = imx_ssi_dai_probe,
 	.playback = {
-		.channels_min = 2,
+		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
-		.channels_min = 2,
+		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c
index 30fdb15..46fadf4 100644
--- a/sound/soc/imx/wm1133-ev1.c
+++ b/sound/soc/imx/wm1133-ev1.c
@@ -213,11 +213,12 @@
 static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets,
+	snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
 				  ARRAY_SIZE(wm1133_ev1_widgets));
 
-	snd_soc_dapm_add_routes(codec, wm1133_ev1_map,
+	snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
 				ARRAY_SIZE(wm1133_ev1_map));
 
 	/* Headphone jack detection */
@@ -234,7 +235,7 @@
 	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
 			       SND_JACK_BTN_0);
 
-	snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
+	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
 
 	return 0;
 }
diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c
index ef1a99e..70afbfa 100644
--- a/sound/soc/jz4740/qi_lb60.c
+++ b/sound/soc/jz4740/qi_lb60.c
@@ -59,10 +59,11 @@
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	snd_soc_dapm_nc_pin(codec, "LIN");
-	snd_soc_dapm_nc_pin(codec, "RIN");
+	snd_soc_dapm_nc_pin(dapm, "LIN");
+	snd_soc_dapm_nc_pin(dapm, "RIN");
 
 	ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT);
 	if (ret < 0) {
@@ -70,9 +71,11 @@
 		return ret;
 	}
 
-	snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets));
-	snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes));
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_new_controls(dapm, qi_lb60_widgets,
+				  ARRAY_SIZE(qi_lb60_widgets));
+	snd_soc_dapm_add_routes(dapm, qi_lb60_routes,
+				ARRAY_SIZE(qi_lb60_routes));
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 16ec2a2..8f49e16 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -11,10 +11,19 @@
 
 config SND_KIRKWOOD_SOC_OPENRD
 	tristate "SoC Audio support for Kirkwood Openrd Client"
-	depends on SND_KIRKWOOD_SOC && MACH_OPENRD_CLIENT
+	depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE)
 	select SND_KIRKWOOD_SOC_I2S
 	select SND_SOC_CS42L51
 	help
 	  Say Y if you want to add support for SoC audio on
 	  Openrd Client.
 
+config SND_KIRKWOOD_SOC_T5325
+	tristate "SoC Audio support for HP t5325"
+	depends on SND_KIRKWOOD_SOC && MACH_T5325
+	select SND_KIRKWOOD_SOC_I2S
+	select SND_SOC_ALC5623
+	help
+	  Say Y if you want to add support for SoC audio on
+	  the HP t5325 thin client.
+
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index 33a16dc..3e62ae9 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -5,5 +5,7 @@
 obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
 
 snd-soc-openrd-objs := kirkwood-openrd.o
+snd-soc-t5325-objs := kirkwood-t5325.o
 
 obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
+obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o
diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c
index 9d7c81e..d863afb 100644
--- a/sound/soc/kirkwood/kirkwood-openrd.c
+++ b/sound/soc/kirkwood/kirkwood-openrd.c
@@ -86,7 +86,7 @@
 {
 	int ret;
 
-	if (!machine_is_openrd_client())
+	if (!machine_is_openrd_client() && !machine_is_openrd_ultimate())
 		return 0;
 
 	openrd_client_snd_device = platform_device_alloc("soc-audio", -1);
diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c
new file mode 100644
index 0000000..07b6eca
--- /dev/null
+++ b/sound/soc/kirkwood/kirkwood-t5325.c
@@ -0,0 +1,142 @@
+/*
+ * kirkwood-t5325.c
+ *
+ * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <mach/kirkwood.h>
+#include <plat/audio.h>
+#include <asm/mach-types.h>
+#include "../codecs/alc5623.h"
+
+static int t5325_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+	unsigned int freq, fmt;
+
+	fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	freq = params_rate(params) * 256;
+
+	return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
+
+}
+
+static struct snd_soc_ops t5325_ops = {
+	.hw_params = t5325_hw_params,
+};
+
+static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route t5325_route[] = {
+	{ "Headphone Jack",	NULL,	"HPL" },
+	{ "Headphone Jack",	NULL,	"HPR" },
+
+	{"Speaker",		NULL,	"SPKOUT"},
+	{"Speaker",		NULL,	"SPKOUTN"},
+
+	{ "MIC1",		NULL,	"Mic Jack" },
+	{ "MIC2",		NULL,	"Mic Jack" },
+};
+
+static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_new_controls(dapm, t5325_dapm_widgets,
+				ARRAY_SIZE(t5325_dapm_widgets));
+
+	snd_soc_dapm_add_routes(dapm, t5325_route, ARRAY_SIZE(t5325_route));
+
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Speaker");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link t5325_dai[] = {
+{
+	.name = "ALC5621",
+	.stream_name = "ALC5621 HiFi",
+	.cpu_dai_name = "kirkwood-i2s",
+	.platform_name = "kirkwood-pcm-audio",
+	.codec_dai_name = "alc5621-hifi",
+	.codec_name = "alc562x-codec.0-001a",
+	.ops = &t5325_ops,
+	.init = t5325_dai_init,
+},
+};
+
+
+static struct snd_soc_card t5325 = {
+	.name = "t5325",
+	.dai_link = t5325_dai,
+	.num_links = ARRAY_SIZE(t5325_dai),
+};
+
+static struct platform_device *t5325_snd_device;
+
+static int __init t5325_init(void)
+{
+	int ret;
+
+	if (!machine_is_t5325())
+		return 0;
+
+	t5325_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!t5325_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(t5325_snd_device,
+			&t5325);
+
+	ret = platform_device_add(t5325_snd_device);
+	if (ret) {
+		printk(KERN_ERR "%s: platform_device_add failed\n", __func__);
+		platform_device_put(t5325_snd_device);
+	}
+
+	return ret;
+}
+module_init(t5325_init);
+
+static void __exit t5325_exit(void)
+{
+	platform_device_unregister(t5325_snd_device);
+}
+module_exit(t5325_exit);
+
+MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
+MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 979dd50..668773d 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -114,20 +114,21 @@
 static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add am3517-evm specific widgets */
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
 				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
 
 	/* Set up davinci-evm specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* always connected */
-	snd_soc_dapm_enable_pin(codec, "Line Out");
-	snd_soc_dapm_enable_pin(codec, "Line In");
-	snd_soc_dapm_enable_pin(codec, "Mic In");
+	snd_soc_dapm_enable_pin(dapm, "Line Out");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Mic In");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 438146a..2101bdc 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -26,7 +26,7 @@
 #include <linux/spinlock.h>
 #include <linux/tty.h>
 
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
 #include <sound/jack.h>
 
 #include <asm/mach-types.h>
@@ -94,6 +94,7 @@
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
 	unsigned short pins;
 	int pin, changed = 0;
@@ -112,48 +113,48 @@
 
 	/* Setup pins after corresponding bits if changed */
 	pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
-	if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) {
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
 		changed = 1;
 		if (pin)
-			snd_soc_dapm_enable_pin(codec, "Mouthpiece");
+			snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
 		else
-			snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+			snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
 	}
 	pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
-	if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) {
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
 		changed = 1;
 		if (pin)
-			snd_soc_dapm_enable_pin(codec, "Earpiece");
+			snd_soc_dapm_enable_pin(dapm, "Earpiece");
 		else
-			snd_soc_dapm_disable_pin(codec, "Earpiece");
+			snd_soc_dapm_disable_pin(dapm, "Earpiece");
 	}
 	pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
-	if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) {
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
 		changed = 1;
 		if (pin)
-			snd_soc_dapm_enable_pin(codec, "Microphone");
+			snd_soc_dapm_enable_pin(dapm, "Microphone");
 		else
-			snd_soc_dapm_disable_pin(codec, "Microphone");
+			snd_soc_dapm_disable_pin(dapm, "Microphone");
 	}
 	pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
-	if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) {
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
 		changed = 1;
 		if (pin)
-			snd_soc_dapm_enable_pin(codec, "Speaker");
+			snd_soc_dapm_enable_pin(dapm, "Speaker");
 		else
-			snd_soc_dapm_disable_pin(codec, "Speaker");
+			snd_soc_dapm_disable_pin(dapm, "Speaker");
 	}
 	pin = !!(pins & (1 << AMS_DELTA_AGC));
 	if (pin != ams_delta_audio_agc) {
 		ams_delta_audio_agc = pin;
 		changed = 1;
 		if (pin)
-			snd_soc_dapm_enable_pin(codec, "AGCIN");
+			snd_soc_dapm_enable_pin(dapm, "AGCIN");
 		else
-			snd_soc_dapm_disable_pin(codec, "AGCIN");
+			snd_soc_dapm_disable_pin(dapm, "AGCIN");
 	}
 	if (changed)
-		snd_soc_dapm_sync(codec);
+		snd_soc_dapm_sync(dapm);
 
 	mutex_unlock(&codec->mutex);
 
@@ -164,19 +165,20 @@
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	unsigned short pins, mode;
 
-	pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") <<
+	pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
 							AMS_DELTA_MOUTHPIECE) |
-			(snd_soc_dapm_get_pin_status(codec, "Earpiece") <<
+			(snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
 							AMS_DELTA_EARPIECE));
 	if (pins)
-		pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+		pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
 							AMS_DELTA_MICROPHONE);
 	else
-		pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+		pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
 							AMS_DELTA_MICROPHONE) |
-			(snd_soc_dapm_get_pin_status(codec, "Speaker") <<
+			(snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
 							AMS_DELTA_SPEAKER) |
 			(ams_delta_audio_agc << AMS_DELTA_AGC));
 
@@ -300,6 +302,7 @@
 static void cx81801_close(struct tty_struct *tty)
 {
 	struct snd_soc_codec *codec = tty->disc_data;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	del_timer_sync(&cx81801_timer);
 
@@ -312,12 +315,12 @@
 	v253_ops.close(tty);
 
 	/* Revert back to default audio input/output constellation */
-	snd_soc_dapm_disable_pin(codec, "Mouthpiece");
-	snd_soc_dapm_enable_pin(codec, "Earpiece");
-	snd_soc_dapm_enable_pin(codec, "Microphone");
-	snd_soc_dapm_disable_pin(codec, "Speaker");
-	snd_soc_dapm_disable_pin(codec, "AGCIN");
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+	snd_soc_dapm_enable_pin(dapm, "Earpiece");
+	snd_soc_dapm_enable_pin(dapm, "Microphone");
+	snd_soc_dapm_disable_pin(dapm, "Speaker");
+	snd_soc_dapm_disable_pin(dapm, "AGCIN");
+	snd_soc_dapm_sync(dapm);
 }
 
 /* Line discipline .hangup() */
@@ -432,16 +435,16 @@
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF)
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
 			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
 						AMS_DELTA_LATCH2_MODEM_NRESET);
 		break;
 	case SND_SOC_BIAS_OFF:
-		if (codec->bias_level != SND_SOC_BIAS_OFF)
+		if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
 			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
 						0);
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -492,6 +495,7 @@
 static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_card *card = rtd->card;
 	int ret;
@@ -541,7 +545,7 @@
 	}
 
 	/* Add board specific DAPM widgets and routes */
-	ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
 					ARRAY_SIZE(ams_delta_dapm_widgets));
 	if (ret) {
 		dev_warn(card->dev,
@@ -550,7 +554,7 @@
 		return 0;
 	}
 
-	ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
+	ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
 					ARRAY_SIZE(ams_delta_audio_map));
 	if (ret) {
 		dev_warn(card->dev,
@@ -560,13 +564,13 @@
 	}
 
 	/* Set up initial pin constellation */
-	snd_soc_dapm_disable_pin(codec, "Mouthpiece");
-	snd_soc_dapm_enable_pin(codec, "Earpiece");
-	snd_soc_dapm_enable_pin(codec, "Microphone");
-	snd_soc_dapm_disable_pin(codec, "Speaker");
-	snd_soc_dapm_disable_pin(codec, "AGCIN");
-	snd_soc_dapm_disable_pin(codec, "AGCOUT");
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+	snd_soc_dapm_enable_pin(dapm, "Earpiece");
+	snd_soc_dapm_enable_pin(dapm, "Microphone");
+	snd_soc_dapm_disable_pin(dapm, "Speaker");
+	snd_soc_dapm_disable_pin(dapm, "AGCIN");
+	snd_soc_dapm_disable_pin(dapm, "AGCOUT");
+	snd_soc_dapm_sync(dapm);
 
 	/* Add virtual switch */
 	ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index a3b6d89..296cd9b 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -58,6 +58,7 @@
 
 static void n810_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int hp = 0, line1l = 0;
 
 	switch (n810_jack_func) {
@@ -72,25 +73,25 @@
 	}
 
 	if (n810_spk_func)
-		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 	else
-		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
 
 	if (hp)
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 	else
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
 	if (line1l)
-		snd_soc_dapm_enable_pin(codec, "LINE1L");
+		snd_soc_dapm_enable_pin(dapm, "LINE1L");
 	else
-		snd_soc_dapm_disable_pin(codec, "LINE1L");
+		snd_soc_dapm_disable_pin(dapm, "LINE1L");
 
 	if (n810_dmic_func)
-		snd_soc_dapm_enable_pin(codec, "DMic");
+		snd_soc_dapm_enable_pin(dapm, "DMic");
 	else
-		snd_soc_dapm_disable_pin(codec, "DMic");
+		snd_soc_dapm_disable_pin(dapm, "DMic");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int n810_startup(struct snd_pcm_substream *substream)
@@ -274,17 +275,18 @@
 static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* Not connected */
-	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
-	snd_soc_dapm_nc_pin(codec, "HPLCOM");
-	snd_soc_dapm_nc_pin(codec, "HPRCOM");
-	snd_soc_dapm_nc_pin(codec, "MIC3L");
-	snd_soc_dapm_nc_pin(codec, "MIC3R");
-	snd_soc_dapm_nc_pin(codec, "LINE1R");
-	snd_soc_dapm_nc_pin(codec, "LINE2L");
-	snd_soc_dapm_nc_pin(codec, "LINE2R");
+	snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(dapm, "HPLCOM");
+	snd_soc_dapm_nc_pin(dapm, "HPRCOM");
+	snd_soc_dapm_nc_pin(dapm, "MIC3L");
+	snd_soc_dapm_nc_pin(dapm, "MIC3R");
+	snd_soc_dapm_nc_pin(dapm, "LINE1R");
+	snd_soc_dapm_nc_pin(dapm, "LINE2L");
+	snd_soc_dapm_nc_pin(dapm, "LINE2R");
 
 	/* Add N810 specific controls */
 	err = snd_soc_add_controls(codec, aic33_n810_controls,
@@ -293,13 +295,13 @@
 		return err;
 
 	/* Add N810 specific widgets */
-	snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets,
 				  ARRAY_SIZE(aic33_dapm_widgets));
 
 	/* Set up N810 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index dbd9d96..93e83c0 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -170,51 +170,53 @@
 static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* All TWL4030 output pins are floating */
-	snd_soc_dapm_nc_pin(codec, "EARPIECE");
-	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
-	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
-	snd_soc_dapm_nc_pin(codec, "HSOL");
-	snd_soc_dapm_nc_pin(codec, "HSOR");
-	snd_soc_dapm_nc_pin(codec, "CARKITL");
-	snd_soc_dapm_nc_pin(codec, "CARKITR");
-	snd_soc_dapm_nc_pin(codec, "HFL");
-	snd_soc_dapm_nc_pin(codec, "HFR");
-	snd_soc_dapm_nc_pin(codec, "VIBRA");
+	snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+	snd_soc_dapm_nc_pin(dapm, "HSOL");
+	snd_soc_dapm_nc_pin(dapm, "HSOR");
+	snd_soc_dapm_nc_pin(dapm, "CARKITL");
+	snd_soc_dapm_nc_pin(dapm, "CARKITR");
+	snd_soc_dapm_nc_pin(dapm, "HFL");
+	snd_soc_dapm_nc_pin(dapm, "HFR");
+	snd_soc_dapm_nc_pin(dapm, "VIBRA");
 
-	ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, omap3pandora_out_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
 	if (ret < 0)
 		return ret;
 
-	snd_soc_dapm_add_routes(codec, omap3pandora_out_map,
+	snd_soc_dapm_add_routes(dapm, omap3pandora_out_map,
 		ARRAY_SIZE(omap3pandora_out_map));
 
-	return snd_soc_dapm_sync(codec);
+	return snd_soc_dapm_sync(dapm);
 }
 
 static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* Not comnnected */
-	snd_soc_dapm_nc_pin(codec, "HSMIC");
-	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
-	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
-	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+	snd_soc_dapm_nc_pin(dapm, "HSMIC");
+	snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
 
-	ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, omap3pandora_in_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_in_dapm_widgets));
 	if (ret < 0)
 		return ret;
 
-	snd_soc_dapm_add_routes(codec, omap3pandora_in_map,
+	snd_soc_dapm_add_routes(dapm, omap3pandora_in_map,
 		ARRAY_SIZE(omap3pandora_in_map));
 
-	return snd_soc_dapm_sync(codec);
+	return snd_soc_dapm_sync(dapm);
 }
 
 static struct snd_soc_ops omap3pandora_ops = {
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index f0e6625..c2a5420 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -116,19 +116,20 @@
 static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add osk5912 specific widgets */
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
 				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
 
 	/* Set up osk5912 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Line In");
-	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 04b5723..62fc7a4 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -58,19 +58,21 @@
 
 static void rx51_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	if (rx51_spk_func)
-		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 	else
-		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
 	if (rx51_dmic_func)
-		snd_soc_dapm_enable_pin(codec, "DMic");
+		snd_soc_dapm_enable_pin(dapm, "DMic");
 	else
-		snd_soc_dapm_disable_pin(codec, "DMic");
+		snd_soc_dapm_disable_pin(dapm, "DMic");
 
 	gpio_set_value(RX51_TVOUT_SEL_GPIO,
 		       rx51_jack_func == RX51_JACK_TVOUT);
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int rx51_startup(struct snd_pcm_substream *substream)
@@ -244,12 +246,13 @@
 static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* Set up NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "MIC3L");
-	snd_soc_dapm_nc_pin(codec, "MIC3R");
-	snd_soc_dapm_nc_pin(codec, "LINE1R");
+	snd_soc_dapm_nc_pin(dapm, "MIC3L");
+	snd_soc_dapm_nc_pin(dapm, "MIC3R");
+	snd_soc_dapm_nc_pin(dapm, "LINE1R");
 
 	/* Add RX-51 specific controls */
 	err = snd_soc_add_controls(codec, aic34_rx51_controls,
@@ -258,13 +261,13 @@
 		return err;
 
 	/* Add RX-51 specific widgets */
-	snd_soc_dapm_new_controls(codec, aic34_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, aic34_dapm_widgets,
 				  ARRAY_SIZE(aic34_dapm_widgets));
 
 	/* Set up RX-51 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	/* AV jack detection */
 	err = snd_soc_jack_new(codec, "AV Jack",
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 07fbcf7..a3dd07a 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -191,39 +191,40 @@
 static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* Add SDP3430 specific widgets */
-	ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, sdp3430_twl4030_dapm_widgets,
 				ARRAY_SIZE(sdp3430_twl4030_dapm_widgets));
 	if (ret)
 		return ret;
 
 	/* Set up SDP3430 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* SDP3430 connected pins */
-	snd_soc_dapm_enable_pin(codec, "Ext Mic");
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
-	snd_soc_dapm_disable_pin(codec, "Headset Mic");
-	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
 
 	/* TWL4030 not connected pins */
-	snd_soc_dapm_nc_pin(codec, "AUXL");
-	snd_soc_dapm_nc_pin(codec, "AUXR");
-	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
-	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
-	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+	snd_soc_dapm_nc_pin(dapm, "AUXL");
+	snd_soc_dapm_nc_pin(dapm, "AUXR");
+	snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
 
-	snd_soc_dapm_nc_pin(codec, "OUTL");
-	snd_soc_dapm_nc_pin(codec, "OUTR");
-	snd_soc_dapm_nc_pin(codec, "EARPIECE");
-	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
-	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
-	snd_soc_dapm_nc_pin(codec, "CARKITL");
-	snd_soc_dapm_nc_pin(codec, "CARKITR");
+	snd_soc_dapm_nc_pin(dapm, "OUTL");
+	snd_soc_dapm_nc_pin(dapm, "OUTR");
+	snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+	snd_soc_dapm_nc_pin(dapm, "CARKITL");
+	snd_soc_dapm_nc_pin(dapm, "CARKITR");
 
-	ret = snd_soc_dapm_sync(codec);
+	ret = snd_soc_dapm_sync(dapm);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
index 4b4463d..3ce1731 100644
--- a/sound/soc/omap/sdp4430.c
+++ b/sound/soc/omap/sdp4430.c
@@ -129,6 +129,7 @@
 static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* Add SDP4430 specific controls */
@@ -138,25 +139,25 @@
 		return ret;
 
 	/* Add SDP4430 specific widgets */
-	ret = snd_soc_dapm_new_controls(codec, sdp4430_twl6040_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets,
 				ARRAY_SIZE(sdp4430_twl6040_dapm_widgets));
 	if (ret)
 		return ret;
 
 	/* Set up SDP4430 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* SDP4430 connected pins */
-	snd_soc_dapm_enable_pin(codec, "Ext Mic");
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
-	snd_soc_dapm_enable_pin(codec, "Headset Mic");
-	snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
 
 	/* TWL6040 not connected pins */
-	snd_soc_dapm_nc_pin(codec, "AFML");
-	snd_soc_dapm_nc_pin(codec, "AFMR");
+	snd_soc_dapm_nc_pin(dapm, "AFML");
+	snd_soc_dapm_nc_pin(dapm, "AFMR");
 
-	ret = snd_soc_dapm_sync(codec);
+	ret = snd_soc_dapm_sync(dapm);
 
 	return ret;
 }
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
index 718031e..cc5bc523b 100644
--- a/sound/soc/omap/zoom2.c
+++ b/sound/soc/omap/zoom2.c
@@ -162,35 +162,36 @@
 static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* Add Zoom2 specific widgets */
-	ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, zoom2_twl4030_dapm_widgets,
 				ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
 	if (ret)
 		return ret;
 
 	/* Set up Zoom2 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* Zoom2 connected pins */
-	snd_soc_dapm_enable_pin(codec, "Ext Mic");
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
-	snd_soc_dapm_enable_pin(codec, "Headset Mic");
-	snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
-	snd_soc_dapm_enable_pin(codec, "Aux In");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Aux In");
 
 	/* TWL4030 not connected pins */
-	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
-	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
-	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
-	snd_soc_dapm_nc_pin(codec, "EARPIECE");
-	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
-	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
-	snd_soc_dapm_nc_pin(codec, "CARKITL");
-	snd_soc_dapm_nc_pin(codec, "CARKITR");
+	snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
+	snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+	snd_soc_dapm_nc_pin(dapm, "CARKITL");
+	snd_soc_dapm_nc_pin(dapm, "CARKITR");
 
-	ret = snd_soc_dapm_sync(codec);
+	ret = snd_soc_dapm_sync(dapm);
 
 	return ret;
 }
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index f451acd..85956ff 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -48,51 +48,53 @@
 
 static void corgi_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	/* set up jack connection */
 	switch (corgi_jack_func) {
 	case CORGI_HP:
 		/* set = unmute headphone */
 		gpio_set_value(CORGI_GPIO_MUTE_L, 1);
 		gpio_set_value(CORGI_GPIO_MUTE_R, 1);
-		snd_soc_dapm_disable_pin(codec, "Mic Jack");
-		snd_soc_dapm_disable_pin(codec, "Line Jack");
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
 		break;
 	case CORGI_MIC:
 		/* reset = mute headphone */
 		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
 		gpio_set_value(CORGI_GPIO_MUTE_R, 0);
-		snd_soc_dapm_enable_pin(codec, "Mic Jack");
-		snd_soc_dapm_disable_pin(codec, "Line Jack");
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin(dapm, "Line Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
 		break;
 	case CORGI_LINE:
 		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
 		gpio_set_value(CORGI_GPIO_MUTE_R, 0);
-		snd_soc_dapm_disable_pin(codec, "Mic Jack");
-		snd_soc_dapm_enable_pin(codec, "Line Jack");
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_enable_pin(dapm, "Line Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
 		break;
 	case CORGI_HEADSET:
 		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
 		gpio_set_value(CORGI_GPIO_MUTE_R, 1);
-		snd_soc_dapm_enable_pin(codec, "Mic Jack");
-		snd_soc_dapm_disable_pin(codec, "Line Jack");
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_enable_pin(codec, "Headset Jack");
+		snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin(dapm, "Line Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headset Jack");
 		break;
 	}
 
 	if (corgi_spk_func == CORGI_SPK_ON)
-		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 	else
-		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
 
 	/* signal a DAPM event */
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int corgi_startup(struct snd_pcm_substream *substream)
@@ -279,10 +281,11 @@
 static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
-	snd_soc_dapm_nc_pin(codec, "LLINEIN");
-	snd_soc_dapm_nc_pin(codec, "RLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "LLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "RLINEIN");
 
 	/* Add corgi specific controls */
 	err = snd_soc_add_controls(codec, wm8731_corgi_controls,
@@ -291,13 +294,13 @@
 		return err;
 
 	/* Add corgi specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets,
 				  ARRAY_SIZE(wm8731_dapm_widgets));
 
 	/* Set up corgi specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index c82cedb..38a84b8 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -92,23 +92,24 @@
 static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_nc_pin(codec, "HPOUTL");
-	snd_soc_dapm_nc_pin(codec, "HPOUTR");
-	snd_soc_dapm_nc_pin(codec, "PHONE");
-	snd_soc_dapm_nc_pin(codec, "LINEINL");
-	snd_soc_dapm_nc_pin(codec, "LINEINR");
-	snd_soc_dapm_nc_pin(codec, "CDINL");
-	snd_soc_dapm_nc_pin(codec, "CDINR");
-	snd_soc_dapm_nc_pin(codec, "PCBEEP");
-	snd_soc_dapm_nc_pin(codec, "MIC2");
+	snd_soc_dapm_nc_pin(dapm, "HPOUTL");
+	snd_soc_dapm_nc_pin(dapm, "HPOUTR");
+	snd_soc_dapm_nc_pin(dapm, "PHONE");
+	snd_soc_dapm_nc_pin(dapm, "LINEINL");
+	snd_soc_dapm_nc_pin(dapm, "LINEINR");
+	snd_soc_dapm_nc_pin(dapm, "CDINL");
+	snd_soc_dapm_nc_pin(dapm, "CDINR");
+	snd_soc_dapm_nc_pin(dapm, "PCBEEP");
+	snd_soc_dapm_nc_pin(dapm, "MIC2");
 
-	snd_soc_dapm_new_controls(codec, e740_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, e740_dapm_widgets,
 					ARRAY_SIZE(e740_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 4c14380..2bc97e9 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -74,23 +74,24 @@
 static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_nc_pin(codec, "LOUT");
-	snd_soc_dapm_nc_pin(codec, "ROUT");
-	snd_soc_dapm_nc_pin(codec, "PHONE");
-	snd_soc_dapm_nc_pin(codec, "LINEINL");
-	snd_soc_dapm_nc_pin(codec, "LINEINR");
-	snd_soc_dapm_nc_pin(codec, "CDINL");
-	snd_soc_dapm_nc_pin(codec, "CDINR");
-	snd_soc_dapm_nc_pin(codec, "PCBEEP");
-	snd_soc_dapm_nc_pin(codec, "MIC2");
+	snd_soc_dapm_nc_pin(dapm, "LOUT");
+	snd_soc_dapm_nc_pin(dapm, "ROUT");
+	snd_soc_dapm_nc_pin(dapm, "PHONE");
+	snd_soc_dapm_nc_pin(dapm, "LINEINL");
+	snd_soc_dapm_nc_pin(dapm, "LINEINR");
+	snd_soc_dapm_nc_pin(dapm, "CDINL");
+	snd_soc_dapm_nc_pin(dapm, "CDINR");
+	snd_soc_dapm_nc_pin(dapm, "PCBEEP");
+	snd_soc_dapm_nc_pin(dapm, "MIC2");
 
-	snd_soc_dapm_new_controls(codec, e750_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, e750_dapm_widgets,
 					ARRAY_SIZE(e750_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index d42e5fe..eac846c 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -75,12 +75,13 @@
 static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, e800_dapm_widgets,
 					ARRAY_SIZE(e800_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 5ef0526..98cb990 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -44,27 +44,29 @@
 
 static void magician_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	if (magician_spk_switch)
-		snd_soc_dapm_enable_pin(codec, "Speaker");
+		snd_soc_dapm_enable_pin(dapm, "Speaker");
 	else
-		snd_soc_dapm_disable_pin(codec, "Speaker");
+		snd_soc_dapm_disable_pin(dapm, "Speaker");
 	if (magician_hp_switch)
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 	else
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
 
 	switch (magician_in_sel) {
 	case MAGICIAN_MIC:
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_enable_pin(codec, "Call Mic");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_enable_pin(dapm, "Call Mic");
 		break;
 	case MAGICIAN_MIC_EXT:
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
-		snd_soc_dapm_enable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
+		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
 		break;
 	}
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int magician_startup(struct snd_pcm_substream *substream)
@@ -399,15 +401,16 @@
 static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "VOUTLHP");
-	snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+	snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
+	snd_soc_dapm_nc_pin(dapm, "VOUTRHP");
 
 	/* FIXME: is anything connected here? */
-	snd_soc_dapm_nc_pin(codec, "VINL");
-	snd_soc_dapm_nc_pin(codec, "VINR");
+	snd_soc_dapm_nc_pin(dapm, "VINL");
+	snd_soc_dapm_nc_pin(dapm, "VINR");
 
 	/* Add magician specific controls */
 	err = snd_soc_add_controls(codec, uda1380_magician_controls,
@@ -416,13 +419,13 @@
 		return err;
 
 	/* Add magician specific widgets */
-	snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
 				  ARRAY_SIZE(uda1380_dapm_widgets));
 
 	/* Set up magician specific audio path interconnects */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index f284cc5..f7a1e8f 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -130,13 +130,14 @@
 static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	unsigned short reg;
 
 	/* Add mioa701 specific widgets */
-	snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets));
+	snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets));
 
 	/* Set up mioa701 specific audio path audio_mapnects */
-	snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map));
 
 	/* Prepare GPIO8 for rear speaker amplifier */
 	reg = codec->driver->read(codec, AC97_GPIO_CFG);
@@ -146,12 +147,12 @@
 	reg = codec->driver->read(codec, AC97_3D_CONTROL);
 	codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000);
 
-	snd_soc_dapm_enable_pin(codec, "Front Speaker");
-	snd_soc_dapm_enable_pin(codec, "Rear Speaker");
-	snd_soc_dapm_enable_pin(codec, "Front Mic");
-	snd_soc_dapm_enable_pin(codec, "GSM Line In");
-	snd_soc_dapm_enable_pin(codec, "GSM Line Out");
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_enable_pin(dapm, "Front Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Rear Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Front Mic");
+	snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+	snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 13f6d48..530064d 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -77,37 +77,38 @@
 static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* add palm27x specific widgets */
-	err = snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+	err = snd_soc_dapm_new_controls(dapm, palm27x_dapm_widgets,
 				ARRAY_SIZE(palm27x_dapm_widgets));
 	if (err)
 		return err;
 
 	/* set up palm27x specific audio path audio_map */
-	err = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	err = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 	if (err)
 		return err;
 
 	/* connected pins */
 	if (machine_is_palmld())
-		snd_soc_dapm_enable_pin(codec, "MIC1");
-	snd_soc_dapm_enable_pin(codec, "HPOUTL");
-	snd_soc_dapm_enable_pin(codec, "HPOUTR");
-	snd_soc_dapm_enable_pin(codec, "LOUT2");
-	snd_soc_dapm_enable_pin(codec, "ROUT2");
+		snd_soc_dapm_enable_pin(dapm, "MIC1");
+	snd_soc_dapm_enable_pin(dapm, "HPOUTL");
+	snd_soc_dapm_enable_pin(dapm, "HPOUTR");
+	snd_soc_dapm_enable_pin(dapm, "LOUT2");
+	snd_soc_dapm_enable_pin(dapm, "ROUT2");
 
 	/* not connected pins */
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "MONOOUT");
-	snd_soc_dapm_nc_pin(codec, "LINEINL");
-	snd_soc_dapm_nc_pin(codec, "LINEINR");
-	snd_soc_dapm_nc_pin(codec, "PCBEEP");
-	snd_soc_dapm_nc_pin(codec, "PHONE");
-	snd_soc_dapm_nc_pin(codec, "MIC2");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "MONOOUT");
+	snd_soc_dapm_nc_pin(dapm, "LINEINL");
+	snd_soc_dapm_nc_pin(dapm, "LINEINR");
+	snd_soc_dapm_nc_pin(dapm, "PCBEEP");
+	snd_soc_dapm_nc_pin(dapm, "PHONE");
+	snd_soc_dapm_nc_pin(dapm, "MIC2");
 
-	err = snd_soc_dapm_sync(codec);
+	err = snd_soc_dapm_sync(dapm);
 	if (err)
 		return err;
 
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 84edd03..f45ea408 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -46,6 +46,8 @@
 
 static void poodle_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	/* set up jack connection */
 	if (poodle_jack_func == POODLE_HP) {
 		/* set = unmute headphone */
@@ -53,23 +55,23 @@
 			POODLE_LOCOMO_GPIO_MUTE_L, 1);
 		locomo_gpio_write(&poodle_locomo_device.dev,
 			POODLE_LOCOMO_GPIO_MUTE_R, 1);
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 	} else {
 		locomo_gpio_write(&poodle_locomo_device.dev,
 			POODLE_LOCOMO_GPIO_MUTE_L, 0);
 		locomo_gpio_write(&poodle_locomo_device.dev,
 			POODLE_LOCOMO_GPIO_MUTE_R, 0);
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
 	}
 
 	/* set the enpoints to their new connetion states */
 	if (poodle_spk_func == POODLE_SPK_ON)
-		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 	else
-		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
 
 	/* signal a DAPM event */
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int poodle_startup(struct snd_pcm_substream *substream)
@@ -244,11 +246,12 @@
 static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
-	snd_soc_dapm_nc_pin(codec, "LLINEIN");
-	snd_soc_dapm_nc_pin(codec, "RLINEIN");
-	snd_soc_dapm_enable_pin(codec, "MICIN");
+	snd_soc_dapm_nc_pin(dapm, "LLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+	snd_soc_dapm_enable_pin(dapm, "MICIN");
 
 	/* Add poodle specific controls */
 	err = snd_soc_add_controls(codec, wm8731_poodle_controls,
@@ -257,13 +260,13 @@
 		return err;
 
 	/* Add poodle specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets,
 				  ARRAY_SIZE(wm8731_dapm_widgets));
 
 	/* Set up poodle specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c
index d63cb47..ee06f99 100644
--- a/sound/soc/pxa/saarb.c
+++ b/sound/soc/pxa/saarb.c
@@ -133,20 +133,21 @@
 static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	snd_soc_dapm_new_controls(codec, saarb_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets,
 				  ARRAY_SIZE(saarb_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* connected pins */
-	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
-	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
-	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
-	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
-	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
 
-	ret = snd_soc_dapm_sync(codec);
+	ret = snd_soc_dapm_sync(dapm);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 0b30d7d..7e13440 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -46,61 +46,63 @@
 
 static void spitz_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	if (spitz_spk_func == SPITZ_SPK_ON)
-		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 	else
-		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
 
 	/* set up jack connection */
 	switch (spitz_jack_func) {
 	case SPITZ_HP:
 		/* enable and unmute hp jack, disable mic bias */
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
-		snd_soc_dapm_disable_pin(codec, "Mic Jack");
-		snd_soc_dapm_disable_pin(codec, "Line Jack");
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 		gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
 		gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
 		break;
 	case SPITZ_MIC:
 		/* enable mic jack and bias, mute hp */
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
-		snd_soc_dapm_disable_pin(codec, "Line Jack");
-		snd_soc_dapm_enable_pin(codec, "Mic Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
 		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
 		break;
 	case SPITZ_LINE:
 		/* enable line jack, disable mic bias and mute hp */
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
-		snd_soc_dapm_disable_pin(codec, "Mic Jack");
-		snd_soc_dapm_enable_pin(codec, "Line Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_enable_pin(dapm, "Line Jack");
 		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
 		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
 		break;
 	case SPITZ_HEADSET:
 		/* enable and unmute headset jack enable mic bias, mute L hp */
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_enable_pin(codec, "Mic Jack");
-		snd_soc_dapm_disable_pin(codec, "Line Jack");
-		snd_soc_dapm_enable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headset Jack");
 		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
 		gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
 		break;
 	case SPITZ_HP_OFF:
 
 		/* jack removed, everything off */
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
-		snd_soc_dapm_disable_pin(codec, "Mic Jack");
-		snd_soc_dapm_disable_pin(codec, "Line Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin(dapm, "Line Jack");
 		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
 		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
 		break;
 	}
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int spitz_startup(struct snd_pcm_substream *substream)
@@ -281,16 +283,17 @@
 static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "RINPUT1");
-	snd_soc_dapm_nc_pin(codec, "LINPUT2");
-	snd_soc_dapm_nc_pin(codec, "RINPUT2");
-	snd_soc_dapm_nc_pin(codec, "LINPUT3");
-	snd_soc_dapm_nc_pin(codec, "RINPUT3");
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "MONO1");
+	snd_soc_dapm_nc_pin(dapm, "RINPUT1");
+	snd_soc_dapm_nc_pin(dapm, "LINPUT2");
+	snd_soc_dapm_nc_pin(dapm, "RINPUT2");
+	snd_soc_dapm_nc_pin(dapm, "LINPUT3");
+	snd_soc_dapm_nc_pin(dapm, "RINPUT3");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "MONO1");
 
 	/* Add spitz specific controls */
 	err = snd_soc_add_controls(codec, wm8750_spitz_controls,
@@ -299,13 +302,13 @@
 		return err;
 
 	/* Add spitz specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
 				  ARRAY_SIZE(wm8750_dapm_widgets));
 
 	/* Set up spitz specific audio paths */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c
index 248c283..18cbe0e 100644
--- a/sound/soc/pxa/tavorevb3.c
+++ b/sound/soc/pxa/tavorevb3.c
@@ -133,20 +133,21 @@
 static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	snd_soc_dapm_new_controls(codec, evb3_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, evb3_dapm_widgets,
 				  ARRAY_SIZE(evb3_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* connected pins */
-	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
-	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
-	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
-	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
-	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
 
-	ret = snd_soc_dapm_sync(codec);
+	ret = snd_soc_dapm_sync(dapm);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 7b983f9..7f81f82 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -49,31 +49,33 @@
 
 static void tosa_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	/* set up jack connection */
 	switch (tosa_jack_func) {
 	case TOSA_HP:
-		snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
 		break;
 	case TOSA_MIC_INT:
-		snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_enable_pin(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
 		break;
 	case TOSA_HEADSET:
-		snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_enable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headset Jack");
 		break;
 	}
 
 	if (tosa_spk_func == TOSA_SPK_ON)
-		snd_soc_dapm_enable_pin(codec, "Speaker");
+		snd_soc_dapm_enable_pin(dapm, "Speaker");
 	else
-		snd_soc_dapm_disable_pin(codec, "Speaker");
+		snd_soc_dapm_disable_pin(dapm, "Speaker");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int tosa_startup(struct snd_pcm_substream *substream)
@@ -191,10 +193,11 @@
 static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "MONOOUT");
 
 	/* add tosa specific controls */
 	err = snd_soc_add_controls(codec, tosa_controls,
@@ -203,13 +206,13 @@
 		return err;
 
 	/* add tosa specific widgets */
-	snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets,
 				  ARRAY_SIZE(tosa_dapm_widgets));
 
 	/* set up tosa specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 4cc841b..cacbcd4 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -140,22 +140,23 @@
 static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* NC codec pins */
-	snd_soc_dapm_disable_pin(codec, "LINPUT3");
-	snd_soc_dapm_disable_pin(codec, "RINPUT3");
-	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "MONO");
+	snd_soc_dapm_disable_pin(dapm, "LINPUT3");
+	snd_soc_dapm_disable_pin(dapm, "RINPUT3");
+	snd_soc_dapm_disable_pin(dapm, "OUT3");
+	snd_soc_dapm_disable_pin(dapm, "MONO");
 
 	/* Add z2 specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
 				 ARRAY_SIZE(wm8750_dapm_widgets));
 
 	/* Set up z2 specific audio paths */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	ret = snd_soc_dapm_sync(codec);
+	ret = snd_soc_dapm_sync(dapm);
 	if (ret)
 		goto err;
 
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index d27e05a..c74eac3 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -73,21 +73,22 @@
 static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	if (clk_pout)
 		snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
 				    clk_get_rate(pout), 0);
 
-	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, zylonite_dapm_widgets,
 				  ARRAY_SIZE(zylonite_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* Static setup for now */
-	snd_soc_dapm_enable_pin(codec, "Headphone");
-	snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
+	snd_soc_dapm_enable_pin(dapm, "Headphone");
+	snd_soc_dapm_enable_pin(dapm, "Headset Earpiece");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c
index 235d197..33bebda 100644
--- a/sound/soc/s3c24xx/aquila_wm8994.c
+++ b/sound/soc/s3c24xx/aquila_wm8994.c
@@ -93,27 +93,28 @@
 static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* add aquila specific widgets */
-	snd_soc_dapm_new_controls(codec, aquila_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, aquila_dapm_widgets,
 			ARRAY_SIZE(aquila_dapm_widgets));
 
 	/* set up aquila specific audio routes */
-	snd_soc_dapm_add_routes(codec, aquila_dapm_routes,
+	snd_soc_dapm_add_routes(dapm, aquila_dapm_routes,
 			ARRAY_SIZE(aquila_dapm_routes));
 
 	/* set endpoints to not connected */
-	snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN");
-	snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT1N");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT1P");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT2N");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT2P");
-	snd_soc_dapm_nc_pin(codec, "SPKOUTRN");
-	snd_soc_dapm_nc_pin(codec, "SPKOUTRP");
+	snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "SPKOUTRN");
+	snd_soc_dapm_nc_pin(dapm, "SPKOUTRP");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	/* Headset jack detection */
 	ret = snd_soc_jack_new(&aquila, "Headset Jack",
diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c
index 694f702..052729c 100644
--- a/sound/soc/s3c24xx/goni_wm8994.c
+++ b/sound/soc/s3c24xx/goni_wm8994.c
@@ -97,25 +97,26 @@
 static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	/* add goni specific widgets */
-	snd_soc_dapm_new_controls(codec, goni_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, goni_dapm_widgets,
 			ARRAY_SIZE(goni_dapm_widgets));
 
 	/* set up goni specific audio routes */
-	snd_soc_dapm_add_routes(codec, goni_dapm_routes,
+	snd_soc_dapm_add_routes(dapm, goni_dapm_routes,
 			ARRAY_SIZE(goni_dapm_routes));
 
 	/* set endpoints to not connected */
-	snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN");
-	snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT1N");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT1P");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT2N");
-	snd_soc_dapm_nc_pin(codec, "LINEOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	/* Headset jack detection */
 	ret = snd_soc_jack_new(&goni, "Headset Jack",
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 49605cd..e3599e2 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -111,18 +111,19 @@
 static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* These endpoints are not being used. */
-	snd_soc_dapm_nc_pin(codec, "LINPUT2");
-	snd_soc_dapm_nc_pin(codec, "RINPUT2");
-	snd_soc_dapm_nc_pin(codec, "LINPUT3");
-	snd_soc_dapm_nc_pin(codec, "RINPUT3");
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "MONO");
+	snd_soc_dapm_nc_pin(dapm, "LINPUT2");
+	snd_soc_dapm_nc_pin(dapm, "RINPUT2");
+	snd_soc_dapm_nc_pin(dapm, "LINPUT3");
+	snd_soc_dapm_nc_pin(dapm, "RINPUT3");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "MONO");
 
 	/* Add jive specific widgets */
-	err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+	err = snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
 					ARRAY_SIZE(wm8750_dapm_widgets));
 	if (err) {
 		printk(KERN_ERR "%s: failed to add widgets (%d)\n",
@@ -130,8 +131,8 @@
 		return err;
 	}
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
index e97bdf1..c3f63ef 100644
--- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -333,16 +333,17 @@
 static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "OUT4");
-	snd_soc_dapm_nc_pin(codec, "LINE1");
-	snd_soc_dapm_nc_pin(codec, "LINE2");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "OUT4");
+	snd_soc_dapm_nc_pin(dapm, "LINE1");
+	snd_soc_dapm_nc_pin(dapm, "LINE2");
 
 	/* Add neo1973 gta02 specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
 				  ARRAY_SIZE(wm8753_dapm_widgets));
 
 	/* add neo1973 gta02 specific controls */
@@ -353,25 +354,25 @@
 		return err;
 
 	/* set up neo1973 gta02 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* set endpoints to default off mode */
-	snd_soc_dapm_disable_pin(codec, "Stereo Out");
-	snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-	snd_soc_dapm_disable_pin(codec, "GSM Line In");
-	snd_soc_dapm_disable_pin(codec, "Headset Mic");
-	snd_soc_dapm_disable_pin(codec, "Handset Mic");
-	snd_soc_dapm_disable_pin(codec, "Handset Spk");
+	snd_soc_dapm_disable_pin(dapm, "Stereo Out");
+	snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+	snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+	snd_soc_dapm_disable_pin(dapm, "Handset Spk");
 
 	/* allow audio paths from the GSM modem to run during suspend */
-	snd_soc_dapm_ignore_suspend(codec, "Stereo Out");
-	snd_soc_dapm_ignore_suspend(codec, "GSM Line Out");
-	snd_soc_dapm_ignore_suspend(codec, "GSM Line In");
-	snd_soc_dapm_ignore_suspend(codec, "Headset Mic");
-	snd_soc_dapm_ignore_suspend(codec, "Handset Mic");
-	snd_soc_dapm_ignore_suspend(codec, "Handset Spk");
+	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
+	snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
+	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index f4f2ee7..e94ffe0 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -237,81 +237,83 @@
 
 static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	pr_debug("Entered %s\n", __func__);
 
 	switch (neo1973_scenario) {
 	case NEO_AUDIO_OFF:
-		snd_soc_dapm_disable_pin(codec, "Audio Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_disable_pin(dapm, "Audio Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 		break;
 	case NEO_GSM_CALL_AUDIO_HANDSET:
-		snd_soc_dapm_enable_pin(codec, "Audio Out");
-		snd_soc_dapm_enable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_enable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_enable_pin(codec, "Call Mic");
+		snd_soc_dapm_enable_pin(dapm, "Audio Out");
+		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_enable_pin(dapm, "Call Mic");
 		break;
 	case NEO_GSM_CALL_AUDIO_HEADSET:
-		snd_soc_dapm_enable_pin(codec, "Audio Out");
-		snd_soc_dapm_enable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_enable_pin(codec, "GSM Line In");
-		snd_soc_dapm_enable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_enable_pin(dapm, "Audio Out");
+		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 		break;
 	case NEO_GSM_CALL_AUDIO_BLUETOOTH:
-		snd_soc_dapm_disable_pin(codec, "Audio Out");
-		snd_soc_dapm_enable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_enable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_disable_pin(dapm, "Audio Out");
+		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 		break;
 	case NEO_STEREO_TO_SPEAKERS:
-		snd_soc_dapm_enable_pin(codec, "Audio Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_enable_pin(dapm, "Audio Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 		break;
 	case NEO_STEREO_TO_HEADPHONES:
-		snd_soc_dapm_enable_pin(codec, "Audio Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_enable_pin(dapm, "Audio Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 		break;
 	case NEO_CAPTURE_HANDSET:
-		snd_soc_dapm_disable_pin(codec, "Audio Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_enable_pin(codec, "Call Mic");
+		snd_soc_dapm_disable_pin(dapm, "Audio Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_enable_pin(dapm, "Call Mic");
 		break;
 	case NEO_CAPTURE_HEADSET:
-		snd_soc_dapm_disable_pin(codec, "Audio Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line In");
-		snd_soc_dapm_enable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_disable_pin(dapm, "Audio Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 		break;
 	case NEO_CAPTURE_BLUETOOTH:
-		snd_soc_dapm_disable_pin(codec, "Audio Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_disable_pin(dapm, "Audio Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 		break;
 	default:
-		snd_soc_dapm_disable_pin(codec, "Audio Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
-		snd_soc_dapm_disable_pin(codec, "GSM Line In");
-		snd_soc_dapm_disable_pin(codec, "Headset Mic");
-		snd_soc_dapm_disable_pin(codec, "Call Mic");
+		snd_soc_dapm_disable_pin(dapm, "Audio Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
 	}
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -502,20 +504,21 @@
 static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	pr_debug("Entered %s\n", __func__);
 
 	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "LOUT2");
-	snd_soc_dapm_nc_pin(codec, "ROUT2");
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "OUT4");
-	snd_soc_dapm_nc_pin(codec, "LINE1");
-	snd_soc_dapm_nc_pin(codec, "LINE2");
+	snd_soc_dapm_nc_pin(dapm, "LOUT2");
+	snd_soc_dapm_nc_pin(dapm, "ROUT2");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "OUT4");
+	snd_soc_dapm_nc_pin(dapm, "LINE1");
+	snd_soc_dapm_nc_pin(dapm, "LINE2");
 
 	/* Add neo1973 specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
 				  ARRAY_SIZE(wm8753_dapm_widgets));
 
 	/* set endpoints to default mode */
@@ -528,10 +531,10 @@
 		return err;
 
 	/* set up neo1973 specific audio routes */
-	err = snd_soc_dapm_add_routes(codec, dapm_routes,
+	err = snd_soc_dapm_add_routes(dapm, dapm_routes,
 				      ARRAY_SIZE(dapm_routes));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c
index 468cc11..b2a741a 100644
--- a/sound/soc/s3c24xx/rx1950_uda1380.c
+++ b/sound/soc/s3c24xx/rx1950_uda1380.c
@@ -228,26 +228,27 @@
 static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
 	/* Add rx1950 specific widgets */
-	err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+	err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
 				  ARRAY_SIZE(uda1380_dapm_widgets));
 
 	if (err)
 		return err;
 
 	/* Set up rx1950 specific audio path audio_mapnects */
-	err = snd_soc_dapm_add_routes(codec, audio_map,
+	err = snd_soc_dapm_add_routes(dapm, audio_map,
 				      ARRAY_SIZE(audio_map));
 
 	if (err)
 		return err;
 
-	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Speaker");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
 		&hp_jack);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
index f884537..05c7937 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
@@ -76,19 +76,20 @@
 static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, dapm_widgets,
 				  ARRAY_SIZE(dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+	snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map));
 
-	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Line In");
-	snd_soc_dapm_enable_pin(codec, "Line Out");
-	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Line Out");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
 	simtec_audio_init(rtd);
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
index c096759..653dc75 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
@@ -65,19 +65,20 @@
 static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, dapm_widgets,
 				  ARRAY_SIZE(dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+	snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map));
 
-	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Line In");
-	snd_soc_dapm_enable_pin(codec, "Line Out");
-	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Line Out");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
 	simtec_audio_init(rtd);
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c
index dd20ca7..1f6da1e 100644
--- a/sound/soc/s3c24xx/smartq_wm8987.c
+++ b/sound/soc/s3c24xx/smartq_wm8987.c
@@ -158,10 +158,11 @@
 
 static int smartq_wm8987_init(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err = 0;
 
 	/* Add SmartQ specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, wm8987_dapm_widgets,
 				  ARRAY_SIZE(wm8987_dapm_widgets));
 
 	/* add SmartQ specific controls */
@@ -172,20 +173,20 @@
 		return err;
 
 	/* setup SmartQ specific audio path */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* set endpoints to not connected */
-	snd_soc_dapm_nc_pin(codec, "LINPUT1");
-	snd_soc_dapm_nc_pin(codec, "RINPUT1");
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "ROUT1");
+	snd_soc_dapm_nc_pin(dapm, "LINPUT1");
+	snd_soc_dapm_nc_pin(dapm, "RINPUT1");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "ROUT1");
 
 	/* set endpoints to default off mode */
-	snd_soc_dapm_enable_pin(codec, "Internal Speaker");
-	snd_soc_dapm_enable_pin(codec, "Internal Mic");
-	snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Internal Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Internal Mic");
+	snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
 
-	err = snd_soc_dapm_sync(codec);
+	err = snd_soc_dapm_sync(dapm);
 	if (err)
 		return err;
 
diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c
index 052e499..291939c 100644
--- a/sound/soc/s3c24xx/smdk64xx_wm8580.c
+++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c
@@ -182,21 +182,22 @@
 static int smdk64xx_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add smdk64xx specific Capture widgets */
-	snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt,
+	snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_cpt,
 				  ARRAY_SIZE(wm8580_dapm_widgets_cpt));
 
 	/* Set up PAIFTX audio path */
-	snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx));
+	snd_soc_dapm_add_routes(dapm, audio_map_tx, ARRAY_SIZE(audio_map_tx));
 
 	/* Enabling the microphone requires the fitting of a 0R
 	 * resistor to connect the line from the microphone jack.
 	 */
-	snd_soc_dapm_disable_pin(codec, "MicIn");
+	snd_soc_dapm_disable_pin(dapm, "MicIn");
 
 	/* signal a DAPM event */
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -204,16 +205,17 @@
 static int smdk64xx_wm8580_init_paifrx(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add smdk64xx specific Playback widgets */
-	snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk,
+	snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_pbk,
 				  ARRAY_SIZE(wm8580_dapm_widgets_pbk));
 
 	/* Set up PAIFRX audio path */
-	snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx));
+	snd_soc_dapm_add_routes(dapm, audio_map_rx, ARRAY_SIZE(audio_map_rx));
 
 	/* signal a DAPM event */
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 96c05e1..db1803d 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -107,6 +107,7 @@
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = kcontrol->private_data;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	unsigned int val = (ucontrol->value.enumerated.item[0] != 0);
 	char *differential = "Audio Out Differential";
 	char *stereo = "Audio Out Stereo";
@@ -114,10 +115,10 @@
 	if (kcontrol->private_value == val)
 		return 0;
 	kcontrol->private_value = val;
-	snd_soc_dapm_disable_pin(codec, val ? differential : stereo);
-	snd_soc_dapm_sync(codec);
-	snd_soc_dapm_enable_pin(codec, val ? stereo : differential);
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_disable_pin(dapm, val ? differential : stereo);
+	snd_soc_dapm_sync(dapm);
+	snd_soc_dapm_enable_pin(dapm, val ? stereo : differential);
+	snd_soc_dapm_sync(dapm);
 
 	return 1;
 }
@@ -137,35 +138,36 @@
 static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add s6105 specific widgets */
-	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
 				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* Set up s6105 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* not present */
-	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
-	snd_soc_dapm_nc_pin(codec, "LINE2L");
-	snd_soc_dapm_nc_pin(codec, "LINE2R");
+	snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(dapm, "LINE2L");
+	snd_soc_dapm_nc_pin(dapm, "LINE2R");
 
 	/* not connected */
-	snd_soc_dapm_nc_pin(codec, "MIC3L"); /* LINE2L on this chip */
-	snd_soc_dapm_nc_pin(codec, "MIC3R"); /* LINE2R on this chip */
-	snd_soc_dapm_nc_pin(codec, "LLOUT");
-	snd_soc_dapm_nc_pin(codec, "RLOUT");
-	snd_soc_dapm_nc_pin(codec, "HPRCOM");
+	snd_soc_dapm_nc_pin(dapm, "MIC3L"); /* LINE2L on this chip */
+	snd_soc_dapm_nc_pin(dapm, "MIC3R"); /* LINE2R on this chip */
+	snd_soc_dapm_nc_pin(dapm, "LLOUT");
+	snd_soc_dapm_nc_pin(dapm, "RLOUT");
+	snd_soc_dapm_nc_pin(dapm, "HPRCOM");
 
 	/* always connected */
-	snd_soc_dapm_enable_pin(codec, "Audio In");
+	snd_soc_dapm_enable_pin(dapm, "Audio In");
 
 	/* must correspond to audio_out_mux.private_value initializer */
-	snd_soc_dapm_disable_pin(codec, "Audio Out Differential");
-	snd_soc_dapm_sync(codec);
-	snd_soc_dapm_enable_pin(codec, "Audio Out Stereo");
+	snd_soc_dapm_disable_pin(dapm, "Audio Out Differential");
+	snd_soc_dapm_sync(dapm);
+	snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec));
 
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index ac6c49c..c61fc18 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -140,11 +140,12 @@
 static int migor_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(codec, migor_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, migor_dapm_widgets,
 				  ARRAY_SIZE(migor_dapm_widgets));
 
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	return 0;
 }
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index f8e0ab8..105d411 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -23,7 +23,7 @@
 
 static int machine_init(struct snd_soc_pcm_runtime *rtd)
 {
-	snd_soc_dapm_sync(rtd->codec);
+	snd_soc_dapm_sync(&rtd->codec->dapm);
 	return 0;
 }
 
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index d214f02..6c0589e 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -14,27 +14,34 @@
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
 #include <sound/soc.h>
+#include <linux/lzo.h>
+#include <linux/bitmap.h>
+#include <linux/rbtree.h>
 
 static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
 				     unsigned int reg)
 {
-	u16 *cache = codec->reg_cache;
+	int ret;
+	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
 		snd_soc_codec_volatile_register(codec, reg)) {
 			if (codec->cache_only)
 				return -1;
 
+			BUG_ON(!codec->hw_read);
 			return codec->hw_read(codec, reg);
 	}
 
-	return cache[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 *cache = codec->reg_cache;
 	u8 data[2];
 	int ret;
 
@@ -42,16 +49,17 @@
 	data[1] = value & 0x00ff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size)
-			cache[reg] = value;
+		reg < codec->driver->reg_cache_size) {
+		ret = snd_soc_cache_write(codec, reg, value);
+		if (ret < 0)
+			return -1;
+	}
 
 	if (codec->cache_only) {
 		codec->cache_sync = 1;
 		return 0;
 	}
 
-	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
 	ret = codec->hw_write(codec->control_data, data, 2);
 	if (ret == 2)
 		return 0;
@@ -94,23 +102,27 @@
 static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
 				     unsigned int reg)
 {
-	u16 *cache = codec->reg_cache;
+	int ret;
+	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
 		snd_soc_codec_volatile_register(codec, reg)) {
 			if (codec->cache_only)
 				return -1;
 
+			BUG_ON(!codec->hw_read);
 			return codec->hw_read(codec, reg);
 	}
 
-	return cache[reg];
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret < 0)
+		return -1;
+	return val;
 }
 
 static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
 			     unsigned int value)
 {
-	u16 *cache = codec->reg_cache;
 	u8 data[2];
 	int ret;
 
@@ -118,16 +130,17 @@
 	data[1] = value & 0x00ff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size)
-			cache[reg] = value;
+		reg < codec->driver->reg_cache_size) {
+		ret = snd_soc_cache_write(codec, reg, value);
+		if (ret < 0)
+			return -1;
+	}
 
 	if (codec->cache_only) {
 		codec->cache_sync = 1;
 		return 0;
 	}
 
-	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
 	ret = codec->hw_write(codec->control_data, data, 2);
 	if (ret == 2)
 		return 0;
@@ -170,24 +183,25 @@
 static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
 			     unsigned int value)
 {
-	u8 *cache = codec->reg_cache;
 	u8 data[2];
+	int ret;
 
 	reg &= 0xff;
 	data[0] = reg;
 	data[1] = value & 0xff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size)
-			cache[reg] = value;
+		reg < codec->driver->reg_cache_size) {
+		ret = snd_soc_cache_write(codec, reg, value);
+		if (ret < 0)
+			return -1;
+	}
 
 	if (codec->cache_only) {
 		codec->cache_sync = 1;
 		return 0;
 	}
 
-	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
 	if (codec->hw_write(codec->control_data, data, 2) == 2)
 		return 0;
 	else
@@ -197,7 +211,8 @@
 static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
 				     unsigned int reg)
 {
-	u8 *cache = codec->reg_cache;
+	int ret;
+	unsigned int val;
 
 	reg &= 0xff;
 	if (reg >= codec->driver->reg_cache_size ||
@@ -205,10 +220,14 @@
 			if (codec->cache_only)
 				return -1;
 
+			BUG_ON(!codec->hw_read);
 			return codec->hw_read(codec, reg);
 	}
 
-	return cache[reg];
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret < 0)
+		return -1;
+	return val;
 }
 
 #if defined(CONFIG_SPI_MASTER)
@@ -244,24 +263,25 @@
 static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
 			      unsigned int value)
 {
-	u16 *reg_cache = codec->reg_cache;
 	u8 data[3];
+	int ret;
 
 	data[0] = reg;
 	data[1] = (value >> 8) & 0xff;
 	data[2] = value & 0xff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-	    reg < codec->driver->reg_cache_size)
-		reg_cache[reg] = value;
+		reg < codec->driver->reg_cache_size) {
+		ret = snd_soc_cache_write(codec, reg, value);
+		if (ret < 0)
+			return -1;
+	}
 
 	if (codec->cache_only) {
 		codec->cache_sync = 1;
 		return 0;
 	}
 
-	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
 	if (codec->hw_write(codec->control_data, data, 3) == 3)
 		return 0;
 	else
@@ -271,17 +291,22 @@
 static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
 				      unsigned int reg)
 {
-	u16 *cache = codec->reg_cache;
+	int ret;
+	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
 	    snd_soc_codec_volatile_register(codec, reg)) {
 		if (codec->cache_only)
 			return -1;
 
+		BUG_ON(!codec->hw_read);
 		return codec->hw_read(codec, reg);
-	} else {
-		return cache[reg];
 	}
+
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret < 0)
+		return -1;
+	return val;
 }
 
 #if defined(CONFIG_SPI_MASTER)
@@ -420,7 +445,8 @@
 static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec,
 				     unsigned int reg)
 {
-	u8 *cache = codec->reg_cache;
+	int ret;
+	unsigned int val;
 
 	reg &= 0xff;
 	if (reg >= codec->driver->reg_cache_size ||
@@ -428,16 +454,19 @@
 			if (codec->cache_only)
 				return -1;
 
+			BUG_ON(!codec->hw_read);
 			return codec->hw_read(codec, reg);
 	}
 
-	return cache[reg];
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret < 0)
+		return -1;
+	return val;
 }
 
 static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
 			     unsigned int value)
 {
-	u8 *cache = codec->reg_cache;
 	u8 data[3];
 	int ret;
 
@@ -447,16 +476,17 @@
 
 	reg &= 0xff;
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size)
-			cache[reg] = value;
+		reg < codec->driver->reg_cache_size) {
+		ret = snd_soc_cache_write(codec, reg, value);
+		if (ret < 0)
+			return -1;
+	}
 
 	if (codec->cache_only) {
 		codec->cache_sync = 1;
 		return 0;
 	}
 
-	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
 	ret = codec->hw_write(codec->control_data, data, 3);
 	if (ret == 3)
 		return 0;
@@ -534,23 +564,28 @@
 static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec,
 				       unsigned int reg)
 {
-	u16 *cache = codec->reg_cache;
+	int ret;
+	unsigned int val;
 
 	if (reg >= codec->driver->reg_cache_size ||
 	    snd_soc_codec_volatile_register(codec, reg)) {
 		if (codec->cache_only)
 			return -1;
 
+		BUG_ON(!codec->hw_read);
 		return codec->hw_read(codec, reg);
 	}
 
-	return cache[reg];
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret < 0)
+		return -1;
+
+	return val;
 }
 
 static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
 			       unsigned int value)
 {
-	u16 *cache = codec->reg_cache;
 	u8 data[4];
 	int ret;
 
@@ -560,16 +595,17 @@
 	data[3] = value & 0xff;
 
 	if (!snd_soc_codec_volatile_register(codec, reg) &&
-		reg < codec->driver->reg_cache_size)
-			cache[reg] = value;
+		reg < codec->driver->reg_cache_size) {
+		ret = snd_soc_cache_write(codec, reg, value);
+		if (ret < 0)
+			return -1;
+	}
 
 	if (codec->cache_only) {
 		codec->cache_sync = 1;
 		return 0;
 	}
 
-	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
 	ret = codec->hw_write(codec->control_data, data, 4);
 	if (ret == 4)
 		return 0;
@@ -724,3 +760,883 @@
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
+
+struct snd_soc_rbtree_node {
+	struct rb_node node;
+	unsigned int reg;
+	unsigned int value;
+	unsigned int defval;
+} __attribute__ ((packed));
+
+struct snd_soc_rbtree_ctx {
+	struct rb_root root;
+};
+
+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;
+
+	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
+			return rbnode;
+	}
+
+	return NULL;
+}
+
+
+static int snd_soc_rbtree_insert(struct rb_root *root,
+				 struct snd_soc_rbtree_node *rbnode)
+{
+	struct rb_node **new, *parent;
+	struct snd_soc_rbtree_node *rbnode_tmp;
+
+	parent = NULL;
+	new = &root->rb_node;
+	while (*new) {
+		rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node,
+					  node);
+		parent = *new;
+		if (rbnode_tmp->reg < rbnode->reg)
+			new = &((*new)->rb_left);
+		else if (rbnode_tmp->reg > rbnode->reg)
+			new = &((*new)->rb_right);
+		else
+			return 0;
+	}
+
+	/* insert the node into the rbtree */
+	rb_link_node(&rbnode->node, parent, new);
+	rb_insert_color(&rbnode->node, root);
+
+	return 1;
+}
+
+static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec)
+{
+	struct snd_soc_rbtree_ctx *rbtree_ctx;
+	struct rb_node *node;
+	struct snd_soc_rbtree_node *rbnode;
+	unsigned int val;
+
+	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;
+		snd_soc_cache_read(codec, rbnode->reg, &val);
+		snd_soc_write(codec, rbnode->reg, val);
+		dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
+			rbnode->reg, val);
+	}
+
+	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;
+
+	rbtree_ctx = codec->reg_cache;
+	rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
+	if (rbnode) {
+		if (rbnode->value == value)
+			return 0;
+		rbnode->value = value;
+	} 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.
+		 */
+		rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL);
+		if (!rbnode)
+			return -ENOMEM;
+		rbnode->reg = reg;
+		rbnode->value = value;
+		snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode);
+	}
+
+	return 0;
+}
+
+static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec,
+				     unsigned int reg, unsigned int *value)
+{
+	struct snd_soc_rbtree_ctx *rbtree_ctx;
+	struct snd_soc_rbtree_node *rbnode;
+
+	rbtree_ctx = codec->reg_cache;
+	rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
+	if (rbnode) {
+		*value = rbnode->value;
+	} else {
+		/* uninitialized registers default to 0 */
+		*value = 0;
+	}
+
+	return 0;
+}
+
+static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec)
+{
+	struct rb_node *next;
+	struct snd_soc_rbtree_ctx *rbtree_ctx;
+	struct snd_soc_rbtree_node *rbtree_node;
+
+	/* if we've already been called then just return */
+	rbtree_ctx = codec->reg_cache;
+	if (!rbtree_ctx)
+		return 0;
+
+	/* free up the rbtree */
+	next = rb_first(&rbtree_ctx->root);
+	while (next) {
+		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);
+	}
+
+	/* release the resources */
+	kfree(codec->reg_cache);
+	codec->reg_cache = NULL;
+
+	return 0;
+}
+
+static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
+{
+	struct snd_soc_rbtree_ctx *rbtree_ctx;
+
+	codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL);
+	if (!codec->reg_cache)
+		return -ENOMEM;
+
+	rbtree_ctx = codec->reg_cache;
+	rbtree_ctx->root = RB_ROOT;
+
+	if (!codec->driver->reg_cache_default)
+		return 0;
+
+/*
+ * populate the rbtree with the initialized registers.  All other
+ * registers will be inserted into the tree when they are first written.
+ *
+ * The reasoning behind this, is that we need to step through and
+ * dereference the cache in u8/u16 increments without sacrificing
+ * portability.  This could also be done using memcpy() but that would
+ * be slightly more cryptic.
+ */
+#define snd_soc_rbtree_populate(cache)					\
+({									\
+	int ret, i;							\
+	struct snd_soc_rbtree_node *rbtree_node;			\
+									\
+	ret = 0;							\
+	cache = codec->driver->reg_cache_default;			\
+	for (i = 0; i < codec->driver->reg_cache_size; ++i) {		\
+		if (!cache[i])						\
+			continue;					\
+		rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL);	\
+		if (!rbtree_node) {					\
+			ret = -ENOMEM;					\
+			snd_soc_cache_exit(codec);			\
+			break;						\
+		}							\
+		rbtree_node->reg = i;					\
+		rbtree_node->value = cache[i];				\
+		rbtree_node->defval = cache[i];				\
+		snd_soc_rbtree_insert(&rbtree_ctx->root,		\
+				      rbtree_node);			\
+	}								\
+	ret;								\
+})
+
+	switch (codec->driver->reg_word_size) {
+	case 1: {
+		const u8 *cache;
+
+		return snd_soc_rbtree_populate(cache);
+	}
+	case 2: {
+		const u16 *cache;
+
+		return snd_soc_rbtree_populate(cache);
+	}
+	default:
+		BUG();
+	}
+
+	return 0;
+}
+
+struct snd_soc_lzo_ctx {
+	void *wmem;
+	void *dst;
+	const void *src;
+	size_t src_len;
+	size_t dst_len;
+	size_t decompressed_size;
+	unsigned long *sync_bmp;
+	int sync_bmp_nbits;
+};
+
+#define LZO_BLOCK_NUM 8
+static int snd_soc_lzo_block_count(void)
+{
+	return LZO_BLOCK_NUM;
+}
+
+static int snd_soc_lzo_prepare(struct snd_soc_lzo_ctx *lzo_ctx)
+{
+	lzo_ctx->wmem = kmalloc(LZO1X_MEM_COMPRESS, GFP_KERNEL);
+	if (!lzo_ctx->wmem)
+		return -ENOMEM;
+	return 0;
+}
+
+static int snd_soc_lzo_compress(struct snd_soc_lzo_ctx *lzo_ctx)
+{
+	size_t compress_size;
+	int ret;
+
+	ret = lzo1x_1_compress(lzo_ctx->src, lzo_ctx->src_len,
+			       lzo_ctx->dst, &compress_size, lzo_ctx->wmem);
+	if (ret != LZO_E_OK || compress_size > lzo_ctx->dst_len)
+		return -EINVAL;
+	lzo_ctx->dst_len = compress_size;
+	return 0;
+}
+
+static int snd_soc_lzo_decompress(struct snd_soc_lzo_ctx *lzo_ctx)
+{
+	size_t dst_len;
+	int ret;
+
+	dst_len = lzo_ctx->dst_len;
+	ret = lzo1x_decompress_safe(lzo_ctx->src, lzo_ctx->src_len,
+				    lzo_ctx->dst, &dst_len);
+	if (ret != LZO_E_OK || dst_len != lzo_ctx->dst_len)
+		return -EINVAL;
+	return 0;
+}
+
+static int snd_soc_lzo_compress_cache_block(struct snd_soc_codec *codec,
+		struct snd_soc_lzo_ctx *lzo_ctx)
+{
+	int ret;
+
+	lzo_ctx->dst_len = lzo1x_worst_compress(PAGE_SIZE);
+	lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL);
+	if (!lzo_ctx->dst) {
+		lzo_ctx->dst_len = 0;
+		return -ENOMEM;
+	}
+
+	ret = snd_soc_lzo_compress(lzo_ctx);
+	if (ret < 0)
+		return ret;
+	return 0;
+}
+
+static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec,
+		struct snd_soc_lzo_ctx *lzo_ctx)
+{
+	int ret;
+
+	lzo_ctx->dst_len = lzo_ctx->decompressed_size;
+	lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL);
+	if (!lzo_ctx->dst) {
+		lzo_ctx->dst_len = 0;
+		return -ENOMEM;
+	}
+
+	ret = snd_soc_lzo_decompress(lzo_ctx);
+	if (ret < 0)
+		return ret;
+	return 0;
+}
+
+static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec,
+		unsigned int reg)
+{
+	struct snd_soc_codec_driver *codec_drv;
+	size_t reg_size;
+
+	codec_drv = codec->driver;
+	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+	return (reg * codec_drv->reg_word_size) /
+	       DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count());
+}
+
+static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec,
+		unsigned int reg)
+{
+	struct snd_soc_codec_driver *codec_drv;
+	size_t reg_size;
+
+	codec_drv = codec->driver;
+	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+	return reg % (DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()) /
+		      codec_drv->reg_word_size);
+}
+
+static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec)
+{
+	struct snd_soc_codec_driver *codec_drv;
+	size_t reg_size;
+
+	codec_drv = codec->driver;
+	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+	return DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count());
+}
+
+static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
+{
+	struct snd_soc_lzo_ctx **lzo_blocks;
+	unsigned int val;
+	int i;
+
+	lzo_blocks = codec->reg_cache;
+	for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) {
+		snd_soc_cache_read(codec, i, &val);
+		snd_soc_write(codec, i, val);
+		dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
+			i, val);
+	}
+
+	return 0;
+}
+
+static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec,
+				   unsigned int reg, unsigned int value)
+{
+	struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks;
+	int ret, blkindex, blkpos;
+	size_t blksize, tmp_dst_len;
+	void *tmp_dst;
+
+	/* index of the compressed lzo block */
+	blkindex = snd_soc_lzo_get_blkindex(codec, reg);
+	/* register index within the decompressed block */
+	blkpos = snd_soc_lzo_get_blkpos(codec, reg);
+	/* size of the compressed block */
+	blksize = snd_soc_lzo_get_blksize(codec);
+	lzo_blocks = codec->reg_cache;
+	lzo_block = lzo_blocks[blkindex];
+
+	/* save the pointer and length of the compressed block */
+	tmp_dst = lzo_block->dst;
+	tmp_dst_len = lzo_block->dst_len;
+
+	/* prepare the source to be the compressed block */
+	lzo_block->src = lzo_block->dst;
+	lzo_block->src_len = lzo_block->dst_len;
+
+	/* decompress the block */
+	ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block);
+	if (ret < 0) {
+		kfree(lzo_block->dst);
+		goto out;
+	}
+
+	/* write the new value to the cache */
+	switch (codec->driver->reg_word_size) {
+	case 1: {
+		u8 *cache;
+		cache = lzo_block->dst;
+		if (cache[blkpos] == value) {
+			kfree(lzo_block->dst);
+			goto out;
+		}
+		cache[blkpos] = value;
+	}
+	break;
+	case 2: {
+		u16 *cache;
+		cache = lzo_block->dst;
+		if (cache[blkpos] == value) {
+			kfree(lzo_block->dst);
+			goto out;
+		}
+		cache[blkpos] = value;
+	}
+	break;
+	default:
+		BUG();
+	}
+
+	/* prepare the source to be the decompressed block */
+	lzo_block->src = lzo_block->dst;
+	lzo_block->src_len = lzo_block->dst_len;
+
+	/* compress the block */
+	ret = snd_soc_lzo_compress_cache_block(codec, lzo_block);
+	if (ret < 0) {
+		kfree(lzo_block->dst);
+		kfree(lzo_block->src);
+		goto out;
+	}
+
+	/* set the bit so we know we have to sync this register */
+	set_bit(reg, lzo_block->sync_bmp);
+	kfree(tmp_dst);
+	kfree(lzo_block->src);
+	return 0;
+out:
+	lzo_block->dst = tmp_dst;
+	lzo_block->dst_len = tmp_dst_len;
+	return ret;
+}
+
+static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec,
+				  unsigned int reg, unsigned int *value)
+{
+	struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks;
+	int ret, blkindex, blkpos;
+	size_t blksize, tmp_dst_len;
+	void *tmp_dst;
+
+	*value = 0;
+	/* index of the compressed lzo block */
+	blkindex = snd_soc_lzo_get_blkindex(codec, reg);
+	/* register index within the decompressed block */
+	blkpos = snd_soc_lzo_get_blkpos(codec, reg);
+	/* size of the compressed block */
+	blksize = snd_soc_lzo_get_blksize(codec);
+	lzo_blocks = codec->reg_cache;
+	lzo_block = lzo_blocks[blkindex];
+
+	/* save the pointer and length of the compressed block */
+	tmp_dst = lzo_block->dst;
+	tmp_dst_len = lzo_block->dst_len;
+
+	/* prepare the source to be the compressed block */
+	lzo_block->src = lzo_block->dst;
+	lzo_block->src_len = lzo_block->dst_len;
+
+	/* decompress the block */
+	ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block);
+	if (ret >= 0) {
+		/* fetch the value from the cache */
+		switch (codec->driver->reg_word_size) {
+		case 1: {
+			u8 *cache;
+			cache = lzo_block->dst;
+			*value = cache[blkpos];
+		}
+		break;
+		case 2: {
+			u16 *cache;
+			cache = lzo_block->dst;
+			*value = cache[blkpos];
+		}
+		break;
+		default:
+			BUG();
+		}
+	}
+
+	kfree(lzo_block->dst);
+	/* restore the pointer and length of the compressed block */
+	lzo_block->dst = tmp_dst;
+	lzo_block->dst_len = tmp_dst_len;
+	return 0;
+}
+
+static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec)
+{
+	struct snd_soc_lzo_ctx **lzo_blocks;
+	int i, blkcount;
+
+	lzo_blocks = codec->reg_cache;
+	if (!lzo_blocks)
+		return 0;
+
+	blkcount = snd_soc_lzo_block_count();
+	/*
+	 * the pointer to the bitmap used for syncing the cache
+	 * is shared amongst all lzo_blocks.  Ensure it is freed
+	 * only once.
+	 */
+	if (lzo_blocks[0])
+		kfree(lzo_blocks[0]->sync_bmp);
+	for (i = 0; i < blkcount; ++i) {
+		if (lzo_blocks[i]) {
+			kfree(lzo_blocks[i]->wmem);
+			kfree(lzo_blocks[i]->dst);
+		}
+		/* each lzo_block is a pointer returned by kmalloc or NULL */
+		kfree(lzo_blocks[i]);
+	}
+	kfree(lzo_blocks);
+	codec->reg_cache = NULL;
+	return 0;
+}
+
+static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
+{
+	struct snd_soc_lzo_ctx **lzo_blocks;
+	size_t reg_size, bmp_size;
+	struct snd_soc_codec_driver *codec_drv;
+	int ret, tofree, i, blksize, blkcount;
+	const char *p, *end;
+	unsigned long *sync_bmp;
+
+	ret = 0;
+	codec_drv = codec->driver;
+	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+
+	/*
+	 * If we have not been given a default register cache
+	 * then allocate a dummy zero-ed out region, compress it
+	 * and remember to free it afterwards.
+	 */
+	tofree = 0;
+	if (!codec_drv->reg_cache_default)
+		tofree = 1;
+
+	if (!codec_drv->reg_cache_default) {
+		codec_drv->reg_cache_default = kzalloc(reg_size,
+						       GFP_KERNEL);
+		if (!codec_drv->reg_cache_default)
+			return -ENOMEM;
+	}
+
+	blkcount = snd_soc_lzo_block_count();
+	codec->reg_cache = kzalloc(blkcount * sizeof *lzo_blocks,
+				   GFP_KERNEL);
+	if (!codec->reg_cache) {
+		ret = -ENOMEM;
+		goto err_tofree;
+	}
+	lzo_blocks = codec->reg_cache;
+
+	/*
+	 * allocate a bitmap to be used when syncing the cache with
+	 * the hardware.  Each time a register is modified, the corresponding
+	 * bit is set in the bitmap, so we know that we have to sync
+	 * that register.
+	 */
+	bmp_size = codec_drv->reg_cache_size;
+	sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof (long),
+			   GFP_KERNEL);
+	if (!sync_bmp) {
+		ret = -ENOMEM;
+		goto err;
+	}
+	bitmap_zero(sync_bmp, reg_size);
+
+	/* allocate the lzo blocks and initialize them */
+	for (i = 0; i < blkcount; ++i) {
+		lzo_blocks[i] = kzalloc(sizeof **lzo_blocks,
+					GFP_KERNEL);
+		if (!lzo_blocks[i]) {
+			kfree(sync_bmp);
+			ret = -ENOMEM;
+			goto err;
+		}
+		lzo_blocks[i]->sync_bmp = sync_bmp;
+		lzo_blocks[i]->sync_bmp_nbits = reg_size;
+		/* alloc the working space for the compressed block */
+		ret = snd_soc_lzo_prepare(lzo_blocks[i]);
+		if (ret < 0)
+			goto err;
+	}
+
+	blksize = snd_soc_lzo_get_blksize(codec);
+	p = codec_drv->reg_cache_default;
+	end = codec_drv->reg_cache_default + reg_size;
+	/* compress the register map and fill the lzo blocks */
+	for (i = 0; i < blkcount; ++i, p += blksize) {
+		lzo_blocks[i]->src = p;
+		if (p + blksize > end)
+			lzo_blocks[i]->src_len = end - p;
+		else
+			lzo_blocks[i]->src_len = blksize;
+		ret = snd_soc_lzo_compress_cache_block(codec,
+						       lzo_blocks[i]);
+		if (ret < 0)
+			goto err;
+		lzo_blocks[i]->decompressed_size =
+			lzo_blocks[i]->src_len;
+	}
+
+	if (tofree)
+		kfree(codec_drv->reg_cache_default);
+	return 0;
+err:
+	snd_soc_cache_exit(codec);
+err_tofree:
+	if (tofree)
+		kfree(codec_drv->reg_cache_default);
+	return ret;
+}
+
+static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
+{
+	int i;
+	struct snd_soc_codec_driver *codec_drv;
+	unsigned int val;
+
+	codec_drv = codec->driver;
+	for (i = 0; i < codec_drv->reg_cache_size; ++i) {
+		snd_soc_cache_read(codec, i, &val);
+		if (codec_drv->reg_cache_default) {
+			switch (codec_drv->reg_word_size) {
+			case 1: {
+				const u8 *cache;
+
+				cache = codec_drv->reg_cache_default;
+				if (cache[i] == val)
+					continue;
+			}
+			break;
+			case 2: {
+				const u16 *cache;
+
+				cache = codec_drv->reg_cache_default;
+				if (cache[i] == val)
+					continue;
+			}
+			break;
+			default:
+				BUG();
+			}
+		}
+		snd_soc_write(codec, i, val);
+		dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
+			i, val);
+	}
+	return 0;
+}
+
+static int snd_soc_flat_cache_write(struct snd_soc_codec *codec,
+				    unsigned int reg, unsigned int value)
+{
+	switch (codec->driver->reg_word_size) {
+	case 1: {
+		u8 *cache;
+
+		cache = codec->reg_cache;
+		cache[reg] = value;
+	}
+	break;
+	case 2: {
+		u16 *cache;
+
+		cache = codec->reg_cache;
+		cache[reg] = value;
+	}
+	break;
+	default:
+		BUG();
+	}
+
+	return 0;
+}
+
+static int snd_soc_flat_cache_read(struct snd_soc_codec *codec,
+				   unsigned int reg, unsigned int *value)
+{
+	switch (codec->driver->reg_word_size) {
+	case 1: {
+		u8 *cache;
+
+		cache = codec->reg_cache;
+		*value = cache[reg];
+	}
+	break;
+	case 2: {
+		u16 *cache;
+
+		cache = codec->reg_cache;
+		*value = cache[reg];
+	}
+	break;
+	default:
+		BUG();
+	}
+
+	return 0;
+}
+
+static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec)
+{
+	if (!codec->reg_cache)
+		return 0;
+	kfree(codec->reg_cache);
+	codec->reg_cache = NULL;
+	return 0;
+}
+
+static int snd_soc_flat_cache_init(struct snd_soc_codec *codec)
+{
+	struct snd_soc_codec_driver *codec_drv;
+	size_t reg_size;
+
+	codec_drv = codec->driver;
+	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+
+	if (codec_drv->reg_cache_default)
+		codec->reg_cache = kmemdup(codec_drv->reg_cache_default,
+					   reg_size, GFP_KERNEL);
+	else
+		codec->reg_cache = kzalloc(reg_size, GFP_KERNEL);
+	if (!codec->reg_cache)
+		return -ENOMEM;
+
+	return 0;
+}
+
+/* an array of all supported compression types */
+static const struct snd_soc_cache_ops cache_types[] = {
+	{
+		.id = SND_SOC_NO_COMPRESSION,
+		.init = snd_soc_flat_cache_init,
+		.exit = snd_soc_flat_cache_exit,
+		.read = snd_soc_flat_cache_read,
+		.write = snd_soc_flat_cache_write,
+		.sync = snd_soc_flat_cache_sync
+	},
+	{
+		.id = SND_SOC_LZO_COMPRESSION,
+		.init = snd_soc_lzo_cache_init,
+		.exit = snd_soc_lzo_cache_exit,
+		.read = snd_soc_lzo_cache_read,
+		.write = snd_soc_lzo_cache_write,
+		.sync = snd_soc_lzo_cache_sync
+	},
+	{
+		.id = SND_SOC_RBTREE_COMPRESSION,
+		.init = snd_soc_rbtree_cache_init,
+		.exit = snd_soc_rbtree_cache_exit,
+		.read = snd_soc_rbtree_cache_read,
+		.write = snd_soc_rbtree_cache_write,
+		.sync = snd_soc_rbtree_cache_sync
+	}
+};
+
+int snd_soc_cache_init(struct snd_soc_codec *codec)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(cache_types); ++i)
+		if (cache_types[i].id == codec->driver->compress_type)
+			break;
+	if (i == ARRAY_SIZE(cache_types)) {
+		dev_err(codec->dev, "Could not match compress type: %d\n",
+			codec->driver->compress_type);
+		return -EINVAL;
+	}
+
+	mutex_init(&codec->cache_rw_mutex);
+	codec->cache_ops = &cache_types[i];
+
+	if (codec->cache_ops->init)
+		return codec->cache_ops->init(codec);
+	return -EINVAL;
+}
+
+/*
+ * NOTE: keep in mind that this function might be called
+ * multiple times.
+ */
+int snd_soc_cache_exit(struct snd_soc_codec *codec)
+{
+	if (codec->cache_ops && codec->cache_ops->exit)
+		return codec->cache_ops->exit(codec);
+	return -EINVAL;
+}
+
+/**
+ * snd_soc_cache_read: Fetch the value of a given register from the cache.
+ *
+ * @codec: CODEC to configure.
+ * @reg: The register index.
+ * @value: The value to be returned.
+ */
+int snd_soc_cache_read(struct snd_soc_codec *codec,
+		       unsigned int reg, unsigned int *value)
+{
+	int ret;
+
+	mutex_lock(&codec->cache_rw_mutex);
+
+	if (value && codec->cache_ops && codec->cache_ops->read) {
+		ret = codec->cache_ops->read(codec, reg, value);
+		mutex_unlock(&codec->cache_rw_mutex);
+		return ret;
+	}
+
+	mutex_unlock(&codec->cache_rw_mutex);
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_cache_read);
+
+/**
+ * snd_soc_cache_write: Set the value of a given register in the cache.
+ *
+ * @codec: CODEC to configure.
+ * @reg: The register index.
+ * @value: The new register value.
+ */
+int snd_soc_cache_write(struct snd_soc_codec *codec,
+			unsigned int reg, unsigned int value)
+{
+	int ret;
+
+	mutex_lock(&codec->cache_rw_mutex);
+
+	if (codec->cache_ops && codec->cache_ops->write) {
+		ret = codec->cache_ops->write(codec, reg, value);
+		mutex_unlock(&codec->cache_rw_mutex);
+		return ret;
+	}
+
+	mutex_unlock(&codec->cache_rw_mutex);
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_cache_write);
+
+/**
+ * snd_soc_cache_sync: Sync the register cache with the hardware.
+ *
+ * @codec: CODEC to configure.
+ *
+ * Any registers that should not be synced should be marked as
+ * volatile.  In general drivers can choose not to use the provided
+ * syncing functionality if they so require.
+ */
+int snd_soc_cache_sync(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	if (!codec->cache_sync) {
+		return 0;
+	}
+
+	if (codec->cache_ops && codec->cache_ops->sync) {
+		ret = codec->cache_ops->sync(codec);
+		if (!ret)
+			codec->cache_sync = 0;
+		return ret;
+	}
+
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_cache_sync);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 441285a..2540efd 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -39,6 +39,9 @@
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
 
+#define CREATE_TRACE_POINTS
+#include <trace/events/asoc.h>
+
 #define NAME_SIZE	32
 
 static DEFINE_MUTEX(pcm_mutex);
@@ -238,8 +241,10 @@
 
 static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
 {
-	codec->debugfs_codec_root = debugfs_create_dir(codec->name ,
-						       debugfs_root);
+	struct dentry *debugfs_card_root = codec->card->debugfs_card_root;
+
+	codec->debugfs_codec_root = debugfs_create_dir(codec->name,
+						       debugfs_card_root);
 	if (!codec->debugfs_codec_root) {
 		printk(KERN_WARNING
 		       "ASoC: Failed to create codec debugfs directory\n");
@@ -253,20 +258,13 @@
 		printk(KERN_WARNING
 		       "ASoC: Failed to create codec register debugfs file\n");
 
-	codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
-						     codec->debugfs_codec_root,
-						     &codec->pop_time);
-	if (!codec->debugfs_pop_time)
-		printk(KERN_WARNING
-		       "Failed to create pop time debugfs file\n");
-
-	codec->debugfs_dapm = debugfs_create_dir("dapm",
+	codec->dapm.debugfs_dapm = debugfs_create_dir("dapm",
 						 codec->debugfs_codec_root);
-	if (!codec->debugfs_dapm)
+	if (!codec->dapm.debugfs_dapm)
 		printk(KERN_WARNING
 		       "Failed to create DAPM debugfs directory\n");
 
-	snd_soc_dapm_debugfs_init(codec);
+	snd_soc_dapm_debugfs_init(&codec->dapm);
 }
 
 static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
@@ -374,6 +372,29 @@
 	.llseek = default_llseek,/* read accesses f_pos */
 };
 
+static void soc_init_card_debugfs(struct snd_soc_card *card)
+{
+	card->debugfs_card_root = debugfs_create_dir(card->name,
+						     debugfs_root);
+	if (!card->debugfs_card_root) {
+		dev_warn(card->dev,
+			 "ASoC: Failed to create codec debugfs directory\n");
+		return;
+	}
+
+	card->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
+						    card->debugfs_card_root,
+						    &card->pop_time);
+	if (!card->debugfs_pop_time)
+		dev_warn(card->dev,
+		       "Failed to create pop time debugfs file\n");
+}
+
+static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
+{
+	debugfs_remove_recursive(card->debugfs_card_root);
+}
+
 #else
 
 static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
@@ -383,6 +404,14 @@
 static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
 {
 }
+
+static inline void soc_init_card_debugfs(struct snd_soc_card *card)
+{
+}
+
+static inline void soc_cleanup_card_debugfs(struct snd_soc_card *card)
+{
+}
 #endif
 
 #ifdef CONFIG_SND_SOC_AC97_BUS
@@ -1017,7 +1046,7 @@
 	/* close any waiting streams and save state */
 	for (i = 0; i < card->num_rtd; i++) {
 		run_delayed_work(&card->rtd[i].delayed_work);
-		card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level;
+		card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level;
 	}
 
 	for (i = 0; i < card->num_rtd; i++) {
@@ -1041,7 +1070,7 @@
 		/* If there are paths active then the CODEC will be held with
 		 * bias _ON and should not be suspended. */
 		if (!codec->suspended && codec->driver->suspend) {
-			switch (codec->bias_level) {
+			switch (codec->dapm.bias_level) {
 			case SND_SOC_BIAS_STANDBY:
 			case SND_SOC_BIAS_OFF:
 				codec->driver->suspend(codec, PMSG_SUSPEND);
@@ -1110,7 +1139,7 @@
 		 * resume.  Otherwise the suspend was suppressed.
 		 */
 		if (codec->driver->resume && codec->suspended) {
-			switch (codec->bias_level) {
+			switch (codec->dapm.bias_level) {
 			case SND_SOC_BIAS_STANDBY:
 			case SND_SOC_BIAS_OFF:
 				codec->driver->resume(codec);
@@ -1346,7 +1375,7 @@
 		}
 
 		/* Make sure all DAPM widgets are freed */
-		snd_soc_dapm_free(codec);
+		snd_soc_dapm_free(&codec->dapm);
 
 		soc_cleanup_codec_debugfs(codec);
 		device_remove_file(&rtd->dev, &dev_attr_codec_reg);
@@ -1368,6 +1397,23 @@
 	}
 }
 
+static void soc_set_name_prefix(struct snd_soc_card *card,
+				struct snd_soc_codec *codec)
+{
+	int i;
+
+	if (card->prefix_map == NULL)
+		return;
+
+	for (i = 0; i < card->num_prefixes; i++) {
+		struct snd_soc_prefix_map *map = &card->prefix_map[i];
+		if (map->dev_name && !strcmp(codec->name, map->dev_name)) {
+			codec->name_prefix = map->name_prefix;
+			break;
+		}
+	}
+}
+
 static void rtd_release(struct device *dev) {}
 
 static int soc_probe_dai_link(struct snd_soc_card *card, int num)
@@ -1377,6 +1423,7 @@
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
+	const char *temp;
 	int ret;
 
 	dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num);
@@ -1410,6 +1457,8 @@
 
 	/* probe the CODEC */
 	if (!codec->probed) {
+		codec->dapm.card = card;
+		soc_set_name_prefix(card, codec);
 		if (codec->driver->probe) {
 			ret = codec->driver->probe(codec);
 			if (ret < 0) {
@@ -1462,16 +1511,20 @@
 
 	/* now that all clients have probed, initialise the DAI link */
 	if (dai_link->init) {
+		/* machine controls, routes and widgets are not prefixed */
+		temp = rtd->codec->name_prefix;
+		rtd->codec->name_prefix = NULL;
 		ret = dai_link->init(rtd);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: failed to init %s\n", dai_link->stream_name);
 			return ret;
 		}
+		rtd->codec->name_prefix = temp;
 	}
 
 	/* Make sure all DAPM widgets are instantiated */
-	snd_soc_dapm_new_widgets(codec);
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_new_widgets(&codec->dapm);
+	snd_soc_dapm_sync(&codec->dapm);
 
 	/* register the rtd device */
 	rtd->dev.release = rtd_release;
@@ -1667,6 +1720,8 @@
 	INIT_LIST_HEAD(&card->codec_dev_list);
 	INIT_LIST_HEAD(&card->platform_dev_list);
 
+	soc_init_card_debugfs(card);
+
 	ret = snd_soc_register_card(card);
 	if (ret != 0) {
 		dev_err(&pdev->dev, "Failed to register card\n");
@@ -1694,6 +1749,8 @@
 		for (i = 0; i < card->num_rtd; i++)
 			soc_remove_dai_link(card, i);
 
+		soc_cleanup_card_debugfs(card);
+
 		/* remove the card */
 		if (card->remove)
 			card->remove(pdev);
@@ -1877,6 +1934,27 @@
 }
 EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
 
+unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	unsigned int ret;
+
+	ret = codec->driver->read(codec, reg);
+	dev_dbg(codec->dev, "read %x => %x\n", reg, ret);
+	trace_snd_soc_reg_read(codec, reg, ret);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_read);
+
+unsigned int snd_soc_write(struct snd_soc_codec *codec,
+			   unsigned int reg, unsigned int val)
+{
+	dev_dbg(codec->dev, "write %x = %x\n", reg, val);
+	trace_snd_soc_reg_write(codec, reg, val);
+	return codec->driver->write(codec, reg, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_write);
+
 /**
  * snd_soc_update_bits - update codec register bits
  * @codec: audio codec
@@ -2017,14 +2095,22 @@
 	const struct snd_kcontrol_new *controls, int num_controls)
 {
 	struct snd_card *card = codec->card->snd_card;
+	char prefixed_name[44], *name;
 	int err, i;
 
 	for (i = 0; i < num_controls; i++) {
 		const struct snd_kcontrol_new *control = &controls[i];
-		err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
+		if (codec->name_prefix) {
+			snprintf(prefixed_name, sizeof(prefixed_name), "%s %s",
+				 codec->name_prefix, control->name);
+			name = prefixed_name;
+		} else {
+			name = control->name;
+		}
+		err = snd_ctl_add(card, snd_soc_cnew(control, codec, name));
 		if (err < 0) {
 			dev_err(codec->dev, "%s: Failed to add %s: %d\n",
-				codec->name, control->name, err);
+				codec->name, name, err);
 			return err;
 		}
 	}
@@ -3219,30 +3305,25 @@
 		return -ENOMEM;
 	}
 
-	/* allocate CODEC register cache */
-	if (codec_drv->reg_cache_size && codec_drv->reg_word_size) {
-
-		if (codec_drv->reg_cache_default)
-			codec->reg_cache = kmemdup(codec_drv->reg_cache_default,
-				codec_drv->reg_cache_size * codec_drv->reg_word_size, GFP_KERNEL);
-		else
-			codec->reg_cache = kzalloc(codec_drv->reg_cache_size *
-				codec_drv->reg_word_size, GFP_KERNEL);
-
-		if (codec->reg_cache == NULL) {
-			kfree(codec->name);
-			kfree(codec);
-			return -ENOMEM;
-		}
-	}
-
+	INIT_LIST_HEAD(&codec->dapm.widgets);
+	INIT_LIST_HEAD(&codec->dapm.paths);
+	codec->dapm.bias_level = SND_SOC_BIAS_OFF;
+	codec->dapm.dev = dev;
+	codec->dapm.codec = codec;
 	codec->dev = dev;
 	codec->driver = codec_drv;
-	codec->bias_level = SND_SOC_BIAS_OFF;
 	codec->num_dai = num_dai;
 	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	/* allocate CODEC register cache */
+	if (codec_drv->reg_cache_size && codec_drv->reg_word_size) {
+		ret = snd_soc_cache_init(codec);
+		if (ret < 0) {
+			dev_err(codec->dev, "Failed to set cache compression type: %d\n",
+				ret);
+			goto error_cache;
+		}
+	}
 
 	for (i = 0; i < num_dai; i++) {
 		fixup_codec_formats(&dai_drv[i].playback);
@@ -3253,7 +3334,7 @@
 	if (num_dai) {
 		ret = snd_soc_register_dais(dev, dai_drv, num_dai);
 		if (ret < 0)
-			goto error;
+			goto error_dais;
 	}
 
 	mutex_lock(&client_mutex);
@@ -3264,9 +3345,9 @@
 	pr_debug("Registered codec '%s'\n", codec->name);
 	return 0;
 
-error:
-	if (codec->reg_cache)
-		kfree(codec->reg_cache);
+error_dais:
+	snd_soc_cache_exit(codec);
+error_cache:
 	kfree(codec->name);
 	kfree(codec);
 	return ret;
@@ -3300,8 +3381,7 @@
 
 	pr_debug("Unregistered codec '%s'\n", codec->name);
 
-	if (codec->reg_cache)
-		kfree(codec->reg_cache);
+	snd_soc_cache_exit(codec);
 	kfree(codec->name);
 	kfree(codec);
 }
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 75ed649..02b6ac4 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -42,9 +42,12 @@
 #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 <trace/events/asoc.h>
+
 /* dapm power sequences - make this per codec in the future */
 static int dapm_up_seq[] = {
 	[snd_soc_dapm_pre] = 0,
@@ -90,17 +93,24 @@
 		schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time));
 }
 
-static void pop_dbg(u32 pop_time, const char *fmt, ...)
+static void pop_dbg(struct device *dev, u32 pop_time, const char *fmt, ...)
 {
 	va_list args;
+	char *buf;
+
+	if (!pop_time)
+		return;
+
+	buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	if (buf == NULL)
+		return;
 
 	va_start(args, fmt);
-
-	if (pop_time) {
-		vprintk(fmt, args);
-	}
-
+	vsnprintf(buf, PAGE_SIZE, fmt, args);
+	dev_info(dev, buf);
 	va_end(args);
+
+	kfree(buf);
 }
 
 /* create a new dapm widget */
@@ -120,37 +130,42 @@
  * Returns 0 for success else error.
  */
 static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card,
-		struct snd_soc_codec *codec, enum snd_soc_bias_level level)
+				       struct snd_soc_dapm_context *dapm,
+				       enum snd_soc_bias_level level)
 {
 	int ret = 0;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		dev_dbg(codec->dev, "Setting full bias\n");
+		dev_dbg(dapm->dev, "Setting full bias\n");
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		dev_dbg(codec->dev, "Setting bias prepare\n");
+		dev_dbg(dapm->dev, "Setting bias prepare\n");
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		dev_dbg(codec->dev, "Setting standby bias\n");
+		dev_dbg(dapm->dev, "Setting standby bias\n");
 		break;
 	case SND_SOC_BIAS_OFF:
-		dev_dbg(codec->dev, "Setting bias off\n");
+		dev_dbg(dapm->dev, "Setting bias off\n");
 		break;
 	default:
-		dev_err(codec->dev, "Setting invalid bias %d\n", level);
+		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 (codec->driver->set_bias_level)
-			ret = codec->driver->set_bias_level(codec, level);
+		if (dapm->codec && dapm->codec->driver->set_bias_level)
+			ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
 		else
-			codec->bias_level = level;
+			dapm->bias_level = level;
 	}
 
+	trace_snd_soc_bias_level_done(card, level);
+
 	return ret;
 }
 
@@ -241,7 +256,7 @@
 }
 
 /* connect mux widget to its interconnecting audio paths */
-static int dapm_connect_mux(struct snd_soc_codec *codec,
+static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
 	struct snd_soc_dapm_path *path, const char *control_name,
 	const struct snd_kcontrol_new *kcontrol)
@@ -251,7 +266,7 @@
 
 	for (i = 0; i < e->max; i++) {
 		if (!(strcmp(control_name, e->texts[i]))) {
-			list_add(&path->list, &codec->dapm_paths);
+			list_add(&path->list, &dapm->paths);
 			list_add(&path->list_sink, &dest->sources);
 			list_add(&path->list_source, &src->sinks);
 			path->name = (char*)e->texts[i];
@@ -264,7 +279,7 @@
 }
 
 /* connect mixer widget to its interconnecting audio paths */
-static int dapm_connect_mixer(struct snd_soc_codec *codec,
+static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
 	struct snd_soc_dapm_path *path, const char *control_name)
 {
@@ -273,7 +288,7 @@
 	/* search for mixer kcontrol */
 	for (i = 0; i < dest->num_kcontrols; i++) {
 		if (!strcmp(control_name, dest->kcontrols[i].name)) {
-			list_add(&path->list, &codec->dapm_paths);
+			list_add(&path->list, &dapm->paths);
 			list_add(&path->list_sink, &dest->sources);
 			list_add(&path->list_source, &src->sinks);
 			path->name = dest->kcontrols[i].name;
@@ -290,6 +305,8 @@
 	int change, power;
 	unsigned int old, new;
 	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_dapm_context *dapm = widget->dapm;
+	struct snd_soc_card *card = dapm->card;
 
 	/* check for valid widgets */
 	if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
@@ -309,24 +326,26 @@
 
 	change = old != new;
 	if (change) {
-		pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n",
+		pop_dbg(dapm->dev, card->pop_time,
+			"pop test %s : %s in %d ms\n",
 			widget->name, widget->power ? "on" : "off",
-			codec->pop_time);
-		pop_wait(codec->pop_time);
+			card->pop_time);
+		pop_wait(card->pop_time);
 		snd_soc_write(codec, widget->reg, new);
 	}
-	pr_debug("reg %x old %x new %x change %d\n", widget->reg,
-		 old, new, change);
+	dev_dbg(dapm->dev, "reg %x old %x new %x change %d\n", widget->reg,
+		old, new, change);
 	return change;
 }
 
 /* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_codec *codec,
+static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_dapm_widget *w)
 {
 	int i, ret = 0;
 	size_t name_len;
 	struct snd_soc_dapm_path *path;
+	struct snd_card *card = dapm->codec->card->snd_card;
 
 	/* add kcontrol */
 	for (i = 0; i < w->num_kcontrols; i++) {
@@ -368,11 +387,11 @@
 
 			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
 				path->long_name);
-			ret = snd_ctl_add(codec->card->snd_card, path->kcontrol);
+			ret = snd_ctl_add(card, path->kcontrol);
 			if (ret < 0) {
-				printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n",
-				       path->long_name,
-				       ret);
+				dev_err(dapm->dev,
+					"asoc: failed to add dapm kcontrol %s: %d\n",
+					path->long_name, ret);
 				kfree(path->long_name);
 				path->long_name = NULL;
 				return ret;
@@ -383,20 +402,22 @@
 }
 
 /* create new dapm mux control */
-static int dapm_new_mux(struct snd_soc_codec *codec,
+static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_dapm_widget *w)
 {
 	struct snd_soc_dapm_path *path = NULL;
 	struct snd_kcontrol *kcontrol;
+	struct snd_card *card = dapm->codec->card->snd_card;
 	int ret = 0;
 
 	if (!w->num_kcontrols) {
-		printk(KERN_ERR "asoc: mux %s has no controls\n", w->name);
+		dev_err(dapm->dev, "asoc: mux %s has no controls\n", w->name);
 		return -EINVAL;
 	}
 
 	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
-	ret = snd_ctl_add(codec->card->snd_card, kcontrol);
+	ret = snd_ctl_add(card, kcontrol);
+
 	if (ret < 0)
 		goto err;
 
@@ -406,26 +427,27 @@
 	return ret;
 
 err:
-	printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
+	dev_err(dapm->dev, "asoc: failed to add kcontrol %s\n", w->name);
 	return ret;
 }
 
 /* create new dapm volume control */
-static int dapm_new_pga(struct snd_soc_codec *codec,
+static int dapm_new_pga(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_dapm_widget *w)
 {
 	if (w->num_kcontrols)
-		pr_err("asoc: PGA controls not supported: '%s'\n", w->name);
+		dev_err(w->dapm->dev,
+			"asoc: PGA controls not supported: '%s'\n", w->name);
 
 	return 0;
 }
 
 /* reset 'walked' bit for each dapm path */
-static inline void dapm_clear_walk(struct snd_soc_codec *codec)
+static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_path *p;
 
-	list_for_each_entry(p, &codec->dapm_paths, list)
+	list_for_each_entry(p, &dapm->paths, list)
 		p->walked = 0;
 }
 
@@ -435,13 +457,14 @@
  */
 static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
 {
-	int level = snd_power_get_state(widget->codec->card->snd_card);
+	int level = snd_power_get_state(widget->dapm->codec->card->snd_card);
 
 	switch (level) {
 	case SNDRV_CTL_POWER_D3hot:
 	case SNDRV_CTL_POWER_D3cold:
 		if (widget->ignore_suspend)
-			pr_debug("%s ignoring suspend\n", widget->name);
+			dev_dbg(widget->dapm->dev, "%s ignoring suspend\n",
+				widget->name);
 		return widget->ignore_suspend;
 	default:
 		return 1;
@@ -572,7 +595,7 @@
 
 	/* call any power change event handlers */
 	if (w->event)
-		pr_debug("power %s event for %s flags %x\n",
+		dev_dbg(w->dapm->dev, "power %s event for %s flags %x\n",
 			 w->power ? "on" : "off",
 			 w->name, w->event_flags);
 
@@ -621,9 +644,9 @@
 	int in, out;
 
 	in = is_connected_input_ep(w);
-	dapm_clear_walk(w->codec);
+	dapm_clear_walk(w->dapm);
 	out = is_connected_output_ep(w);
-	dapm_clear_walk(w->codec);
+	dapm_clear_walk(w->dapm);
 	return out != 0 && in != 0;
 }
 
@@ -634,7 +657,7 @@
 
 	if (w->active) {
 		in = is_connected_input_ep(w);
-		dapm_clear_walk(w->codec);
+		dapm_clear_walk(w->dapm);
 		return in != 0;
 	} else {
 		return dapm_generic_check_power(w);
@@ -648,7 +671,7 @@
 
 	if (w->active) {
 		out = is_connected_output_ep(w);
-		dapm_clear_walk(w->codec);
+		dapm_clear_walk(w->dapm);
 		return out != 0;
 	} else {
 		return dapm_generic_check_power(w);
@@ -674,7 +697,7 @@
 		}
 	}
 
-	dapm_clear_walk(w->codec);
+	dapm_clear_walk(w->dapm);
 
 	return power;
 }
@@ -687,8 +710,8 @@
 		return sort[a->id] - sort[b->id];
 	if (a->reg != b->reg)
 		return a->reg - b->reg;
-	if (a->codec != b->codec)
-		return (unsigned long)a->codec - (unsigned long)b->codec;
+	if (a->dapm != b->dapm)
+		return (unsigned long)a->dapm - (unsigned long)b->dapm;
 
 	return 0;
 }
@@ -709,12 +732,57 @@
 	list_add_tail(&new_widget->power_list, list);
 }
 
+static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm,
+				 struct snd_soc_dapm_widget *w, int event)
+{
+	struct snd_soc_card *card = dapm->card;
+	const char *ev_name;
+	int power, ret;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		ev_name = "PRE_PMU";
+		power = 1;
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		ev_name = "POST_PMU";
+		power = 1;
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		ev_name = "PRE_PMD";
+		power = 0;
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		ev_name = "POST_PMD";
+		power = 0;
+		break;
+	default:
+		BUG();
+		return;
+	}
+
+	if (w->power != power)
+		return;
+
+	if (w->event && (w->event_flags & event)) {
+		pop_dbg(dapm->dev, card->pop_time, "pop test : %s %s\n",
+			w->name, ev_name);
+		trace_snd_soc_dapm_widget_event_start(w, event);
+		ret = w->event(w, NULL, event);
+		trace_snd_soc_dapm_widget_event_done(w, event);
+		if (ret < 0)
+			pr_err("%s: %s event failed: %d\n",
+			       ev_name, w->name, ret);
+	}
+}
+
 /* Apply the coalesced changes from a DAPM sequence */
-static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
+static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
 				   struct list_head *pending)
 {
+	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
-	int reg, power, ret;
+	int reg, power;
 	unsigned int value = 0;
 	unsigned int mask = 0;
 	unsigned int cur_mask;
@@ -735,64 +803,26 @@
 		if (power)
 			value |= cur_mask;
 
-		pop_dbg(codec->pop_time,
+		pop_dbg(dapm->dev, card->pop_time,
 			"pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n",
 			w->name, reg, value, mask);
 
-		/* power up pre event */
-		if (w->power && w->event &&
-		    (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
-			pop_dbg(codec->pop_time, "pop test : %s PRE_PMU\n",
-				w->name);
-			ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
-			if (ret < 0)
-				pr_err("%s: pre event failed: %d\n",
-				       w->name, ret);
-		}
-
-		/* power down pre event */
-		if (!w->power && w->event &&
-		    (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
-			pop_dbg(codec->pop_time, "pop test : %s PRE_PMD\n",
-				w->name);
-			ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
-			if (ret < 0)
-				pr_err("%s: pre event failed: %d\n",
-				       w->name, ret);
-		}
+		/* Check for events */
+		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMU);
+		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMD);
 	}
 
 	if (reg >= 0) {
-		pop_dbg(codec->pop_time,
+		pop_dbg(dapm->dev, card->pop_time,
 			"pop test : Applying 0x%x/0x%x to %x in %dms\n",
-			value, mask, reg, codec->pop_time);
-		pop_wait(codec->pop_time);
-		snd_soc_update_bits(codec, reg, mask, value);
+			value, mask, reg, card->pop_time);
+		pop_wait(card->pop_time);
+		snd_soc_update_bits(dapm->codec, reg, mask, value);
 	}
 
 	list_for_each_entry(w, pending, power_list) {
-		/* power up post event */
-		if (w->power && w->event &&
-		    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
-			pop_dbg(codec->pop_time, "pop test : %s POST_PMU\n",
-				w->name);
-			ret = w->event(w,
-				       NULL, SND_SOC_DAPM_POST_PMU);
-			if (ret < 0)
-				pr_err("%s: post event failed: %d\n",
-				       w->name, ret);
-		}
-
-		/* power down post event */
-		if (!w->power && w->event &&
-		    (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
-			pop_dbg(codec->pop_time, "pop test : %s POST_PMD\n",
-				w->name);
-			ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
-			if (ret < 0)
-				pr_err("%s: post event failed: %d\n",
-				       w->name, ret);
-		}
+		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMU);
+		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMD);
 	}
 }
 
@@ -804,8 +834,8 @@
  * Currently anything that requires more than a single write is not
  * handled.
  */
-static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list,
-			 int event, int sort[])
+static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
+			 struct list_head *list, int event, int sort[])
 {
 	struct snd_soc_dapm_widget *w, *n;
 	LIST_HEAD(pending);
@@ -819,7 +849,7 @@
 		/* Do we need to apply any queued changes? */
 		if (sort[w->id] != cur_sort || w->reg != cur_reg) {
 			if (!list_empty(&pending))
-				dapm_seq_run_coalesced(codec, &pending);
+				dapm_seq_run_coalesced(dapm, &pending);
 
 			INIT_LIST_HEAD(&pending);
 			cur_sort = -1;
@@ -872,12 +902,12 @@
 		}
 
 		if (ret < 0)
-			pr_err("Failed to apply widget power: %d\n",
-			       ret);
+			dev_err(w->dapm->dev,
+				"Failed to apply widget power: %d\n", ret);
 	}
 
 	if (!list_empty(&pending))
-		dapm_seq_run_coalesced(codec, &pending);
+		dapm_seq_run_coalesced(dapm, &pending);
 }
 
 /*
@@ -889,9 +919,9 @@
  *  o Input pin to Output pin (bypass, sidetone)
  *  o DAC to ADC (loopback).
  */
-static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
+static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 {
-	struct snd_soc_card *card = codec->card;
+	struct snd_soc_card *card = dapm->codec->card;
 	struct snd_soc_dapm_widget *w;
 	LIST_HEAD(up_list);
 	LIST_HEAD(down_list);
@@ -899,10 +929,12 @@
 	int power;
 	int sys_power = 0;
 
+	trace_snd_soc_dapm_start(card);
+
 	/* Check which widgets we need to power and store them in
 	 * lists indicating if they should be powered up or down.
 	 */
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 		switch (w->id) {
 		case snd_soc_dapm_pre:
 			dapm_seq_insert(w, &down_list, dapm_down_seq);
@@ -925,6 +957,8 @@
 			if (w->power == power)
 				continue;
 
+			trace_snd_soc_dapm_widget_power(w, power);
+
 			if (power)
 				dapm_seq_insert(w, &up_list, dapm_up_seq);
 			else
@@ -938,7 +972,7 @@
 	/* If there are no DAPM widgets then try to figure out power from the
 	 * event type.
 	 */
-	if (list_empty(&codec->dapm_widgets)) {
+	if (list_empty(&dapm->widgets)) {
 		switch (event) {
 		case SND_SOC_DAPM_STREAM_START:
 		case SND_SOC_DAPM_STREAM_RESUME:
@@ -948,7 +982,7 @@
 			sys_power = 0;
 			break;
 		case SND_SOC_DAPM_STREAM_NOP:
-			switch (codec->bias_level) {
+			switch (dapm->bias_level) {
 				case SND_SOC_BIAS_STANDBY:
 				case SND_SOC_BIAS_OFF:
 					sys_power = 0;
@@ -963,52 +997,59 @@
 		}
 	}
 
-	if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) {
-		ret = snd_soc_dapm_set_bias_level(card, codec,
+	if (sys_power && dapm->bias_level == SND_SOC_BIAS_OFF) {
+		ret = snd_soc_dapm_set_bias_level(card, dapm,
 						  SND_SOC_BIAS_STANDBY);
 		if (ret != 0)
-			pr_err("Failed to turn on bias: %d\n", ret);
+			dev_err(dapm->dev,
+				"Failed to turn on bias: %d\n", ret);
 	}
 
 	/* If we're changing to all on or all off then prepare */
-	if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) ||
-	    (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) {
-		ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_PREPARE);
+	if ((sys_power && dapm->bias_level == SND_SOC_BIAS_STANDBY) ||
+	    (!sys_power && dapm->bias_level == SND_SOC_BIAS_ON)) {
+		ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_PREPARE);
 		if (ret != 0)
-			pr_err("Failed to prepare bias: %d\n", ret);
+			dev_err(dapm->dev,
+				"Failed to prepare bias: %d\n", ret);
 	}
 
 	/* Power down widgets first; try to avoid amplifying pops. */
-	dapm_seq_run(codec, &down_list, event, dapm_down_seq);
+	dapm_seq_run(dapm, &down_list, event, dapm_down_seq);
 
 	/* Now power up. */
-	dapm_seq_run(codec, &up_list, event, dapm_up_seq);
+	dapm_seq_run(dapm, &up_list, event, dapm_up_seq);
 
 	/* If we just powered the last thing off drop to standby bias */
-	if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) {
-		ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_STANDBY);
+	if (dapm->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) {
+		ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
 		if (ret != 0)
-			pr_err("Failed to apply standby bias: %d\n", ret);
+			dev_err(dapm->dev,
+				"Failed to apply standby bias: %d\n", ret);
 	}
 
 	/* If we're in standby and can support bias off then do that */
-	if (codec->bias_level == SND_SOC_BIAS_STANDBY &&
-	    codec->idle_bias_off) {
-		ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF);
+	if (dapm->bias_level == SND_SOC_BIAS_STANDBY &&
+	    dapm->idle_bias_off) {
+		ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_OFF);
 		if (ret != 0)
-			pr_err("Failed to turn off bias: %d\n", ret);
+			dev_err(dapm->dev,
+				"Failed to turn off bias: %d\n", ret);
 	}
 
 	/* If we just powered up then move to active bias */
-	if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) {
-		ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_ON);
+	if (dapm->bias_level == SND_SOC_BIAS_PREPARE && sys_power) {
+		ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_ON);
 		if (ret != 0)
-			pr_err("Failed to apply active bias: %d\n", ret);
+			dev_err(dapm->dev,
+				"Failed to apply active bias: %d\n", ret);
 	}
 
-	pop_dbg(codec->pop_time, "DAPM sequencing finished, waiting %dms\n",
-		codec->pop_time);
-	pop_wait(codec->pop_time);
+	pop_dbg(dapm->dev, card->pop_time,
+		"DAPM sequencing finished, waiting %dms\n", card->pop_time);
+	pop_wait(card->pop_time);
+
+	trace_snd_soc_dapm_done(card);
 
 	return 0;
 }
@@ -1035,9 +1076,9 @@
 		return -ENOMEM;
 
 	in = is_connected_input_ep(w);
-	dapm_clear_walk(w->codec);
+	dapm_clear_walk(w->dapm);
 	out = is_connected_output_ep(w);
-	dapm_clear_walk(w->codec);
+	dapm_clear_walk(w->dapm);
 
 	ret = snprintf(buf, PAGE_SIZE, "%s: %s  in %d out %d",
 		       w->name, w->power ? "On" : "Off", in, out);
@@ -1087,29 +1128,29 @@
 	.llseek = default_llseek,
 };
 
-void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_widget *w;
 	struct dentry *d;
 
-	if (!codec->debugfs_dapm)
+	if (!dapm->debugfs_dapm)
 		return;
 
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 		if (!w->name)
 			continue;
 
 		d = debugfs_create_file(w->name, 0444,
-					codec->debugfs_dapm, w,
+					dapm->debugfs_dapm, w,
 					&dapm_widget_power_fops);
 		if (!d)
-			printk(KERN_WARNING
-			       "ASoC: Failed to create %s debugfs file\n",
-			       w->name);
+			dev_warn(w->dapm->dev,
+				"ASoC: Failed to create %s debugfs file\n",
+				w->name);
 	}
 }
 #else
-void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm)
 {
 }
 #endif
@@ -1130,7 +1171,7 @@
 		return 0;
 
 	/* find dapm widget path assoc with kcontrol */
-	list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+	list_for_each_entry(path, &widget->dapm->paths, list) {
 		if (path->kcontrol != kcontrol)
 			continue;
 
@@ -1146,7 +1187,7 @@
 	}
 
 	if (found)
-		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+		dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
 
 	return 0;
 }
@@ -1164,7 +1205,7 @@
 		return -ENODEV;
 
 	/* find dapm widget path assoc with kcontrol */
-	list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+	list_for_each_entry(path, &widget->dapm->paths, list) {
 		if (path->kcontrol != kcontrol)
 			continue;
 
@@ -1175,7 +1216,7 @@
 	}
 
 	if (found)
-		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+		dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
 
 	return 0;
 }
@@ -1191,7 +1232,7 @@
 	int count = 0;
 	char *state = "not set";
 
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &codec->dapm.widgets, list) {
 
 		/* only display widgets that burnm power */
 		switch (w->id) {
@@ -1215,7 +1256,7 @@
 		}
 	}
 
-	switch (codec->bias_level) {
+	switch (codec->dapm.bias_level) {
 	case SND_SOC_BIAS_ON:
 		state = "On";
 		break;
@@ -1247,31 +1288,33 @@
 }
 
 /* free all dapm widgets and resources */
-static void dapm_free_widgets(struct snd_soc_codec *codec)
+static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_widget *w, *next_w;
 	struct snd_soc_dapm_path *p, *next_p;
 
-	list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) {
+	list_for_each_entry_safe(w, next_w, &dapm->widgets, list) {
 		list_del(&w->list);
+		kfree(w->name);
 		kfree(w);
 	}
 
-	list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) {
+	list_for_each_entry_safe(p, next_p, &dapm->paths, list) {
 		list_del(&p->list);
 		kfree(p->long_name);
 		kfree(p);
 	}
 }
 
-static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
+static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
 				const char *pin, int status)
 {
 	struct snd_soc_dapm_widget *w;
 
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 		if (!strcmp(w->name, pin)) {
-			pr_debug("dapm: %s: pin %s\n", codec->name, pin);
+			dev_dbg(w->dapm->dev, "dapm: pin %s = %d\n",
+				pin, status);
 			w->connected = status;
 			/* Allow disabling of forced pins */
 			if (status == 0)
@@ -1280,37 +1323,51 @@
 		}
 	}
 
-	pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+	dev_err(dapm->dev, "dapm: unknown pin %s\n", pin);
 	return -EINVAL;
 }
 
 /**
  * snd_soc_dapm_sync - scan and power dapm paths
- * @codec: audio codec
+ * @dapm: DAPM context
  *
  * Walks all dapm audio paths and powers widgets according to their
  * stream or path usage.
  *
  * Returns 0 for success.
  */
-int snd_soc_dapm_sync(struct snd_soc_codec *codec)
+int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
 {
-	return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+	return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
 
-static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
+static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 				  const struct snd_soc_dapm_route *route)
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
-	const char *sink = route->sink;
+	const char *sink;
 	const char *control = route->control;
-	const char *source = route->source;
+	const char *source;
+	char prefixed_sink[80];
+	char prefixed_source[80];
 	int ret = 0;
 
+	if (dapm->codec->name_prefix) {
+		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
+			 dapm->codec->name_prefix, route->sink);
+		sink = prefixed_sink;
+		snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
+			 dapm->codec->name_prefix, route->source);
+		source = prefixed_source;
+	} else {
+		sink = route->sink;
+		source = route->source;
+	}
+
 	/* find src and dest widgets */
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 
 		if (!wsink && !(strcmp(w->name, sink))) {
 			wsink = w;
@@ -1353,7 +1410,7 @@
 
 	/* connect static paths */
 	if (control == NULL) {
-		list_add(&path->list, &codec->dapm_paths);
+		list_add(&path->list, &dapm->paths);
 		list_add(&path->list_sink, &wsink->sources);
 		list_add(&path->list_source, &wsource->sinks);
 		path->connect = 1;
@@ -1374,14 +1431,14 @@
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
-		list_add(&path->list, &codec->dapm_paths);
+		list_add(&path->list, &dapm->paths);
 		list_add(&path->list_sink, &wsink->sources);
 		list_add(&path->list_source, &wsource->sinks);
 		path->connect = 1;
 		return 0;
 	case snd_soc_dapm_mux:
 	case snd_soc_dapm_value_mux:
-		ret = dapm_connect_mux(codec, wsource, wsink, path, control,
+		ret = dapm_connect_mux(dapm, wsource, wsink, path, control,
 			&wsink->kcontrols[0]);
 		if (ret != 0)
 			goto err;
@@ -1389,7 +1446,7 @@
 	case snd_soc_dapm_switch:
 	case snd_soc_dapm_mixer:
 	case snd_soc_dapm_mixer_named_ctl:
-		ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
+		ret = dapm_connect_mixer(dapm, wsource, wsink, path, control);
 		if (ret != 0)
 			goto err;
 		break;
@@ -1397,7 +1454,7 @@
 	case snd_soc_dapm_mic:
 	case snd_soc_dapm_line:
 	case snd_soc_dapm_spk:
-		list_add(&path->list, &codec->dapm_paths);
+		list_add(&path->list, &dapm->paths);
 		list_add(&path->list_sink, &wsink->sources);
 		list_add(&path->list_source, &wsource->sinks);
 		path->connect = 0;
@@ -1406,15 +1463,15 @@
 	return 0;
 
 err:
-	printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source,
-		control, sink);
+	dev_warn(dapm->dev, "asoc: no dapm match for %s --> %s --> %s\n",
+		 source, control, sink);
 	kfree(path);
 	return ret;
 }
 
 /**
  * snd_soc_dapm_add_routes - Add routes between DAPM widgets
- * @codec: codec
+ * @dapm: DAPM context
  * @route: audio routes
  * @num: number of routes
  *
@@ -1425,17 +1482,16 @@
  * Returns 0 for success else error. On error all resources can be freed
  * with a call to snd_soc_card_free().
  */
-int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num)
 {
 	int i, ret;
 
 	for (i = 0; i < num; i++) {
-		ret = snd_soc_dapm_add_route(codec, route);
+		ret = snd_soc_dapm_add_route(dapm, route);
 		if (ret < 0) {
-			printk(KERN_ERR "Failed to add route %s->%s\n",
-			       route->source,
-			       route->sink);
+			dev_err(dapm->dev, "Failed to add route %s->%s\n",
+				route->source, route->sink);
 			return ret;
 		}
 		route++;
@@ -1447,17 +1503,17 @@
 
 /**
  * snd_soc_dapm_new_widgets - add new dapm widgets
- * @codec: audio codec
+ * @dapm: DAPM context
  *
  * Checks the codec for any new dapm widgets and creates them if found.
  *
  * Returns 0 for success.
  */
-int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
+int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_widget *w;
 
-	list_for_each_entry(w, &codec->dapm_widgets, list)
+	list_for_each_entry(w, &dapm->widgets, list)
 	{
 		if (w->new)
 			continue;
@@ -1467,12 +1523,12 @@
 		case snd_soc_dapm_mixer:
 		case snd_soc_dapm_mixer_named_ctl:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_mixer(codec, w);
+			dapm_new_mixer(dapm, w);
 			break;
 		case snd_soc_dapm_mux:
 		case snd_soc_dapm_value_mux:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_mux(codec, w);
+			dapm_new_mux(dapm, w);
 			break;
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_aif_out:
@@ -1484,7 +1540,7 @@
 			break;
 		case snd_soc_dapm_pga:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_pga(codec, w);
+			dapm_new_pga(dapm, w);
 			break;
 		case snd_soc_dapm_input:
 		case snd_soc_dapm_output:
@@ -1505,7 +1561,7 @@
 		w->new = 1;
 	}
 
-	dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+	dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
@@ -1889,7 +1945,7 @@
 	mutex_lock(&codec->mutex);
 
 	ucontrol->value.integer.value[0] =
-		snd_soc_dapm_get_pin_status(codec, pin);
+		snd_soc_dapm_get_pin_status(&codec->dapm, pin);
 
 	mutex_unlock(&codec->mutex);
 
@@ -1912,11 +1968,11 @@
 	mutex_lock(&codec->mutex);
 
 	if (ucontrol->value.integer.value[0])
-		snd_soc_dapm_enable_pin(codec, pin);
+		snd_soc_dapm_enable_pin(&codec->dapm, pin);
 	else
-		snd_soc_dapm_disable_pin(codec, pin);
+		snd_soc_dapm_disable_pin(&codec->dapm, pin);
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(&codec->dapm);
 
 	mutex_unlock(&codec->mutex);
 
@@ -1926,26 +1982,42 @@
 
 /**
  * snd_soc_dapm_new_control - create new dapm control
- * @codec: audio codec
+ * @dapm: DAPM context
  * @widget: widget template
  *
  * Creates a new dapm control based upon the template.
  *
  * Returns 0 for success else error.
  */
-int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	const struct snd_soc_dapm_widget *widget)
 {
 	struct snd_soc_dapm_widget *w;
+	size_t name_len;
 
 	if ((w = dapm_cnew_widget(widget)) == NULL)
 		return -ENOMEM;
 
-	w->codec = codec;
+	name_len = strlen(widget->name) + 1;
+	if (dapm->codec->name_prefix)
+		name_len += 1 + strlen(dapm->codec->name_prefix);
+	w->name = kmalloc(name_len, GFP_KERNEL);
+	if (w->name == NULL) {
+		kfree(w);
+		return -ENOMEM;
+	}
+	if (dapm->codec->name_prefix)
+		snprintf(w->name, name_len, "%s %s",
+			dapm->codec->name_prefix, widget->name);
+	else
+		snprintf(w->name, name_len, "%s", widget->name);
+
+	w->dapm = dapm;
+	w->codec = dapm->codec;
 	INIT_LIST_HEAD(&w->sources);
 	INIT_LIST_HEAD(&w->sinks);
 	INIT_LIST_HEAD(&w->list);
-	list_add(&w->list, &codec->dapm_widgets);
+	list_add(&w->list, &dapm->widgets);
 
 	/* machine layer set ups unconnected pins and insertions */
 	w->connected = 1;
@@ -1955,7 +2027,7 @@
 
 /**
  * snd_soc_dapm_new_controls - create new dapm controls
- * @codec: audio codec
+ * @dapm: DAPM context
  * @widget: widget array
  * @num: number of widgets
  *
@@ -1963,18 +2035,18 @@
  *
  * Returns 0 for success else error.
  */
-int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 	const struct snd_soc_dapm_widget *widget,
 	int num)
 {
 	int i, ret;
 
 	for (i = 0; i < num; i++) {
-		ret = snd_soc_dapm_new_control(codec, widget);
+		ret = snd_soc_dapm_new_control(dapm, widget);
 		if (ret < 0) {
-			printk(KERN_ERR
-			       "ASoC: Failed to create DAPM control %s: %d\n",
-			       widget->name, ret);
+			dev_err(dapm->dev,
+				"ASoC: Failed to create DAPM control %s: %d\n",
+				widget->name, ret);
 			return ret;
 		}
 		widget++;
@@ -1983,34 +2055,17 @@
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 
-
-/**
- * snd_soc_dapm_stream_event - send a stream event to the dapm core
- * @codec: audio codec
- * @stream: stream name
- * @event: stream event
- *
- * Sends a stream event to the dapm core. The core then makes any
- * necessary widget power changes.
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
+static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
 	const char *stream, int event)
 {
-	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_widget *w;
 
-	if (stream == NULL)
-		return 0;
-
-	mutex_lock(&codec->mutex);
-	list_for_each_entry(w, &codec->dapm_widgets, list)
+	list_for_each_entry(w, &dapm->widgets, list)
 	{
 		if (!w->sname)
 			continue;
-		pr_debug("widget %s\n %s stream %s event %d\n",
-			 w->name, w->sname, stream, event);
+		dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
+			w->name, w->sname, stream, event);
 		if (strstr(w->sname, stream)) {
 			switch(event) {
 			case SND_SOC_DAPM_STREAM_START:
@@ -2028,7 +2083,30 @@
 		}
 	}
 
-	dapm_power_widgets(codec, event);
+	dapm_power_widgets(dapm, event);
+}
+
+/**
+ * snd_soc_dapm_stream_event - send a stream event to the dapm core
+ * @rtd: PCM runtime data
+ * @stream: stream name
+ * @event: stream event
+ *
+ * Sends a stream event to the dapm core. The core then makes any
+ * necessary widget power changes.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
+	const char *stream, int event)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+
+	if (stream == NULL)
+		return 0;
+
+	mutex_lock(&codec->mutex);
+	soc_dapm_stream_event(&codec->dapm, stream, event);
 	mutex_unlock(&codec->mutex);
 	return 0;
 }
@@ -2036,7 +2114,7 @@
 
 /**
  * snd_soc_dapm_enable_pin - enable pin.
- * @codec: SoC codec
+ * @dapm: DAPM context
  * @pin: pin name
  *
  * Enables input/output pin and its parents or children widgets iff there is
@@ -2044,15 +2122,15 @@
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin)
 {
-	return snd_soc_dapm_set_pin(codec, pin, 1);
+	return snd_soc_dapm_set_pin(dapm, pin, 1);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
 
 /**
  * snd_soc_dapm_force_enable_pin - force a pin to be enabled
- * @codec: SoC codec
+ * @dapm: DAPM context
  * @pin: pin name
  *
  * Enables input/output pin regardless of any other state.  This is
@@ -2062,42 +2140,45 @@
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
+				  const char *pin)
 {
 	struct snd_soc_dapm_widget *w;
 
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 		if (!strcmp(w->name, pin)) {
-			pr_debug("dapm: %s: pin %s\n", codec->name, pin);
+			dev_dbg(w->dapm->dev,
+				"dapm: force enable pin %s\n", pin);
 			w->connected = 1;
 			w->force = 1;
 			return 0;
 		}
 	}
 
-	pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+	dev_err(dapm->dev, "dapm: unknown pin %s\n", pin);
 	return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin);
 
 /**
  * snd_soc_dapm_disable_pin - disable pin.
- * @codec: SoC codec
+ * @dapm: DAPM context
  * @pin: pin name
  *
  * Disables input/output pin and its parents or children widgets.
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
+			     const char *pin)
 {
-	return snd_soc_dapm_set_pin(codec, pin, 0);
+	return snd_soc_dapm_set_pin(dapm, pin, 0);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
 
 /**
  * snd_soc_dapm_nc_pin - permanently disable pin.
- * @codec: SoC codec
+ * @dapm: DAPM context
  * @pin: pin name
  *
  * Marks the specified pin as being not connected, disabling it along
@@ -2109,26 +2190,27 @@
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin)
 {
-	return snd_soc_dapm_set_pin(codec, pin, 0);
+	return snd_soc_dapm_set_pin(dapm, pin, 0);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
 
 /**
  * snd_soc_dapm_get_pin_status - get audio pin status
- * @codec: audio codec
+ * @dapm: DAPM context
  * @pin: audio signal pin endpoint (or start point)
  *
  * Get audio pin status - connected or disconnected.
  *
  * Returns 1 for connected otherwise 0.
  */
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm,
+				const char *pin)
 {
 	struct snd_soc_dapm_widget *w;
 
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 		if (!strcmp(w->name, pin))
 			return w->connected;
 	}
@@ -2139,7 +2221,7 @@
 
 /**
  * snd_soc_dapm_ignore_suspend - ignore suspend status for DAPM endpoint
- * @codec: audio codec
+ * @dapm: DAPM context
  * @pin: audio signal pin endpoint (or start point)
  *
  * Mark the given endpoint or pin as ignoring suspend.  When the
@@ -2148,18 +2230,19 @@
  * normal means at suspend time, it will not be turned on if it was not
  * already enabled.
  */
-int snd_soc_dapm_ignore_suspend(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
+				const char *pin)
 {
 	struct snd_soc_dapm_widget *w;
 
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 		if (!strcmp(w->name, pin)) {
 			w->ignore_suspend = 1;
 			return 0;
 		}
 	}
 
-	pr_err("Unknown DAPM pin: %s\n", pin);
+	dev_err(dapm->dev, "dapm: unknown pin %s\n", pin);
 	return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
@@ -2170,20 +2253,20 @@
  *
  * Free all dapm widgets and resources.
  */
-void snd_soc_dapm_free(struct snd_soc_codec *codec)
+void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm)
 {
-	snd_soc_dapm_sys_remove(codec->dev);
-	dapm_free_widgets(codec);
+	snd_soc_dapm_sys_remove(dapm->dev);
+	dapm_free_widgets(dapm);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
 
-static void soc_dapm_shutdown_codec(struct snd_soc_codec *codec)
+static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_widget *w;
 	LIST_HEAD(down_list);
 	int powerdown = 0;
 
-	list_for_each_entry(w, &codec->dapm_widgets, list) {
+	list_for_each_entry(w, &dapm->widgets, list) {
 		if (w->power) {
 			dapm_seq_insert(w, &down_list, dapm_down_seq);
 			w->power = 0;
@@ -2195,9 +2278,9 @@
 	 * standby.
 	 */
 	if (powerdown) {
-		snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_PREPARE);
-		dapm_seq_run(codec, &down_list, 0, dapm_down_seq);
-		snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_STANDBY);
+		snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE);
+		dapm_seq_run(dapm, &down_list, 0, dapm_down_seq);
+		snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY);
 	}
 }
 
@@ -2208,10 +2291,10 @@
 {
 	struct snd_soc_codec *codec;
 
-	list_for_each_entry(codec, &card->codec_dev_list, list)
-		soc_dapm_shutdown_codec(codec);
-
-	snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF);
+	list_for_each_entry(codec, &card->codec_dev_list, list) {
+		soc_dapm_shutdown_codec(&codec->dapm);
+		snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF);
+	}
 }
 
 /* Module information */
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 8a0a920..619061d 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -60,6 +60,7 @@
 void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 {
 	struct snd_soc_codec *codec;
+	struct snd_soc_dapm_context *dapm;
 	struct snd_soc_jack_pin *pin;
 	int enable;
 	int oldstatus;
@@ -68,6 +69,7 @@
 		return;
 
 	codec = jack->codec;
+	dapm =  &codec->dapm;
 
 	mutex_lock(&codec->mutex);
 
@@ -88,15 +90,15 @@
 			enable = !enable;
 
 		if (enable)
-			snd_soc_dapm_enable_pin(codec, pin->pin);
+			snd_soc_dapm_enable_pin(dapm, pin->pin);
 		else
-			snd_soc_dapm_disable_pin(codec, pin->pin);
+			snd_soc_dapm_disable_pin(dapm, pin->pin);
 	}
 
 	/* Report before the DAPM sync to help users updating micbias status */
 	blocking_notifier_call_chain(&jack->notifier, status, NULL);
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	snd_jack_report(jack->jack, status);
 
@@ -207,6 +209,10 @@
 static irqreturn_t gpio_handler(int irq, void *data)
 {
 	struct snd_soc_jack_gpio *gpio = data;
+	struct device *dev = gpio->jack->codec->card->dev;
+
+	if (device_may_wakeup(dev))
+		pm_wakeup_event(dev, gpio->debounce_time + 50);
 
 	schedule_delayed_work(&gpio->work,
 			      msecs_to_jiffies(gpio->debounce_time));
@@ -263,11 +269,12 @@
 		INIT_DELAYED_WORK(&gpios[i].work, gpio_work);
 		gpios[i].jack = jack;
 
-		ret = request_irq(gpio_to_irq(gpios[i].gpio),
-				gpio_handler,
-				IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
-				jack->codec->dev->driver->name,
-				&gpios[i]);
+		ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio),
+					      gpio_handler,
+					      IRQF_TRIGGER_RISING |
+					      IRQF_TRIGGER_FALLING,
+					      jack->codec->dev->driver->name,
+					      &gpios[i]);
 		if (ret)
 			goto err;