Merge branch 'topic/asoc' into for-linus

* topic/asoc: (135 commits)
  ASoC: Apostrophe patrol
  ASoC: codec tlv320aic23 fix bogus divide by 0 message
  ASoC: fix NULL pointer dereference in soc_suspend()
  ASoC: Fix build error in twl4030.c
  ASoC: SSM2602: assign last substream to the master when shutting down
  ASoC: Blackfin: document how anomaly 05000250 is handled
  ASoC: Blackfin: set the transfer size according the ac97_frame size
  ASoC: SSM2602: remove unsupported sample rates
  ASoC: TWL4030: Check the interface format for 4 channel mode
  ASoC: TWL4030: Use reg_cache in twl4030_init_chip
  ASoC: Initialise dev for the dummy S/PDIF DAI
  ASoC: Add dummy S/PDIF codec support
  ASoC: correct print specifiers for unsigneds
  ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
  ASoC: Switch FSL SSI DAI over to symmetric_rates
  ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
  ASoC: Fabric bindings for STAC9766 on the Efika
  ASoC: Support for AC97 on Phytec pmc030 base board.
  ASoC: AC97 driver for mpc5200
  ASoC: Main rewite of the mpc5200 audio DMA code
  ...

101 files changed, 12466 insertions, 1646 deletions

diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 9e6763264a2..9ac842be9b4 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -62,6 +62,7 @@ Audio DAPM widgets fall into a number of types:-
  o Mic        - Mic (and optional Jack)
  o Line       - Line Input/Output (and optional Jack)
  o Speaker    - Speaker
+ o Supply     - Power or clock supply widget used by other widgets.
  o Pre        - Special PRE widget (exec before all others)
  o Post       - Special POST widget (exec after all others)
 
diff --git a/MAINTAINERS b/MAINTAINERS
index cf4abddfc8a..e8cb115e33b 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -4574,7 +4574,8 @@ F:	drivers/pcmcia/pxa2xx*
 F:	drivers/spi/pxa2xx*
 F:	drivers/usb/gadget/pxa2*
 F:	include/sound/pxa2xx-lib.h
-F:	sound/soc/pxa/pxa2xx*
+F:	sound/arm/pxa*
+F:	sound/soc/pxa
 
 PXA168 SUPPORT
 P:	Eric Miao
@@ -5302,11 +5303,12 @@ P:	Liam Girdwood
 M:	lrg@slimlogic.co.uk
 P:	Mark Brown
 M:	broonie@opensource.wolfsonmicro.com
-T:	git git://opensource.wolfsonmicro.com/linux-2.6-asoc
+T:	git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
 L:	alsa-devel@alsa-project.org (subscribers-only)
 W:	http://alsa-project.org/main/index.php/ASoC
 S:	Supported
 F:	sound/soc/
+F:	include/sound/soc*
 
 SPARC + UltraSPARC (sparc/sparc64)
 P:	David S. Miller
diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h
index a218da6bec7..fb841205745 100644
--- a/arch/powerpc/include/asm/mpc52xx_psc.h
+++ b/arch/powerpc/include/asm/mpc52xx_psc.h
@@ -28,6 +28,10 @@
 #define MPC52xx_PSC_MAXNUM	6
 
 /* Programmable Serial Controller (PSC) status register bits */
+#define MPC52xx_PSC_SR_UNEX_RX	0x0001
+#define MPC52xx_PSC_SR_DATA_VAL	0x0002
+#define MPC52xx_PSC_SR_DATA_OVR	0x0004
+#define MPC52xx_PSC_SR_CMDSEND	0x0008
 #define MPC52xx_PSC_SR_CDE	0x0080
 #define MPC52xx_PSC_SR_RXRDY	0x0100
 #define MPC52xx_PSC_SR_RXFULL	0x0200
@@ -61,6 +65,12 @@
 #define MPC52xx_PSC_RXTX_FIFO_EMPTY	0x0001
 
 /* PSC interrupt status/mask bits */
+#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001
+#define MPC52xx_PSC_IMR_DATA_VALID	0x0002
+#define MPC52xx_PSC_IMR_DATA_OVR	0x0004
+#define MPC52xx_PSC_IMR_CMD_SEND	0x0008
+#define MPC52xx_PSC_IMR_ERROR		0x0040
+#define MPC52xx_PSC_IMR_DEOF		0x0080
 #define MPC52xx_PSC_IMR_TXRDY		0x0100
 #define MPC52xx_PSC_IMR_RXRDY		0x0200
 #define MPC52xx_PSC_IMR_DB		0x0400
@@ -117,6 +127,7 @@
 #define MPC52xx_PSC_SICR_SIM_FIR		(0x6 << 24)
 #define MPC52xx_PSC_SICR_SIM_CODEC_24		(0x7 << 24)
 #define MPC52xx_PSC_SICR_SIM_CODEC_32		(0xf << 24)
+#define MPC52xx_PSC_SICR_AWR			(1 << 30)
 #define MPC52xx_PSC_SICR_GENCLK			(1 << 23)
 #define MPC52xx_PSC_SICR_I2S			(1 << 22)
 #define MPC52xx_PSC_SICR_CLKPOL			(1 << 21)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 13676472ddf..352d7eee9b6 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -45,24 +45,6 @@ struct snd_pcm_substream;
 #define SND_SOC_DAIFMT_GATED		(1 << 4) /* clock is gated */
 
 /*
- * DAI Left/Right Clocks.
- *
- * Specifies whether the DAI can support different samples for similtanious
- * playback and capture. This usually requires a seperate physical frame
- * clock for playback and capture.
- */
-#define SND_SOC_DAIFMT_SYNC		(0 << 5) /* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC		(1 << 5) /* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- *
- * Time Division Multiplexing. Allows PCM data to be multplexed with other
- * data on the DAI.
- */
-#define SND_SOC_DAIFMT_TDM		(1 << 6)
-
-/*
  * DAI hardware signal inversions.
  *
  * Specifies whether the DAI can also support inverted clocks for the specified
@@ -96,6 +78,10 @@ struct snd_pcm_substream;
 #define SND_SOC_CLOCK_IN		0
 #define SND_SOC_CLOCK_OUT		1
 
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+                               SNDRV_PCM_FMTBIT_S32_LE |\
+                               SNDRV_PCM_FMTBIT_S32_BE)
+
 struct snd_soc_dai_ops;
 struct snd_soc_dai;
 struct snd_ac97_bus_ops;
@@ -208,6 +194,7 @@ struct snd_soc_dai {
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
 	struct snd_soc_pcm_stream playback;
+	unsigned int symmetric_rates:1;
 
 	/* DAI runtime info */
 	struct snd_pcm_runtime *runtime;
@@ -219,11 +206,8 @@ struct snd_soc_dai {
 	/* DAI private data */
 	void *private_data;
 
-	/* parent codec/platform */
-	union {
-		struct snd_soc_codec *codec;
-		struct snd_soc_platform *platform;
-	};
+	/* parent platform */
+	struct snd_soc_platform *platform;
 
 	struct list_head list;
 };
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index a7def6a9a03..ec8a45f9a06 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -140,16 +140,30 @@
 #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
 	.shift = wshift, .invert = winvert}
+#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \
+			   wevent, wflags)				\
+{	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
+	.shift = wshift, .invert = winvert, \
+	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
 	.shift = wshift, .invert = winvert}
+#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \
+			   wevent, wflags)				\
+{	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
+	.shift = wshift, .invert = winvert, \
+	.event = wevent, .event_flags = wflags}
 
-/* generic register modifier widget */
+/* generic widgets */
 #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
 {	.id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \
 	.reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \
 	.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
 	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
+#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
+{	.id = snd_soc_dapm_supply, .name = wname, .reg = wreg,	\
+	.shift = wshift, .invert = winvert, .event = wevent, \
+	.event_flags = wflags}
 
 /* dapm kcontrol types */
 #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
@@ -265,8 +279,6 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
 	int event);
-int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
-	enum snd_soc_bias_level level);
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);
@@ -298,6 +310,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_vmid,			/* codec bias/vmid - to minimise pops */
 	snd_soc_dapm_pre,			/* machine specific pre widget - exec first */
 	snd_soc_dapm_post,			/* machine specific post widget - exec last */
+	snd_soc_dapm_supply,		/* power/clock supply */
 };
 
 /*
@@ -357,6 +370,8 @@ struct snd_soc_dapm_widget {
 	unsigned char suspend:1;		/* was active before suspend */
 	unsigned char pmdown:1;			/* waiting for timeout */
 
+	int (*power_check)(struct snd_soc_dapm_widget *w);
+
 	/* external events */
 	unsigned short event_flags;		/* flags to specify event types */
 	int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);
@@ -368,6 +383,9 @@ struct snd_soc_dapm_widget {
 	/* widget input and outputs */
 	struct list_head sources;
 	struct list_head sinks;
+
+	/* used during DAPM updates */
+	struct list_head power_list;
 };
 
 #endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index a40bc6f316f..cf6111d72b1 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -118,6 +118,14 @@
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+	 xhandler_get, xhandler_put) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.info = snd_soc_info_volsw, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+		 .max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -206,10 +214,6 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
 			struct snd_soc_jack_gpio *gpios);
 #endif
 
-/* codec IO */
-#define snd_soc_read(codec, reg) codec->read(codec, reg)
-#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value)
-
 /* codec register bit access */
 int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
 				unsigned short mask, unsigned short value);
@@ -331,6 +335,7 @@ struct snd_soc_codec {
 	struct module *owner;
 	struct mutex mutex;
 	struct device *dev;
+	struct snd_soc_device *socdev;
 
 	struct list_head list;
 
@@ -364,6 +369,8 @@ struct snd_soc_codec {
 	enum snd_soc_bias_level bias_level;
 	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
+	struct list_head up_list;
+	struct list_head down_list;
 
 	/* codec DAI's */
 	struct snd_soc_dai *dai;
@@ -417,6 +424,12 @@ struct snd_soc_dai_link  {
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_codec *codec);
 
+	/* Symmetry requirements */
+	unsigned int symmetric_rates:1;
+
+	/* Symmetry data - only valid if symmetry is being enforced */
+	unsigned int rate;
+
 	/* DAI pcm */
 	struct snd_pcm *pcm;
 };
@@ -490,6 +503,19 @@ struct soc_enum {
 	void *dapm;
 };
 
+/* codec IO */
+static inline unsigned int snd_soc_read(struct snd_soc_codec *codec,
+					unsigned int reg)
+{
+	return codec->read(codec, reg);
+}
+
+static inline unsigned int snd_soc_write(struct snd_soc_codec *codec,
+					 unsigned int reg, unsigned int val)
+{
+	return codec->write(codec, reg, val);
+}
+
 #include <sound/soc-dai.h>
 
 #endif
diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h
new file mode 100644
index 00000000000..e173ddbf6bd
--- /dev/null
+++ b/include/sound/wm9081.h
@@ -0,0 +1,25 @@
+/*
+ * linux/sound/wm9081.h -- Platform data for WM9081
+ *
+ * Copyright 2009 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM_9081_H
+#define __LINUX_SND_WM_9081_H
+
+struct wm9081_retune_mobile_setting {
+	const char *name;
+	unsigned int rate;
+	u16 config[20];
+};
+
+struct wm9081_retune_mobile_config {
+	struct wm9081_retune_mobile_setting *configs;
+	int num_configs;
+};
+
+#endif
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index fbf5c933baa..586965f9605 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1037,7 +1037,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 	}
 	ldev->selfptr_headphone.ptr = ldev;
 	ldev->selfptr_lineout.ptr = ldev;
-	sdev->ofdev.dev.driver_data = ldev;
+	dev_set_drvdata(&sdev->ofdev.dev, ldev);
 	list_add(&ldev->list, &layouts_list);
 	layouts_list_items++;
 
@@ -1081,7 +1081,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 
 static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
 {
-	struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
 	int i;
 
 	for (i=0; i<MAX_CODECS_PER_BUS; i++) {
@@ -1114,7 +1114,7 @@ static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
 #ifdef CONFIG_PM
 static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
 {
-	struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
 
 	if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
 		ldev->gpio.methods->all_amps_off(&ldev->gpio);
@@ -1124,7 +1124,7 @@ static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t sta
 
 static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
 {
-	struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
 
 	if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
 		ldev->gpio.methods->all_amps_restore(&ldev->gpio);
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 418c84c99d6..4e3b819d499 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -358,14 +358,14 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match)
 		return -ENODEV;
 	}
 
-	dev->ofdev.dev.driver_data = control;
+	dev_set_drvdata(&dev->ofdev.dev, control);
 
 	return 0;
 }
 
 static int i2sbus_remove(struct macio_dev* dev)
 {
-	struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
 	struct i2sbus_dev *i2sdev, *tmp;
 
 	list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
@@ -377,7 +377,7 @@ static int i2sbus_remove(struct macio_dev* dev)
 #ifdef CONFIG_PM
 static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
 {
-	struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
 	struct codec_info_item *cii;
 	struct i2sbus_dev* i2sdev;
 	int err, ret = 0;
@@ -407,7 +407,7 @@ static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
 
 static int i2sbus_resume(struct macio_dev* dev)
 {
-	struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
 	struct codec_info_item *cii;
 	struct i2sbus_dev* i2sdev;
 	int err, ret = 0;
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 3d2bb6fc6dc..d3e786a9a0a 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -32,7 +32,9 @@ source "sound/soc/fsl/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
+source "sound/soc/s6000/Kconfig"
 source "sound/soc/sh/Kconfig"
+source "sound/soc/txx9/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 0237879fd41..6f1e28de23c 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -10,4 +10,6 @@ obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)	+= omap/
 obj-$(CONFIG_SND_SOC)	+= pxa/
 obj-$(CONFIG_SND_SOC)	+= s3c24xx/
+obj-$(CONFIG_SND_SOC)	+= s6000/
 obj-$(CONFIG_SND_SOC)	+= sh/
+obj-$(CONFIG_SND_SOC)	+= txx9/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index a608d7009db..e720d5e6f04 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
           and FRAME signals on the PlayPaq.  Unless you want to play
           with the AT32 as the SSC master, you probably want to say N here,
           as this will give you better sound quality.
+
+config SND_AT91_SOC_AFEB9260
+	tristate "SoC Audio support for AFEB9260 board"
+	depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index f54a7cc68e6..e7ea56bd5f8 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
+obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 70657534e6b..9eb610c2ba9 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
 	 * Find actual rate, compare to requested rate
 	 */
 	actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
-	pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
+	pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
 		 rate, actual_rate);
 
 
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
new file mode 100644
index 00000000000..23349de2731
--- /dev/null
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -0,0 +1,203 @@
+/*
+ * afeb9260.c  --  SoC audio for AFEB9260
+ *
+ * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+#define CODEC_CLOCK 	12000000
+
+static int afeb9260_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S|
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return err;
+	}
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return err;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	err =
+	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return err;
+	}
+
+	return err;
+}
+
+static struct snd_soc_ops afeb9260_ops = {
+	.hw_params = afeb9260_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic Jack"},
+};
+
+static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+	/* Add afeb9260 specific widgets */
+	snd_soc_dapm_new_controls(codec, 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_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &atmel_ssc_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = afeb9260_tlv320aic23_init,
+	.ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+	.name = "AFEB9260",
+	.platform = &atmel_soc_platform,
+	.dai_link = &afeb9260_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device afeb9260_snd_devdata = {
+	.card = &snd_soc_machine_afeb9260,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+	int err;
+	struct device *dev;
+	struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
+	struct ssc_device *ssc = NULL;
+
+	if (!(machine_is_afeb9260()))
+		return -ENODEV;
+
+	ssc = ssc_request(0);
+	if (IS_ERR(ssc)) {
+		printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
+		err = PTR_ERR(ssc);
+		ssc = NULL;
+		goto err_ssc;
+	}
+	ssc_p->ssc = ssc;
+
+	afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!afeb9260_snd_device) {
+		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
+	afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
+	err = platform_device_add(afeb9260_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &afeb9260_snd_device->dev;
+
+	return 0;
+err1:
+	platform_device_del(afeb9260_snd_device);
+	platform_device_put(afeb9260_snd_device);
+err_ssc:
+	return err;
+
+}
+
+static void __exit afeb9260_soc_exit(void)
+{
+	platform_device_unregister(afeb9260_snd_device);
+}
+
+module_init(afeb9260_soc_init);
+module_exit(afeb9260_soc_exit);
+
+MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
+MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 8a935f2d176..b1ed423fabd 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -31,6 +31,15 @@
 #include "bf5xx-sport.h"
 #include "bf5xx-ac97.h"
 
+/* Anomaly notes:
+ *  05000250 -	AD1980 is running in TDM mode and RFS/TFS are generated by SPORT
+ *		contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1,
+ *		while the max AC97 data size is 13*16. The DIV is always larger
+ *		than data size. AD73311 and ad2602 are not running in TDM mode.
+ *		AD1836 and AD73322 depend on external RFS/TFS only. So, this
+ *		anomaly does not affect blackfin sound drivers.
+*/
+
 static int *cmd_count;
 static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
 
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index b7953c8cf83..469ce7fab20 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -190,7 +190,7 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport)
 	desc = get_dma_next_desc_ptr(sport->dma_rx_chan);
 	/* Copy the descriptor which will be damaged to backup */
 	temp_desc = *desc;
-	desc->x_count = 0xa;
+	desc->x_count = sport->dummy_count / 2;
 	desc->y_count = 0;
 	desc->next_desc_addr = sport->dummy_rx_desc;
 	local_irq_restore(flags);
@@ -309,7 +309,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport)
 	desc = get_dma_next_desc_ptr(sport->dma_tx_chan);
 	/* Store the descriptor which will be damaged */
 	temp_desc = *desc;
-	desc->x_count = 0xa;
+	desc->x_count = sport->dummy_count / 2;
 	desc->y_count = 0;
 	desc->next_desc_addr = sport->dummy_tx_desc;
 	local_irq_restore(flags);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b6c7f7a01cb..bbc97fd7664 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -18,7 +18,9 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_PCM3008
+	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if I2C
+	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
 	select SND_SOC_TLV320AIC3X if I2C
@@ -35,8 +37,12 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
+	select SND_SOC_WM8940 if I2C
+	select SND_SOC_WM8960 if I2C
 	select SND_SOC_WM8971 if I2C
+	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8990 if I2C
+	select SND_SOC_WM9081 if I2C
 	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9713 if SND_SOC_AC97_BUS
@@ -86,9 +92,15 @@ config SND_SOC_L3
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_SPDIF
+	tristate
+
 config SND_SOC_SSM2602
 	tristate
 
+config SND_SOC_STAC9766
+	tristate
+
 config SND_SOC_TLV320AIC23
 	tristate
 
@@ -138,12 +150,24 @@ config SND_SOC_WM8900
 config SND_SOC_WM8903
 	tristate
 
+config SND_SOC_WM8940
+        tristate
+
+config SND_SOC_WM8960
+	tristate
+
 config SND_SOC_WM8971
 	tristate
 
+config SND_SOC_WM8988
+	tristate
+
 config SND_SOC_WM8990
 	tristate
 
+config SND_SOC_WM9081
+	tristate
+
 config SND_SOC_WM9705
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index f2653803ede..8b7530546f4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -6,7 +6,9 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -23,8 +25,12 @@ snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8940-objs := wm8940.o
+snd-soc-wm8960-objs := wm8960.o
 snd-soc-wm8971-objs := wm8971.o
+snd-soc-wm8988-objs := wm8988.o
 snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9081-objs := wm9081.o
 snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
@@ -37,7 +43,9 @@ obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
@@ -55,7 +63,11 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8940)	+= snd-soc-wm8940.o
+obj-$(CONFIG_SND_SOC_WM8960)	+= snd-soc-wm8960.o
+obj-$(CONFIG_SND_SOC_WM8988)	+= snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM9081)	+= snd-soc-wm9081.o
 obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index b0d4af145b8..932299bb5d1 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -53,13 +53,13 @@ struct snd_soc_dai ac97_dai = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.capture = {
 		.stream_name = "AC97 Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ac97_dai);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index ddb3b08ac23..d7440a982d2 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -137,13 +137,13 @@ struct snd_soc_dai ad1980_dai = {
 		.channels_min = 2,
 		.channels_max = 6,
 		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+		.formats = SND_SOC_STD_AC97_FMTS, },
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+		.formats = SND_SOC_STD_AC97_FMTS, },
 };
 EXPORT_SYMBOL_GPL(ad1980_dai);
 
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 7fa09a38762..a32b8226c8a 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -18,7 +18,7 @@
  * - The machine driver's 'startup' function must call
  *   cs4270_set_dai_sysclk() with the value of MCLK.
  * - Only I2S and left-justified modes are supported
- * - Power management is not supported
+ * - Power management is supported
  */
 
 #include <linux/module.h>
@@ -27,6 +27,7 @@
 #include <sound/soc.h>
 #include <sound/initval.h>
 #include <linux/i2c.h>
+#include <linux/delay.h>
 
 #include "cs4270.h"
 
@@ -56,6 +57,7 @@
 #define CS4270_FIRSTREG	0x01
 #define CS4270_LASTREG	0x08
 #define CS4270_NUMREGS	(CS4270_LASTREG - CS4270_FIRSTREG + 1)
+#define CS4270_I2C_INCR	0x80
 
 /* Bit masks for the CS4270 registers */
 #define CS4270_CHIPID_ID	0xF0
@@ -64,6 +66,8 @@
 #define CS4270_PWRCTL_PDN_ADC	0x20
 #define CS4270_PWRCTL_PDN_DAC	0x02
 #define CS4270_PWRCTL_PDN	0x01
+#define CS4270_PWRCTL_PDN_ALL	\
+	(CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN)
 #define CS4270_MODE_SPEED_MASK	0x30
 #define CS4270_MODE_1X		0x00
 #define CS4270_MODE_2X		0x10
@@ -109,6 +113,7 @@ struct cs4270_private {
 	unsigned int mclk; /* Input frequency of the MCLK pin */
 	unsigned int mode; /* The mode (I2S or left-justified) */
 	unsigned int slave_mode;
+	unsigned int manual_mute;
 };
 
 /**
@@ -295,7 +300,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 	s32 length;
 
 	length = i2c_smbus_read_i2c_block_data(i2c_client,
-		CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
+		CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
 
 	if (length != CS4270_NUMREGS) {
 		dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
@@ -453,7 +458,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 }
 
 /**
- * cs4270_mute - enable/disable the CS4270 external mute
+ * cs4270_dai_mute - enable/disable the CS4270 external mute
  * @dai: the SOC DAI
  * @mute: 0 = disable mute, 1 = enable mute
  *
@@ -462,21 +467,52 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
  * board does not have the MUTEA or MUTEB pins connected to such circuitry,
  * then this function will do nothing.
  */
-static int cs4270_mute(struct snd_soc_dai *dai, int mute)
+static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct cs4270_private *cs4270 = codec->private_data;
 	int reg6;
 
 	reg6 = snd_soc_read(codec, CS4270_MUTE);
 
 	if (mute)
 		reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
-	else
+	else {
 		reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+		reg6 |= cs4270->manual_mute;
+	}
 
 	return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
 
+/**
+ * cs4270_soc_put_mute - put callback for the 'Master Playback switch'
+ * 			 alsa control.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * This function basically passes the arguments on to the generic
+ * snd_soc_put_volsw() function and saves the mute information in
+ * our private data structure. This is because we want to prevent
+ * cs4270_dai_mute() neglecting the user's decision to manually
+ * mute the codec's output.
+ *
+ * Returns 0 for success.
+ */
+static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs4270_private *cs4270 = codec->private_data;
+	int left = !ucontrol->value.integer.value[0];
+	int right = !ucontrol->value.integer.value[1];
+
+	cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) |
+			      (right ? CS4270_MUTE_DAC_B : 0);
+
+	return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_DOUBLE_R("Master Playback Volume",
@@ -486,7 +522,9 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
 	SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
 	SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
-	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
+	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1),
+	SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1,
+		snd_soc_get_volsw, cs4270_soc_put_mute),
 };
 
 /*
@@ -506,7 +544,7 @@ static struct snd_soc_dai_ops cs4270_dai_ops = {
 	.hw_params	= cs4270_hw_params,
 	.set_sysclk	= cs4270_set_dai_sysclk,
 	.set_fmt	= cs4270_set_dai_fmt,
-	.digital_mute	= cs4270_mute,
+	.digital_mute	= cs4270_dai_mute,
 };
 
 struct snd_soc_dai cs4270_dai = {
@@ -753,6 +791,57 @@ static struct i2c_device_id cs4270_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, cs4270_id);
 
+#ifdef CONFIG_PM
+
+/* This suspend/resume implementation can handle both - a simple standby
+ * where the codec remains powered, and a full suspend, where the voltage
+ * domain the codec is connected to is teared down and/or any other hardware
+ * reset condition is asserted.
+ *
+ * The codec's own power saving features are enabled in the suspend callback,
+ * and all registers are written back to the hardware when resuming.
+ */
+
+static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg)
+{
+	struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+	struct snd_soc_codec *codec = &cs4270->codec;
+	int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+
+	return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+
+static int cs4270_i2c_resume(struct i2c_client *client)
+{
+	struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+	struct snd_soc_codec *codec = &cs4270->codec;
+	int reg;
+
+	/* In case the device was put to hard reset during sleep, we need to
+	 * wait 500ns here before any I2C communication. */
+	ndelay(500);
+
+	/* first restore the entire register cache ... */
+	for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
+		u8 val = snd_soc_read(codec, reg);
+
+		if (i2c_smbus_write_byte_data(client, reg, val)) {
+			dev_err(codec->dev, "i2c write failed\n");
+			return -EIO;
+		}
+	}
+
+	/* ... then disable the power-down bits */
+	reg = snd_soc_read(codec, CS4270_PWRCTL);
+	reg &= ~CS4270_PWRCTL_PDN_ALL;
+
+	return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+#else
+#define cs4270_i2c_suspend	NULL
+#define cs4270_i2c_resume	NULL
+#endif /* CONFIG_PM */
+
 /*
  * cs4270_i2c_driver - I2C device identification
  *
@@ -767,6 +856,8 @@ static struct i2c_driver cs4270_i2c_driver = {
 	.id_table = cs4270_id,
 	.probe = cs4270_i2c_probe,
 	.remove = cs4270_i2c_remove,
+	.suspend = cs4270_i2c_suspend,
+	.resume = cs4270_i2c_resume,
 };
 
 /*
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
new file mode 100644
index 00000000000..218b33adad9
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -0,0 +1,71 @@
+/*
+ * ALSA SoC SPDIF DIT driver
+ *
+ *  This driver is used by controllers which can operate in DIT (SPDI/F) where
+ *  no codec is needed.  This file provides stub codec that can be used
+ *  in these configurations. TI DaVinci Audio controller uses this driver.
+ *
+ * Author:      Steve Chen,  <schen@mvista.com>
+ * Copyright:   (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright:   (C) 2009  Texas Instruments, India
+ *
+ * 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 <sound/soc.h>
+#include <sound/pcm.h>
+
+#include "spdif_transciever.h"
+
+#define STUB_RATES	SNDRV_PCM_RATE_8000_96000
+#define STUB_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+
+struct snd_soc_dai dit_stub_dai = {
+	.name		= "DIT",
+	.playback 	= {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates		= STUB_RATES,
+		.formats	= STUB_FORMATS,
+	},
+};
+
+static int spdif_dit_probe(struct platform_device *pdev)
+{
+	dit_stub_dai.dev = &pdev->dev;
+	return snd_soc_register_dai(&dit_stub_dai);
+}
+
+static int spdif_dit_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&dit_stub_dai);
+	return 0;
+}
+
+static struct platform_driver spdif_dit_driver = {
+	.probe		= spdif_dit_probe,
+	.remove		= spdif_dit_remove,
+	.driver		= {
+		.name	= "spdif-dit",
+		.owner	= THIS_MODULE,
+	},
+};
+
+static int __init dit_modinit(void)
+{
+	return platform_driver_register(&spdif_dit_driver);
+}
+
+static void __exit dit_exit(void)
+{
+	platform_driver_unregister(&spdif_dit_driver);
+}
+
+module_init(dit_modinit);
+module_exit(dit_exit);
+
diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h
new file mode 100644
index 00000000000..296f2eb6c4e
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.h
@@ -0,0 +1,17 @@
+/*
+ * ALSA SoC DIT/DIR driver header
+ *
+ * Author:      Steve Chen,  <schen@mvista.com>
+ * Copyright:   (C) 2008 MontaVista Software, Inc., <source@mvista.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 CODEC_STUBS_H
+#define CODEC_STUBS_H
+
+extern struct snd_soc_dai dit_stub_dai;
+
+#endif /* CODEC_STUBS_H */
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 87f606c7682..1fc4c8e0899 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -336,15 +336,17 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 			master_runtime->sample_bits,
 			master_runtime->rate);
 
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_RATE,
-					     master_runtime->rate,
-					     master_runtime->rate);
-
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     master_runtime->sample_bits,
-					     master_runtime->sample_bits);
+		if (master_runtime->rate != 0)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+						     SNDRV_PCM_HW_PARAM_RATE,
+						     master_runtime->rate,
+						     master_runtime->rate);
+
+		if (master_runtime->sample_bits != 0)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+						     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+						     master_runtime->sample_bits,
+						     master_runtime->sample_bits);
 
 		ssm2602->slave_substream = substream;
 	} else
@@ -372,6 +374,11 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
+
+	if (ssm2602->master_substream == substream)
+		ssm2602->master_substream = ssm2602->slave_substream;
+
+	ssm2602->slave_substream = NULL;
 	/* deactivate */
 	if (!codec->active)
 		ssm2602_write(codec, SSM2602_ACTIVE, 0);
@@ -497,11 +504,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
-		SNDRV_PCM_RATE_96000)
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\
+		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+		SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
new file mode 100644
index 00000000000..8ad4b7b3e3b
--- /dev/null
+++ b/sound/soc/codecs/stac9766.c
@@ -0,0 +1,463 @@
+/*
+ * stac9766.c  --  ALSA SoC STAC9766 codec support
+ *
+ * Copyright 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Features:-
+ *
+ *   o Support for AC97 Codec, S/PDIF
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-of-simple.h>
+
+#include "stac9766.h"
+
+#define STAC9766_VERSION "0.10"
+
+/*
+ * STAC9766 register cache
+ */
+static const u16 stac9766_reg[] = {
+	0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
+	0x0000, 0x0000, 0x8008, 0x8008, /* e */
+	0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
+	0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+	0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+	0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
+	0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
+	0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+	0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
+	0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+	0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
+};
+
+static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
+			"Line", "Stereo Mix", "Mono Mix", "Phone"};
+static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
+static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
+static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
+static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
+static const char *stac9766_record_all_mux[] = {"All analog",
+	"Analog plus DAC"};
+static const char *stac9766_boost1[] = {"0dB", "10dB"};
+static const char *stac9766_boost2[] = {"0dB", "20dB"};
+static const char *stac9766_stereo_mic[] = {"Off", "On"};
+
+static const struct soc_enum stac9766_record_enum =
+	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
+static const struct soc_enum stac9766_mono_enum =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
+static const struct soc_enum stac9766_mic_enum =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
+static const struct soc_enum stac9766_SPDIF_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
+static const struct soc_enum stac9766_popbypass_enum =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
+static const struct soc_enum stac9766_record_all_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
+			stac9766_record_all_mux);
+static const struct soc_enum stac9766_boost1_enum =
+	SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
+static const struct soc_enum stac9766_boost2_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
+static const struct soc_enum stac9766_stereo_mic_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+
+static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
+static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
+static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
+static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
+
+static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
+	SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
+	SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
+		       master_tlv),
+	SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
+	SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
+		       master_tlv),
+	SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+	SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
+	SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
+
+
+	SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
+	SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
+	SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
+	SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
+
+	SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
+	SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
+	SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+	SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
+
+	SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
+	SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
+	SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
+	SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
+
+	SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
+	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
+	SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
+	SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+
+	SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
+	SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
+	SOC_ENUM("Record All Mux", stac9766_record_all_enum),
+	SOC_ENUM("Record Mux", stac9766_record_enum),
+	SOC_ENUM("Mono Mux", stac9766_mono_enum),
+	SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
+};
+
+static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+			       unsigned int val)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg > AC97_STAC_PAGE0) {
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+		soc_ac97_ops.write(codec->ac97, reg, val);
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+		return 0;
+	}
+	if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+		return -EIO;
+
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	cache[reg / 2] = val;
+	return 0;
+}
+
+static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
+				       unsigned int reg)
+{
+	u16 val = 0, *cache = codec->reg_cache;
+
+	if (reg > AC97_STAC_PAGE0) {
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+		val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+		return val;
+	}
+	if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+		return -EIO;
+
+	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+		reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
+		reg == AC97_VENDOR_ID2) {
+
+		val = soc_ac97_ops.read(codec->ac97, reg);
+		return val;
+	}
+	return cache[reg / 2];
+}
+
+static int ac97_analog_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned short reg, vra;
+
+	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+
+	vra |= 0x1; /* enable variable rate audio */
+
+	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = AC97_PCM_FRONT_DAC_RATE;
+	else
+		reg = AC97_PCM_LR_ADC_RATE;
+
+	return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned short reg, vra;
+
+	stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
+
+	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+	vra |= 0x5; /* Enable VRA and SPDIF out */
+
+	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+	reg = AC97_PCM_FRONT_DAC_RATE;
+
+	return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_trigger(struct snd_pcm_substream *substream,
+				int cmd, struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned short vra;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
+		vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+		vra &= !0x04;
+		stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+		break;
+	}
+	return 0;
+}
+
+static int stac9766_set_bias_level(struct snd_soc_codec *codec,
+				   enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON: /* full On */
+	case SND_SOC_BIAS_PREPARE: /* partial On */
+	case SND_SOC_BIAS_STANDBY: /* Off, with power */
+		stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
+		break;
+	case SND_SOC_BIAS_OFF: /* Off, without power */
+		/* disable everything including AC link */
+		stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
+{
+	if (try_warm && soc_ac97_ops.warm_reset) {
+		soc_ac97_ops.warm_reset(codec->ac97);
+		if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
+			return 1;
+	}
+
+	soc_ac97_ops.reset(codec->ac97);
+	if (soc_ac97_ops.warm_reset)
+		soc_ac97_ops.warm_reset(codec->ac97);
+	if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
+		return -EIO;
+	return 0;
+}
+
+static int stac9766_codec_suspend(struct platform_device *pdev,
+				  pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int stac9766_codec_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u16 id, reset;
+
+	reset = 0;
+	/* give the codec an AC97 warm reset to start the link */
+reset:
+	if (reset > 5) {
+		printk(KERN_ERR "stac9766 failed to resume");
+		return -EIO;
+	}
+	codec->ac97->bus->ops->warm_reset(codec->ac97);
+	id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
+	if (id != 0x4c13) {
+		stac9766_reset(codec, 0);
+		reset++;
+		goto reset;
+	}
+	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
+	.prepare = ac97_analog_prepare,
+};
+
+static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
+	.prepare = ac97_digital_prepare,
+	.trigger = ac97_digital_trigger,
+};
+
+struct snd_soc_dai stac9766_dai[] = {
+{
+	.name = "stac9766 analog",
+	.id = 0,
+	.ac97_control = 1,
+
+	/* stream cababilities */
+	.playback = {
+		.stream_name = "stac9766 analog",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SND_SOC_STD_AC97_FMTS,
+	},
+	.capture = {
+		.stream_name = "stac9766 analog",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SND_SOC_STD_AC97_FMTS,
+	},
+	/* alsa ops */
+	.ops = &stac9766_dai_ops_analog,
+},
+{
+	.name = "stac9766 IEC958",
+	.id = 1,
+	.ac97_control = 1,
+
+	/* stream cababilities */
+	.playback = {
+		.stream_name = "stac9766 IEC958",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_32000 | \
+			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
+	},
+	/* alsa ops */
+	.ops = &stac9766_dai_ops_digital,
+}
+};
+EXPORT_SYMBOL_GPL(stac9766_dai);
+
+static int stac9766_codec_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
+
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->card->codec == NULL)
+		return -ENOMEM;
+	codec = socdev->card->codec;
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto cache_err;
+	}
+	codec->reg_cache_size = sizeof(stac9766_reg);
+	codec->reg_cache_step = 2;
+
+	codec->name = "STAC9766";
+	codec->owner = THIS_MODULE;
+	codec->dai = stac9766_dai;
+	codec->num_dai = ARRAY_SIZE(stac9766_dai);
+	codec->write = stac9766_ac97_write;
+	codec->read = stac9766_ac97_read;
+	codec->set_bias_level = stac9766_set_bias_level;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+	if (ret < 0)
+		goto codec_err;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0)
+		goto pcm_err;
+
+	/* do a cold reset for the controller and then try
+	 * a warm reset followed by an optional cold reset for codec */
+	stac9766_reset(codec, 0);
+	ret = stac9766_reset(codec, 1);
+	if (ret < 0) {
+		printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
+		goto reset_err;
+	}
+
+	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
+			     ARRAY_SIZE(stac9766_snd_ac97_controls));
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0)
+		goto reset_err;
+	return 0;
+
+reset_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	snd_soc_free_ac97_codec(codec);
+codec_err:
+	kfree(codec->private_data);
+cache_err:
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
+	return ret;
+}
+
+static int stac9766_codec_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	if (codec == NULL)
+		return 0;
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_free_ac97_codec(codec);
+	kfree(codec->reg_cache);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_stac9766 = {
+	.probe = stac9766_codec_probe,
+	.remove = stac9766_codec_remove,
+	.suspend = stac9766_codec_suspend,
+	.resume = stac9766_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
+
+MODULE_DESCRIPTION("ASoC stac9766 driver");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h
new file mode 100644
index 00000000000..65642eb8393
--- /dev/null
+++ b/sound/soc/codecs/stac9766.h
@@ -0,0 +1,21 @@
+/*
+ * stac9766.h  --  STAC9766 Soc Audio driver
+ */
+
+#ifndef _STAC9766_H
+#define _STAC9766_H
+
+#define AC97_STAC_PAGE0 0x1000
+#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
+#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
+#define AC97_STAC_STEREO_MIC 0x78
+
+/* STAC9766 DAI ID's */
+#define STAC9766_DAI_AC97_ANALOG		0
+#define STAC9766_DAI_AC97_DIGITAL		1
+
+extern struct snd_soc_dai stac9766_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_stac9766;
+
+
+#endif
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index c3f4afb5d01..0b8dcb5cd72 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -86,7 +86,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
 	 */
 
 	if ((reg < 0 || reg > 9) && (reg != 15)) {
-		printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+		printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
 		return -1;
 	}
 
@@ -98,7 +98,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (codec->hw_write(codec->control_data, data, 2) == 2)
 		return 0;
 
-	printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+	printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__,
 	       value, reg);
 
 	return -EIO;
@@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = {
  * Every divisor is a factor of 11*12
  */
 #define SR_MULT (11*12)
-#define A(x) (x) ? (SR_MULT/x) : 0
+#define A(x) (SR_MULT/x)
 static const unsigned char sr_adc_mult_table[] = {
-	A(2), A(2), A(12), A(12),  A(0), A(0), A(3), A(1),
-	A(2), A(2), A(11), A(11),  A(0), A(0), A(0), A(1)
+	A(2), A(2), A(12), A(12),  0, 0, A(3), A(1),
+	A(2), A(2), A(11), A(11),  0, 0, 0, A(1)
 };
 static const unsigned char sr_dac_mult_table[] = {
-	A(2), A(12), A(2), A(12),  A(0), A(0), A(3), A(1),
-	A(2), A(11), A(2), A(11),  A(0), A(0), A(0), A(1)
+	A(2), A(12), A(2), A(12),  0, 0, A(3), A(1),
+	A(2), A(11), A(2), A(11),  0, 0, 0, A(1)
 };
 
 static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
@@ -523,6 +523,8 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	case SND_SOC_DAIFMT_I2S:
 		iface_reg |= TLV320AIC23_FOR_I2S;
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface_reg |= TLV320AIC23_LRP_ON;
 	case SND_SOC_DAIFMT_DSP_B:
 		iface_reg |= TLV320AIC23_FOR_DSP;
 		break;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index df7c8c281d2..4dbb853eef5 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -115,6 +115,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* REG_VIBRA_PWM_SET	(0x47)	*/
 	0x00, /* REG_ANAMIC_GAIN	(0x48)	*/
 	0x00, /* REG_MISC_SET_2		(0x49)	*/
+	0x00, /* REG_SW_SHADOW		(0x4A)	- Shadow, non HW register */
 };
 
 /* codec private data */
@@ -125,6 +126,17 @@ struct twl4030_priv {
 
 	struct snd_pcm_substream *master_substream;
 	struct snd_pcm_substream *slave_substream;
+
+	unsigned int configured;
+	unsigned int rate;
+	unsigned int sample_bits;
+	unsigned int channels;
+
+	unsigned int sysclk;
+
+	/* Headset output state handling */
+	unsigned int hsl_enabled;
+	unsigned int hsr_enabled;
 };
 
 /*
@@ -161,7 +173,11 @@ static int twl4030_write(struct snd_soc_codec *codec,
 			unsigned int reg, unsigned int value)
 {
 	twl4030_write_reg_cache(codec, reg, value);
-	return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+	if (likely(reg < TWL4030_REG_SW_SHADOW))
+		return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
+					    reg);
+	else
+		return 0;
 }
 
 static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
@@ -188,6 +204,7 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 
 static void twl4030_init_chip(struct snd_soc_codec *codec)
 {
+	u8 *cache = codec->reg_cache;
 	int i;
 
 	/* clear CODECPDZ prior to setting register defaults */
@@ -195,7 +212,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 	/* set all audio section registers to reasonable defaults */
 	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
-		twl4030_write(codec, i,	twl4030_reg[i]);
+		twl4030_write(codec, i,	cache[i]);
 
 }
 
@@ -232,7 +249,7 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute)
 					TWL4030_REG_PRECKL_CTL);
 		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
 		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
-					reg_val & (~TWL4030_PRECKL_GAIN),
+					reg_val & (~TWL4030_PRECKR_GAIN),
 					TWL4030_REG_PRECKR_CTL);
 
 		/* Disable PLL */
@@ -316,104 +333,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec)
 }
 
 /* Earpiece */
-static const char *twl4030_earpiece_texts[] =
-		{"Off", "DACL1", "DACL2", "DACR1"};
-
-static const unsigned int twl4030_earpiece_values[] =
-		{0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_earpiece_enum =
-	SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7,
-			ARRAY_SIZE(twl4030_earpiece_texts),
-			twl4030_earpiece_texts,
-			twl4030_earpiece_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum);
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = {
+	SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0),
+	SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0),
+	SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0),
+};
 
 /* PreDrive Left */
-static const char *twl4030_predrivel_texts[] =
-		{"Off", "DACL1", "DACL2", "DACR2"};
-
-static const unsigned int twl4030_predrivel_values[] =
-		{0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predrivel_enum =
-	SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7,
-			ARRAY_SIZE(twl4030_predrivel_texts),
-			twl4030_predrivel_texts,
-			twl4030_predrivel_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = {
+	SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0),
+	SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0),
+	SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0),
+};
 
 /* PreDrive Right */
-static const char *twl4030_predriver_texts[] =
-		{"Off", "DACR1", "DACR2", "DACL2"};
-
-static const unsigned int twl4030_predriver_values[] =
-		{0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predriver_enum =
-	SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7,
-			ARRAY_SIZE(twl4030_predriver_texts),
-			twl4030_predriver_texts,
-			twl4030_predriver_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = {
+	SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0),
+	SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0),
+	SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0),
+};
 
 /* Headset Left */
-static const char *twl4030_hsol_texts[] =
-		{"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_hsol_enum =
-	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
-			ARRAY_SIZE(twl4030_hsol_texts),
-			twl4030_hsol_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
-SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = {
+	SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0),
+	SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0),
+};
 
 /* Headset Right */
-static const char *twl4030_hsor_texts[] =
-		{"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_hsor_enum =
-	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
-			ARRAY_SIZE(twl4030_hsor_texts),
-			twl4030_hsor_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
-SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = {
+	SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0),
+	SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0),
+	SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0),
+};
 
 /* Carkit Left */
-static const char *twl4030_carkitl_texts[] =
-		{"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_carkitl_enum =
-	SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
-			ARRAY_SIZE(twl4030_carkitl_texts),
-			twl4030_carkitl_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = {
+	SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0),
+	SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0),
+};
 
 /* Carkit Right */
-static const char *twl4030_carkitr_texts[] =
-		{"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_carkitr_enum =
-	SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
-			ARRAY_SIZE(twl4030_carkitr_texts),
-			twl4030_carkitr_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = {
+	SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0),
+	SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0),
+};
 
 /* Handsfree Left */
 static const char *twl4030_handsfreel_texts[] =
-		{"Voice", "DACL1", "DACL2", "DACR2"};
+		{"Voice", "AudioL1", "AudioL2", "AudioR2"};
 
 static const struct soc_enum twl4030_handsfreel_enum =
 	SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
@@ -423,9 +396,13 @@ static const struct soc_enum twl4030_handsfreel_enum =
 static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
 SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
 
+/* Handsfree Left virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0);
+
 /* Handsfree Right */
 static const char *twl4030_handsfreer_texts[] =
-		{"Voice", "DACR1", "DACR2", "DACL2"};
+		{"Voice", "AudioR1", "AudioR2", "AudioL2"};
 
 static const struct soc_enum twl4030_handsfreer_enum =
 	SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
@@ -435,37 +412,48 @@ static const struct soc_enum twl4030_handsfreer_enum =
 static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
 SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
 
-/* Left analog microphone selection */
-static const char *twl4030_analoglmic_texts[] =
-		{"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
+/* Handsfree Right virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0);
 
-static const unsigned int twl4030_analoglmic_values[] =
-		{0x0, 0x1, 0x2, 0x4, 0x8};
+/* Vibra */
+/* Vibra audio path selection */
+static const char *twl4030_vibra_texts[] =
+		{"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
 
-static const struct soc_enum twl4030_analoglmic_enum =
-	SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
-			ARRAY_SIZE(twl4030_analoglmic_texts),
-			twl4030_analoglmic_texts,
-			twl4030_analoglmic_values);
+static const struct soc_enum twl4030_vibra_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
+			ARRAY_SIZE(twl4030_vibra_texts),
+			twl4030_vibra_texts);
 
-static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
+static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
+SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
 
-/* Right analog microphone selection */
-static const char *twl4030_analogrmic_texts[] =
-		{"Off", "Sub mic", "AUXR"};
+/* Vibra path selection: local vibrator (PWM) or audio driven */
+static const char *twl4030_vibrapath_texts[] =
+		{"Local vibrator", "Audio"};
 
-static const unsigned int twl4030_analogrmic_values[] =
-		{0x0, 0x1, 0x4};
+static const struct soc_enum twl4030_vibrapath_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
+			ARRAY_SIZE(twl4030_vibrapath_texts),
+			twl4030_vibrapath_texts);
 
-static const struct soc_enum twl4030_analogrmic_enum =
-	SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
-			ARRAY_SIZE(twl4030_analogrmic_texts),
-			twl4030_analogrmic_texts,
-			twl4030_analogrmic_values);
+static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
+SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
 
-static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
+/* Left analog microphone selection */
+static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
+	SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
+};
+
+/* Right analog microphone selection */
+static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
+	SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
+	SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0),
+};
 
 /* TX1 L/R Analog/Digital microphone selection */
 static const char *twl4030_micpathtx1_texts[] =
@@ -507,6 +495,10 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control =
 static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
 	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
 
+/* Analog bypass for Voice */
+static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
+
 /* Digital bypass gain, 0 mutes the bypass */
 static const unsigned int twl4030_dapm_dbypass_tlv[] = {
 	TLV_DB_RANGE_HEAD(2),
@@ -526,6 +518,18 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control =
 			TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
 			twl4030_dapm_dbypass_tlv);
 
+/*
+ * Voice Sidetone GAIN volume control:
+ * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB)
+ */
+static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1);
+
+/* Digital bypass voice: sidetone (VUL -> VDL)*/
+static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
+	SOC_DAPM_SINGLE_TLV("Volume",
+			TWL4030_REG_VSTPGA, 0, 0x29, 0,
+			twl4030_dapm_dbypassv_tlv);
+
 static int micpath_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -556,63 +560,143 @@ static int micpath_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int handsfree_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
 {
-	struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
 	unsigned char hs_ctl;
 
-	hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+	hs_ctl = twl4030_read_reg_cache(codec, reg);
 
-	if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+	if (ramp) {
+		/* HF ramp-up */
+		hs_ctl |= TWL4030_HF_CTL_REF_EN;
+		twl4030_write(codec, reg, hs_ctl);
+		udelay(10);
 		hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
-		twl4030_write(w->codec, e->reg, hs_ctl);
+		twl4030_write(codec, reg, hs_ctl);
+		udelay(40);
 		hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
-		twl4030_write(w->codec, e->reg, hs_ctl);
 		hs_ctl |= TWL4030_HF_CTL_HB_EN;
-		twl4030_write(w->codec, e->reg, hs_ctl);
+		twl4030_write(codec, reg, hs_ctl);
 	} else {
-		hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
-				| TWL4030_HF_CTL_HB_EN);
-		twl4030_write(w->codec, e->reg, hs_ctl);
+		/* HF ramp-down */
+		hs_ctl &= ~TWL4030_HF_CTL_LOOP_EN;
+		hs_ctl &= ~TWL4030_HF_CTL_HB_EN;
+		twl4030_write(codec, reg, hs_ctl);
+		hs_ctl &= ~TWL4030_HF_CTL_RAMP_EN;
+		twl4030_write(codec, reg, hs_ctl);
+		udelay(40);
+		hs_ctl &= ~TWL4030_HF_CTL_REF_EN;
+		twl4030_write(codec, reg, hs_ctl);
 	}
+}
 
+static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0);
+		break;
+	}
 	return 0;
 }
 
-static int headsetl_event(struct snd_soc_dapm_widget *w,
+static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
 		struct snd_kcontrol *kcontrol, int event)
 {
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0);
+		break;
+	}
+	return 0;
+}
+
+static void headset_ramp(struct snd_soc_codec *codec, int ramp)
+{
 	unsigned char hs_gain, hs_pop;
+	struct twl4030_priv *twl4030 = codec->private_data;
+	/* Base values for ramp delay calculation: 2^19 - 2^26 */
+	unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304,
+				    8388608, 16777216, 33554432, 67108864};
 
-	/* Save the current volume */
-	hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
-	hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
+	hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET);
+	hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
 
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		/* Do the anti-pop/bias ramp enable according to the TRM */
+	if (ramp) {
+		/* Headset ramp-up according to the TRM */
 		hs_pop |= TWL4030_VMID_EN;
-		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-		/* Is this needed? Can we just use whatever gain here? */
-		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET,
-				(hs_gain & (~0x0f)) | 0x0a);
+		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
 		hs_pop |= TWL4030_RAMP_EN;
-		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-
-		/* Restore the original volume */
-		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
-		break;
-	case SND_SOC_DAPM_POST_PMD:
-		/* Do the anti-pop/bias ramp disable according to the TRM */
+		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+	} else {
+		/* Headset ramp-down _not_ according to
+		 * the TRM, but in a way that it is working */
 		hs_pop &= ~TWL4030_RAMP_EN;
-		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		/* Wait ramp delay time + 1, so the VMID can settle */
+		mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
+			twl4030->sysclk) + 1);
 		/* Bypass the reg_cache to mute the headset */
 		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
 					hs_gain & (~0x0f),
 					TWL4030_REG_HS_GAIN_SET);
+
 		hs_pop &= ~TWL4030_VMID_EN;
-		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+	}
+}
+
+static int headsetlpga_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct twl4030_priv *twl4030 = w->codec->private_data;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* Do the ramp-up only once */
+		if (!twl4030->hsr_enabled)
+			headset_ramp(w->codec, 1);
+
+		twl4030->hsl_enabled = 1;
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* Do the ramp-down only if both headsetL/R is disabled */
+		if (!twl4030->hsr_enabled)
+			headset_ramp(w->codec, 0);
+
+		twl4030->hsl_enabled = 0;
+		break;
+	}
+	return 0;
+}
+
+static int headsetrpga_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct twl4030_priv *twl4030 = w->codec->private_data;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* Do the ramp-up only once */
+		if (!twl4030->hsl_enabled)
+			headset_ramp(w->codec, 1);
+
+		twl4030->hsr_enabled = 1;
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* Do the ramp-down only if both headsetL/R is disabled */
+		if (!twl4030->hsl_enabled)
+			headset_ramp(w->codec, 0);
+
+		twl4030->hsr_enabled = 0;
 		break;
 	}
 	return 0;
@@ -624,7 +708,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
 	struct soc_mixer_control *m =
 		(struct soc_mixer_control *)w->kcontrols->private_value;
 	struct twl4030_priv *twl4030 = w->codec->private_data;
-	unsigned char reg;
+	unsigned char reg, misc;
 
 	reg = twl4030_read_reg_cache(w->codec, m->reg);
 
@@ -636,14 +720,34 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
 		else
 			twl4030->bypass_state &=
 				~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+	} else if (m->reg == TWL4030_REG_VDL_APGA_CTL) {
+		/* Analog voice bypass */
+		if (reg & (1 << m->shift))
+			twl4030->bypass_state |= (1 << 4);
+		else
+			twl4030->bypass_state &= ~(1 << 4);
+	} else if (m->reg == TWL4030_REG_VSTPGA) {
+		/* Voice digital bypass */
+		if (reg)
+			twl4030->bypass_state |= (1 << 5);
+		else
+			twl4030->bypass_state &= ~(1 << 5);
 	} else {
 		/* Digital bypass */
 		if (reg & (0x7 << m->shift))
-			twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+			twl4030->bypass_state |= (1 << (m->shift ? 7 : 6));
 		else
-			twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+			twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6));
 	}
 
+	/* Enable master analog loopback mode if any analog switch is enabled*/
+	misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1);
+	if (twl4030->bypass_state & 0x1F)
+		misc |= TWL4030_FMLOOP_EN;
+	else
+		misc &= ~TWL4030_FMLOOP_EN;
+	twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc);
+
 	if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
 		if (twl4030->bypass_state)
 			twl4030_codec_mute(w->codec, 0);
@@ -810,6 +914,48 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
 	return err;
 }
 
+/* Codec operation modes */
+static const char *twl4030_op_modes_texts[] = {
+	"Option 2 (voice/audio)", "Option 1 (audio)"
+};
+
+static const struct soc_enum twl4030_op_modes_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0,
+			ARRAY_SIZE(twl4030_op_modes_texts),
+			twl4030_op_modes_texts);
+
+int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct twl4030_priv *twl4030 = codec->private_data;
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned short val;
+	unsigned short mask, bitmask;
+
+	if (twl4030->configured) {
+		printk(KERN_ERR "twl4030 operation mode cannot be "
+			"changed on-the-fly\n");
+		return -EBUSY;
+	}
+
+	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+		;
+	if (ucontrol->value.enumerated.item[0] > e->max - 1)
+		return -EINVAL;
+
+	val = ucontrol->value.enumerated.item[0] << e->shift_l;
+	mask = (bitmask - 1) << e->shift_l;
+	if (e->shift_l != e->shift_r) {
+		if (ucontrol->value.enumerated.item[1] > e->max - 1)
+			return -EINVAL;
+		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+		mask |= (bitmask - 1) << e->shift_r;
+	}
+
+	return snd_soc_update_bits(codec, e->reg, mask, val);
+}
+
 /*
  * FGAIN volume control:
  * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
@@ -824,6 +970,12 @@ static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1);
 static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
 
 /*
+ * Voice Downlink GAIN volume control:
+ * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1);
+
+/*
  * Analog playback gain
  * -24 dB to 12 dB in 2 dB steps
  */
@@ -864,7 +1016,32 @@ static const struct soc_enum twl4030_rampdelay_enum =
 			ARRAY_SIZE(twl4030_rampdelay_texts),
 			twl4030_rampdelay_texts);
 
+/* Vibra H-bridge direction mode */
+static const char *twl4030_vibradirmode_texts[] = {
+	"Vibra H-bridge direction", "Audio data MSB",
+};
+
+static const struct soc_enum twl4030_vibradirmode_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
+			ARRAY_SIZE(twl4030_vibradirmode_texts),
+			twl4030_vibradirmode_texts);
+
+/* Vibra H-bridge direction */
+static const char *twl4030_vibradir_texts[] = {
+	"Positive polarity", "Negative polarity",
+};
+
+static const struct soc_enum twl4030_vibradir_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
+			ARRAY_SIZE(twl4030_vibradir_texts),
+			twl4030_vibradir_texts);
+
 static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+	/* Codec operation mode control */
+	SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum,
+		snd_soc_get_enum_double,
+		snd_soc_put_twl4030_opmode_enum_double),
+
 	/* Common playback gain controls */
 	SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
 		TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
@@ -893,6 +1070,16 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
 		1, 1, 0),
 
+	/* Common voice downlink gain controls */
+	SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume",
+		TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv),
+
+	SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume",
+		TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv),
+
+	SOC_SINGLE("DAC Voice Analog Downlink Switch",
+		TWL4030_REG_VDL_APGA_CTL, 1, 1, 0),
+
 	/* Separate output gain controls */
 	SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
 		TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
@@ -920,6 +1107,9 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		0, 3, 5, 0, input_gain_tlv),
 
 	SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
+
+	SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
+	SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum),
 };
 
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
@@ -947,26 +1137,19 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("CARKITR"),
 	SND_SOC_DAPM_OUTPUT("HFL"),
 	SND_SOC_DAPM_OUTPUT("HFR"),
+	SND_SOC_DAPM_OUTPUT("VIBRA"),
 
 	/* DACs */
-	SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+	SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
 			SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+	SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback",
 			SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+	SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback",
 			SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+	SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback",
+			SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback",
 			SND_SOC_NOPM, 0, 0),
-
-	/* Analog PGAs */
-	SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
-			0, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
-			0, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
-			0, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
-			0, 0, NULL, 0),
 
 	/* Analog bypasses */
 	SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
@@ -981,6 +1164,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
 			&twl4030_dapm_abypassl2_control,
 			bypass_event, SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassv_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
 
 	/* Digital bypasses */
 	SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
@@ -989,43 +1175,88 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
 			&twl4030_dapm_dbypassr_control, bypass_event,
 			SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassv_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
 
-	SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-			0, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-			1, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-			2, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-			3, 0, NULL, 0),
-
-	/* Output MUX controls */
+	/* Digital mixers, power control for the physical DACs */
+	SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer",
+			TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer",
+			TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer",
+			TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer",
+			TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer",
+			TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0),
+
+	/* Analog mixers, power control for the physical PGAs */
+	SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer",
+			TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer",
+			TWL4030_REG_ARXL1_APGA_CTL, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer",
+			TWL4030_REG_ARXR2_APGA_CTL, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer",
+			TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer",
+			TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0),
+
+	/* Output MIXER controls */
 	/* Earpiece */
-	SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_earpiece_control),
+	SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_earpiece_controls[0],
+			ARRAY_SIZE(twl4030_dapm_earpiece_controls)),
 	/* PreDrivL/R */
-	SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_predrivel_control),
-	SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_predriver_control),
+	SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_predrivel_controls[0],
+			ARRAY_SIZE(twl4030_dapm_predrivel_controls)),
+	SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_predriver_controls[0],
+			ARRAY_SIZE(twl4030_dapm_predriver_controls)),
 	/* HeadsetL/R */
-	SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_hsol_control, headsetl_event,
-		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
-	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_hsor_control),
+	SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_hsol_controls[0],
+			ARRAY_SIZE(twl4030_dapm_hsol_controls)),
+	SND_SOC_DAPM_PGA_E("HeadsetL PGA", SND_SOC_NOPM,
+			0, 0, NULL, 0, headsetlpga_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_hsor_controls[0],
+			ARRAY_SIZE(twl4030_dapm_hsor_controls)),
+	SND_SOC_DAPM_PGA_E("HeadsetR PGA", SND_SOC_NOPM,
+			0, 0, NULL, 0, headsetrpga_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
 	/* CarkitL/R */
-	SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_carkitl_control),
-	SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_carkitr_control),
+	SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_carkitl_controls[0],
+			ARRAY_SIZE(twl4030_dapm_carkitl_controls)),
+	SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_carkitr_controls[0],
+			ARRAY_SIZE(twl4030_dapm_carkitr_controls)),
+
+	/* Output MUX controls */
 	/* HandsfreeL/R */
-	SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
-		&twl4030_dapm_handsfreel_control, handsfree_event,
-		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
-	SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
-		&twl4030_dapm_handsfreer_control, handsfree_event,
-		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_handsfreel_control),
+	SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_handsfreelmute_control),
+	SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM,
+			0, 0, NULL, 0, handsfreelpga_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0,
+		&twl4030_dapm_handsfreer_control),
+	SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_handsfreermute_control),
+	SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM,
+			0, 0, NULL, 0, handsfreerpga_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	/* Vibra */
+	SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0,
+		&twl4030_dapm_vibra_control),
+	SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_vibrapath_control),
 
 	/* Introducing four virtual ADC, since TWL4030 have four channel for
 	   capture */
@@ -1050,11 +1281,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
 		SND_SOC_DAPM_POST_REG),
 
-	/* Analog input muxes with switch for the capture amplifiers */
-	SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
-		TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
-	SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
-		TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
+	/* Analog input mixers for the capture amplifiers */
+	SND_SOC_DAPM_MIXER("Analog Left Capture Route",
+		TWL4030_REG_ANAMICL, 4, 0,
+		&twl4030_dapm_analoglmic_controls[0],
+		ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
+	SND_SOC_DAPM_MIXER("Analog Right Capture Route",
+		TWL4030_REG_ANAMICR, 4, 0,
+		&twl4030_dapm_analogrmic_controls[0],
+		ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
 
 	SND_SOC_DAPM_PGA("ADC Physical Left",
 		TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
@@ -1073,62 +1308,86 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
-	{"Analog L1 Playback Mixer", NULL, "DAC Left1"},
-	{"Analog R1 Playback Mixer", NULL, "DAC Right1"},
-	{"Analog L2 Playback Mixer", NULL, "DAC Left2"},
-	{"Analog R2 Playback Mixer", NULL, "DAC Right2"},
-
-	{"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
-	{"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
-	{"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
-	{"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
+	{"Digital L1 Playback Mixer", NULL, "DAC Left1"},
+	{"Digital R1 Playback Mixer", NULL, "DAC Right1"},
+	{"Digital L2 Playback Mixer", NULL, "DAC Left2"},
+	{"Digital R2 Playback Mixer", NULL, "DAC Right2"},
+	{"Digital Voice Playback Mixer", NULL, "DAC Voice"},
+
+	{"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"},
+	{"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"},
+	{"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"},
+	{"Analog R2 Playback Mixer", NULL, "Digital R2 Playback Mixer"},
+	{"Analog Voice Playback Mixer", NULL, "Digital Voice Playback Mixer"},
 
 	/* Internal playback routings */
 	/* Earpiece */
-	{"Earpiece Mux", "DACL1", "ARXL1_APGA"},
-	{"Earpiece Mux", "DACL2", "ARXL2_APGA"},
-	{"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+	{"Earpiece Mixer", "Voice", "Analog Voice Playback Mixer"},
+	{"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+	{"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+	{"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"},
 	/* PreDrivL */
-	{"PredriveL Mux", "DACL1", "ARXL1_APGA"},
-	{"PredriveL Mux", "DACL2", "ARXL2_APGA"},
-	{"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+	{"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"},
+	{"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+	{"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+	{"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"},
 	/* PreDrivR */
-	{"PredriveR Mux", "DACR1", "ARXR1_APGA"},
-	{"PredriveR Mux", "DACR2", "ARXR2_APGA"},
-	{"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+	{"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"},
+	{"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+	{"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+	{"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"},
 	/* HeadsetL */
-	{"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
-	{"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+	{"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"},
+	{"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+	{"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+	{"HeadsetL PGA", NULL, "HeadsetL Mixer"},
 	/* HeadsetR */
-	{"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
-	{"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+	{"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"},
+	{"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+	{"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+	{"HeadsetR PGA", NULL, "HeadsetR Mixer"},
 	/* CarkitL */
-	{"CarkitL Mux", "DACL1", "ARXL1_APGA"},
-	{"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+	{"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"},
+	{"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+	{"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
 	/* CarkitR */
-	{"CarkitR Mux", "DACR1", "ARXR1_APGA"},
-	{"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+	{"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"},
+	{"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+	{"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
 	/* HandsfreeL */
-	{"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
-	{"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
-	{"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+	{"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"},
+	{"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"},
+	{"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"},
+	{"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"},
+	{"HandsfreeL Switch", "Switch", "HandsfreeL Mux"},
+	{"HandsfreeL PGA", NULL, "HandsfreeL Switch"},
 	/* HandsfreeR */
-	{"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
-	{"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
-	{"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+	{"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"},
+	{"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"},
+	{"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"},
+	{"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"},
+	{"HandsfreeR Switch", "Switch", "HandsfreeR Mux"},
+	{"HandsfreeR PGA", NULL, "HandsfreeR Switch"},
+	/* Vibra */
+	{"Vibra Mux", "AudioL1", "DAC Left1"},
+	{"Vibra Mux", "AudioR1", "DAC Right1"},
+	{"Vibra Mux", "AudioL2", "DAC Left2"},
+	{"Vibra Mux", "AudioR2", "DAC Right2"},
 
 	/* outputs */
-	{"OUTL", NULL, "ARXL2_APGA"},
-	{"OUTR", NULL, "ARXR2_APGA"},
-	{"EARPIECE", NULL, "Earpiece Mux"},
-	{"PREDRIVEL", NULL, "PredriveL Mux"},
-	{"PREDRIVER", NULL, "PredriveR Mux"},
-	{"HSOL", NULL, "HeadsetL Mux"},
-	{"HSOR", NULL, "HeadsetR Mux"},
-	{"CARKITL", NULL, "CarkitL Mux"},
-	{"CARKITR", NULL, "CarkitR Mux"},
-	{"HFL", NULL, "HandsfreeL Mux"},
-	{"HFR", NULL, "HandsfreeR Mux"},
+	{"OUTL", NULL, "Analog L2 Playback Mixer"},
+	{"OUTR", NULL, "Analog R2 Playback Mixer"},
+	{"EARPIECE", NULL, "Earpiece Mixer"},
+	{"PREDRIVEL", NULL, "PredriveL Mixer"},
+	{"PREDRIVER", NULL, "PredriveR Mixer"},
+	{"HSOL", NULL, "HeadsetL PGA"},
+	{"HSOR", NULL, "HeadsetR PGA"},
+	{"CARKITL", NULL, "CarkitL Mixer"},
+	{"CARKITR", NULL, "CarkitR Mixer"},
+	{"HFL", NULL, "HandsfreeL PGA"},
+	{"HFR", NULL, "HandsfreeR PGA"},
+	{"Vibra Route", "Audio", "Vibra Mux"},
+	{"VIBRA", NULL, "Vibra Route"},
 
 	/* Capture path */
 	{"Analog Left Capture Route", "Main mic", "MAINMIC"},
@@ -1168,18 +1427,22 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
 	{"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
 	{"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+	{"Voice Analog Loopback", "Switch", "Analog Left Capture Route"},
 
 	{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
 	{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
 	{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
 	{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+	{"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"},
 
 	/* Digital bypass routes */
 	{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
 	{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+	{"Voice Digital Loopback", "Volume", "TX2 Capture Route"},
 
-	{"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
-	{"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+	{"Digital R2 Playback Mixer", NULL, "Right Digital Loopback"},
+	{"Digital L2 Playback Mixer", NULL, "Left Digital Loopback"},
+	{"Digital Voice Playback Mixer", NULL, "Voice Digital Loopback"},
 
 };
 
@@ -1226,6 +1489,58 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+static void twl4030_constraints(struct twl4030_priv *twl4030,
+				struct snd_pcm_substream *mst_substream)
+{
+	struct snd_pcm_substream *slv_substream;
+
+	/* Pick the stream, which need to be constrained */
+	if (mst_substream == twl4030->master_substream)
+		slv_substream = twl4030->slave_substream;
+	else if (mst_substream == twl4030->slave_substream)
+		slv_substream = twl4030->master_substream;
+	else /* This should not happen.. */
+		return;
+
+	/* Set the constraints according to the already configured stream */
+	snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+				SNDRV_PCM_HW_PARAM_RATE,
+				twl4030->rate,
+				twl4030->rate);
+
+	snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+				SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+				twl4030->sample_bits,
+				twl4030->sample_bits);
+
+	snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+				SNDRV_PCM_HW_PARAM_CHANNELS,
+				twl4030->channels,
+				twl4030->channels);
+}
+
+/* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for
+ * capture has to be enabled/disabled. */
+static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
+				int enable)
+{
+	u8 reg, mask;
+
+	reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+		mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN;
+	else
+		mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+	if (enable)
+		reg |= mask;
+	else
+		reg &= ~mask;
+
+	twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
 static int twl4030_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
@@ -1234,26 +1549,25 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct twl4030_priv *twl4030 = codec->private_data;
 
-	/* If we already have a playback or capture going then constrain
-	 * this substream to match it.
-	 */
 	if (twl4030->master_substream) {
-		struct snd_pcm_runtime *master_runtime;
-		master_runtime = twl4030->master_substream->runtime;
-
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_RATE,
-					     master_runtime->rate,
-					     master_runtime->rate);
-
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     master_runtime->sample_bits,
-					     master_runtime->sample_bits);
-
 		twl4030->slave_substream = substream;
-	} else
+		/* The DAI has one configuration for playback and capture, so
+		 * if the DAI has been already configured then constrain this
+		 * substream to match it. */
+		if (twl4030->configured)
+			twl4030_constraints(twl4030, twl4030->master_substream);
+	} else {
+		if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) &
+			TWL4030_OPTION_1)) {
+			/* In option2 4 channel is not supported, set the
+			 * constraint for the first stream for channels, the
+			 * second stream will 'inherit' this cosntraint */
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_CHANNELS,
+						2, 2);
+		}
 		twl4030->master_substream = substream;
+	}
 
 	return 0;
 }
@@ -1270,6 +1584,17 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream,
 		twl4030->master_substream = twl4030->slave_substream;
 
 	twl4030->slave_substream = NULL;
+
+	/* If all streams are closed, or the remaining stream has not yet
+	 * been configured than set the DAI as not configured. */
+	if (!twl4030->master_substream)
+		twl4030->configured = 0;
+	 else if (!twl4030->master_substream->runtime->channels)
+		twl4030->configured = 0;
+
+	 /* If the closing substream had 4 channel, do the necessary cleanup */
+	if (substream->runtime->channels == 4)
+		twl4030_tdm_enable(codec, substream->stream, 0);
 }
 
 static int twl4030_hw_params(struct snd_pcm_substream *substream,
@@ -1282,8 +1607,24 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	struct twl4030_priv *twl4030 = codec->private_data;
 	u8 mode, old_mode, format, old_format;
 
-	if (substream == twl4030->slave_substream)
-		/* Ignoring hw_params for slave substream */
+	 /* If the substream has 4 channel, do the necessary setup */
+	if (params_channels(params) == 4) {
+		u8 format, mode;
+
+		format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+		mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+
+		/* Safety check: are we in the correct operating mode and
+		 * the interface is in TDM mode? */
+		if ((mode & TWL4030_OPTION_1) &&
+		    ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM))
+			twl4030_tdm_enable(codec, substream->stream, 1);
+		else
+			return -EINVAL;
+	}
+
+	if (twl4030->configured)
+		/* Ignoring hw_params for already configured DAI */
 		return 0;
 
 	/* bit rate */
@@ -1363,6 +1704,21 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 		/* set CODECPDZ afterwards */
 		twl4030_codec_enable(codec, 1);
 	}
+
+	/* Store the important parameters for the DAI configuration and set
+	 * the DAI as configured */
+	twl4030->configured = 1;
+	twl4030->rate = params_rate(params);
+	twl4030->sample_bits = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
+	twl4030->channels = params_channels(params);
+
+	/* If both playback and capture streams are open, and one of them
+	 * is setting the hw parameters right now (since we are here), set
+	 * constraints to the other stream to match the current one. */
+	if (twl4030->slave_substream)
+		twl4030_constraints(twl4030, substream);
+
 	return 0;
 }
 
@@ -1370,17 +1726,21 @@ static int twl4030_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 twl4030_priv *twl4030 = codec->private_data;
 	u8 infreq;
 
 	switch (freq) {
 	case 19200000:
 		infreq = TWL4030_APLL_INFREQ_19200KHZ;
+		twl4030->sysclk = 19200;
 		break;
 	case 26000000:
 		infreq = TWL4030_APLL_INFREQ_26000KHZ;
+		twl4030->sysclk = 26000;
 		break;
 	case 38400000:
 		infreq = TWL4030_APLL_INFREQ_38400KHZ;
+		twl4030->sysclk = 38400;
 		break;
 	default:
 		printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
@@ -1424,6 +1784,9 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	case SND_SOC_DAIFMT_I2S:
 		format |= TWL4030_AIF_FORMAT_CODEC;
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		format |= TWL4030_AIF_FORMAT_TDM;
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -1443,6 +1806,180 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
+/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R
+ * (VTXL, VTXR) for uplink has to be enabled/disabled. */
+static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
+				int enable)
+{
+	u8 reg, mask;
+
+	reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+		mask = TWL4030_ARXL1_VRX_EN;
+	else
+		mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+	if (enable)
+		reg |= mask;
+	else
+		reg &= ~mask;
+
+	twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u8 infreq;
+	u8 mode;
+
+	/* If the system master clock is not 26MHz, the voice PCM interface is
+	 * not avilable.
+	 */
+	infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
+		& TWL4030_APLL_INFREQ;
+
+	if (infreq != TWL4030_APLL_INFREQ_26000KHZ) {
+		printk(KERN_ERR "TWL4030 voice startup: "
+			"MCLK is not 26MHz, call set_sysclk() on init\n");
+		return -EINVAL;
+	}
+
+	/* If the codec mode is not option2, the voice PCM interface is not
+	 * avilable.
+	 */
+	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+		& TWL4030_OPT_MODE;
+
+	if (mode != TWL4030_OPTION_2) {
+		printk(KERN_ERR "TWL4030 voice startup: "
+			"the codec mode is not option2\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	/* Enable voice digital filters */
+	twl4030_voice_enable(codec, substream->stream, 0);
+}
+
+static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u8 old_mode, mode;
+
+	/* Enable voice digital filters */
+	twl4030_voice_enable(codec, substream->stream, 1);
+
+	/* bit rate */
+	old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+		& ~(TWL4030_CODECPDZ);
+	mode = old_mode;
+
+	switch (params_rate(params)) {
+	case 8000:
+		mode &= ~(TWL4030_SEL_16K);
+		break;
+	case 16000:
+		mode |= TWL4030_SEL_16K;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	if (mode != old_mode) {
+		/* change rate and set CODECPDZ */
+		twl4030_codec_enable(codec, 0);
+		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+		twl4030_codec_enable(codec, 1);
+	}
+
+	return 0;
+}
+
+static int twl4030_voice_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;
+	u8 infreq;
+
+	switch (freq) {
+	case 26000000:
+		infreq = TWL4030_APLL_INFREQ_26000KHZ;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n",
+			freq);
+		return -EINVAL;
+	}
+
+	infreq |= TWL4030_APLL_EN;
+	twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+	return 0;
+}
+
+static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 old_format, format;
+
+	/* get format */
+	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
+	format = old_format;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFM:
+		format &= ~(TWL4030_VIF_SLAVE_EN);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		format |= TWL4030_VIF_SLAVE_EN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_NF:
+		format &= ~(TWL4030_VIF_FORMAT);
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		format |= TWL4030_VIF_FORMAT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (format != old_format) {
+		/* change format and set CODECPDZ */
+		twl4030_codec_enable(codec, 0);
+		twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+		twl4030_codec_enable(codec, 1);
+	}
+
+	return 0;
+}
+
 #define TWL4030_RATES	 (SNDRV_PCM_RATE_8000_48000)
 #define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
 
@@ -1454,21 +1991,47 @@ static struct snd_soc_dai_ops twl4030_dai_ops = {
 	.set_fmt	= twl4030_set_dai_fmt,
 };
 
-struct snd_soc_dai twl4030_dai = {
+static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
+	.startup	= twl4030_voice_startup,
+	.shutdown	= twl4030_voice_shutdown,
+	.hw_params	= twl4030_voice_hw_params,
+	.set_sysclk	= twl4030_voice_set_dai_sysclk,
+	.set_fmt	= twl4030_voice_set_dai_fmt,
+};
+
+struct snd_soc_dai twl4030_dai[] = {
+{
 	.name = "twl4030",
 	.playback = {
-		.stream_name = "Playback",
+		.stream_name = "HiFi Playback",
 		.channels_min = 2,
-		.channels_max = 2,
+		.channels_max = 4,
 		.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
 		.formats = TWL4030_FORMATS,},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 2,
-		.channels_max = 2,
+		.channels_max = 4,
 		.rates = TWL4030_RATES,
 		.formats = TWL4030_FORMATS,},
 	.ops = &twl4030_dai_ops,
+},
+{
+	.name = "twl4030 Voice",
+	.playback = {
+		.stream_name = "Voice Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &twl4030_dai_voice_ops,
+},
 };
 EXPORT_SYMBOL_GPL(twl4030_dai);
 
@@ -1500,6 +2063,8 @@ static int twl4030_resume(struct platform_device *pdev)
 static int twl4030_init(struct snd_soc_device *socdev)
 {
 	struct snd_soc_codec *codec = socdev->card->codec;
+	struct twl4030_setup_data *setup = socdev->codec_data;
+	struct twl4030_priv *twl4030 = codec->private_data;
 	int ret = 0;
 
 	printk(KERN_INFO "TWL4030 Audio Codec init \n");
@@ -1509,14 +2074,31 @@ static int twl4030_init(struct snd_soc_device *socdev)
 	codec->read = twl4030_read_reg_cache;
 	codec->write = twl4030_write;
 	codec->set_bias_level = twl4030_set_bias_level;
-	codec->dai = &twl4030_dai;
-	codec->num_dai = 1;
+	codec->dai = twl4030_dai;
+	codec->num_dai = ARRAY_SIZE(twl4030_dai),
 	codec->reg_cache_size = sizeof(twl4030_reg);
 	codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
 					GFP_KERNEL);
 	if (codec->reg_cache == NULL)
 		return -ENOMEM;
 
+	/* Configuration for headset ramp delay from setup data */
+	if (setup) {
+		unsigned char hs_pop;
+
+		if (setup->sysclk)
+			twl4030->sysclk = setup->sysclk;
+		else
+			twl4030->sysclk = 26000;
+
+		hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+		hs_pop &= ~TWL4030_RAMP_DELAY;
+		hs_pop |= (setup->ramp_delay_value << 2);
+		twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+	} else {
+		twl4030->sysclk = 26000;
+	}
+
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -1604,13 +2186,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
 
 static int __init twl4030_modinit(void)
 {
-	return snd_soc_register_dai(&twl4030_dai);
+	return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
 }
 module_init(twl4030_modinit);
 
 static void __exit twl4030_exit(void)
 {
-	snd_soc_unregister_dai(&twl4030_dai);
+	snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
 }
 module_exit(twl4030_exit);
 
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index cb63765db1d..fe5f395d9e4 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -92,8 +92,9 @@
 #define TWL4030_REG_VIBRA_PWM_SET	0x47
 #define TWL4030_REG_ANAMIC_GAIN		0x48
 #define TWL4030_REG_MISC_SET_2		0x49
+#define TWL4030_REG_SW_SHADOW		0x4A
 
-#define TWL4030_CACHEREGNUM	(TWL4030_REG_MISC_SET_2 + 1)
+#define TWL4030_CACHEREGNUM	(TWL4030_REG_SW_SHADOW + 1)
 
 /* Bitfield Definitions */
 
@@ -110,9 +111,22 @@
 #define TWL4030_APLL_RATE_44100		0x90
 #define TWL4030_APLL_RATE_48000		0xA0
 #define TWL4030_APLL_RATE_96000		0xE0
-#define TWL4030_SEL_16K			0x04
+#define TWL4030_SEL_16K			0x08
 #define TWL4030_CODECPDZ		0x02
 #define TWL4030_OPT_MODE		0x01
+#define TWL4030_OPTION_1		(1 << 0)
+#define TWL4030_OPTION_2		(0 << 0)
+
+/* TWL4030_OPTION (0x02) Fields */
+
+#define TWL4030_ATXL1_EN		(1 << 0)
+#define TWL4030_ATXR1_EN		(1 << 1)
+#define TWL4030_ATXL2_VTXL_EN		(1 << 2)
+#define TWL4030_ATXR2_VTXR_EN		(1 << 3)
+#define TWL4030_ARXL1_VRX_EN		(1 << 4)
+#define TWL4030_ARXR1_EN		(1 << 5)
+#define TWL4030_ARXL2_EN		(1 << 6)
+#define TWL4030_ARXR2_EN		(1 << 7)
 
 /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
 
@@ -171,6 +185,17 @@
 #define TWL4030_CLK256FS_EN		0x02
 #define TWL4030_AIF_EN			0x01
 
+/* VOICE_IF (0x0F) Fields */
+
+#define TWL4030_VIF_SLAVE_EN		0x80
+#define TWL4030_VIF_DIN_EN		0x40
+#define TWL4030_VIF_DOUT_EN		0x20
+#define TWL4030_VIF_SWAP		0x10
+#define TWL4030_VIF_FORMAT		0x08
+#define TWL4030_VIF_TRI_EN		0x04
+#define TWL4030_VIF_SUB_EN		0x02
+#define TWL4030_VIF_EN			0x01
+
 /* EAR_CTL (0x21) */
 #define TWL4030_EAR_GAIN		0x30
 
@@ -236,7 +261,19 @@
 #define TWL4030_SMOOTH_ANAVOL_EN	0x02
 #define TWL4030_DIGMIC_LR_SWAP_EN	0x01
 
-extern struct snd_soc_dai twl4030_dai;
+/* TWL4030_REG_SW_SHADOW (0x4A) Fields */
+#define TWL4030_HFL_EN			0x01
+#define TWL4030_HFR_EN			0x02
+
+#define TWL4030_DAI_HIFI		0
+#define TWL4030_DAI_VOICE		1
+
+extern struct snd_soc_dai twl4030_dai[2];
 extern struct snd_soc_codec_device soc_codec_dev_twl4030;
 
+struct twl4030_setup_data {
+	unsigned int ramp_delay_value;
+	unsigned int sysclk;
+};
+
 #endif	/* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index ddefb8f8014..269b108e1de 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
 	pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
 
 	if (reg >= UDA134X_REGS_NUM) {
-		printk(KERN_ERR "%s unkown register: reg: %d",
+		printk(KERN_ERR "%s unkown register: reg: %u",
 		       __func__, reg);
 		return -EINVAL;
 	}
@@ -296,7 +296,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 
-	pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+	pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__,
 		 clk_id, freq, dir);
 
 	/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 0275321ff8a..e7348d341b7 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
 	if (ret < 0)
 		return ret;
 	dev_dbg(wm8350->dev,
-		"FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+		"FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d",
 		freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
 		fll_div.ratio);
 
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index d11bd9288cf..d088eb4b88b 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -13,6 +13,7 @@
 #define _WM8350_H
 
 #include <sound/soc.h>
+#include <linux/mfd/wm8350/audio.h>
 
 extern struct snd_soc_dai wm8350_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8350;
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 510efa60400..502eefac1ec 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -954,7 +954,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
 		factors->outdiv *= 2;
 		if (factors->outdiv > 32) {
 			dev_err(wm8400->wm8400->dev,
-				"Unsupported FLL output frequency %dHz\n",
+				"Unsupported FLL output frequency %uHz\n",
 				Fout);
 			return -EINVAL;
 		}
@@ -1003,7 +1003,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
 	factors->k = K / 10;
 
 	dev_dbg(wm8400->wm8400->dev,
-		"FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
+		"FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
 		Fref, Fout,
 		factors->n, factors->k, factors->fratio, factors->outdiv);
 
@@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev)
 
 	codec = &priv->codec;
 	codec->private_data = priv;
-	codec->control_data = dev->dev.driver_data;
-	priv->wm8400 = dev->dev.driver_data;
+	codec->control_data = dev_get_drvdata(&dev->dev);
+	priv->wm8400 = dev_get_drvdata(&dev->dev);
 
 	ret = regulator_bulk_get(priv->wm8400->dev,
 				 ARRAY_SIZE(power), &power[0]);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 6a4cea09c45..c8b8dba8589 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -298,7 +298,7 @@ static void pll_factors(unsigned int target, unsigned int source)
 
 	if ((Ndiv < 6) || (Ndiv > 12))
 		printk(KERN_WARNING
-			"WM8510 N value %d outwith recommended range!d\n",
+			"WM8510 N value %u outwith recommended range!d\n",
 			Ndiv);
 
 	pll_div.n = Ndiv;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 9f6be3d31ac..86c4b24db81 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -415,7 +415,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target,
 	unsigned int K, Ndiv, Nmod;
 	int i;
 
-	pr_debug("wm8580: PLL %dHz->%dHz\n", source, target);
+	pr_debug("wm8580: PLL %uHz->%uHz\n", source, target);
 
 	/* Scale the output frequency up; the PLL should run in the
 	 * region of 90-100MHz.
@@ -447,7 +447,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target,
 
 	if ((Ndiv < 5) || (Ndiv > 13)) {
 		printk(KERN_ERR
-			"WM8580 N=%d outside supported range\n", Ndiv);
+			"WM8580 N=%u outside supported range\n", Ndiv);
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index e043e3f6000..7a205876ef4 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
 	codec->hw_write = (hw_write_t)wm8731_spi_write;
 	codec->dev = &spi->dev;
 
-	spi->dev.driver_data = wm8731;
+	dev_set_drvdata(&spi->dev, wm8731);
 
 	return wm8731_register(wm8731);
 }
 
 static int __devexit wm8731_spi_remove(struct spi_device *spi)
 {
-	struct wm8731_priv *wm8731 = spi->dev.driver_data;
+	struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev);
 
 	wm8731_unregister(wm8731);
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a6e8f3f7f05..d28eeaceb85 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -703,7 +703,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 
 	if ((Ndiv < 6) || (Ndiv > 12))
 		printk(KERN_WARNING
-			"wm8753: unsupported N = %d\n", Ndiv);
+			"wm8753: unsupported N = %u\n", Ndiv);
 
 	pll_div->n = Ndiv;
 	Nmod = target % source;
@@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi)
 	codec->hw_write = (hw_write_t)wm8753_spi_write;
 	codec->dev = &spi->dev;
 
-	spi->dev.driver_data = wm8753;
+	dev_set_drvdata(&spi->dev, wm8753);
 
 	return wm8753_register(wm8753);
 }
 
 static int __devexit wm8753_spi_remove(struct spi_device *spi)
 {
-	struct wm8753_priv *wm8753 = spi->dev.driver_data;
+	struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev);
 	wm8753_unregister(wm8753);
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 46c5ea1ff92..3c78945244b 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -778,11 +778,11 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 	}
 
 	if (target > 100000000)
-		printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d"
-		       " Fout=%d\n", target, Fref, Fout);
+		printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u"
+		       " Fout=%u\n", target, Fref, Fout);
 	if (div > 32) {
 		printk(KERN_ERR "wm8900: Invalid FLL division rate %u, "
-		       "Fref=%d, Fout=%d, target=%d\n",
+		       "Fref=%u, Fout=%u, target=%u\n",
 		       div, Fref, Fout, target);
 		return -EINVAL;
 	}
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 8cf571f1a80..d8a9222fbf7 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -217,7 +217,6 @@ struct wm8903_priv {
 	int sysclk;
 
 	/* Reference counts */
-	int charge_pump_users;
 	int class_w_users;
 	int playback_active;
 	int capture_active;
@@ -373,6 +372,15 @@ static void wm8903_reset(struct snd_soc_codec *codec)
 #define WM8903_OUTPUT_INT   0x2
 #define WM8903_OUTPUT_IN    0x1
 
+static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	WARN_ON(event != SND_SOC_DAPM_POST_PMU);
+	mdelay(4);
+
+	return 0;
+}
+
 /*
  * Event for headphone and line out amplifier power changes.  Special
  * power up/down sequences are required in order to maximise pop/click
@@ -382,19 +390,20 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 			       struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm8903_priv *wm8903 = codec->private_data;
-	struct i2c_client *i2c = codec->control_data;
 	u16 val;
 	u16 reg;
+	u16 dcs_reg;
+	u16 dcs_bit;
 	int shift;
-	u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0);
 
 	switch (w->reg) {
 	case WM8903_POWER_MANAGEMENT_2:
 		reg = WM8903_ANALOGUE_HP_0;
+		dcs_bit = 0 + w->shift;
 		break;
 	case WM8903_POWER_MANAGEMENT_3:
 		reg = WM8903_ANALOGUE_LINEOUT_0;
+		dcs_bit = 2 + w->shift;
 		break;
 	default:
 		BUG();
@@ -419,18 +428,6 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 		/* Short the output */
 		val &= ~(WM8903_OUTPUT_SHORT << shift);
 		wm8903_write(codec, reg, val);
-
-		wm8903->charge_pump_users++;
-
-		dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
-			wm8903->charge_pump_users);
-
-		if (wm8903->charge_pump_users == 1) {
-			dev_dbg(&i2c->dev, "Enabling charge pump\n");
-			wm8903_write(codec, WM8903_CHARGE_PUMP_0,
-				     cp_reg | WM8903_CP_ENA);
-			mdelay(4);
-		}
 	}
 
 	if (event & SND_SOC_DAPM_POST_PMU) {
@@ -446,6 +443,11 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 		val |= (WM8903_OUTPUT_OUT << shift);
 		wm8903_write(codec, reg, val);
 
+		/* Enable the DC servo */
+		dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+		dcs_reg |= dcs_bit;
+		wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
 		/* Remove the short */
 		val |= (WM8903_OUTPUT_SHORT << shift);
 		wm8903_write(codec, reg, val);
@@ -458,25 +460,17 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 		val &= ~(WM8903_OUTPUT_SHORT << shift);
 		wm8903_write(codec, reg, val);
 
+		/* Disable the DC servo */
+		dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+		dcs_reg &= ~dcs_bit;
+		wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
 		/* Then disable the intermediate and output stages */
 		val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
 			  WM8903_OUTPUT_IN) << shift);
 		wm8903_write(codec, reg, val);
 	}
 
-	if (event & SND_SOC_DAPM_POST_PMD) {
-		wm8903->charge_pump_users--;
-
-		dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
-			wm8903->charge_pump_users);
-
-		if (wm8903->charge_pump_users == 0) {
-			dev_dbg(&i2c->dev, "Disabling charge pump\n");
-			wm8903_write(codec, WM8903_CHARGE_PUMP_0,
-				     cp_reg & ~WM8903_CP_ENA);
-		}
-	}
-
 	return 0;
 }
 
@@ -539,6 +533,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
 /* ALSA can only do steps of .01dB */
 static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
 
+static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
 
 static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0);
@@ -657,6 +652,16 @@ static const struct soc_enum rinput_inv_enum =
 	SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
 
 
+static const char *sidetone_text[] = {
+	"None", "Left", "Right"
+};
+
+static const struct soc_enum lsidetone_enum =
+	SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text);
+
+static const struct soc_enum rsidetone_enum =
+	SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
+
 static const struct snd_kcontrol_new wm8903_snd_controls[] = {
 
 /* Input PGAs - No TLV since the scale depends on PGA mode */
@@ -700,6 +705,9 @@ SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
 SOC_ENUM("ADC Companding Mode", adc_companding),
 SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
 
+SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8,
+	       12, 0, digital_sidetone_tlv),
+
 /* DAC */
 SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT,
 		 WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
@@ -762,6 +770,12 @@ static const struct snd_kcontrol_new rinput_mux =
 static const struct snd_kcontrol_new rinput_inv_mux =
 	SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum);
 
+static const struct snd_kcontrol_new lsidetone_mux =
+	SOC_DAPM_ENUM("DACL Sidetone Mux", lsidetone_enum);
+
+static const struct snd_kcontrol_new rsidetone_mux =
+	SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
+
 static const struct snd_kcontrol_new left_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
@@ -828,6 +842,9 @@ SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0),
 SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
 SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
 
+SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
+SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
+
 SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
 SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
 
@@ -844,26 +861,29 @@ SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
 SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
 		   1, 0, NULL, 0, wm8903_output_event,
 		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+		   SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
 		   0, 0, NULL, 0, wm8903_output_event,
 		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+		   SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
 		   NULL, 0, wm8903_output_event,
 		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+		   SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
 		   NULL, 0, wm8903_output_event,
 		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+		   SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
 		 NULL, 0),
 SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0,
 		 NULL, 0),
 
+SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
+		    wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
@@ -909,7 +929,19 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Right Input PGA", NULL, "Right Input Mode Mux" },
 
 	{ "ADCL", NULL, "Left Input PGA" },
+	{ "ADCL", NULL, "CLK_DSP" },
 	{ "ADCR", NULL, "Right Input PGA" },
+	{ "ADCR", NULL, "CLK_DSP" },
+
+	{ "DACL Sidetone", "Left", "ADCL" },
+	{ "DACL Sidetone", "Right", "ADCR" },
+	{ "DACR Sidetone", "Left", "ADCL" },
+	{ "DACR Sidetone", "Right", "ADCR" },
+
+	{ "DACL", NULL, "DACL Sidetone" },
+	{ "DACL", NULL, "CLK_DSP" },
+	{ "DACR", NULL, "DACR Sidetone" },
+	{ "DACR", NULL, "CLK_DSP" },
 
 	{ "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" },
 	{ "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" },
@@ -951,6 +983,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 
 	{ "ROP", NULL, "Right Speaker PGA" },
 	{ "RON", NULL, "Right Speaker PGA" },
+
+	{ "Left Headphone Output PGA", NULL, "Charge Pump" },
+	{ "Right Headphone Output PGA", NULL, "Charge Pump" },
+	{ "Left Line Output PGA", NULL, "Charge Pump" },
+	{ "Right Line Output PGA", NULL, "Charge Pump" },
 };
 
 static int wm8903_add_widgets(struct snd_soc_codec *codec)
@@ -985,6 +1022,11 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 			wm8903_write(codec, WM8903_CLOCK_RATES_2,
 				     WM8903_CLK_SYS_ENA);
 
+			/* Change DC servo dither level in startup sequence */
+			wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
+			wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
+			wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
+
 			wm8903_run_sequence(codec, 0);
 			wm8903_sync_reg_cache(codec, codec->reg_cache);
 
@@ -1277,14 +1319,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream,
 	if (wm8903->master_substream) {
 		master_runtime = wm8903->master_substream->runtime;
 
-		dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
-			master_runtime->sample_bits,
-			master_runtime->rate);
-
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_RATE,
-					     master_runtime->rate,
-					     master_runtime->rate);
+		dev_dbg(&i2c->dev, "Constraining to %d bits\n",
+			master_runtime->sample_bits);
 
 		snd_pcm_hw_constraint_minmax(substream->runtime,
 					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
@@ -1523,6 +1559,7 @@ struct snd_soc_dai wm8903_dai = {
 		 .formats = WM8903_FORMATS,
 	 },
 	.ops = &wm8903_dai_ops,
+	.symmetric_rates = 1,
 };
 EXPORT_SYMBOL_GPL(wm8903_dai);
 
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
new file mode 100644
index 00000000000..b8e17d6bc1f
--- /dev/null
+++ b/sound/soc/codecs/wm8940.c
@@ -0,0 +1,955 @@
+/*
+ * wm8940.c  --  WM8940 ALSA Soc Audio driver
+ *
+ * Author: Jonathan Cameron <jic23@cam.ac.uk>
+ *
+ * Based on wm8510.c
+ *    Copyright  2006 Wolfson Microelectronics PLC.
+ *    Author:  Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Not currently handled:
+ * Notch filter control
+ * AUXMode (inverting vs mixer)
+ * No means to obtain current gain if alc enabled.
+ * No use made of gpio
+ * Fast VMID discharge for power down
+ * Soft Start
+ * DLR and ALR Swaps not enabled
+ * Digital Sidetone not supported
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.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 "wm8940.h"
+
+struct wm8940_priv {
+	unsigned int sysclk;
+	u16 reg_cache[WM8940_CACHEREGNUM];
+	struct snd_soc_codec codec;
+};
+
+static u16 wm8940_reg_defaults[] = {
+	0x8940, /* Soft Reset */
+	0x0000, /* Power 1 */
+	0x0000, /* Power 2 */
+	0x0000, /* Power 3 */
+	0x0010, /* Interface Control */
+	0x0000, /* Companding Control */
+	0x0140, /* Clock Control */
+	0x0000, /* Additional Controls */
+	0x0000, /* GPIO Control */
+	0x0002, /* Auto Increment Control */
+	0x0000, /* DAC Control */
+	0x00FF, /* DAC Volume */
+	0,
+	0,
+	0x0100, /* ADC Control */
+	0x00FF, /* ADC Volume */
+	0x0000, /* Notch Filter 1 Control 1 */
+	0x0000, /* Notch Filter 1 Control 2 */
+	0x0000, /* Notch Filter 2 Control 1 */
+	0x0000, /* Notch Filter 2 Control 2 */
+	0x0000, /* Notch Filter 3 Control 1 */
+	0x0000, /* Notch Filter 3 Control 2 */
+	0x0000, /* Notch Filter 4 Control 1 */
+	0x0000, /* Notch Filter 4 Control 2 */
+	0x0032, /* DAC Limit Control 1 */
+	0x0000, /* DAC Limit Control 2 */
+	0,
+	0,
+	0,
+	0,
+	0,
+	0,
+	0x0038, /* ALC Control 1 */
+	0x000B, /* ALC Control 2 */
+	0x0032, /* ALC Control 3 */
+	0x0000, /* Noise Gate */
+	0x0041, /* PLLN */
+	0x000C, /* PLLK1 */
+	0x0093, /* PLLK2 */
+	0x00E9, /* PLLK3 */
+	0,
+	0,
+	0x0030, /* ALC Control 4 */
+	0,
+	0x0002, /* Input Control */
+	0x0050, /* PGA Gain */
+	0,
+	0x0002, /* ADC Boost Control */
+	0,
+	0x0002, /* Output Control */
+	0x0000, /* Speaker Mixer Control */
+	0,
+	0,
+	0,
+	0x0079, /* Speaker Volume */
+	0,
+	0x0000, /* Mono Mixer Control */
+};
+
+static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec,
+						 unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+		return -1;
+
+	return cache[reg];
+}
+
+static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec,
+					  u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+		return -1;
+
+	cache[reg] = value;
+
+	return 0;
+}
+
+static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg,
+			unsigned int value)
+{
+	int ret;
+	u8 data[3] = { reg,
+		       (value & 0xff00) >> 8,
+		       (value & 0x00ff)
+	};
+
+	wm8940_write_reg_cache(codec, reg, value);
+
+	ret = codec->hw_write(codec->control_data, data, 3);
+
+	if (ret < 0)
+		return ret;
+	else if (ret != 3)
+		return -EIO;
+	return 0;
+}
+
+static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const struct soc_enum wm8940_adc_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
+static const struct soc_enum wm8940_dac_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding);
+
+static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"};
+static const struct soc_enum wm8940_alc_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text);
+
+static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"};
+static const struct soc_enum wm8940_mic_bias_level_enum
+= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text);
+
+static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
+static const struct soc_enum wm8940_filter_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
+
+static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0);
+
+static const struct snd_kcontrol_new wm8940_snd_controls[] = {
+	SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL,
+		   6, 1, 0),
+	SOC_ENUM("DAC Companding", wm8940_dac_companding_enum),
+	SOC_ENUM("ADC Companding", wm8940_adc_companding_enum),
+
+	SOC_ENUM("ALC Mode", wm8940_alc_mode_enum),
+	SOC_SINGLE("ALC Switch", WM8940_ALC1, 8, 1, 0),
+	SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1,
+		       3, 7, 1, wm8940_alc_max_tlv),
+	SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1,
+		       0, 7, 0, wm8940_alc_min_tlv),
+	SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2,
+		       0, 14, 0, wm8940_alc_tar_tlv),
+	SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0),
+	SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0),
+	SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0),
+	SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0),
+	SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE,
+		   3, 1, 0),
+	SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE,
+		   0, 7, 0),
+
+	SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0),
+	SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0),
+	SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0),
+	SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2,
+		       4, 9, 1, wm8940_lim_thresh_tlv),
+	SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2,
+		       0, 12, 0, wm8940_lim_boost_tlv),
+
+	SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0),
+	SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN,
+		       0, 63, 0, wm8940_pga_vol_tlv),
+	SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL,
+		       0, 255, 0, wm8940_adc_tlv),
+	SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL,
+		       0, 255, 0, wm8940_adc_tlv),
+	SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum),
+	SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST,
+		       8, 1, 0, wm8940_capture_boost_vol_tlv),
+	SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL,
+		       0, 63, 0, wm8940_spk_vol_tlv),
+	SOC_SINGLE("Speaker Playback Switch", WM8940_SPKVOL,  6, 1, 1),
+
+	SOC_SINGLE_TLV("Speaker Mixer Line Bypass Volume", WM8940_SPKVOL,
+		       8, 1, 1, wm8940_att_tlv),
+	SOC_SINGLE("Speaker Playback ZC Switch", WM8940_SPKVOL, 7, 1, 0),
+
+	SOC_SINGLE("Mono Out Switch", WM8940_MONOMIX, 6, 1, 1),
+	SOC_SINGLE_TLV("Mono Mixer Line Bypass Volume", WM8940_MONOMIX,
+		       7, 1, 1, wm8940_att_tlv),
+
+	SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0),
+	SOC_ENUM("High Pass Filter Mode", wm8940_filter_mode_enum),
+	SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0),
+	SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0),
+	SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0),
+	SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0),
+	SOC_SINGLE("ZC Timeout Clock Switch", WM8940_ADDCNTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0),
+	SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0),
+	SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0),
+	SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0),
+	SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1);
+static const struct snd_kcontrol_new wm8940_input_boost_controls[] = {
+	SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1),
+	SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST,
+			    0, 7, 0, wm8940_boost_vol_tlv),
+	SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST,
+			    4, 7, 0, wm8940_boost_vol_tlv),
+};
+
+static const struct snd_kcontrol_new wm8940_micpga_controls[] = {
+	SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0),
+	SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0),
+	SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = {
+	SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0,
+			   &wm8940_speaker_mixer_controls[0],
+			   ARRAY_SIZE(wm8940_speaker_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0,
+			   &wm8940_mono_mixer_controls[0],
+			   ARRAY_SIZE(wm8940_mono_mixer_controls)),
+	SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0),
+
+	SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("MONOOUT"),
+	SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+	SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+
+	SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0),
+	SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0),
+	SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0,
+			   &wm8940_micpga_controls[0],
+			   ARRAY_SIZE(wm8940_micpga_controls)),
+	SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0,
+			   &wm8940_input_boost_controls[0],
+			   ARRAY_SIZE(wm8940_input_boost_controls)),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0),
+
+	SND_SOC_DAPM_INPUT("MICN"),
+	SND_SOC_DAPM_INPUT("MICP"),
+	SND_SOC_DAPM_INPUT("AUX"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Mono output mixer */
+	{"Mono Mixer", "PCM Playback Switch", "DAC"},
+	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+	{"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+	/* Speaker output mixer */
+	{"Speaker Mixer", "PCM Playback Switch", "DAC"},
+	{"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+	{"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+	/* Outputs */
+	{"Mono Out", NULL, "Mono Mixer"},
+	{"MONOOUT", NULL, "Mono Out"},
+	{"SpkN Out", NULL, "Speaker Mixer"},
+	{"SpkP Out", NULL, "Speaker Mixer"},
+	{"SPKOUTN", NULL, "SpkN Out"},
+	{"SPKOUTP", NULL, "SpkP Out"},
+
+	/*  Microphone PGA */
+	{"Mic PGA", "MICN Switch", "MICN"},
+	{"Mic PGA", "MICP Switch", "MICP"},
+	{"Mic PGA", "AUX Switch", "AUX"},
+
+	/* Boost Mixer */
+	{"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+	{"Boost Mixer", "Mic Volume",  "MICP"},
+	{"Boost Mixer", "Aux Volume", "Aux Input"},
+
+	{"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8940_add_widgets(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec, 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));
+	if (ret)
+		goto error_ret;
+	ret = snd_soc_dapm_new_widgets(codec);
+
+error_ret:
+	return ret;
+}
+
+#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0);
+
+static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67;
+	u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		clk |= 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+	wm8940_write(codec, WM8940_CLOCK, clk);
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= (2 << 3);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= (1 << 3);
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= (3 << 3);
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= (3 << 3) | (1 << 7);
+		break;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= (1 << 7);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= (1 << 8);
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= (1 << 8) | (1 << 7);
+		break;
+	}
+
+	wm8940_write(codec, WM8940_IFACE, iface);
+
+	return 0;
+}
+
+static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F;
+	u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1;
+	u16 companding =  wm8940_read_reg_cache(codec,
+						WM8940_COMPANDINGCTL) & 0xFFDF;
+	int ret;
+
+	/* LoutR control */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE
+	    && params_channels(params) == 2)
+		iface |= (1 << 9);
+
+	switch (params_rate(params)) {
+	case SNDRV_PCM_RATE_8000:
+		addcntrl |= (0x5 << 1);
+		break;
+	case SNDRV_PCM_RATE_11025:
+		addcntrl |= (0x4 << 1);
+		break;
+	case SNDRV_PCM_RATE_16000:
+		addcntrl |= (0x3 << 1);
+		break;
+	case SNDRV_PCM_RATE_22050:
+		addcntrl |= (0x2 << 1);
+		break;
+	case SNDRV_PCM_RATE_32000:
+		addcntrl |= (0x1 << 1);
+		break;
+	case SNDRV_PCM_RATE_44100:
+	case SNDRV_PCM_RATE_48000:
+		break;
+	}
+	ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl);
+	if (ret)
+		goto error_ret;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		companding = companding | (1 << 5);
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= (1 << 5);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= (2 << 5);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= (3 << 5);
+		break;
+	}
+	ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding);
+	if (ret)
+		goto error_ret;
+	ret = wm8940_write(codec, WM8940_IFACE, iface);
+
+error_ret:
+	return ret;
+}
+
+static int wm8940_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf;
+
+	if (mute)
+		mute_reg |= 0x40;
+
+	return wm8940_write(codec, WM8940_DAC, mute_reg);
+}
+
+static int wm8940_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 val;
+	u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0;
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* ensure bufioen and biasen */
+		pwr_reg |= (1 << 2) | (1 << 3);
+		/* Enable thermal shutdown */
+		val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+		ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2);
+		if (ret)
+			break;
+		/* set vmid to 75k */
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* ensure bufioen and biasen */
+		pwr_reg |= (1 << 2) | (1 << 3);
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* ensure bufioen and biasen */
+		pwr_reg |= (1 << 2) | (1 << 3);
+		/* set vmid to 300k for standby */
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2);
+		break;
+	case SND_SOC_BIAS_OFF:
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg);
+		break;
+	}
+
+	return ret;
+}
+
+struct pll_ {
+	unsigned int pre_scale:2;
+	unsigned int n:4;
+	unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+static void pll_factors(unsigned int target, unsigned int source)
+{
+	unsigned long long Kpart;
+	unsigned int K, Ndiv, Nmod;
+	/* The left shift ist to avoid accuracy loss when right shifting */
+	Ndiv = target / source;
+
+	if (Ndiv > 12) {
+		source <<= 1;
+		/* Multiply by 2 */
+		pll_div.pre_scale = 0;
+		Ndiv = target / source;
+	} else if (Ndiv < 3) {
+		source >>= 2;
+		/* Divide by 4 */
+		pll_div.pre_scale = 3;
+		Ndiv = target / source;
+	} else if (Ndiv < 6) {
+		source >>= 1;
+		/* divide by 2 */
+		pll_div.pre_scale = 2;
+		Ndiv = target / source;
+	} else
+		pll_div.pre_scale = 1;
+
+	if ((Ndiv < 6) || (Ndiv > 12))
+		printk(KERN_WARNING
+			"WM8940 N value %d outwith recommended range!d\n",
+			Ndiv);
+
+	pll_div.n = Ndiv;
+	Nmod = target % source;
+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, source);
+
+	K = Kpart & 0xFFFFFFFF;
+
+	/* Check if we need to round */
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	K /= 10;
+
+	pll_div.k = K;
+}
+
+/* Untested at the moment */
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
+		int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	/* Turn off PLL */
+	reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+	wm8940_write(codec, WM8940_POWER1, reg & 0x1df);
+
+	if (freq_in == 0 || freq_out == 0) {
+		/* Clock CODEC directly from MCLK */
+		reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+		wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff);
+		/* Pll power down */
+		wm8940_write(codec, WM8940_PLLN, (1 << 7));
+		return 0;
+	}
+
+	/* Pll is followed by a frequency divide by 4 */
+	pll_factors(freq_out*4, freq_in);
+	if (pll_div.k)
+		wm8940_write(codec, WM8940_PLLN,
+			     (pll_div.pre_scale << 4) | pll_div.n | (1 << 6));
+	else /* No factional component */
+		wm8940_write(codec, WM8940_PLLN,
+			     (pll_div.pre_scale << 4) | pll_div.n);
+	wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18);
+	wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff);
+	wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff);
+	/* Enable the PLL */
+	reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+	wm8940_write(codec, WM8940_POWER1, reg | 0x020);
+
+	/* Run CODEC from PLL instead of MCLK */
+	reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+	wm8940_write(codec, WM8940_CLOCK, reg | 0x100);
+
+	return 0;
+}
+
+static int wm8940_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 wm8940_priv *wm8940 = codec->private_data;
+
+	switch (freq) {
+	case 11289600:
+	case 12000000:
+	case 12288000:
+	case 16934400:
+	case 18432000:
+		wm8940->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+				 int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+	int ret = 0;
+
+	switch (div_id) {
+	case WM8940_BCLKDIV:
+		reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3;
+		ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2));
+		break;
+	case WM8940_MCLKDIV:
+		reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F;
+		ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5));
+		break;
+	case WM8940_OPCLKDIV:
+		reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF;
+		ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4));
+		break;
+	}
+	return ret;
+}
+
+#define WM8940_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 |				\
+			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 wm8940_dai_ops = {
+	.hw_params = wm8940_i2s_hw_params,
+	.set_sysclk = wm8940_set_dai_sysclk,
+	.digital_mute = wm8940_mute,
+	.set_fmt = wm8940_set_dai_fmt,
+	.set_clkdiv = wm8940_set_dai_clkdiv,
+	.set_pll = wm8940_set_dai_pll,
+};
+
+struct snd_soc_dai wm8940_dai = {
+	.name = "WM8940",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8940_RATES,
+		.formats = WM8940_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8940_RATES,
+		.formats = WM8940_FORMATS,
+	},
+	.ops = &wm8940_dai_ops,
+	.symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8940_dai);
+
+static int wm8940_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8940_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int i;
+	int ret;
+	u8 data[3];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware
+	 * Could use auto incremented writes to speed this up
+	 */
+	for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) {
+		data[0] = i;
+		data[1] = (cache[i] & 0xFF00) >> 8;
+		data[2] = cache[i] & 0x00FF;
+		ret = codec->hw_write(codec->control_data, data, 3);
+		if (ret < 0)
+			goto error_ret;
+		else if (ret != 3) {
+			ret = -EIO;
+			goto error_ret;
+		}
+	}
+	ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	if (ret)
+		goto error_ret;
+	ret = wm8940_set_bias_level(codec, codec->suspend_bias_level);
+
+error_ret:
+	return ret;
+}
+
+static struct snd_soc_codec *wm8940_codec;
+
+static int wm8940_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+
+	int ret = 0;
+
+	if (wm8940_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm8940_codec;
+	codec = wm8940_codec;
+
+	mutex_init(&codec->mutex);
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	ret = snd_soc_add_controls(codec, wm8940_snd_controls,
+			     ARRAY_SIZE(wm8940_snd_controls));
+	if (ret)
+		goto error_free_pcms;
+	ret = wm8940_add_widgets(codec);
+	if (ret)
+		goto error_free_pcms;
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
+		goto error_free_pcms;
+	}
+
+	return ret;
+
+error_free_pcms:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	return ret;
+}
+
+static int wm8940_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8940 = {
+	.probe = wm8940_probe,
+	.remove = wm8940_remove,
+	.suspend = wm8940_suspend,
+	.resume = wm8940_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940);
+
+static int wm8940_register(struct wm8940_priv *wm8940)
+{
+	struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data;
+	struct snd_soc_codec *codec = &wm8940->codec;
+	int ret;
+	u16 reg;
+	if (wm8940_codec) {
+		dev_err(codec->dev, "Another WM8940 is registered\n");
+		return -EINVAL;
+	}
+
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data = wm8940;
+	codec->name = "WM8940";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8940_read_reg_cache;
+	codec->write = wm8940_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8940_set_bias_level;
+	codec->dai = &wm8940_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults);
+	codec->reg_cache = &wm8940->reg_cache;
+
+	memcpy(codec->reg_cache, wm8940_reg_defaults,
+	       sizeof(wm8940_reg_defaults));
+
+	ret = wm8940_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
+	}
+
+	wm8940_dai.dev = codec->dev;
+
+	wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = wm8940_write(codec, WM8940_POWER1, 0x180);
+	if (ret < 0)
+		return ret;
+
+	if (!pdata)
+		dev_warn(codec->dev, "No platform data supplied\n");
+	else {
+		reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+		ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi);
+		if (ret < 0)
+			return ret;
+	}
+
+
+	wm8940_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm8940_dai);
+	if (ret) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
+
+	return 0;
+}
+
+static void wm8940_unregister(struct wm8940_priv *wm8940)
+{
+	wm8940_set_bias_level(&wm8940->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm8940_dai);
+	snd_soc_unregister_codec(&wm8940->codec);
+	kfree(wm8940);
+	wm8940_codec = NULL;
+}
+
+static int wm8940_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct wm8940_priv *wm8940;
+	struct snd_soc_codec *codec;
+
+	wm8940 = kzalloc(sizeof *wm8940, GFP_KERNEL);
+	if (wm8940 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8940->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+	i2c_set_clientdata(i2c, wm8940);
+	codec->control_data = i2c;
+	codec->dev = &i2c->dev;
+
+	return wm8940_register(wm8940);
+}
+
+static int __devexit wm8940_i2c_remove(struct i2c_client *client)
+{
+	struct wm8940_priv *wm8940 = i2c_get_clientdata(client);
+
+	wm8940_unregister(wm8940);
+
+	return 0;
+}
+
+static const struct i2c_device_id wm8940_i2c_id[] = {
+	{ "wm8940", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id);
+
+static struct i2c_driver wm8940_i2c_driver = {
+	.driver = {
+		.name = "WM8940 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8940_i2c_probe,
+	.remove = __devexit_p(wm8940_i2c_remove),
+	.id_table = wm8940_i2c_id,
+};
+
+static int __init wm8940_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&wm8940_i2c_driver);
+	if (ret)
+		printk(KERN_ERR "Failed to register WM8940 I2C driver: %d\n",
+		       ret);
+	return ret;
+}
+module_init(wm8940_modinit);
+
+static void __exit wm8940_exit(void)
+{
+	i2c_del_driver(&wm8940_i2c_driver);
+}
+module_exit(wm8940_exit);
+
+MODULE_DESCRIPTION("ASoC WM8940 driver");
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h
new file mode 100644
index 00000000000..8410eed3ef8
--- /dev/null
+++ b/sound/soc/codecs/wm8940.h
@@ -0,0 +1,104 @@
+/*
+ * wm8940.h -- WM8940 Soc Audio driver
+ *
+ * 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 _WM8940_H
+#define _WM8940_H
+
+struct wm8940_setup_data {
+	/* Vref to analogue output resistance */
+#define WM8940_VROI_1K 0
+#define WM8940_VROI_30K 1
+	unsigned int vroi:1;
+};
+extern struct snd_soc_dai wm8940_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8940;
+
+/* WM8940 register space */
+#define WM8940_SOFTRESET	0x00
+#define WM8940_POWER1		0x01
+#define WM8940_POWER2		0x02
+#define WM8940_POWER3		0x03
+#define WM8940_IFACE		0x04
+#define WM8940_COMPANDINGCTL	0x05
+#define WM8940_CLOCK		0x06
+#define WM8940_ADDCNTRL		0x07
+#define WM8940_GPIO		0x08
+#define WM8940_CTLINT		0x09
+#define WM8940_DAC		0x0A
+#define WM8940_DACVOL		0x0B
+
+#define WM8940_ADC		0x0E
+#define WM8940_ADCVOL		0x0F
+#define WM8940_NOTCH1		0x10
+#define WM8940_NOTCH2		0x11
+#define WM8940_NOTCH3		0x12
+#define WM8940_NOTCH4		0x13
+#define WM8940_NOTCH5		0x14
+#define WM8940_NOTCH6		0x15
+#define WM8940_NOTCH7		0x16
+#define WM8940_NOTCH8		0x17
+#define WM8940_DACLIM1		0x18
+#define WM8940_DACLIM2		0x19
+
+#define WM8940_ALC1		0x20
+#define WM8940_ALC2		0x21
+#define WM8940_ALC3		0x22
+#define WM8940_NOISEGATE	0x23
+#define WM8940_PLLN		0x24
+#define WM8940_PLLK1		0x25
+#define WM8940_PLLK2		0x26
+#define WM8940_PLLK3		0x27
+
+#define WM8940_ALC4		0x2A
+
+#define WM8940_INPUTCTL		0x2C
+#define WM8940_PGAGAIN		0x2D
+
+#define WM8940_ADCBOOST		0x2F
+
+#define WM8940_OUTPUTCTL	0x31
+#define WM8940_SPKMIX		0x32
+
+#define WM8940_SPKVOL		0x36
+
+#define WM8940_MONOMIX		0x38
+
+#define WM8940_CACHEREGNUM  0x57
+
+
+/* Clock divider Id's */
+#define WM8940_BCLKDIV 0
+#define WM8940_MCLKDIV 1
+#define WM8940_OPCLKDIV 2
+
+/* MCLK clock dividers */
+#define WM8940_MCLKDIV_1	0
+#define WM8940_MCLKDIV_1_5	1
+#define WM8940_MCLKDIV_2	2
+#define WM8940_MCLKDIV_3	3
+#define WM8940_MCLKDIV_4	4
+#define WM8940_MCLKDIV_6	5
+#define WM8940_MCLKDIV_8	6
+#define WM8940_MCLKDIV_12	7
+
+/* BCLK clock dividers */
+#define WM8940_BCLKDIV_1 0
+#define WM8940_BCLKDIV_2 1
+#define WM8940_BCLKDIV_4 2
+#define WM8940_BCLKDIV_8 3
+#define WM8940_BCLKDIV_16 4
+#define WM8940_BCLKDIV_32 5
+
+/* PLL Out Dividers */
+#define WM8940_OPCLKDIV_1 0
+#define WM8940_OPCLKDIV_2 1
+#define WM8940_OPCLKDIV_3 2
+#define WM8940_OPCLKDIV_4 3
+
+#endif /* _WM8940_H */
+
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
new file mode 100644
index 00000000000..e224d8add17
--- /dev/null
+++ b/sound/soc/codecs/wm8960.c
@@ -0,0 +1,969 @@
+/*
+ * wm8960.c  --  WM8960 ALSA SoC Audio driver
+ *
+ * Author: Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <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 "wm8960.h"
+
+#define AUDIO_NAME "wm8960"
+
+struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+/* R25 - Power 1 */
+#define WM8960_VREF      0x40
+
+/* R28 - Anti-pop 1 */
+#define WM8960_POBCTRL   0x80
+#define WM8960_BUFDCOPEN 0x10
+#define WM8960_BUFIOEN   0x08
+#define WM8960_SOFT_ST   0x04
+#define WM8960_HPSTBY    0x01
+
+/* R29 - Anti-pop 2 */
+#define WM8960_DISOP     0x40
+
+/*
+ * wm8960 register cache
+ * We can't read the WM8960 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8960_reg[WM8960_CACHEREGNUM] = {
+	0x0097, 0x0097, 0x0000, 0x0000,
+	0x0000, 0x0008, 0x0000, 0x000a,
+	0x01c0, 0x0000, 0x00ff, 0x00ff,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x007b, 0x0100, 0x0032,
+	0x0000, 0x00c3, 0x00c3, 0x01c0,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0100, 0x0100, 0x0050, 0x0050,
+	0x0050, 0x0050, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0040, 0x0000,
+	0x0000, 0x0050, 0x0050, 0x0000,
+	0x0002, 0x0037, 0x004d, 0x0080,
+	0x0008, 0x0031, 0x0026, 0x00e9,
+};
+
+struct wm8960_priv {
+	u16 reg_cache[WM8960_CACHEREGNUM];
+	struct snd_soc_codec codec;
+};
+
+/*
+ * read wm8960 register cache
+ */
+static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg == WM8960_RESET)
+		return 0;
+	if (reg >= WM8960_CACHEREGNUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write wm8960 register cache
+ */
+static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec,
+	u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= WM8960_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+static inline unsigned int wm8960_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	return wm8960_read_reg_cache(codec, reg);
+}
+
+/*
+ * write to the WM8960 register space
+ */
+static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D9 WM8960 register offset
+	 *   D8...D0 register data
+	 */
+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+	data[1] = value & 0x00ff;
+
+	wm8960_write_reg_cache(codec, reg, value);
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+#define wm8960_reset(c)	wm8960_write(c, WM8960_RESET, 0)
+
+/* enumerated controls */
+static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
+	"Right Inverted", "Stereo Inversion"};
+static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
+static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"};
+static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
+static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
+
+static const struct soc_enum wm8960_enum[] = {
+	SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
+	SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
+	SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
+	SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
+	SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff),
+	SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc),
+	SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
+};
+
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+
+static const struct snd_kcontrol_new wm8960_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
+		 0, 63, 0, adc_tlv),
+SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
+	6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
+	7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
+		 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1,
+		 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1,
+	7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2,
+		 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2,
+	7, 1, 0),
+SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
+SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
+
+SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
+SOC_ENUM("ADC Polarity", wm8960_enum[1]),
+SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
+SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
+
+SOC_ENUM("DAC Polarity", wm8960_enum[2]),
+
+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
+SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
+SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
+
+SOC_ENUM("ALC Function", wm8960_enum[5]),
+SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
+SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
+SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
+SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
+SOC_ENUM("ALC Mode", wm8960_enum[6]),
+SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
+
+SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
+SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
+
+SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH,
+	0, 127, 0),
+
+SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume",
+	       WM8960_BYPASS1, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume",
+	       WM8960_LOUTMIX, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume",
+	       WM8960_BYPASS2, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume",
+	       WM8960_ROUTMIX, 4, 7, 1, bypass_tlv),
+};
+
+static const struct snd_kcontrol_new wm8960_lin_boost[] = {
+SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_lin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin_boost[] = {
+SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_loutput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_routput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_mono_out[] = {
+SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("LINPUT1"),
+SND_SOC_DAPM_INPUT("RINPUT1"),
+SND_SOC_DAPM_INPUT("LINPUT2"),
+SND_SOC_DAPM_INPUT("RINPUT2"),
+SND_SOC_DAPM_INPUT("LINPUT3"),
+SND_SOC_DAPM_INPUT("RINPUT3"),
+
+SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0),
+
+SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0,
+		   wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)),
+SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0,
+		   wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)),
+
+SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
+		   wm8960_lin, ARRAY_SIZE(wm8960_lin)),
+SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
+		   wm8960_rin, ARRAY_SIZE(wm8960_rin)),
+
+SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0),
+
+SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
+
+SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0,
+	&wm8960_loutput_mixer[0],
+	ARRAY_SIZE(wm8960_loutput_mixer)),
+SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0,
+	&wm8960_routput_mixer[0],
+	ARRAY_SIZE(wm8960_routput_mixer)),
+
+SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0,
+	&wm8960_mono_out[0],
+	ARRAY_SIZE(wm8960_mono_out)),
+
+SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("SPK_LP"),
+SND_SOC_DAPM_OUTPUT("SPK_LN"),
+SND_SOC_DAPM_OUTPUT("HP_L"),
+SND_SOC_DAPM_OUTPUT("HP_R"),
+SND_SOC_DAPM_OUTPUT("SPK_RP"),
+SND_SOC_DAPM_OUTPUT("SPK_RN"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+};
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+	{ "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" },
+	{ "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" },
+	{ "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" },
+
+	{ "Left Input Mixer", "Boost Switch", "Left Boost Mixer", },
+	{ "Left Input Mixer", NULL, "LINPUT1", },  /* Really Boost Switch */
+	{ "Left Input Mixer", NULL, "LINPUT2" },
+	{ "Left Input Mixer", NULL, "LINPUT3" },
+
+	{ "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" },
+	{ "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" },
+	{ "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" },
+
+	{ "Right Input Mixer", "Boost Switch", "Right Boost Mixer", },
+	{ "Right Input Mixer", NULL, "RINPUT1", },  /* Really Boost Switch */
+	{ "Right Input Mixer", NULL, "RINPUT2" },
+	{ "Right Input Mixer", NULL, "LINPUT3" },
+
+	{ "Left ADC", NULL, "Left Input Mixer" },
+	{ "Right ADC", NULL, "Right Input Mixer" },
+
+	{ "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" },
+	{ "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} ,
+	{ "Left Output Mixer", "PCM Playback Switch", "Left DAC" },
+
+	{ "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" },
+	{ "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } ,
+	{ "Right Output Mixer", "PCM Playback Switch", "Right DAC" },
+
+	{ "Mono Output Mixer", "Left Switch", "Left Output Mixer" },
+	{ "Mono Output Mixer", "Right Switch", "Right Output Mixer" },
+
+	{ "LOUT1 PGA", NULL, "Left Output Mixer" },
+	{ "ROUT1 PGA", NULL, "Right Output Mixer" },
+
+	{ "HP_L", NULL, "LOUT1 PGA" },
+	{ "HP_R", NULL, "ROUT1 PGA" },
+
+	{ "Left Speaker PGA", NULL, "Left Output Mixer" },
+	{ "Right Speaker PGA", NULL, "Right Output Mixer" },
+
+	{ "Left Speaker Output", NULL, "Left Speaker PGA" },
+	{ "Right Speaker Output", NULL, "Right Speaker PGA" },
+
+	{ "SPK_LN", NULL, "Left Speaker Output" },
+	{ "SPK_LP", NULL, "Left Speaker Output" },
+	{ "SPK_RN", NULL, "Right Speaker Output" },
+	{ "SPK_RP", NULL, "Right Speaker Output" },
+
+	{ "OUT3", NULL, "Mono Output Mixer", }
+};
+
+static int wm8960_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
+				  ARRAY_SIZE(wm8960_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static int wm8960_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 |= 0x0040;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x0002;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x0001;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x0003;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x0013;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0x0090;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= 0x0080;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x0010;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set iface */
+	wm8960_write(codec, WM8960_IFACE1, iface);
+	return 0;
+}
+
+static int wm8960_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x0004;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x0008;
+		break;
+	}
+
+	/* set iface */
+	wm8960_write(codec, WM8960_IFACE1, iface);
+	return 0;
+}
+
+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7;
+
+	if (mute)
+		wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8);
+	else
+		wm8960_write(codec, WM8960_DACCTL1, mute_reg);
+	return 0;
+}
+
+static int wm8960_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct wm8960_data *pdata = codec->dev->platform_data;
+	u16 reg;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* Set VMID to 2x50k */
+		reg = wm8960_read(codec, WM8960_POWER1);
+		reg &= ~0x180;
+		reg |= 0x80;
+		wm8960_write(codec, WM8960_POWER1, reg);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Enable anti-pop features */
+			wm8960_write(codec, WM8960_APOP1,
+				     WM8960_POBCTRL | WM8960_SOFT_ST |
+				     WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+			/* Discharge HP output */
+			reg = WM8960_DISOP;
+			if (pdata)
+				reg |= pdata->dres << 4;
+			wm8960_write(codec, WM8960_APOP2, reg);
+
+			msleep(400);
+
+			wm8960_write(codec, WM8960_APOP2, 0);
+
+			/* Enable & ramp VMID at 2x50k */
+			reg = wm8960_read(codec, WM8960_POWER1);
+			reg |= 0x80;
+			wm8960_write(codec, WM8960_POWER1, reg);
+			msleep(100);
+
+			/* Enable VREF */
+			wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF);
+
+			/* Disable anti-pop features */
+			wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN);
+		}
+
+		/* Set VMID to 2x250k */
+		reg = wm8960_read(codec, WM8960_POWER1);
+		reg &= ~0x180;
+		reg |= 0x100;
+		wm8960_write(codec, WM8960_POWER1, reg);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* Enable anti-pop features */
+		wm8960_write(codec, WM8960_APOP1,
+			     WM8960_POBCTRL | WM8960_SOFT_ST |
+			     WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+		/* Disable VMID and VREF, let them discharge */
+		wm8960_write(codec, WM8960_POWER1, 0);
+		msleep(600);
+
+		wm8960_write(codec, WM8960_APOP1, 0);
+		break;
+	}
+
+	codec->bias_level = level;
+
+	return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+	u32 pre_div:1;
+	u32 n:4;
+	u32 k:24;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+
+static int pll_factors(unsigned int source, unsigned int target,
+		       struct _pll_div *pll_div)
+{
+	unsigned long long Kpart;
+	unsigned int K, Ndiv, Nmod;
+
+	pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target);
+
+	/* Scale up target to PLL operating frequency */
+	target *= 4;
+
+	Ndiv = target / source;
+	if (Ndiv < 6) {
+		source >>= 1;
+		pll_div->pre_div = 1;
+		Ndiv = target / source;
+	} else
+		pll_div->pre_div = 0;
+
+	if ((Ndiv < 6) || (Ndiv > 12)) {
+		pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv);
+		return -EINVAL;
+	}
+
+	pll_div->n = Ndiv;
+	Nmod = target % source;
+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, source);
+
+	K = Kpart & 0xFFFFFFFF;
+
+	/* Check if we need to round */
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	K /= 10;
+
+	pll_div->k = K;
+
+	pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n",
+		 pll_div->n, pll_div->k, pll_div->pre_div);
+
+	return 0;
+}
+
+static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai,
+		int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+	static struct _pll_div pll_div;
+	int ret;
+
+	if (freq_in && freq_out) {
+		ret = pll_factors(freq_in, freq_out, &pll_div);
+		if (ret != 0)
+			return ret;
+	}
+
+	/* Disable the PLL: even if we are changing the frequency the
+	 * PLL needs to be disabled while we do so. */
+	wm8960_write(codec, WM8960_CLOCK1,
+		     wm8960_read(codec, WM8960_CLOCK1) & ~1);
+	wm8960_write(codec, WM8960_POWER2,
+		     wm8960_read(codec, WM8960_POWER2) & ~1);
+
+	if (!freq_in || !freq_out)
+		return 0;
+
+	reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f;
+	reg |= pll_div.pre_div << 4;
+	reg |= pll_div.n;
+
+	if (pll_div.k) {
+		reg |= 0x20;
+
+		wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
+		wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
+		wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+	}
+	wm8960_write(codec, WM8960_PLL1, reg);
+
+	/* Turn it on */
+	wm8960_write(codec, WM8960_POWER2,
+		     wm8960_read(codec, WM8960_POWER2) | 1);
+	msleep(250);
+	wm8960_write(codec, WM8960_CLOCK1,
+		     wm8960_read(codec, WM8960_CLOCK1) | 1);
+
+	return 0;
+}
+
+static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+		int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	switch (div_id) {
+	case WM8960_SYSCLKSEL:
+		reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe;
+		wm8960_write(codec, WM8960_CLOCK1, reg | div);
+		break;
+	case WM8960_SYSCLKDIV:
+		reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9;
+		wm8960_write(codec, WM8960_CLOCK1, reg | div);
+		break;
+	case WM8960_DACDIV:
+		reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7;
+		wm8960_write(codec, WM8960_CLOCK1, reg | div);
+		break;
+	case WM8960_OPCLKDIV:
+		reg = wm8960_read(codec, WM8960_PLL1) & 0x03f;
+		wm8960_write(codec, WM8960_PLL1, reg | div);
+		break;
+	case WM8960_DCLKDIV:
+		reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f;
+		wm8960_write(codec, WM8960_CLOCK2, reg | div);
+		break;
+	case WM8960_TOCLKSEL:
+		reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd;
+		wm8960_write(codec, WM8960_ADDCTL1, reg | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+#define WM8960_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8960_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8960_dai_ops = {
+	.hw_params = wm8960_hw_params,
+	.digital_mute = wm8960_mute,
+	.set_fmt = wm8960_set_dai_fmt,
+	.set_clkdiv = wm8960_set_dai_clkdiv,
+	.set_pll = wm8960_set_dai_pll,
+};
+
+struct snd_soc_dai wm8960_dai = {
+	.name = "WM8960",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8960_RATES,
+		.formats = WM8960_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8960_RATES,
+		.formats = WM8960_FORMATS,},
+	.ops = &wm8960_dai_ops,
+	.symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8960_dai);
+
+static int wm8960_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8960_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int i;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) {
+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+
+	wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	wm8960_set_bias_level(codec, codec->suspend_bias_level);
+	return 0;
+}
+
+static struct snd_soc_codec *wm8960_codec;
+
+static int wm8960_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (wm8960_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm8960_codec;
+	codec = wm8960_codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, wm8960_snd_controls,
+			     ARRAY_SIZE(wm8960_snd_controls));
+	wm8960_add_widgets(codec);
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	return ret;
+}
+
+/* power down chip */
+static int wm8960_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8960 = {
+	.probe = 	wm8960_probe,
+	.remove = 	wm8960_remove,
+	.suspend = 	wm8960_suspend,
+	.resume =	wm8960_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960);
+
+static int wm8960_register(struct wm8960_priv *wm8960)
+{
+	struct wm8960_data *pdata = wm8960->codec.dev->platform_data;
+	struct snd_soc_codec *codec = &wm8960->codec;
+	int ret;
+	u16 reg;
+
+	if (wm8960_codec) {
+		dev_err(codec->dev, "Another WM8960 is registered\n");
+		return -EINVAL;
+	}
+
+	if (!pdata) {
+		dev_warn(codec->dev, "No platform data supplied\n");
+	} else {
+		if (pdata->dres > WM8960_DRES_MAX) {
+			dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres);
+			pdata->dres = 0;
+		}
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data = wm8960;
+	codec->name = "WM8960";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8960_read_reg_cache;
+	codec->write = wm8960_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8960_set_bias_level;
+	codec->dai = &wm8960_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8960_CACHEREGNUM;
+	codec->reg_cache = &wm8960->reg_cache;
+
+	memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg));
+
+	ret = wm8960_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
+	}
+
+	wm8960_dai.dev = codec->dev;
+
+	wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* Latch the update bits */
+	reg = wm8960_read(codec, WM8960_LINVOL);
+	wm8960_write(codec, WM8960_LINVOL, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_RINVOL);
+	wm8960_write(codec, WM8960_RINVOL, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_LADC);
+	wm8960_write(codec, WM8960_LADC, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_RADC);
+	wm8960_write(codec, WM8960_RADC, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_LDAC);
+	wm8960_write(codec, WM8960_LDAC, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_RDAC);
+	wm8960_write(codec, WM8960_RDAC, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_LOUT1);
+	wm8960_write(codec, WM8960_LOUT1, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_ROUT1);
+	wm8960_write(codec, WM8960_ROUT1, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_LOUT2);
+	wm8960_write(codec, WM8960_LOUT2, reg | 0x100);
+	reg = wm8960_read(codec, WM8960_ROUT2);
+	wm8960_write(codec, WM8960_ROUT2, reg | 0x100);
+
+	wm8960_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm8960_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
+
+	return 0;
+}
+
+static void wm8960_unregister(struct wm8960_priv *wm8960)
+{
+	wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm8960_dai);
+	snd_soc_unregister_codec(&wm8960->codec);
+	kfree(wm8960);
+	wm8960_codec = NULL;
+}
+
+static __devinit int wm8960_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct wm8960_priv *wm8960;
+	struct snd_soc_codec *codec;
+
+	wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL);
+	if (wm8960 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8960->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(i2c, wm8960);
+	codec->control_data = i2c;
+
+	codec->dev = &i2c->dev;
+
+	return wm8960_register(wm8960);
+}
+
+static __devexit int wm8960_i2c_remove(struct i2c_client *client)
+{
+	struct wm8960_priv *wm8960 = i2c_get_clientdata(client);
+	wm8960_unregister(wm8960);
+	return 0;
+}
+
+static const struct i2c_device_id wm8960_i2c_id[] = {
+	{ "wm8960", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
+
+static struct i2c_driver wm8960_i2c_driver = {
+	.driver = {
+		.name = "WM8960 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8960_i2c_probe,
+	.remove =   __devexit_p(wm8960_i2c_remove),
+	.id_table = wm8960_i2c_id,
+};
+
+static int __init wm8960_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&wm8960_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n",
+		       ret);
+	}
+
+	return ret;
+}
+module_init(wm8960_modinit);
+
+static void __exit wm8960_exit(void)
+{
+	i2c_del_driver(&wm8960_i2c_driver);
+}
+module_exit(wm8960_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8960 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h
new file mode 100644
index 00000000000..c9af56c9d9d
--- /dev/null
+++ b/sound/soc/codecs/wm8960.h
@@ -0,0 +1,127 @@
+/*
+ * wm8960.h  --  WM8960 Soc Audio driver
+ *
+ * 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 _WM8960_H
+#define _WM8960_H
+
+/* WM8960 register space */
+
+
+#define WM8960_CACHEREGNUM 	56
+
+#define WM8960_LINVOL		0x0
+#define WM8960_RINVOL		0x1
+#define WM8960_LOUT1		0x2
+#define WM8960_ROUT1		0x3
+#define WM8960_CLOCK1		0x4
+#define WM8960_DACCTL1		0x5
+#define WM8960_DACCTL2		0x6
+#define WM8960_IFACE1		0x7
+#define WM8960_CLOCK2		0x8
+#define WM8960_IFACE2		0x9
+#define WM8960_LDAC		0xa
+#define WM8960_RDAC		0xb
+
+#define WM8960_RESET		0xf
+#define WM8960_3D		0x10
+#define WM8960_ALC1		0x11
+#define WM8960_ALC2		0x12
+#define WM8960_ALC3		0x13
+#define WM8960_NOISEG		0x14
+#define WM8960_LADC		0x15
+#define WM8960_RADC		0x16
+#define WM8960_ADDCTL1		0x17
+#define WM8960_ADDCTL2		0x18
+#define WM8960_POWER1		0x19
+#define WM8960_POWER2		0x1a
+#define WM8960_ADDCTL3		0x1b
+#define WM8960_APOP1		0x1c
+#define WM8960_APOP2		0x1d
+
+#define WM8960_LINPATH		0x20
+#define WM8960_RINPATH		0x21
+#define WM8960_LOUTMIX		0x22
+
+#define WM8960_ROUTMIX		0x25
+#define WM8960_MONOMIX1		0x26
+#define WM8960_MONOMIX2		0x27
+#define WM8960_LOUT2		0x28
+#define WM8960_ROUT2		0x29
+#define WM8960_MONO		0x2a
+#define WM8960_INBMIX1		0x2b
+#define WM8960_INBMIX2		0x2c
+#define WM8960_BYPASS1		0x2d
+#define WM8960_BYPASS2		0x2e
+#define WM8960_POWER3		0x2f
+#define WM8960_ADDCTL4		0x30
+#define WM8960_CLASSD1		0x31
+
+#define WM8960_CLASSD3		0x33
+#define WM8960_PLL1		0x34
+#define WM8960_PLL2		0x35
+#define WM8960_PLL3		0x36
+#define WM8960_PLL4		0x37
+
+
+/*
+ * WM8960 Clock dividers
+ */
+#define WM8960_SYSCLKDIV 		0
+#define WM8960_DACDIV			1
+#define WM8960_OPCLKDIV			2
+#define WM8960_DCLKDIV			3
+#define WM8960_TOCLKSEL			4
+#define WM8960_SYSCLKSEL		5
+
+#define WM8960_SYSCLK_DIV_1		(0 << 1)
+#define WM8960_SYSCLK_DIV_2		(2 << 1)
+
+#define WM8960_SYSCLK_MCLK		(0 << 0)
+#define WM8960_SYSCLK_PLL		(1 << 0)
+
+#define WM8960_DAC_DIV_1		(0 << 3)
+#define WM8960_DAC_DIV_1_5		(1 << 3)
+#define WM8960_DAC_DIV_2		(2 << 3)
+#define WM8960_DAC_DIV_3		(3 << 3)
+#define WM8960_DAC_DIV_4		(4 << 3)
+#define WM8960_DAC_DIV_5_5		(5 << 3)
+#define WM8960_DAC_DIV_6		(6 << 3)
+
+#define WM8960_DCLK_DIV_1_5		(0 << 6)
+#define WM8960_DCLK_DIV_2		(1 << 6)
+#define WM8960_DCLK_DIV_3		(2 << 6)
+#define WM8960_DCLK_DIV_4		(3 << 6)
+#define WM8960_DCLK_DIV_6		(4 << 6)
+#define WM8960_DCLK_DIV_8		(5 << 6)
+#define WM8960_DCLK_DIV_12		(6 << 6)
+#define WM8960_DCLK_DIV_16		(7 << 6)
+
+#define WM8960_TOCLK_F19		(0 << 1)
+#define WM8960_TOCLK_F21		(1 << 1)
+
+#define WM8960_OPCLK_DIV_1		(0 << 0)
+#define WM8960_OPCLK_DIV_2		(1 << 0)
+#define WM8960_OPCLK_DIV_3		(2 << 0)
+#define WM8960_OPCLK_DIV_4		(3 << 0)
+#define WM8960_OPCLK_DIV_5_5		(4 << 0)
+#define WM8960_OPCLK_DIV_6		(5 << 0)
+
+extern struct snd_soc_dai wm8960_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+#define WM8960_DRES_400R 0
+#define WM8960_DRES_200R 1
+#define WM8960_DRES_600R 2
+#define WM8960_DRES_150R 3
+#define WM8960_DRES_MAX  3
+
+struct wm8960_data {
+	int dres;
+};
+
+#endif
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
new file mode 100644
index 00000000000..c05f71803aa
--- /dev/null
+++ b/sound/soc/codecs/wm8988.c
@@ -0,0 +1,1097 @@
+/*
+ * wm8988.c -- WM8988 ALSA SoC audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.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 "wm8988.h"
+
+/*
+ * wm8988 register cache
+ * We can't read the WM8988 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8988_reg[] = {
+	0x0097, 0x0097, 0x0079, 0x0079,  /*  0 */
+	0x0000, 0x0008, 0x0000, 0x000a,  /*  4 */
+	0x0000, 0x0000, 0x00ff, 0x00ff,  /*  8 */
+	0x000f, 0x000f, 0x0000, 0x0000,  /* 12 */
+	0x0000, 0x007b, 0x0000, 0x0032,  /* 16 */
+	0x0000, 0x00c3, 0x00c3, 0x00c0,  /* 20 */
+	0x0000, 0x0000, 0x0000, 0x0000,  /* 24 */
+	0x0000, 0x0000, 0x0000, 0x0000,  /* 28 */
+	0x0000, 0x0000, 0x0050, 0x0050,  /* 32 */
+	0x0050, 0x0050, 0x0050, 0x0050,  /* 36 */
+	0x0079, 0x0079, 0x0079,          /* 40 */
+};
+
+/* codec private data */
+struct wm8988_priv {
+	unsigned int sysclk;
+	struct snd_soc_codec codec;
+	struct snd_pcm_hw_constraint_list *sysclk_constraints;
+	u16 reg_cache[WM8988_NUM_REG];
+};
+
+
+/*
+ * read wm8988 register cache
+ */
+static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg > WM8988_NUM_REG)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write wm8988 register cache
+ */
+static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg > WM8988_NUM_REG)
+		return;
+	cache[reg] = value;
+}
+
+static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D9 WM8753 register offset
+	 *   D8...D0 register data
+	 */
+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+	data[1] = value & 0x00ff;
+
+	wm8988_write_reg_cache(codec, reg, value);
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+#define wm8988_reset(c)	wm8988_write(c, WM8988_RESET, 0)
+
+/*
+ * WM8988 Controls
+ */
+
+static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"};
+static const struct soc_enum bass_boost =
+	SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt);
+
+static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
+static const struct soc_enum bass_filter =
+	SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt);
+
+static const char *treble_txt[] = {"8kHz", "4kHz"};
+static const struct soc_enum treble =
+	SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt);
+
+static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"};
+static const struct soc_enum stereo_3d_lc =
+	SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt);
+
+static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"};
+static const struct soc_enum stereo_3d_uc =
+	SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt);
+
+static const char *stereo_3d_func_txt[] = {"Capture", "Playback"};
+static const struct soc_enum stereo_3d_func =
+	SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt);
+
+static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"};
+static const struct soc_enum alc_func =
+	SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt);
+
+static const char *ng_type_txt[] = {"Constant PGA Gain",
+				    "Mute ADC Output"};
+static const struct soc_enum ng_type =
+	SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt);
+
+static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const struct soc_enum deemph =
+	SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt);
+
+static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert",
+				   "L + R Invert"};
+static const struct soc_enum adcpol =
+	SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt);
+
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new wm8988_snd_controls[] = {
+
+SOC_ENUM("Bass Boost", bass_boost),
+SOC_ENUM("Bass Filter", bass_filter),
+SOC_SINGLE("Bass Volume", WM8988_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8988_TREBLE, 0, 15, 0),
+SOC_ENUM("Treble Cut-off", treble),
+
+SOC_SINGLE("3D Switch", WM8988_3D, 0, 1, 0),
+SOC_SINGLE("3D Volume", WM8988_3D, 1, 15, 0),
+SOC_ENUM("3D Lower Cut-off", stereo_3d_lc),
+SOC_ENUM("3D Upper Cut-off", stereo_3d_uc),
+SOC_ENUM("3D Mode", stereo_3d_func),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8988_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8988_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", alc_func),
+SOC_SINGLE("ALC Capture ZC Switch", WM8988_ALC2, 7, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8988_ALC2, 0, 15, 0),
+SOC_SINGLE("ALC Capture Decay Time", WM8988_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack Time", WM8988_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8988_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", ng_type),
+SOC_SINGLE("ALC Capture NG Switch", WM8988_NGATE, 0, 1, 0),
+
+SOC_SINGLE("ZC Timeout Switch", WM8988_ADCTL1, 0, 1, 0),
+
+SOC_DOUBLE_R_TLV("Capture Digital Volume", WM8988_LADC, WM8988_RADC,
+		 0, 255, 0, adc_tlv),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8988_LINVOL, WM8988_RINVOL,
+		 0, 63, 0, pga_tlv),
+SOC_DOUBLE_R("Capture ZC Switch", WM8988_LINVOL, WM8988_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8988_LINVOL, WM8988_RINVOL, 7, 1, 1),
+
+SOC_ENUM("Playback De-emphasis", deemph),
+
+SOC_ENUM("Capture Polarity", adcpol),
+SOC_SINGLE("Playback 6dB Attenuate", WM8988_ADCDAC, 7, 1, 0),
+SOC_SINGLE("Capture 6dB Attenuate", WM8988_ADCDAC, 8, 1, 0),
+
+SOC_DOUBLE_R_TLV("PCM Volume", WM8988_LDAC, WM8988_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", WM8988_LOUTM1, 4, 7, 1,
+	       bypass_tlv),
+SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", WM8988_LOUTM2, 4, 7, 1,
+	       bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", WM8988_ROUTM1, 4, 7, 1,
+	       bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", WM8988_ROUTM2, 4, 7, 1,
+	       bypass_tlv),
+
+SOC_DOUBLE_R("Output 1 Playback ZC Switch", WM8988_LOUT1V,
+	     WM8988_ROUT1V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 1 Playback Volume", WM8988_LOUT1V, WM8988_ROUT1V,
+		 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R("Output 2 Playback ZC Switch", WM8988_LOUT2V,
+	     WM8988_ROUT2V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V,
+		 0, 127, 0, out_tlv),
+
+};
+
+/*
+ * DAPM Controls
+ */
+
+static int wm8988_lrc_control(struct snd_soc_dapm_widget *w,
+			      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2);
+
+	/* Use the DAC to gate LRC if active, otherwise use ADC */
+	if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180)
+		adctl2 &= ~0x4;
+	else
+		adctl2 |= 0x4;
+
+	return wm8988_write(codec, WM8988_ADCTL2, adctl2);
+}
+
+static const char *wm8988_line_texts[] = {
+	"Line 1", "Line 2", "PGA", "Differential"};
+
+static const unsigned int wm8988_line_values[] = {
+	0, 1, 3, 4};
+
+static const struct soc_enum wm8988_lline_enum =
+	SOC_VALUE_ENUM_SINGLE(WM8988_LOUTM1, 0, 7,
+			      ARRAY_SIZE(wm8988_line_texts),
+			      wm8988_line_texts,
+			      wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_left_line_controls =
+	SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+static const struct soc_enum wm8988_rline_enum =
+	SOC_VALUE_ENUM_SINGLE(WM8988_ROUTM1, 0, 7,
+			      ARRAY_SIZE(wm8988_line_texts),
+			      wm8988_line_texts,
+			      wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_right_line_controls =
+	SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch", WM8988_LOUTM1, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_LOUTM1, 7, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch", WM8988_LOUTM2, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8988_right_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left Playback Switch", WM8988_ROUTM1, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_ROUTM1, 7, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", WM8988_ROUTM2, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_ROUTM2, 7, 1, 0),
+};
+
+static const char *wm8988_pga_sel[] = {"Line 1", "Line 2", "Differential"};
+static const unsigned int wm8988_pga_val[] = { 0, 1, 3 };
+
+/* Left PGA Mux */
+static const struct soc_enum wm8988_lpga_enum =
+	SOC_VALUE_ENUM_SINGLE(WM8988_LADCIN, 6, 3,
+			      ARRAY_SIZE(wm8988_pga_sel),
+			      wm8988_pga_sel,
+			      wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_left_pga_controls =
+	SOC_DAPM_VALUE_ENUM("Route", wm8988_lpga_enum);
+
+/* Right PGA Mux */
+static const struct soc_enum wm8988_rpga_enum =
+	SOC_VALUE_ENUM_SINGLE(WM8988_RADCIN, 6, 3,
+			      ARRAY_SIZE(wm8988_pga_sel),
+			      wm8988_pga_sel,
+			      wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_right_pga_controls =
+	SOC_DAPM_VALUE_ENUM("Route", wm8988_rpga_enum);
+
+/* Differential Mux */
+static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"};
+static const struct soc_enum diffmux =
+	SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel);
+static const struct snd_kcontrol_new wm8988_diffmux_controls =
+	SOC_DAPM_ENUM("Route", diffmux);
+
+/* Mono ADC Mux */
+static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)",
+	"Mono (Right)", "Digital Mono"};
+static const struct soc_enum monomux =
+	SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux);
+static const struct snd_kcontrol_new wm8988_monomux_controls =
+	SOC_DAPM_ENUM("Route", monomux);
+
+static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = {
+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0),
+
+	SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+		&wm8988_diffmux_controls),
+	SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+		&wm8988_monomux_controls),
+	SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+		&wm8988_monomux_controls),
+
+	SND_SOC_DAPM_MUX("Left PGA Mux", WM8988_PWR1, 5, 0,
+		&wm8988_left_pga_controls),
+	SND_SOC_DAPM_MUX("Right PGA Mux", WM8988_PWR1, 4, 0,
+		&wm8988_right_pga_controls),
+
+	SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+		&wm8988_left_line_controls),
+	SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+		&wm8988_right_line_controls),
+
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8988_PWR1, 2, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8988_PWR1, 3, 0),
+
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8988_PWR2, 7, 0),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8988_PWR2, 8, 0),
+
+	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+		&wm8988_left_mixer_controls[0],
+		ARRAY_SIZE(wm8988_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+		&wm8988_right_mixer_controls[0],
+		ARRAY_SIZE(wm8988_right_mixer_controls)),
+
+	SND_SOC_DAPM_PGA("Right Out 2", WM8988_PWR2, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 2", WM8988_PWR2, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Out 1", WM8988_PWR2, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 1", WM8988_PWR2, 6, 0, NULL, 0),
+
+	SND_SOC_DAPM_POST("LRC control", wm8988_lrc_control),
+
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+	SND_SOC_DAPM_OUTPUT("VREF"),
+
+	SND_SOC_DAPM_INPUT("LINPUT1"),
+	SND_SOC_DAPM_INPUT("LINPUT2"),
+	SND_SOC_DAPM_INPUT("RINPUT1"),
+	SND_SOC_DAPM_INPUT("RINPUT2"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left PGA Mux", "Line 1", "LINPUT1" },
+	{ "Left PGA Mux", "Line 2", "LINPUT2" },
+	{ "Left PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Right PGA Mux", "Line 1", "RINPUT1" },
+	{ "Right PGA Mux", "Line 2", "RINPUT2" },
+	{ "Right PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Differential Mux", "Line 1", "LINPUT1" },
+	{ "Differential Mux", "Line 1", "RINPUT1" },
+	{ "Differential Mux", "Line 2", "LINPUT2" },
+	{ "Differential Mux", "Line 2", "RINPUT2" },
+
+	{ "Left ADC Mux", "Stereo", "Left PGA Mux" },
+	{ "Left ADC Mux", "Mono (Left)", "Left PGA Mux" },
+	{ "Left ADC Mux", "Digital Mono", "Left PGA Mux" },
+
+	{ "Right ADC Mux", "Stereo", "Right PGA Mux" },
+	{ "Right ADC Mux", "Mono (Right)", "Right PGA Mux" },
+	{ "Right ADC Mux", "Digital Mono", "Right PGA Mux" },
+
+	{ "Left ADC", NULL, "Left ADC Mux" },
+	{ "Right ADC", NULL, "Right ADC Mux" },
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left Mixer", "Playback Switch", "Left DAC" },
+	{ "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Left Mixer", "Right Playback Switch", "Right DAC" },
+	{ "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "Right Mixer", "Left Playback Switch", "Left DAC" },
+	{ "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Right Mixer", "Playback Switch", "Right DAC" },
+	{ "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "Left Out 1", NULL, "Left Mixer" },
+	{ "LOUT1", NULL, "Left Out 1" },
+	{ "Right Out 1", NULL, "Right Mixer" },
+	{ "ROUT1", NULL, "Right Out 1" },
+
+	{ "Left Out 2", NULL, "Left Mixer" },
+	{ "LOUT2", NULL, "Left Out 2" },
+	{ "Right Out 2", NULL, "Right Mixer" },
+	{ "ROUT2", NULL, "Right Out 2" },
+};
+
+struct _coeff_div {
+	u32 mclk;
+	u32 rate;
+	u16 fs;
+	u8 sr:5;
+	u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+	/* 8k */
+	{12288000, 8000, 1536, 0x6, 0x0},
+	{11289600, 8000, 1408, 0x16, 0x0},
+	{18432000, 8000, 2304, 0x7, 0x0},
+	{16934400, 8000, 2112, 0x17, 0x0},
+	{12000000, 8000, 1500, 0x6, 0x1},
+
+	/* 11.025k */
+	{11289600, 11025, 1024, 0x18, 0x0},
+	{16934400, 11025, 1536, 0x19, 0x0},
+	{12000000, 11025, 1088, 0x19, 0x1},
+
+	/* 16k */
+	{12288000, 16000, 768, 0xa, 0x0},
+	{18432000, 16000, 1152, 0xb, 0x0},
+	{12000000, 16000, 750, 0xa, 0x1},
+
+	/* 22.05k */
+	{11289600, 22050, 512, 0x1a, 0x0},
+	{16934400, 22050, 768, 0x1b, 0x0},
+	{12000000, 22050, 544, 0x1b, 0x1},
+
+	/* 32k */
+	{12288000, 32000, 384, 0xc, 0x0},
+	{18432000, 32000, 576, 0xd, 0x0},
+	{12000000, 32000, 375, 0xa, 0x1},
+
+	/* 44.1k */
+	{11289600, 44100, 256, 0x10, 0x0},
+	{16934400, 44100, 384, 0x11, 0x0},
+	{12000000, 44100, 272, 0x11, 0x1},
+
+	/* 48k */
+	{12288000, 48000, 256, 0x0, 0x0},
+	{18432000, 48000, 384, 0x1, 0x0},
+	{12000000, 48000, 250, 0x0, 0x1},
+
+	/* 88.2k */
+	{11289600, 88200, 128, 0x1e, 0x0},
+	{16934400, 88200, 192, 0x1f, 0x0},
+	{12000000, 88200, 136, 0x1f, 0x1},
+
+	/* 96k */
+	{12288000, 96000, 128, 0xe, 0x0},
+	{18432000, 96000, 192, 0xf, 0x0},
+	{12000000, 96000, 125, 0xe, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+			return i;
+	}
+
+	return -EINVAL;
+}
+
+/* The set of rates we can generate from the above for each SYSCLK */
+
+static unsigned int rates_12288[] = {
+	8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12288 = {
+	.count	= ARRAY_SIZE(rates_12288),
+	.list	= rates_12288,
+};
+
+static unsigned int rates_112896[] = {
+	8000, 11025, 22050, 44100,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_112896 = {
+	.count	= ARRAY_SIZE(rates_112896),
+	.list	= rates_112896,
+};
+
+static unsigned int rates_12[] = {
+	8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+	48000, 88235, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12 = {
+	.count	= ARRAY_SIZE(rates_12),
+	.list	= rates_12,
+};
+
+/*
+ * Note that this should be called from init rather than from hw_params.
+ */
+static int wm8988_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 wm8988_priv *wm8988 = codec->private_data;
+
+	switch (freq) {
+	case 11289600:
+	case 18432000:
+	case 22579200:
+	case 36864000:
+		wm8988->sysclk_constraints = &constraints_112896;
+		wm8988->sysclk = freq;
+		return 0;
+
+	case 12288000:
+	case 16934400:
+	case 24576000:
+	case 33868800:
+		wm8988->sysclk_constraints = &constraints_12288;
+		wm8988->sysclk = freq;
+		return 0;
+
+	case 12000000:
+	case 24000000:
+		wm8988->sysclk_constraints = &constraints_12;
+		wm8988->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int wm8988_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 = 0x0040;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x0002;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x0001;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x0003;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x0013;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0x0090;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= 0x0080;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x0010;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8988_write(codec, WM8988_IFACE, iface);
+	return 0;
+}
+
+static int wm8988_pcm_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8988_priv *wm8988 = codec->private_data;
+
+	/* The set of sample rates that can be supported depends on the
+	 * MCLK supplied to the CODEC - enforce this.
+	 */
+	if (!wm8988->sysclk) {
+		dev_err(codec->dev,
+			"No MCLK configured, call set_sysclk() on init\n");
+		return -EINVAL;
+	}
+
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_RATE,
+				   wm8988->sysclk_constraints);
+
+	return 0;
+}
+
+static int wm8988_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_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct wm8988_priv *wm8988 = codec->private_data;
+	u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3;
+	u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180;
+	int coeff;
+
+	coeff = get_coeff(wm8988->sysclk, params_rate(params));
+	if (coeff < 0) {
+		coeff = get_coeff(wm8988->sysclk / 2, params_rate(params));
+		srate |= 0x40;
+	}
+	if (coeff < 0) {
+		dev_err(codec->dev,
+			"Unable to configure sample rate %dHz with %dHz MCLK\n",
+			params_rate(params), wm8988->sysclk);
+		return coeff;
+	}
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x0004;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x0008;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x000c;
+		break;
+	}
+
+	/* set iface & srate */
+	wm8988_write(codec, WM8988_IFACE, iface);
+	if (coeff >= 0)
+		wm8988_write(codec, WM8988_SRATE, srate |
+			(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+	return 0;
+}
+
+static int wm8988_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7;
+
+	if (mute)
+		wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8);
+	else
+		wm8988_write(codec, WM8988_ADCDAC, mute_reg);
+	return 0;
+}
+
+static int wm8988_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VREF, VMID=2x50k, digital enabled */
+		wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* VREF, VMID=2x5k */
+			wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
+
+			/* Charge caps */
+			msleep(100);
+		}
+
+		/* VREF, VMID=2*500k, digital stopped */
+		wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		wm8988_write(codec, WM8988_PWR1, 0x0000);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8988_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8988_ops = {
+	.startup = wm8988_pcm_startup,
+	.hw_params = wm8988_pcm_hw_params,
+	.set_fmt = wm8988_set_dai_fmt,
+	.set_sysclk = wm8988_set_dai_sysclk,
+	.digital_mute = wm8988_mute,
+};
+
+struct snd_soc_dai wm8988_dai = {
+	.name = "WM8988",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8988_RATES,
+		.formats = WM8988_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8988_RATES,
+		.formats = WM8988_FORMATS,
+	 },
+	.ops = &wm8988_ops,
+	.symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8988_dai);
+
+static int wm8988_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8988_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int i;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 0; i < WM8988_NUM_REG; i++) {
+		if (i == WM8988_RESET)
+			continue;
+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+
+	wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+static struct snd_soc_codec *wm8988_codec;
+
+static int wm8988_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (wm8988_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm8988_codec;
+	codec = wm8988_codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, wm8988_snd_controls,
+				ARRAY_SIZE(wm8988_snd_controls));
+	snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets,
+				  ARRAY_SIZE(wm8988_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	return ret;
+}
+
+static int wm8988_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8988 = {
+	.probe = 	wm8988_probe,
+	.remove = 	wm8988_remove,
+	.suspend = 	wm8988_suspend,
+	.resume =	wm8988_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988);
+
+static int wm8988_register(struct wm8988_priv *wm8988)
+{
+	struct snd_soc_codec *codec = &wm8988->codec;
+	int ret;
+	u16 reg;
+
+	if (wm8988_codec) {
+		dev_err(codec->dev, "Another WM8988 is registered\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data = wm8988;
+	codec->name = "WM8988";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8988_read_reg_cache;
+	codec->write = wm8988_write;
+	codec->dai = &wm8988_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache);
+	codec->reg_cache = &wm8988->reg_cache;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8988_set_bias_level;
+
+	memcpy(codec->reg_cache, wm8988_reg,
+	       sizeof(wm8988_reg));
+
+	ret = wm8988_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
+	}
+
+	/* set the update bits (we always update left then right) */
+	reg = wm8988_read_reg_cache(codec, WM8988_RADC);
+	wm8988_write(codec, WM8988_RADC, reg | 0x100);
+	reg = wm8988_read_reg_cache(codec, WM8988_RDAC);
+	wm8988_write(codec, WM8988_RDAC, reg | 0x0100);
+	reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V);
+	wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100);
+	reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V);
+	wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100);
+	reg = wm8988_read_reg_cache(codec, WM8988_RINVOL);
+	wm8988_write(codec, WM8988_RINVOL, reg | 0x0100);
+
+	wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY);
+
+	wm8988_dai.dev = codec->dev;
+
+	wm8988_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm8988_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
+
+	return 0;
+
+err:
+	kfree(wm8988);
+	return ret;
+}
+
+static void wm8988_unregister(struct wm8988_priv *wm8988)
+{
+	wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm8988_dai);
+	snd_soc_unregister_codec(&wm8988->codec);
+	kfree(wm8988);
+	wm8988_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static int wm8988_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct wm8988_priv *wm8988;
+	struct snd_soc_codec *codec;
+
+	wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+	if (wm8988 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8988->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(i2c, wm8988);
+	codec->control_data = i2c;
+
+	codec->dev = &i2c->dev;
+
+	return wm8988_register(wm8988);
+}
+
+static int wm8988_i2c_remove(struct i2c_client *client)
+{
+	struct wm8988_priv *wm8988 = i2c_get_clientdata(client);
+	wm8988_unregister(wm8988);
+	return 0;
+}
+
+static const struct i2c_device_id wm8988_i2c_id[] = {
+	{ "wm8988", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id);
+
+static struct i2c_driver wm8988_i2c_driver = {
+	.driver = {
+		.name = "WM8988",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8988_i2c_probe,
+	.remove = wm8988_i2c_remove,
+	.id_table = wm8988_i2c_id,
+};
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int wm8988_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+
+static int __devinit wm8988_spi_probe(struct spi_device *spi)
+{
+	struct wm8988_priv *wm8988;
+	struct snd_soc_codec *codec;
+
+	wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+	if (wm8988 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8988->codec;
+	codec->hw_write = (hw_write_t)wm8988_spi_write;
+	codec->control_data = spi;
+	codec->dev = &spi->dev;
+
+	spi->dev.driver_data = wm8988;
+
+	return wm8988_register(wm8988);
+}
+
+static int __devexit wm8988_spi_remove(struct spi_device *spi)
+{
+	struct wm8988_priv *wm8988 = spi->dev.driver_data;
+
+	wm8988_unregister(wm8988);
+
+	return 0;
+}
+
+static struct spi_driver wm8988_spi_driver = {
+	.driver = {
+		.name	= "wm8988",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8988_spi_probe,
+	.remove		= __devexit_p(wm8988_spi_remove),
+};
+#endif
+
+static int __init wm8988_modinit(void)
+{
+	int ret;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&wm8988_i2c_driver);
+	if (ret != 0)
+		pr_err("WM8988: Unable to register I2C driver: %d\n", ret);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8988_spi_driver);
+	if (ret != 0)
+		pr_err("WM8988: Unable to register SPI driver: %d\n", ret);
+#endif
+	return ret;
+}
+module_init(wm8988_modinit);
+
+static void __exit wm8988_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8988_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8988_spi_driver);
+#endif
+}
+module_exit(wm8988_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8988 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8988.h b/sound/soc/codecs/wm8988.h
new file mode 100644
index 00000000000..4552d37fdd4
--- /dev/null
+++ b/sound/soc/codecs/wm8988.h
@@ -0,0 +1,60 @@
+/*
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _WM8988_H
+#define _WM8988_H
+
+/* WM8988 register space */
+
+#define WM8988_LINVOL    0x00
+#define WM8988_RINVOL    0x01
+#define WM8988_LOUT1V    0x02
+#define WM8988_ROUT1V    0x03
+#define WM8988_ADCDAC    0x05
+#define WM8988_IFACE     0x07
+#define WM8988_SRATE     0x08
+#define WM8988_LDAC      0x0a
+#define WM8988_RDAC      0x0b
+#define WM8988_BASS      0x0c
+#define WM8988_TREBLE    0x0d
+#define WM8988_RESET     0x0f
+#define WM8988_3D        0x10
+#define WM8988_ALC1      0x11
+#define WM8988_ALC2      0x12
+#define WM8988_ALC3      0x13
+#define WM8988_NGATE     0x14
+#define WM8988_LADC      0x15
+#define WM8988_RADC      0x16
+#define WM8988_ADCTL1    0x17
+#define WM8988_ADCTL2    0x18
+#define WM8988_PWR1      0x19
+#define WM8988_PWR2      0x1a
+#define WM8988_ADCTL3    0x1b
+#define WM8988_ADCIN     0x1f
+#define WM8988_LADCIN    0x20
+#define WM8988_RADCIN    0x21
+#define WM8988_LOUTM1    0x22
+#define WM8988_LOUTM2    0x23
+#define WM8988_ROUTM1    0x24
+#define WM8988_ROUTM2    0x25
+#define WM8988_LOUT2V    0x28
+#define WM8988_ROUT2V    0x29
+#define WM8988_LPPB      0x43
+#define WM8988_NUM_REG   0x44
+
+#define WM8988_SYSCLK	0
+
+extern struct snd_soc_dai wm8988_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8988;
+
+#endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 40cd274eb1e..d029818350e 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -998,7 +998,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 
 	if ((Ndiv < 6) || (Ndiv > 12))
 		printk(KERN_WARNING
-		"WM8990 N value outwith recommended range! N = %d\n", Ndiv);
+		"WM8990 N value outwith recommended range! N = %u\n", Ndiv);
 
 	pll_div->n = Ndiv;
 	Nmod = target % source;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
new file mode 100644
index 00000000000..86fc57e25f9
--- /dev/null
+++ b/sound/soc/codecs/wm9081.c
@@ -0,0 +1,1534 @@
+/*
+ * wm9081.c  --  WM9081 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <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 <sound/wm9081.h>
+#include "wm9081.h"
+
+static u16 wm9081_reg_defaults[] = {
+	0x0000,     /* R0  - Software Reset */
+	0x0000,     /* R1 */
+	0x00B9,     /* R2  - Analogue Lineout */
+	0x00B9,     /* R3  - Analogue Speaker PGA */
+	0x0001,     /* R4  - VMID Control */
+	0x0068,     /* R5  - Bias Control 1 */
+	0x0000,     /* R6 */
+	0x0000,     /* R7  - Analogue Mixer */
+	0x0000,     /* R8  - Anti Pop Control */
+	0x01DB,     /* R9  - Analogue Speaker 1 */
+	0x0018,     /* R10 - Analogue Speaker 2 */
+	0x0180,     /* R11 - Power Management */
+	0x0000,     /* R12 - Clock Control 1 */
+	0x0038,     /* R13 - Clock Control 2 */
+	0x4000,     /* R14 - Clock Control 3 */
+	0x0000,     /* R15 */
+	0x0000,     /* R16 - FLL Control 1 */
+	0x0200,     /* R17 - FLL Control 2 */
+	0x0000,     /* R18 - FLL Control 3 */
+	0x0204,     /* R19 - FLL Control 4 */
+	0x0000,     /* R20 - FLL Control 5 */
+	0x0000,     /* R21 */
+	0x0000,     /* R22 - Audio Interface 1 */
+	0x0002,     /* R23 - Audio Interface 2 */
+	0x0008,     /* R24 - Audio Interface 3 */
+	0x0022,     /* R25 - Audio Interface 4 */
+	0x0000,     /* R26 - Interrupt Status */
+	0x0006,     /* R27 - Interrupt Status Mask */
+	0x0000,     /* R28 - Interrupt Polarity */
+	0x0000,     /* R29 - Interrupt Control */
+	0x00C0,     /* R30 - DAC Digital 1 */
+	0x0008,     /* R31 - DAC Digital 2 */
+	0x09AF,     /* R32 - DRC 1 */
+	0x4201,     /* R33 - DRC 2 */
+	0x0000,     /* R34 - DRC 3 */
+	0x0000,     /* R35 - DRC 4 */
+	0x0000,     /* R36 */
+	0x0000,     /* R37 */
+	0x0000,     /* R38 - Write Sequencer 1 */
+	0x0000,     /* R39 - Write Sequencer 2 */
+	0x0002,     /* R40 - MW Slave 1 */
+	0x0000,     /* R41 */
+	0x0000,     /* R42 - EQ 1 */
+	0x0000,     /* R43 - EQ 2 */
+	0x0FCA,     /* R44 - EQ 3 */
+	0x0400,     /* R45 - EQ 4 */
+	0x00B8,     /* R46 - EQ 5 */
+	0x1EB5,     /* R47 - EQ 6 */
+	0xF145,     /* R48 - EQ 7 */
+	0x0B75,     /* R49 - EQ 8 */
+	0x01C5,     /* R50 - EQ 9 */
+	0x169E,     /* R51 - EQ 10 */
+	0xF829,     /* R52 - EQ 11 */
+	0x07AD,     /* R53 - EQ 12 */
+	0x1103,     /* R54 - EQ 13 */
+	0x1C58,     /* R55 - EQ 14 */
+	0xF373,     /* R56 - EQ 15 */
+	0x0A54,     /* R57 - EQ 16 */
+	0x0558,     /* R58 - EQ 17 */
+	0x0564,     /* R59 - EQ 18 */
+	0x0559,     /* R60 - EQ 19 */
+	0x4000,     /* R61 - EQ 20 */
+};
+
+static struct {
+	int ratio;
+	int clk_sys_rate;
+} clk_sys_rates[] = {
+	{ 64,   0 },
+	{ 128,  1 },
+	{ 192,  2 },
+	{ 256,  3 },
+	{ 384,  4 },
+	{ 512,  5 },
+	{ 768,  6 },
+	{ 1024, 7 },
+	{ 1408, 8 },
+	{ 1536, 9 },
+};
+
+static struct {
+	int rate;
+	int sample_rate;
+} sample_rates[] = {
+	{ 8000,  0  },
+	{ 11025, 1  },
+	{ 12000, 2  },
+	{ 16000, 3  },
+	{ 22050, 4  },
+	{ 24000, 5  },
+	{ 32000, 6  },
+	{ 44100, 7  },
+	{ 48000, 8  },
+	{ 88200, 9  },
+	{ 96000, 10 },
+};
+
+static struct {
+	int div; /* *10 due to .5s */
+	int bclk_div;
+} bclk_divs[] = {
+	{ 10,  0  },
+	{ 15,  1  },
+	{ 20,  2  },
+	{ 30,  3  },
+	{ 40,  4  },
+	{ 50,  5  },
+	{ 55,  6  },
+	{ 60,  7  },
+	{ 80,  8  },
+	{ 100, 9  },
+	{ 110, 10 },
+	{ 120, 11 },
+	{ 160, 12 },
+	{ 200, 13 },
+	{ 220, 14 },
+	{ 240, 15 },
+	{ 250, 16 },
+	{ 300, 17 },
+	{ 320, 18 },
+	{ 440, 19 },
+	{ 480, 20 },
+};
+
+struct wm9081_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM9081_MAX_REGISTER + 1];
+	int sysclk_source;
+	int mclk_rate;
+	int sysclk_rate;
+	int fs;
+	int bclk;
+	int master;
+	int fll_fref;
+	int fll_fout;
+	struct wm9081_retune_mobile_config *retune;
+};
+
+static int wm9081_reg_is_volatile(int reg)
+{
+	switch (reg) {
+	default:
+		return 0;
+	}
+}
+
+static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec,
+					  unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	BUG_ON(reg > WM9081_MAX_REGISTER);
+	return cache[reg];
+}
+
+static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg)
+{
+	struct i2c_msg xfer[2];
+	u16 data;
+	int ret;
+	struct i2c_client *client = codec->control_data;
+
+	BUG_ON(reg > WM9081_MAX_REGISTER);
+
+	/* Write register */
+	xfer[0].addr = client->addr;
+	xfer[0].flags = 0;
+	xfer[0].len = 1;
+	xfer[0].buf = &reg;
+
+	/* Read data */
+	xfer[1].addr = client->addr;
+	xfer[1].flags = I2C_M_RD;
+	xfer[1].len = 2;
+	xfer[1].buf = (u8 *)&data;
+
+	ret = i2c_transfer(client->adapter, xfer, 2);
+	if (ret != 2) {
+		dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
+		return 0;
+	}
+
+	return (data >> 8) | ((data & 0xff) << 8);
+}
+
+static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	if (wm9081_reg_is_volatile(reg))
+		return wm9081_read_hw(codec, reg);
+	else
+		return wm9081_read_reg_cache(codec, reg);
+}
+
+static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg,
+			unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	u8 data[3];
+
+	BUG_ON(reg > WM9081_MAX_REGISTER);
+
+	if (!wm9081_reg_is_volatile(reg))
+		cache[reg] = value;
+
+	data[0] = reg;
+	data[1] = value >> 8;
+	data[2] = value & 0x00ff;
+
+	if (codec->hw_write(codec->control_data, data, 3) == 3)
+		return 0;
+	else
+		return -EIO;
+}
+
+static int wm9081_reset(struct snd_soc_codec *codec)
+{
+	return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_out_tlv, -2250, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
+static unsigned int drc_max_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 0, TLV_DB_SCALE_ITEM(1200, 0, 0),
+	1, 1, TLV_DB_SCALE_ITEM(1800, 0, 0),
+	2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+	3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -300, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+
+static const DECLARE_TLV_DB_SCALE(in_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
+
+static const char *drc_high_text[] = {
+	"1",
+	"1/2",
+	"1/4",
+	"1/8",
+	"1/16",
+	"0",
+};
+
+static const struct soc_enum drc_high =
+	SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text);
+
+static const char *drc_low_text[] = {
+	"1",
+	"1/2",
+	"1/4",
+	"1/8",
+	"0",
+};
+
+static const struct soc_enum drc_low =
+	SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text);
+
+static const char *drc_atk_text[] = {
+	"181us",
+	"181us",
+	"363us",
+	"726us",
+	"1.45ms",
+	"2.9ms",
+	"5.8ms",
+	"11.6ms",
+	"23.2ms",
+	"46.4ms",
+	"92.8ms",
+	"185.6ms",
+};
+
+static const struct soc_enum drc_atk =
+	SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text);
+
+static const char *drc_dcy_text[] = {
+	"186ms",
+	"372ms",
+	"743ms",
+	"1.49s",
+	"2.97s",
+	"5.94s",
+	"11.89s",
+	"23.78s",
+	"47.56s",
+};
+
+static const struct soc_enum drc_dcy =
+	SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text);
+
+static const char *drc_qr_dcy_text[] = {
+	"0.725ms",
+	"1.45ms",
+	"5.8ms",
+};
+
+static const struct soc_enum drc_qr_dcy =
+	SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text);
+
+static const char *dac_deemph_text[] = {
+	"None",
+	"32kHz",
+	"44.1kHz",
+	"48kHz",
+};
+
+static const struct soc_enum dac_deemph =
+	SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text);
+
+static const char *speaker_mode_text[] = {
+	"Class D",
+	"Class AB",
+};
+
+static const struct soc_enum speaker_mode =
+	SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text);
+
+static int speaker_mode_get(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg;
+
+	reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+	if (reg & WM9081_SPK_MODE)
+		ucontrol->value.integer.value[0] = 1;
+	else
+		ucontrol->value.integer.value[0] = 0;
+
+	return 0;
+}
+
+/*
+ * Stop any attempts to change speaker mode while the speaker is enabled.
+ *
+ * We also have some special anti-pop controls dependant on speaker
+ * mode which must be changed along with the mode.
+ */
+static int speaker_mode_put(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+	unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+
+	/* Are we changing anything? */
+	if (ucontrol->value.integer.value[0] ==
+	    ((reg2 & WM9081_SPK_MODE) != 0))
+		return 0;
+
+	/* Don't try to change modes while enabled */
+	if (reg_pwr & WM9081_SPK_ENA)
+		return -EINVAL;
+
+	if (ucontrol->value.integer.value[0]) {
+		/* Class AB */
+		reg2 &= ~(WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL);
+		reg2 |= WM9081_SPK_MODE;
+	} else {
+		/* Class D */
+		reg2 |= WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL;
+		reg2 &= ~WM9081_SPK_MODE;
+	}
+
+	wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2);
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new wm9081_snd_controls[] = {
+SOC_SINGLE_TLV("IN1 Volume", WM9081_ANALOGUE_MIXER, 1, 1, 1, in_tlv),
+SOC_SINGLE_TLV("IN2 Volume", WM9081_ANALOGUE_MIXER, 3, 1, 1, in_tlv),
+
+SOC_SINGLE_TLV("Playback Volume", WM9081_DAC_DIGITAL_1, 1, 96, 0, dac_tlv),
+
+SOC_SINGLE("LINEOUT Switch", WM9081_ANALOGUE_LINEOUT, 7, 1, 1),
+SOC_SINGLE("LINEOUT ZC Switch", WM9081_ANALOGUE_LINEOUT, 6, 1, 0),
+SOC_SINGLE_TLV("LINEOUT Volume", WM9081_ANALOGUE_LINEOUT, 0, 63, 0, out_tlv),
+
+SOC_SINGLE("DRC Switch", WM9081_DRC_1, 15, 1, 0),
+SOC_ENUM("DRC High Slope", drc_high),
+SOC_ENUM("DRC Low Slope", drc_low),
+SOC_SINGLE_TLV("DRC Input Volume", WM9081_DRC_4, 5, 60, 1, drc_in_tlv),
+SOC_SINGLE_TLV("DRC Output Volume", WM9081_DRC_4, 0, 30, 1, drc_out_tlv),
+SOC_SINGLE_TLV("DRC Minimum Volume", WM9081_DRC_2, 2, 3, 1, drc_min_tlv),
+SOC_SINGLE_TLV("DRC Maximum Volume", WM9081_DRC_2, 0, 3, 0, drc_max_tlv),
+SOC_ENUM("DRC Attack", drc_atk),
+SOC_ENUM("DRC Decay", drc_dcy),
+SOC_SINGLE("DRC Quick Release Switch", WM9081_DRC_1, 2, 1, 0),
+SOC_SINGLE_TLV("DRC Quick Release Volume", WM9081_DRC_2, 6, 3, 0, drc_qr_tlv),
+SOC_ENUM("DRC Quick Release Decay", drc_qr_dcy),
+SOC_SINGLE_TLV("DRC Startup Volume", WM9081_DRC_1, 6, 18, 0, drc_startup_tlv),
+
+SOC_SINGLE("EQ Switch", WM9081_EQ_1, 0, 1, 0),
+
+SOC_SINGLE("Speaker DC Volume", WM9081_ANALOGUE_SPEAKER_1, 3, 5, 0),
+SOC_SINGLE("Speaker AC Volume", WM9081_ANALOGUE_SPEAKER_1, 0, 5, 0),
+SOC_SINGLE("Speaker Switch", WM9081_ANALOGUE_SPEAKER_PGA, 7, 1, 1),
+SOC_SINGLE("Speaker ZC Switch", WM9081_ANALOGUE_SPEAKER_PGA, 6, 1, 0),
+SOC_SINGLE_TLV("Speaker Volume", WM9081_ANALOGUE_SPEAKER_PGA, 0, 63, 0,
+	       out_tlv),
+SOC_ENUM("DAC Deemphasis", dac_deemph),
+SOC_ENUM_EXT("Speaker Mode", speaker_mode, speaker_mode_get, speaker_mode_put),
+};
+
+static const struct snd_kcontrol_new wm9081_eq_controls[] = {
+SOC_SINGLE_TLV("EQ1 Volume", WM9081_EQ_1, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 Volume", WM9081_EQ_1, 6, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 Volume", WM9081_EQ_1, 1, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 Volume", WM9081_EQ_2, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ5 Volume", WM9081_EQ_2, 6, 24, 0, eq_tlv),
+};
+
+static const struct snd_kcontrol_new mixer[] = {
+SOC_DAPM_SINGLE("IN1 Switch", WM9081_ANALOGUE_MIXER, 0, 1, 0),
+SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0),
+};
+
+static int speaker_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		reg |= WM9081_SPK_ENA;
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		reg &= ~WM9081_SPK_ENA;
+		break;
+	}
+
+	wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg);
+
+	return 0;
+}
+
+struct _fll_div {
+	u16 fll_fratio;
+	u16 fll_outdiv;
+	u16 fll_clk_ref_div;
+	u16 n;
+	u16 k;
+};
+
+/* The size in bits of the FLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static struct {
+	unsigned int min;
+	unsigned int max;
+	u16 fll_fratio;
+	int ratio;
+} fll_fratios[] = {
+	{       0,    64000, 4, 16 },
+	{   64000,   128000, 3,  8 },
+	{  128000,   256000, 2,  4 },
+	{  256000,  1000000, 1,  2 },
+	{ 1000000, 13500000, 0,  1 },
+};
+
+static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
+		       unsigned int Fout)
+{
+	u64 Kpart;
+	unsigned int K, Ndiv, Nmod, target;
+	unsigned int div;
+	int i;
+
+	/* Fref must be <=13.5MHz */
+	div = 1;
+	while ((Fref / div) > 13500000) {
+		div *= 2;
+
+		if (div > 8) {
+			pr_err("Can't scale %dMHz input down to <=13.5MHz\n",
+			       Fref);
+			return -EINVAL;
+		}
+	}
+	fll_div->fll_clk_ref_div = div / 2;
+
+	pr_debug("Fref=%u Fout=%u\n", Fref, Fout);
+
+	/* Apply the division for our remaining calculations */
+	Fref /= div;
+
+	/* Fvco should be 90-100MHz; don't check the upper bound */
+	div = 0;
+	target = Fout * 2;
+	while (target < 90000000) {
+		div++;
+		target *= 2;
+		if (div > 7) {
+			pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n",
+			       Fout);
+			return -EINVAL;
+		}
+	}
+	fll_div->fll_outdiv = div;
+
+	pr_debug("Fvco=%dHz\n", target);
+
+	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
+			target /= fll_fratios[i].ratio;
+			break;
+		}
+	}
+	if (i == ARRAY_SIZE(fll_fratios)) {
+		pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref);
+		return -EINVAL;
+	}
+
+	/* Now, calculate N.K */
+	Ndiv = target / Fref;
+
+	fll_div->n = Ndiv;
+	Nmod = target % Fref;
+	pr_debug("Nmod=%d\n", Nmod);
+
+	/* Calculate fractional part - scale up so we can round. */
+	Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, Fref);
+
+	K = Kpart & 0xFFFFFFFF;
+
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	fll_div->k = K / 10;
+
+	pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n",
+		 fll_div->n, fll_div->k,
+		 fll_div->fll_fratio, fll_div->fll_outdiv,
+		 fll_div->fll_clk_ref_div);
+
+	return 0;
+}
+
+static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
+			  unsigned int Fref, unsigned int Fout)
+{
+	struct wm9081_priv *wm9081 = codec->private_data;
+	u16 reg1, reg4, reg5;
+	struct _fll_div fll_div;
+	int ret;
+	int clk_sys_reg;
+
+	/* Any change? */
+	if (Fref == wm9081->fll_fref && Fout == wm9081->fll_fout)
+		return 0;
+
+	/* Disable the FLL */
+	if (Fout == 0) {
+		dev_dbg(codec->dev, "FLL disabled\n");
+		wm9081->fll_fref = 0;
+		wm9081->fll_fout = 0;
+
+		return 0;
+	}
+
+	ret = fll_factors(&fll_div, Fref, Fout);
+	if (ret != 0)
+		return ret;
+
+	reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5);
+	reg5 &= ~WM9081_FLL_CLK_SRC_MASK;
+
+	switch (fll_id) {
+	case WM9081_SYSCLK_FLL_MCLK:
+		reg5 |= 0x1;
+		break;
+
+	default:
+		dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id);
+		return -EINVAL;
+	}
+
+	/* Disable CLK_SYS while we reconfigure */
+	clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+	if (clk_sys_reg & WM9081_CLK_SYS_ENA)
+		wm9081_write(codec, WM9081_CLOCK_CONTROL_3,
+			     clk_sys_reg & ~WM9081_CLK_SYS_ENA);
+
+	/* Any FLL configuration change requires that the FLL be
+	 * disabled first. */
+	reg1 = wm9081_read(codec, WM9081_FLL_CONTROL_1);
+	reg1 &= ~WM9081_FLL_ENA;
+	wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+
+	/* Apply the configuration */
+	if (fll_div.k)
+		reg1 |= WM9081_FLL_FRAC_MASK;
+	else
+		reg1 &= ~WM9081_FLL_FRAC_MASK;
+	wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+
+	wm9081_write(codec, WM9081_FLL_CONTROL_2,
+		     (fll_div.fll_outdiv << WM9081_FLL_OUTDIV_SHIFT) |
+		     (fll_div.fll_fratio << WM9081_FLL_FRATIO_SHIFT));
+	wm9081_write(codec, WM9081_FLL_CONTROL_3, fll_div.k);
+
+	reg4 = wm9081_read(codec, WM9081_FLL_CONTROL_4);
+	reg4 &= ~WM9081_FLL_N_MASK;
+	reg4 |= fll_div.n << WM9081_FLL_N_SHIFT;
+	wm9081_write(codec, WM9081_FLL_CONTROL_4, reg4);
+
+	reg5 &= ~WM9081_FLL_CLK_REF_DIV_MASK;
+	reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT;
+	wm9081_write(codec, WM9081_FLL_CONTROL_5, reg5);
+
+	/* Enable the FLL */
+	wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA);
+
+	/* Then bring CLK_SYS up again if it was disabled */
+	if (clk_sys_reg & WM9081_CLK_SYS_ENA)
+		wm9081_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg);
+
+	dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
+
+	wm9081->fll_fref = Fref;
+	wm9081->fll_fout = Fout;
+
+	return 0;
+}
+
+static int configure_clock(struct snd_soc_codec *codec)
+{
+	struct wm9081_priv *wm9081 = codec->private_data;
+	int new_sysclk, i, target;
+	unsigned int reg;
+	int ret = 0;
+	int mclkdiv = 0;
+	int fll = 0;
+
+	switch (wm9081->sysclk_source) {
+	case WM9081_SYSCLK_MCLK:
+		if (wm9081->mclk_rate > 12225000) {
+			mclkdiv = 1;
+			wm9081->sysclk_rate = wm9081->mclk_rate / 2;
+		} else {
+			wm9081->sysclk_rate = wm9081->mclk_rate;
+		}
+		wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK, 0, 0);
+		break;
+
+	case WM9081_SYSCLK_FLL_MCLK:
+		/* If we have a sample rate calculate a CLK_SYS that
+		 * gives us a suitable DAC configuration, plus BCLK.
+		 * Ideally we would check to see if we can clock
+		 * directly from MCLK and only use the FLL if this is
+		 * not the case, though care must be taken with free
+		 * running mode.
+		 */
+		if (wm9081->master && wm9081->bclk) {
+			/* Make sure we can generate CLK_SYS and BCLK
+			 * and that we've got 3MHz for optimal
+			 * performance. */
+			for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) {
+				target = wm9081->fs * clk_sys_rates[i].ratio;
+				new_sysclk = target;
+				if (target >= wm9081->bclk &&
+				    target > 3000000)
+					break;
+			}
+		} else if (wm9081->fs) {
+			for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) {
+				new_sysclk = clk_sys_rates[i].ratio
+					* wm9081->fs;
+				if (new_sysclk > 3000000)
+					break;
+			}
+		} else {
+			new_sysclk = 12288000;
+		}
+
+		ret = wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK,
+				     wm9081->mclk_rate, new_sysclk);
+		if (ret == 0) {
+			wm9081->sysclk_rate = new_sysclk;
+
+			/* Switch SYSCLK over to FLL */
+			fll = 1;
+		} else {
+			wm9081->sysclk_rate = wm9081->mclk_rate;
+		}
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_1);
+	if (mclkdiv)
+		reg |= WM9081_MCLKDIV2;
+	else
+		reg &= ~WM9081_MCLKDIV2;
+	wm9081_write(codec, WM9081_CLOCK_CONTROL_1, reg);
+
+	reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+	if (fll)
+		reg |= WM9081_CLK_SRC_SEL;
+	else
+		reg &= ~WM9081_CLK_SRC_SEL;
+	wm9081_write(codec, WM9081_CLOCK_CONTROL_3, reg);
+
+	dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm9081->sysclk_rate);
+
+	return ret;
+}
+
+static int clk_sys_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm9081_priv *wm9081 = codec->private_data;
+
+	/* This should be done on init() for bypass paths */
+	switch (wm9081->sysclk_source) {
+	case WM9081_SYSCLK_MCLK:
+		dev_dbg(codec->dev, "Using %dHz MCLK\n", wm9081->mclk_rate);
+		break;
+	case WM9081_SYSCLK_FLL_MCLK:
+		dev_dbg(codec->dev, "Using %dHz MCLK with FLL\n",
+			wm9081->mclk_rate);
+		break;
+	default:
+		dev_err(codec->dev, "System clock not configured\n");
+		return -EINVAL;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		configure_clock(codec);
+		break;
+
+	case SND_SOC_DAPM_POST_PMD:
+		/* Disable the FLL if it's running */
+		wm9081_set_fll(codec, 0, 0, 0);
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN1"),
+SND_SOC_DAPM_INPUT("IN2"),
+
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0),
+
+SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
+			     mixer, ARRAY_SIZE(mixer)),
+
+SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0,
+		   speaker_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+SND_SOC_DAPM_OUTPUT("LINEOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+
+SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0),
+};
+
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+	{ "DAC", NULL, "CLK_SYS" },
+	{ "DAC", NULL, "CLK_DSP" },
+
+	{ "Mixer", "IN1 Switch", "IN1" },
+	{ "Mixer", "IN2 Switch", "IN2" },
+	{ "Mixer", "Playback Switch", "DAC" },
+
+	{ "LINEOUT PGA", NULL, "Mixer" },
+	{ "LINEOUT PGA", NULL, "TOCLK" },
+	{ "LINEOUT PGA", NULL, "CLK_SYS" },
+
+	{ "LINEOUT", NULL, "LINEOUT PGA" },
+
+	{ "Speaker PGA", NULL, "Mixer" },
+	{ "Speaker PGA", NULL, "TOCLK" },
+	{ "Speaker PGA", NULL, "CLK_SYS" },
+
+	{ "SPKN", NULL, "Speaker PGA" },
+	{ "SPKP", NULL, "Speaker PGA" },
+};
+
+static int wm9081_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 reg;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VMID=2*40k */
+		reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+		reg &= ~WM9081_VMID_SEL_MASK;
+		reg |= 0x2;
+		wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+		/* Normal bias current */
+		reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+		reg &= ~WM9081_STBY_BIAS_ENA;
+		wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		/* Initial cold start */
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Disable LINEOUT discharge */
+			reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+			reg &= ~WM9081_LINEOUT_DISCH;
+			wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+
+			/* Select startup bias source */
+			reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+			reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA;
+			wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+
+			/* VMID 2*4k; Soft VMID ramp enable */
+			reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+			reg |= WM9081_VMID_RAMP | 0x6;
+			wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+			mdelay(100);
+
+			/* Normal bias enable & soft start off */
+			reg |= WM9081_BIAS_ENA;
+			reg &= ~WM9081_VMID_RAMP;
+			wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+			/* Standard bias source */
+			reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+			reg &= ~WM9081_BIAS_SRC;
+			wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+		}
+
+		/* VMID 2*240k */
+		reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+		reg &= ~WM9081_VMID_SEL_MASK;
+		reg |= 0x40;
+		wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+		/* Standby bias current on */
+		reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+		reg |= WM9081_STBY_BIAS_ENA;
+		wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* Startup bias source */
+		reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+		reg |= WM9081_BIAS_SRC;
+		wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+
+		/* Disable VMID and biases with soft ramping */
+		reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+		reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
+		reg |= WM9081_VMID_RAMP;
+		wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+		/* Actively discharge LINEOUT */
+		reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+		reg |= WM9081_LINEOUT_DISCH;
+		wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+		break;
+	}
+
+	codec->bias_level = level;
+
+	return 0;
+}
+
+static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
+			      unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm9081_priv *wm9081 = codec->private_data;
+	unsigned int aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+
+	aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV |
+		  WM9081_BCLK_DIR | WM9081_LRCLK_DIR | WM9081_AIF_FMT_MASK);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		wm9081->master = 0;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		aif2 |= WM9081_LRCLK_DIR;
+		wm9081->master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		aif2 |= WM9081_BCLK_DIR;
+		wm9081->master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aif2 |= WM9081_LRCLK_DIR | WM9081_BCLK_DIR;
+		wm9081->master = 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_B:
+		aif2 |= WM9081_AIF_LRCLK_INV;
+	case SND_SOC_DAIFMT_DSP_A:
+		aif2 |= 0x3;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		aif2 |= 0x2;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		aif2 |= 0x1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
+		/* frame inversion not valid for DSP modes */
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			aif2 |= WM9081_AIF_BCLK_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_LEFT_J:
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			aif2 |= WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			aif2 |= WM9081_AIF_BCLK_INV;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			aif2 |= WM9081_AIF_LRCLK_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+
+	return 0;
+}
+
+static int wm9081_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm9081_priv *wm9081 = codec->private_data;
+	int ret, i, best, best_val, cur_val;
+	unsigned int clk_ctrl2, aif1, aif2, aif3, aif4;
+
+	clk_ctrl2 = wm9081_read(codec, WM9081_CLOCK_CONTROL_2);
+	clk_ctrl2 &= ~(WM9081_CLK_SYS_RATE_MASK | WM9081_SAMPLE_RATE_MASK);
+
+	aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+
+	aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+	aif2 &= ~WM9081_AIF_WL_MASK;
+
+	aif3 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_3);
+	aif3 &= ~WM9081_BCLK_DIV_MASK;
+
+	aif4 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_4);
+	aif4 &= ~WM9081_LRCLK_RATE_MASK;
+
+	/* What BCLK do we need? */
+	wm9081->fs = params_rate(params);
+	wm9081->bclk = 2 * wm9081->fs;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		wm9081->bclk *= 16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		wm9081->bclk *= 20;
+		aif2 |= 0x4;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		wm9081->bclk *= 24;
+		aif2 |= 0x8;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		wm9081->bclk *= 32;
+		aif2 |= 0xc;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (aif1 & WM9081_AIFDAC_TDM_MODE_MASK) {
+		int slots = ((aif1 & WM9081_AIFDAC_TDM_MODE_MASK) >>
+			     WM9081_AIFDAC_TDM_MODE_SHIFT) + 1;
+		wm9081->bclk *= slots;
+	}
+
+	dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm9081->bclk);
+
+	ret = configure_clock(codec);
+	if (ret != 0)
+		return ret;
+
+	/* Select nearest CLK_SYS_RATE */
+	best = 0;
+	best_val = abs((wm9081->sysclk_rate / clk_sys_rates[0].ratio)
+		       - wm9081->fs);
+	for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
+		cur_val = abs((wm9081->sysclk_rate /
+			       clk_sys_rates[i].ratio) - wm9081->fs);;
+		if (cur_val < best_val) {
+			best = i;
+			best_val = cur_val;
+		}
+	}
+	dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n",
+		clk_sys_rates[best].ratio);
+	clk_ctrl2 |= (clk_sys_rates[best].clk_sys_rate
+		      << WM9081_CLK_SYS_RATE_SHIFT);
+
+	/* SAMPLE_RATE */
+	best = 0;
+	best_val = abs(wm9081->fs - sample_rates[0].rate);
+	for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+		/* Closest match */
+		cur_val = abs(wm9081->fs - sample_rates[i].rate);
+		if (cur_val < best_val) {
+			best = i;
+			best_val = cur_val;
+		}
+	}
+	dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n",
+		sample_rates[best].rate);
+	clk_ctrl2 |= (sample_rates[best].sample_rate
+			<< WM9081_SAMPLE_RATE_SHIFT);
+
+	/* BCLK_DIV */
+	best = 0;
+	best_val = INT_MAX;
+	for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+		cur_val = ((wm9081->sysclk_rate * 10) / bclk_divs[i].div)
+			- wm9081->bclk;
+		if (cur_val < 0) /* Table is sorted */
+			break;
+		if (cur_val < best_val) {
+			best = i;
+			best_val = cur_val;
+		}
+	}
+	wm9081->bclk = (wm9081->sysclk_rate * 10) / bclk_divs[best].div;
+	dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n",
+		bclk_divs[best].div, wm9081->bclk);
+	aif3 |= bclk_divs[best].bclk_div;
+
+	/* LRCLK is a simple fraction of BCLK */
+	dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm9081->bclk / wm9081->fs);
+	aif4 |= wm9081->bclk / wm9081->fs;
+
+	/* Apply a ReTune Mobile configuration if it's in use */
+	if (wm9081->retune) {
+		struct wm9081_retune_mobile_config *retune = wm9081->retune;
+		struct wm9081_retune_mobile_setting *s;
+		int eq1;
+
+		best = 0;
+		best_val = abs(retune->configs[0].rate - wm9081->fs);
+		for (i = 0; i < retune->num_configs; i++) {
+			cur_val = abs(retune->configs[i].rate - wm9081->fs);
+			if (cur_val < best_val) {
+				best_val = cur_val;
+				best = i;
+			}
+		}
+		s = &retune->configs[best];
+
+		dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n",
+			s->name, s->rate);
+
+		/* If the EQ is enabled then disable it while we write out */
+		eq1 = wm9081_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA;
+		if (eq1 & WM9081_EQ_ENA)
+			wm9081_write(codec, WM9081_EQ_1, 0);
+
+		/* Write out the other values */
+		for (i = 1; i < ARRAY_SIZE(s->config); i++)
+			wm9081_write(codec, WM9081_EQ_1 + i, s->config[i]);
+
+		eq1 |= (s->config[0] & ~WM9081_EQ_ENA);
+		wm9081_write(codec, WM9081_EQ_1, eq1);
+	}
+
+	wm9081_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2);
+	wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+	wm9081_write(codec, WM9081_AUDIO_INTERFACE_3, aif3);
+	wm9081_write(codec, WM9081_AUDIO_INTERFACE_4, aif4);
+
+	return 0;
+}
+
+static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	unsigned int reg;
+
+	reg = wm9081_read(codec, WM9081_DAC_DIGITAL_2);
+
+	if (mute)
+		reg |= WM9081_DAC_MUTE;
+	else
+		reg &= ~WM9081_DAC_MUTE;
+
+	wm9081_write(codec, WM9081_DAC_DIGITAL_2, reg);
+
+	return 0;
+}
+
+static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
+			     int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm9081_priv *wm9081 = codec->private_data;
+
+	switch (clk_id) {
+	case WM9081_SYSCLK_MCLK:
+	case WM9081_SYSCLK_FLL_MCLK:
+		wm9081->sysclk_source = clk_id;
+		wm9081->mclk_rate = freq;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
+			       unsigned int mask, int slots)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+
+	aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK);
+
+	if (slots < 1 || slots > 4)
+		return -EINVAL;
+
+	aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT;
+
+	switch (mask) {
+	case 1:
+		break;
+	case 2:
+		aif1 |= 0x10;
+		break;
+	case 4:
+		aif1 |= 0x20;
+		break;
+	case 8:
+		aif1 |= 0x30;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm9081_write(codec, WM9081_AUDIO_INTERFACE_1, aif1);
+
+	return 0;
+}
+
+#define WM9081_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM9081_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 wm9081_dai_ops = {
+	.hw_params = wm9081_hw_params,
+	.set_sysclk = wm9081_set_sysclk,
+	.set_fmt = wm9081_set_dai_fmt,
+	.digital_mute = wm9081_digital_mute,
+	.set_tdm_slot = wm9081_set_tdm_slot,
+};
+
+/* We report two channels because the CODEC processes a stereo signal, even
+ * though it is only capable of handling a mono output.
+ */
+struct snd_soc_dai wm9081_dai = {
+	.name = "WM9081",
+	.playback = {
+		.stream_name = "HiFi Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM9081_RATES,
+		.formats = WM9081_FORMATS,
+	},
+	.ops = &wm9081_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm9081_dai);
+
+
+static struct snd_soc_codec *wm9081_codec;
+
+static int wm9081_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct wm9081_priv *wm9081;
+	int ret = 0;
+
+	if (wm9081_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm9081_codec;
+	codec = wm9081_codec;
+	wm9081 = codec->private_data;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, wm9081_snd_controls,
+			     ARRAY_SIZE(wm9081_snd_controls));
+	if (!wm9081->retune) {
+		dev_dbg(codec->dev,
+			"No ReTune Mobile data, using normal EQ\n");
+		snd_soc_add_controls(codec, wm9081_eq_controls,
+				     ARRAY_SIZE(wm9081_eq_controls));
+	}
+
+	snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets,
+				  ARRAY_SIZE(wm9081_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+	snd_soc_dapm_new_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	return ret;
+}
+
+static int wm9081_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm9081_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int wm9081_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u16 *reg_cache = codec->reg_cache;
+	int i;
+
+	for (i = 0; i < codec->reg_cache_size; i++) {
+		if (i == WM9081_SOFTWARE_RESET)
+			continue;
+
+		wm9081_write(codec, i, reg_cache[i]);
+	}
+
+	wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+#else
+#define wm9081_suspend NULL
+#define wm9081_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_wm9081 = {
+	.probe = 	wm9081_probe,
+	.remove = 	wm9081_remove,
+	.suspend =	wm9081_suspend,
+	.resume =	wm9081_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9081);
+
+static int wm9081_register(struct wm9081_priv *wm9081)
+{
+	struct snd_soc_codec *codec = &wm9081->codec;
+	int ret;
+	u16 reg;
+
+	if (wm9081_codec) {
+		dev_err(codec->dev, "Another WM9081 is registered\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data = wm9081;
+	codec->name = "WM9081";
+	codec->owner = THIS_MODULE;
+	codec->read = wm9081_read;
+	codec->write = wm9081_write;
+	codec->dai = &wm9081_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(wm9081->reg_cache);
+	codec->reg_cache = &wm9081->reg_cache;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm9081_set_bias_level;
+
+	memcpy(codec->reg_cache, wm9081_reg_defaults,
+	       sizeof(wm9081_reg_defaults));
+
+	reg = wm9081_read_hw(codec, WM9081_SOFTWARE_RESET);
+	if (reg != 0x9081) {
+		dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg);
+		ret = -EINVAL;
+		goto err;
+	}
+
+	ret = wm9081_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
+	}
+
+	wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* Enable zero cross by default */
+	reg = wm9081_read(codec, WM9081_ANALOGUE_LINEOUT);
+	wm9081_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC);
+	reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_PGA);
+	wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_PGA,
+		     reg | WM9081_SPKPGAZC);
+
+	wm9081_dai.dev = codec->dev;
+
+	wm9081_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm9081_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
+
+	return 0;
+
+err:
+	kfree(wm9081);
+	return ret;
+}
+
+static void wm9081_unregister(struct wm9081_priv *wm9081)
+{
+	wm9081_set_bias_level(&wm9081->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm9081_dai);
+	snd_soc_unregister_codec(&wm9081->codec);
+	kfree(wm9081);
+	wm9081_codec = NULL;
+}
+
+static __devinit int wm9081_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct wm9081_priv *wm9081;
+	struct snd_soc_codec *codec;
+
+	wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL);
+	if (wm9081 == NULL)
+		return -ENOMEM;
+
+	codec = &wm9081->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+	wm9081->retune = i2c->dev.platform_data;
+
+	i2c_set_clientdata(i2c, wm9081);
+	codec->control_data = i2c;
+
+	codec->dev = &i2c->dev;
+
+	return wm9081_register(wm9081);
+}
+
+static __devexit int wm9081_i2c_remove(struct i2c_client *client)
+{
+	struct wm9081_priv *wm9081 = i2c_get_clientdata(client);
+	wm9081_unregister(wm9081);
+	return 0;
+}
+
+static const struct i2c_device_id wm9081_i2c_id[] = {
+	{ "wm9081", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id);
+
+static struct i2c_driver wm9081_i2c_driver = {
+	.driver = {
+		.name = "wm9081",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm9081_i2c_probe,
+	.remove =   __devexit_p(wm9081_i2c_remove),
+	.id_table = wm9081_i2c_id,
+};
+
+static int __init wm9081_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&wm9081_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n",
+		       ret);
+	}
+
+	return ret;
+}
+module_init(wm9081_modinit);
+
+static void __exit wm9081_exit(void)
+{
+	i2c_del_driver(&wm9081_i2c_driver);
+}
+module_exit(wm9081_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM9081 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9081.h b/sound/soc/codecs/wm9081.h
new file mode 100644
index 00000000000..42d3bc75702
--- /dev/null
+++ b/sound/soc/codecs/wm9081.h
@@ -0,0 +1,787 @@
+#ifndef WM9081_H
+#define WM9081_H
+
+/*
+ * wm9081.c  --  WM9081 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm9081_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm9081;
+
+/*
+ * SYSCLK sources
+ */
+#define WM9081_SYSCLK_MCLK      1   /* Use MCLK without FLL */
+#define WM9081_SYSCLK_FLL_MCLK  2   /* Use MCLK, enabling FLL if required */
+
+/*
+ * Register values.
+ */
+#define WM9081_SOFTWARE_RESET                   0x00
+#define WM9081_ANALOGUE_LINEOUT                 0x02
+#define WM9081_ANALOGUE_SPEAKER_PGA             0x03
+#define WM9081_VMID_CONTROL                     0x04
+#define WM9081_BIAS_CONTROL_1                   0x05
+#define WM9081_ANALOGUE_MIXER                   0x07
+#define WM9081_ANTI_POP_CONTROL                 0x08
+#define WM9081_ANALOGUE_SPEAKER_1               0x09
+#define WM9081_ANALOGUE_SPEAKER_2               0x0A
+#define WM9081_POWER_MANAGEMENT                 0x0B
+#define WM9081_CLOCK_CONTROL_1                  0x0C
+#define WM9081_CLOCK_CONTROL_2                  0x0D
+#define WM9081_CLOCK_CONTROL_3                  0x0E
+#define WM9081_FLL_CONTROL_1                    0x10
+#define WM9081_FLL_CONTROL_2                    0x11
+#define WM9081_FLL_CONTROL_3                    0x12
+#define WM9081_FLL_CONTROL_4                    0x13
+#define WM9081_FLL_CONTROL_5                    0x14
+#define WM9081_AUDIO_INTERFACE_1                0x16
+#define WM9081_AUDIO_INTERFACE_2                0x17
+#define WM9081_AUDIO_INTERFACE_3                0x18
+#define WM9081_AUDIO_INTERFACE_4                0x19
+#define WM9081_INTERRUPT_STATUS                 0x1A
+#define WM9081_INTERRUPT_STATUS_MASK            0x1B
+#define WM9081_INTERRUPT_POLARITY               0x1C
+#define WM9081_INTERRUPT_CONTROL                0x1D
+#define WM9081_DAC_DIGITAL_1                    0x1E
+#define WM9081_DAC_DIGITAL_2                    0x1F
+#define WM9081_DRC_1                            0x20
+#define WM9081_DRC_2                            0x21
+#define WM9081_DRC_3                            0x22
+#define WM9081_DRC_4                            0x23
+#define WM9081_WRITE_SEQUENCER_1                0x26
+#define WM9081_WRITE_SEQUENCER_2                0x27
+#define WM9081_MW_SLAVE_1                       0x28
+#define WM9081_EQ_1                             0x2A
+#define WM9081_EQ_2                             0x2B
+#define WM9081_EQ_3                             0x2C
+#define WM9081_EQ_4                             0x2D
+#define WM9081_EQ_5                             0x2E
+#define WM9081_EQ_6                             0x2F
+#define WM9081_EQ_7                             0x30
+#define WM9081_EQ_8                             0x31
+#define WM9081_EQ_9                             0x32
+#define WM9081_EQ_10                            0x33
+#define WM9081_EQ_11                            0x34
+#define WM9081_EQ_12                            0x35
+#define WM9081_EQ_13                            0x36
+#define WM9081_EQ_14                            0x37
+#define WM9081_EQ_15                            0x38
+#define WM9081_EQ_16                            0x39
+#define WM9081_EQ_17                            0x3A
+#define WM9081_EQ_18                            0x3B
+#define WM9081_EQ_19                            0x3C
+#define WM9081_EQ_20                            0x3D
+
+#define WM9081_REGISTER_COUNT                   55
+#define WM9081_MAX_REGISTER                     0x3D
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Software Reset
+ */
+#define WM9081_SW_RST_DEV_ID1_MASK              0xFFFF  /* SW_RST_DEV_ID1 - [15:0] */
+#define WM9081_SW_RST_DEV_ID1_SHIFT                  0  /* SW_RST_DEV_ID1 - [15:0] */
+#define WM9081_SW_RST_DEV_ID1_WIDTH                 16  /* SW_RST_DEV_ID1 - [15:0] */
+
+/*
+ * R2 (0x02) - Analogue Lineout
+ */
+#define WM9081_LINEOUT_MUTE                     0x0080  /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_MASK                0x0080  /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_SHIFT                    7  /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_WIDTH                    1  /* LINEOUT_MUTE */
+#define WM9081_LINEOUTZC                        0x0040  /* LINEOUTZC */
+#define WM9081_LINEOUTZC_MASK                   0x0040  /* LINEOUTZC */
+#define WM9081_LINEOUTZC_SHIFT                       6  /* LINEOUTZC */
+#define WM9081_LINEOUTZC_WIDTH                       1  /* LINEOUTZC */
+#define WM9081_LINEOUT_VOL_MASK                 0x003F  /* LINEOUT_VOL - [5:0] */
+#define WM9081_LINEOUT_VOL_SHIFT                     0  /* LINEOUT_VOL - [5:0] */
+#define WM9081_LINEOUT_VOL_WIDTH                     6  /* LINEOUT_VOL - [5:0] */
+
+/*
+ * R3 (0x03) - Analogue Speaker PGA
+ */
+#define WM9081_SPKPGA_MUTE                      0x0080  /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_MASK                 0x0080  /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_SHIFT                     7  /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_WIDTH                     1  /* SPKPGA_MUTE */
+#define WM9081_SPKPGAZC                         0x0040  /* SPKPGAZC */
+#define WM9081_SPKPGAZC_MASK                    0x0040  /* SPKPGAZC */
+#define WM9081_SPKPGAZC_SHIFT                        6  /* SPKPGAZC */
+#define WM9081_SPKPGAZC_WIDTH                        1  /* SPKPGAZC */
+#define WM9081_SPKPGA_VOL_MASK                  0x003F  /* SPKPGA_VOL - [5:0] */
+#define WM9081_SPKPGA_VOL_SHIFT                      0  /* SPKPGA_VOL - [5:0] */
+#define WM9081_SPKPGA_VOL_WIDTH                      6  /* SPKPGA_VOL - [5:0] */
+
+/*
+ * R4 (0x04) - VMID Control
+ */
+#define WM9081_VMID_BUF_ENA                     0x0020  /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_MASK                0x0020  /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_SHIFT                    5  /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_WIDTH                    1  /* VMID_BUF_ENA */
+#define WM9081_VMID_RAMP                        0x0008  /* VMID_RAMP */
+#define WM9081_VMID_RAMP_MASK                   0x0008  /* VMID_RAMP */
+#define WM9081_VMID_RAMP_SHIFT                       3  /* VMID_RAMP */
+#define WM9081_VMID_RAMP_WIDTH                       1  /* VMID_RAMP */
+#define WM9081_VMID_SEL_MASK                    0x0006  /* VMID_SEL - [2:1] */
+#define WM9081_VMID_SEL_SHIFT                        1  /* VMID_SEL - [2:1] */
+#define WM9081_VMID_SEL_WIDTH                        2  /* VMID_SEL - [2:1] */
+#define WM9081_VMID_FAST_ST                     0x0001  /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_MASK                0x0001  /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_SHIFT                    0  /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_WIDTH                    1  /* VMID_FAST_ST */
+
+/*
+ * R5 (0x05) - Bias Control 1
+ */
+#define WM9081_BIAS_SRC                         0x0040  /* BIAS_SRC */
+#define WM9081_BIAS_SRC_MASK                    0x0040  /* BIAS_SRC */
+#define WM9081_BIAS_SRC_SHIFT                        6  /* BIAS_SRC */
+#define WM9081_BIAS_SRC_WIDTH                        1  /* BIAS_SRC */
+#define WM9081_STBY_BIAS_LVL                    0x0020  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_MASK               0x0020  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_SHIFT                   5  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_WIDTH                   1  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_ENA                    0x0010  /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_MASK               0x0010  /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_SHIFT                   4  /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_WIDTH                   1  /* STBY_BIAS_ENA */
+#define WM9081_BIAS_LVL_MASK                    0x000C  /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_LVL_SHIFT                        2  /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_LVL_WIDTH                        2  /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_ENA                         0x0002  /* BIAS_ENA */
+#define WM9081_BIAS_ENA_MASK                    0x0002  /* BIAS_ENA */
+#define WM9081_BIAS_ENA_SHIFT                        1  /* BIAS_ENA */
+#define WM9081_BIAS_ENA_WIDTH                        1  /* BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA                 0x0001  /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_MASK            0x0001  /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_SHIFT                0  /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_WIDTH                1  /* STARTUP_BIAS_ENA */
+
+/*
+ * R7 (0x07) - Analogue Mixer
+ */
+#define WM9081_DAC_SEL                          0x0010  /* DAC_SEL */
+#define WM9081_DAC_SEL_MASK                     0x0010  /* DAC_SEL */
+#define WM9081_DAC_SEL_SHIFT                         4  /* DAC_SEL */
+#define WM9081_DAC_SEL_WIDTH                         1  /* DAC_SEL */
+#define WM9081_IN2_VOL                          0x0008  /* IN2_VOL */
+#define WM9081_IN2_VOL_MASK                     0x0008  /* IN2_VOL */
+#define WM9081_IN2_VOL_SHIFT                         3  /* IN2_VOL */
+#define WM9081_IN2_VOL_WIDTH                         1  /* IN2_VOL */
+#define WM9081_IN2_ENA                          0x0004  /* IN2_ENA */
+#define WM9081_IN2_ENA_MASK                     0x0004  /* IN2_ENA */
+#define WM9081_IN2_ENA_SHIFT                         2  /* IN2_ENA */
+#define WM9081_IN2_ENA_WIDTH                         1  /* IN2_ENA */
+#define WM9081_IN1_VOL                          0x0002  /* IN1_VOL */
+#define WM9081_IN1_VOL_MASK                     0x0002  /* IN1_VOL */
+#define WM9081_IN1_VOL_SHIFT                         1  /* IN1_VOL */
+#define WM9081_IN1_VOL_WIDTH                         1  /* IN1_VOL */
+#define WM9081_IN1_ENA                          0x0001  /* IN1_ENA */
+#define WM9081_IN1_ENA_MASK                     0x0001  /* IN1_ENA */
+#define WM9081_IN1_ENA_SHIFT                         0  /* IN1_ENA */
+#define WM9081_IN1_ENA_WIDTH                         1  /* IN1_ENA */
+
+/*
+ * R8 (0x08) - Anti Pop Control
+ */
+#define WM9081_LINEOUT_DISCH                    0x0004  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_MASK               0x0004  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_SHIFT                   2  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_WIDTH                   1  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_VROI                     0x0002  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_MASK                0x0002  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_SHIFT                    1  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_WIDTH                    1  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_CLAMP                    0x0001  /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_MASK               0x0001  /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_SHIFT                   0  /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_WIDTH                   1  /* LINEOUT_CLAMP */
+
+/*
+ * R9 (0x09) - Analogue Speaker 1
+ */
+#define WM9081_SPK_DCGAIN_MASK                  0x0038  /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_DCGAIN_SHIFT                      3  /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_DCGAIN_WIDTH                      3  /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_ACGAIN_MASK                  0x0007  /* SPK_ACGAIN - [2:0] */
+#define WM9081_SPK_ACGAIN_SHIFT                      0  /* SPK_ACGAIN - [2:0] */
+#define WM9081_SPK_ACGAIN_WIDTH                      3  /* SPK_ACGAIN - [2:0] */
+
+/*
+ * R10 (0x0A) - Analogue Speaker 2
+ */
+#define WM9081_SPK_MODE                         0x0040  /* SPK_MODE */
+#define WM9081_SPK_MODE_MASK                    0x0040  /* SPK_MODE */
+#define WM9081_SPK_MODE_SHIFT                        6  /* SPK_MODE */
+#define WM9081_SPK_MODE_WIDTH                        1  /* SPK_MODE */
+#define WM9081_SPK_INV_MUTE                     0x0010  /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_MASK                0x0010  /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_SHIFT                    4  /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_WIDTH                    1  /* SPK_INV_MUTE */
+#define WM9081_OUT_SPK_CTRL                     0x0008  /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_MASK                0x0008  /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_SHIFT                    3  /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_WIDTH                    1  /* OUT_SPK_CTRL */
+
+/*
+ * R11 (0x0B) - Power Management
+ */
+#define WM9081_TSHUT_ENA                        0x0100  /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_MASK                   0x0100  /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_SHIFT                       8  /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_WIDTH                       1  /* TSHUT_ENA */
+#define WM9081_TSENSE_ENA                       0x0080  /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_MASK                  0x0080  /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_SHIFT                      7  /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_WIDTH                      1  /* TSENSE_ENA */
+#define WM9081_TEMP_SHUT                        0x0040  /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_MASK                   0x0040  /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_SHIFT                       6  /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_WIDTH                       1  /* TEMP_SHUT */
+#define WM9081_LINEOUT_ENA                      0x0010  /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_MASK                 0x0010  /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_SHIFT                     4  /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_WIDTH                     1  /* LINEOUT_ENA */
+#define WM9081_SPKPGA_ENA                       0x0004  /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_MASK                  0x0004  /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_SHIFT                      2  /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_WIDTH                      1  /* SPKPGA_ENA */
+#define WM9081_SPK_ENA                          0x0002  /* SPK_ENA */
+#define WM9081_SPK_ENA_MASK                     0x0002  /* SPK_ENA */
+#define WM9081_SPK_ENA_SHIFT                         1  /* SPK_ENA */
+#define WM9081_SPK_ENA_WIDTH                         1  /* SPK_ENA */
+#define WM9081_DAC_ENA                          0x0001  /* DAC_ENA */
+#define WM9081_DAC_ENA_MASK                     0x0001  /* DAC_ENA */
+#define WM9081_DAC_ENA_SHIFT                         0  /* DAC_ENA */
+#define WM9081_DAC_ENA_WIDTH                         1  /* DAC_ENA */
+
+/*
+ * R12 (0x0C) - Clock Control 1
+ */
+#define WM9081_CLK_OP_DIV_MASK                  0x1C00  /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_OP_DIV_SHIFT                     10  /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_OP_DIV_WIDTH                      3  /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_TO_DIV_MASK                  0x0300  /* CLK_TO_DIV - [9:8] */
+#define WM9081_CLK_TO_DIV_SHIFT                      8  /* CLK_TO_DIV - [9:8] */
+#define WM9081_CLK_TO_DIV_WIDTH                      2  /* CLK_TO_DIV - [9:8] */
+#define WM9081_MCLKDIV2                         0x0080  /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_MASK                    0x0080  /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_SHIFT                        7  /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_WIDTH                        1  /* MCLKDIV2 */
+
+/*
+ * R13 (0x0D) - Clock Control 2
+ */
+#define WM9081_CLK_SYS_RATE_MASK                0x00F0  /* CLK_SYS_RATE - [7:4] */
+#define WM9081_CLK_SYS_RATE_SHIFT                    4  /* CLK_SYS_RATE - [7:4] */
+#define WM9081_CLK_SYS_RATE_WIDTH                    4  /* CLK_SYS_RATE - [7:4] */
+#define WM9081_SAMPLE_RATE_MASK                 0x000F  /* SAMPLE_RATE - [3:0] */
+#define WM9081_SAMPLE_RATE_SHIFT                     0  /* SAMPLE_RATE - [3:0] */
+#define WM9081_SAMPLE_RATE_WIDTH                     4  /* SAMPLE_RATE - [3:0] */
+
+/*
+ * R14 (0x0E) - Clock Control 3
+ */
+#define WM9081_CLK_SRC_SEL                      0x2000  /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_MASK                 0x2000  /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_SHIFT                    13  /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_WIDTH                     1  /* CLK_SRC_SEL */
+#define WM9081_CLK_OP_ENA                       0x0020  /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_MASK                  0x0020  /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_SHIFT                      5  /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_WIDTH                      1  /* CLK_OP_ENA */
+#define WM9081_CLK_TO_ENA                       0x0004  /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_MASK                  0x0004  /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_SHIFT                      2  /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_WIDTH                      1  /* CLK_TO_ENA */
+#define WM9081_CLK_DSP_ENA                      0x0002  /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_MASK                 0x0002  /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_SHIFT                     1  /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_WIDTH                     1  /* CLK_DSP_ENA */
+#define WM9081_CLK_SYS_ENA                      0x0001  /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_MASK                 0x0001  /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_SHIFT                     0  /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_WIDTH                     1  /* CLK_SYS_ENA */
+
+/*
+ * R16 (0x10) - FLL Control 1
+ */
+#define WM9081_FLL_HOLD                         0x0008  /* FLL_HOLD */
+#define WM9081_FLL_HOLD_MASK                    0x0008  /* FLL_HOLD */
+#define WM9081_FLL_HOLD_SHIFT                        3  /* FLL_HOLD */
+#define WM9081_FLL_HOLD_WIDTH                        1  /* FLL_HOLD */
+#define WM9081_FLL_FRAC                         0x0004  /* FLL_FRAC */
+#define WM9081_FLL_FRAC_MASK                    0x0004  /* FLL_FRAC */
+#define WM9081_FLL_FRAC_SHIFT                        2  /* FLL_FRAC */
+#define WM9081_FLL_FRAC_WIDTH                        1  /* FLL_FRAC */
+#define WM9081_FLL_ENA                          0x0001  /* FLL_ENA */
+#define WM9081_FLL_ENA_MASK                     0x0001  /* FLL_ENA */
+#define WM9081_FLL_ENA_SHIFT                         0  /* FLL_ENA */
+#define WM9081_FLL_ENA_WIDTH                         1  /* FLL_ENA */
+
+/*
+ * R17 (0x11) - FLL Control 2
+ */
+#define WM9081_FLL_OUTDIV_MASK                  0x0700  /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_OUTDIV_SHIFT                      8  /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_OUTDIV_WIDTH                      3  /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_CTRL_RATE_MASK               0x0070  /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_CTRL_RATE_SHIFT                   4  /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_CTRL_RATE_WIDTH                   3  /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_FRATIO_MASK                  0x0007  /* FLL_FRATIO - [2:0] */
+#define WM9081_FLL_FRATIO_SHIFT                      0  /* FLL_FRATIO - [2:0] */
+#define WM9081_FLL_FRATIO_WIDTH                      3  /* FLL_FRATIO - [2:0] */
+
+/*
+ * R18 (0x12) - FLL Control 3
+ */
+#define WM9081_FLL_K_MASK                       0xFFFF  /* FLL_K - [15:0] */
+#define WM9081_FLL_K_SHIFT                           0  /* FLL_K - [15:0] */
+#define WM9081_FLL_K_WIDTH                          16  /* FLL_K - [15:0] */
+
+/*
+ * R19 (0x13) - FLL Control 4
+ */
+#define WM9081_FLL_N_MASK                       0x7FE0  /* FLL_N - [14:5] */
+#define WM9081_FLL_N_SHIFT                           5  /* FLL_N - [14:5] */
+#define WM9081_FLL_N_WIDTH                          10  /* FLL_N - [14:5] */
+#define WM9081_FLL_GAIN_MASK                    0x000F  /* FLL_GAIN - [3:0] */
+#define WM9081_FLL_GAIN_SHIFT                        0  /* FLL_GAIN - [3:0] */
+#define WM9081_FLL_GAIN_WIDTH                        4  /* FLL_GAIN - [3:0] */
+
+/*
+ * R20 (0x14) - FLL Control 5
+ */
+#define WM9081_FLL_CLK_REF_DIV_MASK             0x0018  /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_REF_DIV_SHIFT                 3  /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_REF_DIV_WIDTH                 2  /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_SRC_MASK                 0x0003  /* FLL_CLK_SRC - [1:0] */
+#define WM9081_FLL_CLK_SRC_SHIFT                     0  /* FLL_CLK_SRC - [1:0] */
+#define WM9081_FLL_CLK_SRC_WIDTH                     2  /* FLL_CLK_SRC - [1:0] */
+
+/*
+ * R22 (0x16) - Audio Interface 1
+ */
+#define WM9081_AIFDAC_CHAN                      0x0040  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_MASK                 0x0040  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_SHIFT                     6  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_WIDTH                     1  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_TDM_SLOT_MASK             0x0030  /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_SLOT_SHIFT                 4  /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_SLOT_WIDTH                 2  /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_MODE_MASK             0x000C  /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_AIFDAC_TDM_MODE_SHIFT                 2  /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_AIFDAC_TDM_MODE_WIDTH                 2  /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_DAC_COMP                         0x0002  /* DAC_COMP */
+#define WM9081_DAC_COMP_MASK                    0x0002  /* DAC_COMP */
+#define WM9081_DAC_COMP_SHIFT                        1  /* DAC_COMP */
+#define WM9081_DAC_COMP_WIDTH                        1  /* DAC_COMP */
+#define WM9081_DAC_COMPMODE                     0x0001  /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_MASK                0x0001  /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_SHIFT                    0  /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_WIDTH                    1  /* DAC_COMPMODE */
+
+/*
+ * R23 (0x17) - Audio Interface 2
+ */
+#define WM9081_AIF_TRIS                         0x0200  /* AIF_TRIS */
+#define WM9081_AIF_TRIS_MASK                    0x0200  /* AIF_TRIS */
+#define WM9081_AIF_TRIS_SHIFT                        9  /* AIF_TRIS */
+#define WM9081_AIF_TRIS_WIDTH                        1  /* AIF_TRIS */
+#define WM9081_DAC_DAT_INV                      0x0100  /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_MASK                 0x0100  /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_SHIFT                     8  /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_WIDTH                     1  /* DAC_DAT_INV */
+#define WM9081_AIF_BCLK_INV                     0x0080  /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_MASK                0x0080  /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_SHIFT                    7  /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_WIDTH                    1  /* AIF_BCLK_INV */
+#define WM9081_BCLK_DIR                         0x0040  /* BCLK_DIR */
+#define WM9081_BCLK_DIR_MASK                    0x0040  /* BCLK_DIR */
+#define WM9081_BCLK_DIR_SHIFT                        6  /* BCLK_DIR */
+#define WM9081_BCLK_DIR_WIDTH                        1  /* BCLK_DIR */
+#define WM9081_LRCLK_DIR                        0x0020  /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_MASK                   0x0020  /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_SHIFT                       5  /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_WIDTH                       1  /* LRCLK_DIR */
+#define WM9081_AIF_LRCLK_INV                    0x0010  /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_MASK               0x0010  /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_SHIFT                   4  /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_WIDTH                   1  /* AIF_LRCLK_INV */
+#define WM9081_AIF_WL_MASK                      0x000C  /* AIF_WL - [3:2] */
+#define WM9081_AIF_WL_SHIFT                          2  /* AIF_WL - [3:2] */
+#define WM9081_AIF_WL_WIDTH                          2  /* AIF_WL - [3:2] */
+#define WM9081_AIF_FMT_MASK                     0x0003  /* AIF_FMT - [1:0] */
+#define WM9081_AIF_FMT_SHIFT                         0  /* AIF_FMT - [1:0] */
+#define WM9081_AIF_FMT_WIDTH                         2  /* AIF_FMT - [1:0] */
+
+/*
+ * R24 (0x18) - Audio Interface 3
+ */
+#define WM9081_BCLK_DIV_MASK                    0x001F  /* BCLK_DIV - [4:0] */
+#define WM9081_BCLK_DIV_SHIFT                        0  /* BCLK_DIV - [4:0] */
+#define WM9081_BCLK_DIV_WIDTH                        5  /* BCLK_DIV - [4:0] */
+
+/*
+ * R25 (0x19) - Audio Interface 4
+ */
+#define WM9081_LRCLK_RATE_MASK                  0x07FF  /* LRCLK_RATE - [10:0] */
+#define WM9081_LRCLK_RATE_SHIFT                      0  /* LRCLK_RATE - [10:0] */
+#define WM9081_LRCLK_RATE_WIDTH                     11  /* LRCLK_RATE - [10:0] */
+
+/*
+ * R26 (0x1A) - Interrupt Status
+ */
+#define WM9081_WSEQ_BUSY_EINT                   0x0004  /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_MASK              0x0004  /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_SHIFT                  2  /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_WIDTH                  1  /* WSEQ_BUSY_EINT */
+#define WM9081_TSHUT_EINT                       0x0001  /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_MASK                  0x0001  /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_SHIFT                      0  /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_WIDTH                      1  /* TSHUT_EINT */
+
+/*
+ * R27 (0x1B) - Interrupt Status Mask
+ */
+#define WM9081_IM_WSEQ_BUSY_EINT                0x0004  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_MASK           0x0004  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_SHIFT               2  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_WIDTH               1  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_TSHUT_EINT                    0x0001  /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_MASK               0x0001  /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_SHIFT                   0  /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_WIDTH                   1  /* IM_TSHUT_EINT */
+
+/*
+ * R28 (0x1C) - Interrupt Polarity
+ */
+#define WM9081_TSHUT_INV                        0x0001  /* TSHUT_INV */
+#define WM9081_TSHUT_INV_MASK                   0x0001  /* TSHUT_INV */
+#define WM9081_TSHUT_INV_SHIFT                       0  /* TSHUT_INV */
+#define WM9081_TSHUT_INV_WIDTH                       1  /* TSHUT_INV */
+
+/*
+ * R29 (0x1D) - Interrupt Control
+ */
+#define WM9081_IRQ_POL                          0x8000  /* IRQ_POL */
+#define WM9081_IRQ_POL_MASK                     0x8000  /* IRQ_POL */
+#define WM9081_IRQ_POL_SHIFT                        15  /* IRQ_POL */
+#define WM9081_IRQ_POL_WIDTH                         1  /* IRQ_POL */
+#define WM9081_IRQ_OP_CTRL                      0x0001  /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_MASK                 0x0001  /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_SHIFT                     0  /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_WIDTH                     1  /* IRQ_OP_CTRL */
+
+/*
+ * R30 (0x1E) - DAC Digital 1
+ */
+#define WM9081_DAC_VOL_MASK                     0x00FF  /* DAC_VOL - [7:0] */
+#define WM9081_DAC_VOL_SHIFT                         0  /* DAC_VOL - [7:0] */
+#define WM9081_DAC_VOL_WIDTH                         8  /* DAC_VOL - [7:0] */
+
+/*
+ * R31 (0x1F) - DAC Digital 2
+ */
+#define WM9081_DAC_MUTERATE                     0x0400  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_MASK                0x0400  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_SHIFT                   10  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_WIDTH                    1  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTEMODE                     0x0200  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_MASK                0x0200  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_SHIFT                    9  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_WIDTH                    1  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTE                         0x0008  /* DAC_MUTE */
+#define WM9081_DAC_MUTE_MASK                    0x0008  /* DAC_MUTE */
+#define WM9081_DAC_MUTE_SHIFT                        3  /* DAC_MUTE */
+#define WM9081_DAC_MUTE_WIDTH                        1  /* DAC_MUTE */
+#define WM9081_DEEMPH_MASK                      0x0006  /* DEEMPH - [2:1] */
+#define WM9081_DEEMPH_SHIFT                          1  /* DEEMPH - [2:1] */
+#define WM9081_DEEMPH_WIDTH                          2  /* DEEMPH - [2:1] */
+
+/*
+ * R32 (0x20) - DRC 1
+ */
+#define WM9081_DRC_ENA                          0x8000  /* DRC_ENA */
+#define WM9081_DRC_ENA_MASK                     0x8000  /* DRC_ENA */
+#define WM9081_DRC_ENA_SHIFT                        15  /* DRC_ENA */
+#define WM9081_DRC_ENA_WIDTH                         1  /* DRC_ENA */
+#define WM9081_DRC_STARTUP_GAIN_MASK            0x07C0  /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_STARTUP_GAIN_SHIFT                6  /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_STARTUP_GAIN_WIDTH                5  /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_FF_DLY                       0x0020  /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_MASK                  0x0020  /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_SHIFT                      5  /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_WIDTH                      1  /* DRC_FF_DLY */
+#define WM9081_DRC_QR                           0x0004  /* DRC_QR */
+#define WM9081_DRC_QR_MASK                      0x0004  /* DRC_QR */
+#define WM9081_DRC_QR_SHIFT                          2  /* DRC_QR */
+#define WM9081_DRC_QR_WIDTH                          1  /* DRC_QR */
+#define WM9081_DRC_ANTICLIP                     0x0002  /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_MASK                0x0002  /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_SHIFT                    1  /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_WIDTH                    1  /* DRC_ANTICLIP */
+
+/*
+ * R33 (0x21) - DRC 2
+ */
+#define WM9081_DRC_ATK_MASK                     0xF000  /* DRC_ATK - [15:12] */
+#define WM9081_DRC_ATK_SHIFT                        12  /* DRC_ATK - [15:12] */
+#define WM9081_DRC_ATK_WIDTH                         4  /* DRC_ATK - [15:12] */
+#define WM9081_DRC_DCY_MASK                     0x0F00  /* DRC_DCY - [11:8] */
+#define WM9081_DRC_DCY_SHIFT                         8  /* DRC_DCY - [11:8] */
+#define WM9081_DRC_DCY_WIDTH                         4  /* DRC_DCY - [11:8] */
+#define WM9081_DRC_QR_THR_MASK                  0x00C0  /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_THR_SHIFT                      6  /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_THR_WIDTH                      2  /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_DCY_MASK                  0x0030  /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_QR_DCY_SHIFT                      4  /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_QR_DCY_WIDTH                      2  /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_MINGAIN_MASK                 0x000C  /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MINGAIN_SHIFT                     2  /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MINGAIN_WIDTH                     2  /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MAXGAIN_MASK                 0x0003  /* DRC_MAXGAIN - [1:0] */
+#define WM9081_DRC_MAXGAIN_SHIFT                     0  /* DRC_MAXGAIN - [1:0] */
+#define WM9081_DRC_MAXGAIN_WIDTH                     2  /* DRC_MAXGAIN - [1:0] */
+
+/*
+ * R34 (0x22) - DRC 3
+ */
+#define WM9081_DRC_HI_COMP_MASK                 0x0038  /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_HI_COMP_SHIFT                     3  /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_HI_COMP_WIDTH                     3  /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_LO_COMP_MASK                 0x0007  /* DRC_LO_COMP - [2:0] */
+#define WM9081_DRC_LO_COMP_SHIFT                     0  /* DRC_LO_COMP - [2:0] */
+#define WM9081_DRC_LO_COMP_WIDTH                     3  /* DRC_LO_COMP - [2:0] */
+
+/*
+ * R35 (0x23) - DRC 4
+ */
+#define WM9081_DRC_KNEE_IP_MASK                 0x07E0  /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_IP_SHIFT                     5  /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_IP_WIDTH                     6  /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_OP_MASK                 0x001F  /* DRC_KNEE_OP - [4:0] */
+#define WM9081_DRC_KNEE_OP_SHIFT                     0  /* DRC_KNEE_OP - [4:0] */
+#define WM9081_DRC_KNEE_OP_WIDTH                     5  /* DRC_KNEE_OP - [4:0] */
+
+/*
+ * R38 (0x26) - Write Sequencer 1
+ */
+#define WM9081_WSEQ_ENA                         0x8000  /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_MASK                    0x8000  /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_SHIFT                       15  /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_WIDTH                        1  /* WSEQ_ENA */
+#define WM9081_WSEQ_ABORT                       0x0200  /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_MASK                  0x0200  /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_SHIFT                      9  /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_WIDTH                      1  /* WSEQ_ABORT */
+#define WM9081_WSEQ_START                       0x0100  /* WSEQ_START */
+#define WM9081_WSEQ_START_MASK                  0x0100  /* WSEQ_START */
+#define WM9081_WSEQ_START_SHIFT                      8  /* WSEQ_START */
+#define WM9081_WSEQ_START_WIDTH                      1  /* WSEQ_START */
+#define WM9081_WSEQ_START_INDEX_MASK            0x007F  /* WSEQ_START_INDEX - [6:0] */
+#define WM9081_WSEQ_START_INDEX_SHIFT                0  /* WSEQ_START_INDEX - [6:0] */
+#define WM9081_WSEQ_START_INDEX_WIDTH                7  /* WSEQ_START_INDEX - [6:0] */
+
+/*
+ * R39 (0x27) - Write Sequencer 2
+ */
+#define WM9081_WSEQ_CURRENT_INDEX_MASK          0x07F0  /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_CURRENT_INDEX_SHIFT              4  /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_CURRENT_INDEX_WIDTH              7  /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_BUSY                        0x0001  /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_MASK                   0x0001  /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_SHIFT                       0  /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_WIDTH                       1  /* WSEQ_BUSY */
+
+/*
+ * R40 (0x28) - MW Slave 1
+ */
+#define WM9081_SPI_CFG                          0x0020  /* SPI_CFG */
+#define WM9081_SPI_CFG_MASK                     0x0020  /* SPI_CFG */
+#define WM9081_SPI_CFG_SHIFT                         5  /* SPI_CFG */
+#define WM9081_SPI_CFG_WIDTH                         1  /* SPI_CFG */
+#define WM9081_SPI_4WIRE                        0x0010  /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_MASK                   0x0010  /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_SHIFT                       4  /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_WIDTH                       1  /* SPI_4WIRE */
+#define WM9081_ARA_ENA                          0x0008  /* ARA_ENA */
+#define WM9081_ARA_ENA_MASK                     0x0008  /* ARA_ENA */
+#define WM9081_ARA_ENA_SHIFT                         3  /* ARA_ENA */
+#define WM9081_ARA_ENA_WIDTH                         1  /* ARA_ENA */
+#define WM9081_AUTO_INC                         0x0002  /* AUTO_INC */
+#define WM9081_AUTO_INC_MASK                    0x0002  /* AUTO_INC */
+#define WM9081_AUTO_INC_SHIFT                        1  /* AUTO_INC */
+#define WM9081_AUTO_INC_WIDTH                        1  /* AUTO_INC */
+
+/*
+ * R42 (0x2A) - EQ 1
+ */
+#define WM9081_EQ_B1_GAIN_MASK                  0xF800  /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B1_GAIN_SHIFT                     11  /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B1_GAIN_WIDTH                      5  /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B2_GAIN_MASK                  0x07C0  /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B2_GAIN_SHIFT                      6  /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B2_GAIN_WIDTH                      5  /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B4_GAIN_MASK                  0x003E  /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_B4_GAIN_SHIFT                      1  /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_B4_GAIN_WIDTH                      5  /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_ENA                           0x0001  /* EQ_ENA */
+#define WM9081_EQ_ENA_MASK                      0x0001  /* EQ_ENA */
+#define WM9081_EQ_ENA_SHIFT                          0  /* EQ_ENA */
+#define WM9081_EQ_ENA_WIDTH                          1  /* EQ_ENA */
+
+/*
+ * R43 (0x2B) - EQ 2
+ */
+#define WM9081_EQ_B3_GAIN_MASK                  0xF800  /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B3_GAIN_SHIFT                     11  /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B3_GAIN_WIDTH                      5  /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B5_GAIN_MASK                  0x07C0  /* EQ_B5_GAIN - [10:6] */
+#define WM9081_EQ_B5_GAIN_SHIFT                      6  /* EQ_B5_GAIN - [10:6] */
+#define WM9081_EQ_B5_GAIN_WIDTH                      5  /* EQ_B5_GAIN - [10:6] */
+
+/*
+ * R44 (0x2C) - EQ 3
+ */
+#define WM9081_EQ_B1_A_MASK                     0xFFFF  /* EQ_B1_A - [15:0] */
+#define WM9081_EQ_B1_A_SHIFT                         0  /* EQ_B1_A - [15:0] */
+#define WM9081_EQ_B1_A_WIDTH                        16  /* EQ_B1_A - [15:0] */
+
+/*
+ * R45 (0x2D) - EQ 4
+ */
+#define WM9081_EQ_B1_B_MASK                     0xFFFF  /* EQ_B1_B - [15:0] */
+#define WM9081_EQ_B1_B_SHIFT                         0  /* EQ_B1_B - [15:0] */
+#define WM9081_EQ_B1_B_WIDTH                        16  /* EQ_B1_B - [15:0] */
+
+/*
+ * R46 (0x2E) - EQ 5
+ */
+#define WM9081_EQ_B1_PG_MASK                    0xFFFF  /* EQ_B1_PG - [15:0] */
+#define WM9081_EQ_B1_PG_SHIFT                        0  /* EQ_B1_PG - [15:0] */
+#define WM9081_EQ_B1_PG_WIDTH                       16  /* EQ_B1_PG - [15:0] */
+
+/*
+ * R47 (0x2F) - EQ 6
+ */
+#define WM9081_EQ_B2_A_MASK                     0xFFFF  /* EQ_B2_A - [15:0] */
+#define WM9081_EQ_B2_A_SHIFT                         0  /* EQ_B2_A - [15:0] */
+#define WM9081_EQ_B2_A_WIDTH                        16  /* EQ_B2_A - [15:0] */
+
+/*
+ * R48 (0x30) - EQ 7
+ */
+#define WM9081_EQ_B2_B_MASK                     0xFFFF  /* EQ_B2_B - [15:0] */
+#define WM9081_EQ_B2_B_SHIFT                         0  /* EQ_B2_B - [15:0] */
+#define WM9081_EQ_B2_B_WIDTH                        16  /* EQ_B2_B - [15:0] */
+
+/*
+ * R49 (0x31) - EQ 8
+ */
+#define WM9081_EQ_B2_C_MASK                     0xFFFF  /* EQ_B2_C - [15:0] */
+#define WM9081_EQ_B2_C_SHIFT                         0  /* EQ_B2_C - [15:0] */
+#define WM9081_EQ_B2_C_WIDTH                        16  /* EQ_B2_C - [15:0] */
+
+/*
+ * R50 (0x32) - EQ 9
+ */
+#define WM9081_EQ_B2_PG_MASK                    0xFFFF  /* EQ_B2_PG - [15:0] */
+#define WM9081_EQ_B2_PG_SHIFT                        0  /* EQ_B2_PG - [15:0] */
+#define WM9081_EQ_B2_PG_WIDTH                       16  /* EQ_B2_PG - [15:0] */
+
+/*
+ * R51 (0x33) - EQ 10
+ */
+#define WM9081_EQ_B4_A_MASK                     0xFFFF  /* EQ_B4_A - [15:0] */
+#define WM9081_EQ_B4_A_SHIFT                         0  /* EQ_B4_A - [15:0] */
+#define WM9081_EQ_B4_A_WIDTH                        16  /* EQ_B4_A - [15:0] */
+
+/*
+ * R52 (0x34) - EQ 11
+ */
+#define WM9081_EQ_B4_B_MASK                     0xFFFF  /* EQ_B4_B - [15:0] */
+#define WM9081_EQ_B4_B_SHIFT                         0  /* EQ_B4_B - [15:0] */
+#define WM9081_EQ_B4_B_WIDTH                        16  /* EQ_B4_B - [15:0] */
+
+/*
+ * R53 (0x35) - EQ 12
+ */
+#define WM9081_EQ_B4_C_MASK                     0xFFFF  /* EQ_B4_C - [15:0] */
+#define WM9081_EQ_B4_C_SHIFT                         0  /* EQ_B4_C - [15:0] */
+#define WM9081_EQ_B4_C_WIDTH                        16  /* EQ_B4_C - [15:0] */
+
+/*
+ * R54 (0x36) - EQ 13
+ */
+#define WM9081_EQ_B4_PG_MASK                    0xFFFF  /* EQ_B4_PG - [15:0] */
+#define WM9081_EQ_B4_PG_SHIFT                        0  /* EQ_B4_PG - [15:0] */
+#define WM9081_EQ_B4_PG_WIDTH                       16  /* EQ_B4_PG - [15:0] */
+
+/*
+ * R55 (0x37) - EQ 14
+ */
+#define WM9081_EQ_B3_A_MASK                     0xFFFF  /* EQ_B3_A - [15:0] */
+#define WM9081_EQ_B3_A_SHIFT                         0  /* EQ_B3_A - [15:0] */
+#define WM9081_EQ_B3_A_WIDTH                        16  /* EQ_B3_A - [15:0] */
+
+/*
+ * R56 (0x38) - EQ 15
+ */
+#define WM9081_EQ_B3_B_MASK                     0xFFFF  /* EQ_B3_B - [15:0] */
+#define WM9081_EQ_B3_B_SHIFT                         0  /* EQ_B3_B - [15:0] */
+#define WM9081_EQ_B3_B_WIDTH                        16  /* EQ_B3_B - [15:0] */
+
+/*
+ * R57 (0x39) - EQ 16
+ */
+#define WM9081_EQ_B3_C_MASK                     0xFFFF  /* EQ_B3_C - [15:0] */
+#define WM9081_EQ_B3_C_SHIFT                         0  /* EQ_B3_C - [15:0] */
+#define WM9081_EQ_B3_C_WIDTH                        16  /* EQ_B3_C - [15:0] */
+
+/*
+ * R58 (0x3A) - EQ 17
+ */
+#define WM9081_EQ_B3_PG_MASK                    0xFFFF  /* EQ_B3_PG - [15:0] */
+#define WM9081_EQ_B3_PG_SHIFT                        0  /* EQ_B3_PG - [15:0] */
+#define WM9081_EQ_B3_PG_WIDTH                       16  /* EQ_B3_PG - [15:0] */
+
+/*
+ * R59 (0x3B) - EQ 18
+ */
+#define WM9081_EQ_B5_A_MASK                     0xFFFF  /* EQ_B5_A - [15:0] */
+#define WM9081_EQ_B5_A_SHIFT                         0  /* EQ_B5_A - [15:0] */
+#define WM9081_EQ_B5_A_WIDTH                        16  /* EQ_B5_A - [15:0] */
+
+/*
+ * R60 (0x3C) - EQ 19
+ */
+#define WM9081_EQ_B5_B_MASK                     0xFFFF  /* EQ_B5_B - [15:0] */
+#define WM9081_EQ_B5_B_SHIFT                         0  /* EQ_B5_B - [15:0] */
+#define WM9081_EQ_B5_B_WIDTH                        16  /* EQ_B5_B - [15:0] */
+
+/*
+ * R61 (0x3D) - EQ 20
+ */
+#define WM9081_EQ_B5_PG_MASK                    0xFFFF  /* EQ_B5_PG - [15:0] */
+#define WM9081_EQ_B5_PG_SHIFT                        0  /* EQ_B5_PG - [15:0] */
+#define WM9081_EQ_B5_PG_WIDTH                       16  /* EQ_B5_PG - [15:0] */
+
+
+#endif
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index c2d1a7a18fa..fa88b463e71 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -282,14 +282,14 @@ struct snd_soc_dai wm9705_dai[] = {
 			.channels_min = 1,
 			.channels_max = 2,
 			.rates = WM9705_AC97_RATES,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.formats = SND_SOC_STD_AC97_FMTS,
 		},
 		.capture = {
 			.stream_name = "HiFi Capture",
 			.channels_min = 1,
 			.channels_max = 2,
 			.rates = WM9705_AC97_RATES,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.formats = SND_SOC_STD_AC97_FMTS,
 		},
 		.ops = &wm9705_dai_ops,
 	},
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 765cf1e7369..1fd4e88f50c 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -534,13 +534,13 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM9712_AC97_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.capture = {
 		.stream_name = "HiFi Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM9712_AC97_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &wm9712_dai_ops_hifi,
 },
 {
@@ -550,7 +550,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_min = 1,
 		.channels_max = 1,
 		.rates = WM9712_AC97_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &wm9712_dai_ops_aux,
 }
 };
@@ -585,6 +585,8 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
 	}
 
 	soc_ac97_ops.reset(codec->ac97);
+	if (soc_ac97_ops.warm_reset)
+		soc_ac97_ops.warm_reset(codec->ac97);
 	if (ac97_read(codec, 0) != wm9712_reg[0])
 		goto err;
 	return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 523bad077fa..abed37acf78 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -189,6 +189,26 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
 SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
 };
 
+static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
+				 struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	u16 status, rate;
+
+	BUG_ON(event != SND_SOC_DAPM_PRE_PMD);
+
+	/* Gracefully shut down the voice interface. */
+	status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000;
+	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
+	schedule_timeout_interruptible(msecs_to_jiffies(1));
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
+	ac97_write(codec, AC97_EXTENDED_MID, status);
+
+	return 0;
+}
+
+
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path using the current
@@ -400,7 +420,8 @@ SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
+SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1,
+		   wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
 SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
@@ -689,7 +710,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source)
 	Ndiv = target / source;
 	if ((Ndiv < 5) || (Ndiv > 12))
 		printk(KERN_WARNING
-			"WM9713 PLL N value %d out of recommended range!\n",
+			"WM9713 PLL N value %u out of recommended range!\n",
 			Ndiv);
 
 	pll_div->n = Ndiv;
@@ -936,21 +957,6 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
-				 struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	u16 status, rate;
-
-	/* Gracefully shut down the voice interface. */
-	status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
-	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
-	schedule_timeout_interruptible(msecs_to_jiffies(1));
-	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
-	ac97_write(codec, AC97_EXTENDED_MID, status);
-}
-
 static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
@@ -1019,7 +1025,6 @@ static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
 
 static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
 	.hw_params	= wm9713_pcm_hw_params,
-	.shutdown	= wm9713_voiceshutdown,
 	.set_clkdiv	= wm9713_set_dai_clkdiv,
 	.set_pll	= wm9713_set_dai_pll,
 	.set_fmt	= wm9713_set_dai_fmt,
@@ -1035,13 +1040,13 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM9713_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.capture = {
 		.stream_name = "HiFi Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM9713_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &wm9713_dai_ops_hifi,
 	},
 	{
@@ -1051,7 +1056,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_min = 1,
 		.channels_max = 1,
 		.rates = WM9713_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &wm9713_dai_ops_aux,
 	},
 	{
@@ -1069,6 +1074,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
 	.ops = &wm9713_dai_ops_voice,
+	.symmetric_rates = 1,
 	},
 };
 EXPORT_SYMBOL_GPL(wm9713_dai);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 9fc90828337..5dbebf82249 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,5 +1,8 @@
 config SND_SOC_OF_SIMPLE
 	tristate
+	
+config SND_MPC52xx_DMA
+	tristate
 
 # ASoC platform support for the Freescale MPC8610 SOC.  This compiles drivers
 # for the SSI and the Elo DMA controller.  You will still need to select
@@ -22,7 +25,34 @@ config SND_SOC_MPC8610_HPCD
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
 	depends on PPC_MPC52xx && PPC_BESTCOMM
-	select SND_SOC_OF_SIMPLE
+	select SND_MPC52xx_DMA
 	select PPC_BESTCOMM_GEN_BD
 	help
 	  Say Y here to support the MPC5200 PSCs in I2S mode.
+
+config SND_SOC_MPC5200_AC97
+	tristate "Freescale MPC5200 PSC in AC97 mode driver"
+	depends on PPC_MPC52xx && PPC_BESTCOMM
+	select AC97_BUS
+	select SND_MPC52xx_DMA
+	select PPC_BESTCOMM_GEN_BD
+	help
+	  Say Y here to support the MPC5200 PSCs in AC97 mode.
+
+config SND_MPC52xx_SOC_PCM030
+	tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
+	depends on PPC_MPC5200_SIMPLE && BROKEN
+	select SND_SOC_MPC5200_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for sound on the Phytec pcm030
+	  baseboard.
+
+config SND_MPC52xx_SOC_EFIKA
+	tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
+	depends on PPC_EFIKA && BROKEN
+	select SND_SOC_MPC5200_AC97
+	select SND_SOC_STAC9766
+	help
+	  Say Y if you want to add support for sound on the Efika.
+
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index f85134c8638..a83a73967ec 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,5 +10,12 @@ snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-dma-objs := fsl_dma.o
 obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
 
+# MPC5200 Platform Support
+obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
+
+# MPC5200 Machine Support
+obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
+obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
 
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
new file mode 100644
index 00000000000..85b0e756950
--- /dev/null
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Efika driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/stac9766.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link efika_fabric_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 Analog",
+	.codec_dai = &stac9766_dai[STAC9766_DAI_AC97_ANALOG],
+	.cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+	.name = "AC97",
+	.stream_name = "AC97 IEC958",
+	.codec_dai = &stac9766_dai[STAC9766_DAI_AC97_DIGITAL],
+	.cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int efika_fabric_init(void)
+{
+	struct platform_device *pdev;
+	int rc;
+
+	if (!machine_is_compatible("bplan,efika"))
+		return -ENODEV;
+
+	card.platform = &mpc5200_audio_dma_platform;
+	card.name = "Efika";
+	card.dai_link = efika_fabric_dai;
+	card.num_links = ARRAY_SIZE(efika_fabric_dai);
+
+	device.card = &card;
+	device.codec_dev = &soc_codec_dev_stac9766;
+
+	pdev = platform_device_alloc("soc-audio", 1);
+	if (!pdev) {
+		pr_err("efika_fabric_init: platform_device_alloc() failed\n");
+		return -ENODEV;
+	}
+
+	platform_set_drvdata(pdev, &device);
+	device.dev = &pdev->dev;
+
+	rc = platform_device_add(pdev);
+	if (rc) {
+		pr_err("efika_fabric_init: platform_device_add() failed\n");
+		return -ENODEV;
+	}
+	return 0;
+}
+
+module_init(efika_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 3711d8454d9..93f0f38a32c 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -375,18 +375,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		struct snd_pcm_runtime *first_runtime =
 			ssi_private->first_stream->runtime;
 
-		if (!first_runtime->rate || !first_runtime->sample_bits) {
+		if (!first_runtime->sample_bits) {
 			dev_err(substream->pcm->card->dev,
-				"set sample rate and size in %s stream first\n",
+				"set sample size in %s stream first\n",
 				substream->stream == SNDRV_PCM_STREAM_PLAYBACK
 				? "capture" : "playback");
 			return -EAGAIN;
 		}
 
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-			SNDRV_PCM_HW_PARAM_RATE,
-			first_runtime->rate, first_runtime->rate);
-
 		/* If we're in synchronous mode, then we need to constrain
 		 * the sample size as well.  We don't support independent sample
 		 * rates in asynchronous mode.
@@ -674,7 +670,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	ssi_private->dev = ssi_info->dev;
 	ssi_private->asynchronous = ssi_info->asynchronous;
 
-	ssi_private->dev->driver_data = fsl_ssi_dai;
+	dev_set_drvdata(ssi_private->dev, fsl_ssi_dai);
 
 	/* Initialize the the device_attribute structure */
 	dev_attr->attr.name = "ssi-stats";
@@ -693,6 +689,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	fsl_ssi_dai->name = ssi_private->name;
 	fsl_ssi_dai->id = ssi_info->id;
 	fsl_ssi_dai->dev = ssi_info->dev;
+	fsl_ssi_dai->symmetric_rates = 1;
 
 	ret = snd_soc_register_dai(fsl_ssi_dai);
 	if (ret != 0) {
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
new file mode 100644
index 00000000000..efec33a1c5b
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -0,0 +1,564 @@
+/*
+ * Freescale MPC5200 PSC DMA
+ * ALSA SoC Platform driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+
+#include <sound/soc.h>
+
+#include <sysdev/bestcomm/bestcomm.h>
+#include <sysdev/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma)
+{
+	struct psc_dma *psc_dma = _psc_dma;
+	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+	u16 isr;
+
+	isr = in_be16(&regs->mpc52xx_psc_isr);
+
+	/* Playback underrun error */
+	if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+		psc_dma->stats.underrun_count++;
+
+	/* Capture overrun error */
+	if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+		psc_dma->stats.overrun_count++;
+
+	out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue.  Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
+{
+	struct bcom_bd *bd;
+
+	/* Prepare and enqueue the next buffer descriptor */
+	bd = bcom_prepare_next_buffer(s->bcom_task);
+	bd->status = s->period_bytes;
+	bd->data[0] = s->period_next_pt;
+	bcom_submit_next_buffer(s->bcom_task, NULL);
+
+	/* Update for next period */
+	s->period_next_pt += s->period_bytes;
+	if (s->period_next_pt >= s->period_end)
+		s->period_next_pt = s->period_start;
+}
+
+static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s)
+{
+	while (s->appl_ptr < s->runtime->control->appl_ptr) {
+
+		if (bcom_queue_full(s->bcom_task))
+			return;
+
+		s->appl_ptr += s->period_size;
+
+		psc_dma_bcom_enqueue_next_buffer(s);
+	}
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream)
+{
+	struct psc_dma_stream *s = _psc_dma_stream;
+
+	spin_lock(&s->psc_dma->lock);
+	/* For each finished period, dequeue the completed period buffer
+	 * and enqueue a new one in it's place. */
+	while (bcom_buffer_done(s->bcom_task)) {
+		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+		s->period_current_pt += s->period_bytes;
+		if (s->period_current_pt >= s->period_end)
+			s->period_current_pt = s->period_start;
+	}
+	psc_dma_bcom_enqueue_tx(s);
+	spin_unlock(&s->psc_dma->lock);
+
+	/* If the stream is active, then also inform the PCM middle layer
+	 * of the period finished event. */
+	if (s->active)
+		snd_pcm_period_elapsed(s->stream);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream)
+{
+	struct psc_dma_stream *s = _psc_dma_stream;
+
+	spin_lock(&s->psc_dma->lock);
+	/* For each finished period, dequeue the completed period buffer
+	 * and enqueue a new one in it's place. */
+	while (bcom_buffer_done(s->bcom_task)) {
+		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+		s->period_current_pt += s->period_bytes;
+		if (s->period_current_pt >= s->period_end)
+			s->period_current_pt = s->period_start;
+
+		psc_dma_bcom_enqueue_next_buffer(s);
+	}
+	spin_unlock(&s->psc_dma->lock);
+
+	/* If the stream is active, then also inform the PCM middle layer
+	 * of the period finished event. */
+	if (s->active)
+		snd_pcm_period_elapsed(s->stream);
+
+	return IRQ_HANDLED;
+}
+
+static int psc_dma_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+	return 0;
+}
+
+/**
+ * psc_dma_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct psc_dma_stream *s;
+	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+	u16 imr;
+	unsigned long flags;
+	int i;
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_dma->capture;
+	else
+		s = &psc_dma->playback;
+
+	dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)"
+		" stream_id=%i\n",
+		substream, cmd, substream->pstr->stream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		s->period_bytes = frames_to_bytes(runtime,
+						  runtime->period_size);
+		s->period_start = virt_to_phys(runtime->dma_area);
+		s->period_end = s->period_start +
+				(s->period_bytes * runtime->periods);
+		s->period_next_pt = s->period_start;
+		s->period_current_pt = s->period_start;
+		s->period_size = runtime->period_size;
+		s->active = 1;
+
+		/* track appl_ptr so that we have a better chance of detecting
+		 * end of stream and not over running it.
+		 */
+		s->runtime = runtime;
+		s->appl_ptr = s->runtime->control->appl_ptr -
+				(runtime->period_size * runtime->periods);
+
+		/* Fill up the bestcomm bd queue and enable DMA.
+		 * This will begin filling the PSC's fifo.
+		 */
+		spin_lock_irqsave(&psc_dma->lock, flags);
+
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			bcom_gen_bd_rx_reset(s->bcom_task);
+			for (i = 0; i < runtime->periods; i++)
+				if (!bcom_queue_full(s->bcom_task))
+					psc_dma_bcom_enqueue_next_buffer(s);
+		} else {
+			bcom_gen_bd_tx_reset(s->bcom_task);
+			psc_dma_bcom_enqueue_tx(s);
+		}
+
+		bcom_enable(s->bcom_task);
+		spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+		out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+		s->active = 0;
+
+		spin_lock_irqsave(&psc_dma->lock, flags);
+		bcom_disable(s->bcom_task);
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+			bcom_gen_bd_rx_reset(s->bcom_task);
+		else
+			bcom_gen_bd_tx_reset(s->bcom_task);
+		spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+		break;
+
+	default:
+		dev_dbg(psc_dma->dev, "invalid command\n");
+		return -EINVAL;
+	}
+
+	/* Update interrupt enable settings */
+	imr = 0;
+	if (psc_dma->playback.active)
+		imr |= MPC52xx_PSC_IMR_TXEMP;
+	if (psc_dma->capture.active)
+		imr |= MPC52xx_PSC_IMR_ORERR;
+	out_be16(&regs->isr_imr.imr, psc_dma->imr | imr);
+
+	return 0;
+}
+
+
+/* ---------------------------------------------------------------------
+ * The PSC DMA 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_dma_hardware = {
+	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_BATCH,
+	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
+		SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+	.rate_min = 8000,
+	.rate_max = 48000,
+	.channels_min = 1,
+	.channels_max = 2,
+	.period_bytes_max	= 1024 * 1024,
+	.period_bytes_min	= 32,
+	.periods_min		= 2,
+	.periods_max		= 256,
+	.buffer_bytes_max	= 2 * 1024 * 1024,
+	.fifo_size		= 512,
+};
+
+static int psc_dma_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	struct psc_dma_stream *s;
+	int rc;
+
+	dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream);
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_dma->capture;
+	else
+		s = &psc_dma->playback;
+
+	snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware);
+
+	rc = snd_pcm_hw_constraint_integer(runtime,
+		SNDRV_PCM_HW_PARAM_PERIODS);
+	if (rc < 0) {
+		dev_err(substream->pcm->card->dev, "invalid buffer size\n");
+		return rc;
+	}
+
+	s->stream = substream;
+	return 0;
+}
+
+static int psc_dma_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	struct psc_dma_stream *s;
+
+	dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_dma->capture;
+	else
+		s = &psc_dma->playback;
+
+	if (!psc_dma->playback.active &&
+	    !psc_dma->capture.active) {
+
+		/* Disable all interrupts and reset the PSC */
+		out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+		out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
+	}
+	s->stream = NULL;
+	return 0;
+}
+
+static snd_pcm_uframes_t
+psc_dma_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	struct psc_dma_stream *s;
+	dma_addr_t count;
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_dma->capture;
+	else
+		s = &psc_dma->playback;
+
+	count = s->period_current_pt - s->period_start;
+
+	return bytes_to_frames(substream->runtime, count);
+}
+
+static int
+psc_dma_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static struct snd_pcm_ops psc_dma_ops = {
+	.open		= psc_dma_open,
+	.close		= psc_dma_close,
+	.hw_free	= psc_dma_hw_free,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.pointer	= psc_dma_pointer,
+	.trigger	= psc_dma_trigger,
+	.hw_params	= psc_dma_hw_params,
+};
+
+static u64 psc_dma_dmamask = 0xffffffff;
+static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
+			   struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	size_t size = psc_dma_hardware.buffer_bytes_max;
+	int rc = 0;
+
+	dev_dbg(rtd->socdev->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n",
+		card, dai, pcm);
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &psc_dma_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = 0xffffffff;
+
+	if (pcm->streams[0].substream) {
+		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+				size, &pcm->streams[0].substream->dma_buffer);
+		if (rc)
+			goto playback_alloc_err;
+	}
+
+	if (pcm->streams[1].substream) {
+		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+				size, &pcm->streams[1].substream->dma_buffer);
+		if (rc)
+			goto capture_alloc_err;
+	}
+
+	if (rtd->socdev->card->codec->ac97)
+		rtd->socdev->card->codec->ac97->private_data = psc_dma;
+
+	return 0;
+
+ capture_alloc_err:
+	if (pcm->streams[0].substream)
+		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+
+ playback_alloc_err:
+	dev_err(card->dev, "Cannot allocate buffer(s)\n");
+
+	return -ENOMEM;
+}
+
+static void psc_dma_free(struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	struct snd_pcm_substream *substream;
+	int stream;
+
+	dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm);
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (substream) {
+			snd_dma_free_pages(&substream->dma_buffer);
+			substream->dma_buffer.area = NULL;
+			substream->dma_buffer.addr = 0;
+		}
+	}
+}
+
+struct snd_soc_platform mpc5200_audio_dma_platform = {
+	.name		= "mpc5200-psc-audio",
+	.pcm_ops	= &psc_dma_ops,
+	.pcm_new	= &psc_dma_new,
+	.pcm_free	= &psc_dma_free,
+};
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform);
+
+int mpc5200_audio_dma_create(struct of_device *op)
+{
+	phys_addr_t fifo;
+	struct psc_dma *psc_dma;
+	struct resource res;
+	int size, irq, rc;
+	const __be32 *prop;
+	void __iomem *regs;
+
+	/* Fetch the registers and IRQ of the PSC */
+	irq = irq_of_parse_and_map(op->node, 0);
+	if (of_address_to_resource(op->node, 0, &res)) {
+		dev_err(&op->dev, "Missing reg property\n");
+		return -ENODEV;
+	}
+	regs = ioremap(res.start, 1 + res.end - res.start);
+	if (!regs) {
+		dev_err(&op->dev, "Could not map registers\n");
+		return -ENODEV;
+	}
+
+	/* Allocate and initialize the driver private data */
+	psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
+	if (!psc_dma) {
+		iounmap(regs);
+		return -ENOMEM;
+	}
+
+	/* Get the PSC ID */
+	prop = of_get_property(op->node, "cell-index", &size);
+	if (!prop || size < sizeof *prop)
+		return -ENODEV;
+
+	spin_lock_init(&psc_dma->lock);
+	psc_dma->id = be32_to_cpu(*prop);
+	psc_dma->irq = irq;
+	psc_dma->psc_regs = regs;
+	psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs;
+	psc_dma->dev = &op->dev;
+	psc_dma->playback.psc_dma = psc_dma;
+	psc_dma->capture.psc_dma = psc_dma;
+	snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id);
+
+	/* Find the address of the fifo data registers and setup the
+	 * DMA tasks */
+	fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
+	psc_dma->capture.bcom_task =
+		bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512);
+	psc_dma->playback.bcom_task =
+		bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo);
+	if (!psc_dma->capture.bcom_task ||
+	    !psc_dma->playback.bcom_task) {
+		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
+		iounmap(regs);
+		kfree(psc_dma);
+		return -ENODEV;
+	}
+
+	/* Disable all interrupts and reset the PSC */
+	out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+	 /* reset receiver */
+	out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX);
+	 /* reset transmitter */
+	out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX);
+	 /* reset error */
+	out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT);
+	 /* reset mode */
+	out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1);
+
+	/* Set up mode register;
+	 * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
+	 * Second write: register Normal mode for non loopback
+	 */
+	out_8(&psc_dma->psc_regs->mode, 0);
+	out_8(&psc_dma->psc_regs->mode, 0);
+
+	/* Set the TX and RX fifo alarm thresholds */
+	out_be16(&psc_dma->fifo_regs->rfalarm, 0x100);
+	out_8(&psc_dma->fifo_regs->rfcntl, 0x4);
+	out_be16(&psc_dma->fifo_regs->tfalarm, 0x100);
+	out_8(&psc_dma->fifo_regs->tfcntl, 0x7);
+
+	/* Lookup the IRQ numbers */
+	psc_dma->playback.irq =
+		bcom_get_task_irq(psc_dma->playback.bcom_task);
+	psc_dma->capture.irq =
+		bcom_get_task_irq(psc_dma->capture.bcom_task);
+
+	rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED,
+			 "psc-dma-status", psc_dma);
+	rc |= request_irq(psc_dma->capture.irq,
+			  &psc_dma_bcom_irq_rx, IRQF_SHARED,
+			  "psc-dma-capture", &psc_dma->capture);
+	rc |= request_irq(psc_dma->playback.irq,
+			  &psc_dma_bcom_irq_tx, IRQF_SHARED,
+			  "psc-dma-playback", &psc_dma->playback);
+	if (rc) {
+		free_irq(psc_dma->irq, psc_dma);
+		free_irq(psc_dma->capture.irq,
+			 &psc_dma->capture);
+		free_irq(psc_dma->playback.irq,
+			 &psc_dma->playback);
+		return -ENODEV;
+	}
+
+	/* Save what we've done so it can be found again later */
+	dev_set_drvdata(&op->dev, psc_dma);
+
+	/* Tell the ASoC OF helpers about it */
+	return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
+
+int mpc5200_audio_dma_destroy(struct of_device *op)
+{
+	struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
+
+	dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n");
+
+	snd_soc_unregister_platform(&mpc5200_audio_dma_platform);
+
+	bcom_gen_bd_rx_release(psc_dma->capture.bcom_task);
+	bcom_gen_bd_tx_release(psc_dma->playback.bcom_task);
+
+	/* Release irqs */
+	free_irq(psc_dma->irq, psc_dma);
+	free_irq(psc_dma->capture.irq, &psc_dma->capture);
+	free_irq(psc_dma->playback.irq, &psc_dma->playback);
+
+	iounmap(psc_dma->psc_regs);
+	kfree(psc_dma);
+	dev_set_drvdata(&op->dev, NULL);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy);
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
new file mode 100644
index 00000000000..2000803f06a
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -0,0 +1,80 @@
+/*
+ * Freescale MPC5200 Audio DMA driver
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__
+#define __SOUND_SOC_FSL_MPC5200_DMA_H__
+
+#define PSC_STREAM_NAME_LEN 32
+
+/**
+ * psc_ac97_stream - Data specific to a single stream (playback or capture)
+ * @active:		flag indicating if the stream is active
+ * @psc_dma:		pointer back to parent psc_dma data structure
+ * @bcom_task:		bestcomm task structure
+ * @irq:		irq number for bestcomm task
+ * @period_start:	physical address of start of DMA region
+ * @period_end:		physical address of end of DMA region
+ * @period_next_pt:	physical address of next DMA buffer to enqueue
+ * @period_bytes:	size of DMA period in bytes
+ */
+struct psc_dma_stream {
+	struct snd_pcm_runtime *runtime;
+	snd_pcm_uframes_t appl_ptr;
+
+	int active;
+	struct psc_dma *psc_dma;
+	struct bcom_task *bcom_task;
+	int irq;
+	struct snd_pcm_substream *stream;
+	dma_addr_t period_start;
+	dma_addr_t period_end;
+	dma_addr_t period_next_pt;
+	dma_addr_t period_current_pt;
+	int period_bytes;
+	int period_size;
+};
+
+/**
+ * psc_dma - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @dai: the CPU DAI for this device
+ * @sicr: Base value used in serial interface control register; mode is ORed
+ *        with this value.
+ * @playback: Playback stream context data
+ * @capture: Capture stream context data
+ */
+struct psc_dma {
+	char name[32];
+	struct mpc52xx_psc __iomem *psc_regs;
+	struct mpc52xx_psc_fifo __iomem *fifo_regs;
+	unsigned int irq;
+	struct device *dev;
+	spinlock_t lock;
+	u32 sicr;
+	uint sysclk;
+	int imr;
+	int id;
+	unsigned int slots;
+
+	/* per-stream data */
+	struct psc_dma_stream playback;
+	struct psc_dma_stream capture;
+
+	/* Statistics */
+	struct {
+		unsigned long overrun_count;
+		unsigned long underrun_count;
+	} stats;
+};
+
+int mpc5200_audio_dma_create(struct of_device *op);
+int mpc5200_audio_dma_destroy(struct of_device *op);
+
+extern struct snd_soc_platform mpc5200_audio_dma_platform;
+
+#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
new file mode 100644
index 00000000000..794a247b3eb
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -0,0 +1,329 @@
+/*
+ * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
+ *
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.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/of_device.h>
+#include <linux/of_platform.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/time.h>
+#include <asm/delay.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+
+#define DRV_NAME "mpc5200-psc-ac97"
+
+/* ALSA only supports a single AC97 device so static is recommend here */
+static struct psc_dma *psc_dma;
+
+static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+	int status;
+	unsigned int val;
+
+	/* Wait for command send status zero = ready */
+	status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+				MPC52xx_PSC_SR_CMDSEND), 100, 0);
+	if (status == 0) {
+		pr_err("timeout on ac97 bus (rdy)\n");
+		return -ENODEV;
+	}
+	/* Send the read */
+	out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
+
+	/* Wait for the answer */
+	status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) &
+				MPC52xx_PSC_SR_DATA_VAL), 100, 0);
+	if (status == 0) {
+		pr_err("timeout on ac97 read (val) %x\n",
+				in_be16(&psc_dma->psc_regs->sr_csr.status));
+		return -ENODEV;
+	}
+	/* Get the data */
+	val = in_be32(&psc_dma->psc_regs->ac97_data);
+	if (((val >> 24) & 0x7f) != reg) {
+		pr_err("reg echo error on ac97 read\n");
+		return -ENODEV;
+	}
+	val = (val >> 8) & 0xffff;
+
+	return (unsigned short) val;
+}
+
+static void psc_ac97_write(struct snd_ac97 *ac97,
+				unsigned short reg, unsigned short val)
+{
+	int status;
+
+	/* Wait for command status zero = ready */
+	status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+				MPC52xx_PSC_SR_CMDSEND), 100, 0);
+	if (status == 0) {
+		pr_err("timeout on ac97 bus (write)\n");
+		return;
+	}
+	/* Write data */
+	out_be32(&psc_dma->psc_regs->ac97_cmd,
+			((reg & 0x7f) << 24) | (val << 8));
+}
+
+static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+	out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR);
+	udelay(3);
+	out_be32(&regs->sicr, psc_dma->sicr);
+}
+
+static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+	/* Do a cold reset */
+	out_8(&regs->op1, MPC52xx_PSC_OP_RES);
+	udelay(10);
+	out_8(&regs->op0, MPC52xx_PSC_OP_RES);
+	udelay(50);
+	psc_ac97_warm_reset(ac97);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+	.read		= psc_ac97_read,
+	.write		= psc_ac97_write,
+	.reset		= psc_ac97_cold_reset,
+	.warm_reset	= psc_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *cpu_dai)
+{
+	struct psc_dma *psc_dma = cpu_dai->private_data;
+
+	dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+		" periods=%i buffer_size=%i  buffer_bytes=%i channels=%i"
+		" rate=%i format=%i\n",
+		__func__, substream, params_period_size(params),
+		params_period_bytes(params), params_periods(params),
+		params_buffer_size(params), params_buffer_bytes(params),
+		params_channels(params), params_rate(params),
+		params_format(params));
+
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		if (params_channels(params) == 1)
+			psc_dma->slots |= 0x00000100;
+		else
+			psc_dma->slots |= 0x00000300;
+	} else {
+		if (params_channels(params) == 1)
+			psc_dma->slots |= 0x01000000;
+		else
+			psc_dma->slots |= 0x03000000;
+	}
+	out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+
+	return 0;
+}
+
+static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *cpu_dai)
+{
+	struct psc_dma *psc_dma = cpu_dai->private_data;
+
+	if (params_channels(params) == 1)
+		out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
+	else
+		out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000);
+
+	return 0;
+}
+
+static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+							struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+			psc_dma->slots &= 0xFFFF0000;
+		else
+			psc_dma->slots &= 0x0000FFFF;
+
+		out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+		break;
+	}
+	return 0;
+}
+
+static int psc_ac97_probe(struct platform_device *pdev,
+					struct snd_soc_dai *cpu_dai)
+{
+	struct psc_dma *psc_dma = cpu_dai->private_data;
+	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+	/* Go */
+	out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+	return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_ac97_dai_template: template CPU Digital Audio Interface
+ */
+static struct snd_soc_dai_ops psc_ac97_analog_ops = {
+	.hw_params	= psc_ac97_hw_analog_params,
+	.trigger	= psc_ac97_trigger,
+};
+
+static struct snd_soc_dai_ops psc_ac97_digital_ops = {
+	.hw_params	= psc_ac97_hw_digital_params,
+};
+
+struct snd_soc_dai psc_ac97_dai[] = {
+{
+	.name   = "AC97",
+	.ac97_control = 1,
+	.probe	= psc_ac97_probe,
+	.playback = {
+		.channels_min   = 1,
+		.channels_max   = 6,
+		.rates          = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S32_BE,
+	},
+	.capture = {
+		.channels_min   = 1,
+		.channels_max   = 2,
+		.rates          = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S32_BE,
+	},
+	.ops = &psc_ac97_analog_ops,
+},
+{
+	.name   = "SPDIF",
+	.ac97_control = 1,
+	.playback = {
+		.channels_min   = 1,
+		.channels_max   = 2,
+		.rates          = SNDRV_PCM_RATE_32000 | \
+			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+	},
+	.ops = &psc_ac97_digital_ops,
+} };
+EXPORT_SYMBOL_GPL(psc_ac97_dai);
+
+
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int __devinit psc_ac97_of_probe(struct of_device *op,
+				      const struct of_device_id *match)
+{
+	int rc, i;
+	struct snd_ac97 ac97;
+	struct mpc52xx_psc __iomem *regs;
+
+	rc = mpc5200_audio_dma_create(op);
+	if (rc != 0)
+		return rc;
+
+	for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+		psc_ac97_dai[i].dev = &op->dev;
+
+	rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
+	if (rc != 0) {
+		dev_err(&op->dev, "Failed to register DAI\n");
+		return rc;
+	}
+
+	psc_dma = dev_get_drvdata(&op->dev);
+	regs = psc_dma->psc_regs;
+	ac97.private_data = psc_dma;
+
+	for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+		psc_ac97_dai[i].private_data = psc_dma;
+
+	psc_dma->imr = 0;
+	out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+
+	/* Configure the serial interface mode to AC97 */
+	psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97;
+	out_be32(&regs->sicr, psc_dma->sicr);
+
+	/* No slots active */
+	out_be32(&regs->ac97_slots, 0x00000000);
+
+	return 0;
+}
+
+static int __devexit psc_ac97_of_remove(struct of_device *op)
+{
+	return mpc5200_audio_dma_destroy(op);
+}
+
+/* Match table for of_platform binding */
+static struct of_device_id psc_ac97_match[] __devinitdata = {
+	{ .compatible = "fsl,mpc5200-psc-ac97", },
+	{ .compatible = "fsl,mpc5200b-psc-ac97", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, psc_ac97_match);
+
+static struct of_platform_driver psc_ac97_driver = {
+	.match_table = psc_ac97_match,
+	.probe = psc_ac97_of_probe,
+	.remove = __devexit_p(psc_ac97_of_remove),
+	.driver = {
+		.name = "mpc5200-psc-ac97",
+		.owner = THIS_MODULE,
+	},
+};
+
+/* ---------------------------------------------------------------------
+ * Module setup and teardown; simply register the of_platform driver
+ * for the PSC in AC97 mode.
+ */
+static int __init psc_ac97_init(void)
+{
+	return of_register_platform_driver(&psc_ac97_driver);
+}
+module_init(psc_ac97_init);
+
+static void __exit psc_ac97_exit(void)
+{
+	of_unregister_platform_driver(&psc_ac97_driver);
+}
+module_exit(psc_ac97_exit);
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION("mpc5200 AC97 module");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h
new file mode 100644
index 00000000000..4bc18c35c36
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.h
@@ -0,0 +1,15 @@
+/*
+ * Freescale MPC5200 PSC in AC97 mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+
+extern struct snd_soc_dai psc_ac97_dai[];
+
+#define MPC5200_AC97_NORMAL 0
+#define MPC5200_AC97_SPDIF 1
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 1111c710118..ce8de90fb94 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -3,31 +3,21 @@
  * ALSA SoC Digital Audio Interface (DAI) driver
  *
  * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
  */
 
-#include <linux/init.h>
 #include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
 #include <linux/of_device.h>
 #include <linux/of_platform.h>
-#include <linux/dma-mapping.h>
 
-#include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-#include <sound/initval.h>
 #include <sound/soc.h>
-#include <sound/soc-of-simple.h>
 
-#include <sysdev/bestcomm/bestcomm.h>
-#include <sysdev/bestcomm/gen_bd.h>
 #include <asm/mpc52xx_psc.h>
 
-MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
-MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
-MODULE_LICENSE("GPL");
+#include "mpc5200_psc_i2s.h"
+#include "mpc5200_dma.h"
 
 /**
  * PSC_I2S_RATES: sample rates supported by the I2S
@@ -44,191 +34,17 @@ MODULE_LICENSE("GPL");
  * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
  */
 #define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
-			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
-			 SNDRV_PCM_FMTBIT_S32_BE)
-
-/**
- * psc_i2s_stream - Data specific to a single stream (playback or capture)
- * @active:		flag indicating if the stream is active
- * @psc_i2s:		pointer back to parent psc_i2s data structure
- * @bcom_task:		bestcomm task structure
- * @irq:		irq number for bestcomm task
- * @period_start:	physical address of start of DMA region
- * @period_end:		physical address of end of DMA region
- * @period_next_pt:	physical address of next DMA buffer to enqueue
- * @period_bytes:	size of DMA period in bytes
- */
-struct psc_i2s_stream {
-	int active;
-	struct psc_i2s *psc_i2s;
-	struct bcom_task *bcom_task;
-	int irq;
-	struct snd_pcm_substream *stream;
-	dma_addr_t period_start;
-	dma_addr_t period_end;
-	dma_addr_t period_next_pt;
-	dma_addr_t period_current_pt;
-	int period_bytes;
-};
-
-/**
- * psc_i2s - Private driver data
- * @name: short name for this device ("PSC0", "PSC1", etc)
- * @psc_regs: pointer to the PSC's registers
- * @fifo_regs: pointer to the PSC's FIFO registers
- * @irq: IRQ of this PSC
- * @dev: struct device pointer
- * @dai: the CPU DAI for this device
- * @sicr: Base value used in serial interface control register; mode is ORed
- *        with this value.
- * @playback: Playback stream context data
- * @capture: Capture stream context data
- */
-struct psc_i2s {
-	char name[32];
-	struct mpc52xx_psc __iomem *psc_regs;
-	struct mpc52xx_psc_fifo __iomem *fifo_regs;
-	unsigned int irq;
-	struct device *dev;
-	struct snd_soc_dai dai;
-	spinlock_t lock;
-	u32 sicr;
-
-	/* per-stream data */
-	struct psc_i2s_stream playback;
-	struct psc_i2s_stream capture;
-
-	/* Statistics */
-	struct {
-		int overrun_count;
-		int underrun_count;
-	} stats;
-};
-
-/*
- * Interrupt handlers
- */
-static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
-{
-	struct psc_i2s *psc_i2s = _psc_i2s;
-	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
-	u16 isr;
-
-	isr = in_be16(&regs->mpc52xx_psc_isr);
-
-	/* Playback underrun error */
-	if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
-		psc_i2s->stats.underrun_count++;
-
-	/* Capture overrun error */
-	if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
-		psc_i2s->stats.overrun_count++;
-
-	out_8(&regs->command, 4 << 4);	/* reset the error status */
-
-	return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- * @s: pointer to stream private data structure
- *
- * Enqueues another audio period buffer into the bestcomm queue.
- *
- * Note: The routine must only be called when there is space available in
- * the queue.  Otherwise the enqueue will fail and the audio ring buffer
- * will get out of sync
- */
-static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
-{
-	struct bcom_bd *bd;
-
-	/* Prepare and enqueue the next buffer descriptor */
-	bd = bcom_prepare_next_buffer(s->bcom_task);
-	bd->status = s->period_bytes;
-	bd->data[0] = s->period_next_pt;
-	bcom_submit_next_buffer(s->bcom_task, NULL);
-
-	/* Update for next period */
-	s->period_next_pt += s->period_bytes;
-	if (s->period_next_pt >= s->period_end)
-		s->period_next_pt = s->period_start;
-}
-
-/* Bestcomm DMA irq handler */
-static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
-{
-	struct psc_i2s_stream *s = _psc_i2s_stream;
-
-	/* For each finished period, dequeue the completed period buffer
-	 * and enqueue a new one in it's place. */
-	while (bcom_buffer_done(s->bcom_task)) {
-		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-		s->period_current_pt += s->period_bytes;
-		if (s->period_current_pt >= s->period_end)
-			s->period_current_pt = s->period_start;
-		psc_i2s_bcom_enqueue_next_buffer(s);
-		bcom_enable(s->bcom_task);
-	}
-
-	/* If the stream is active, then also inform the PCM middle layer
-	 * of the period finished event. */
-	if (s->active)
-		snd_pcm_period_elapsed(s->stream);
-
-	return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_startup: create a new substream
- *
- * This is the first function called when a stream is opened.
- *
- * If this is the first stream open, then grab the IRQ and program most of
- * the PSC registers.
- */
-static int psc_i2s_startup(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	int rc;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
-
-	if (!psc_i2s->playback.active &&
-	    !psc_i2s->capture.active) {
-		/* Setup the IRQs */
-		rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
-				 "psc-i2s-status", psc_i2s);
-		rc |= request_irq(psc_i2s->capture.irq,
-				  &psc_i2s_bcom_irq, IRQF_SHARED,
-				  "psc-i2s-capture", &psc_i2s->capture);
-		rc |= request_irq(psc_i2s->playback.irq,
-				  &psc_i2s_bcom_irq, IRQF_SHARED,
-				  "psc-i2s-playback", &psc_i2s->playback);
-		if (rc) {
-			free_irq(psc_i2s->irq, psc_i2s);
-			free_irq(psc_i2s->capture.irq,
-				 &psc_i2s->capture);
-			free_irq(psc_i2s->playback.irq,
-				 &psc_i2s->playback);
-			return -ENODEV;
-		}
-	}
-
-	return 0;
-}
+			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
 
 static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
 	u32 mode;
 
-	dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+	dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
 		" periods=%i buffer_size=%i  buffer_bytes=%i\n",
 		__func__, substream, params_period_size(params),
 		params_period_bytes(params), params_periods(params),
@@ -248,175 +64,15 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 		mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
 		break;
 	default:
-		dev_dbg(psc_i2s->dev, "invalid format\n");
-		return -EINVAL;
-	}
-	out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode);
-
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
-	return 0;
-}
-
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
-{
-	snd_pcm_set_runtime_buffer(substream, NULL);
-	return 0;
-}
-
-/**
- * psc_i2s_trigger: start and stop the DMA transfer.
- *
- * This function is called by ALSA to start, stop, pause, and resume the DMA
- * transfer of data.
- */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct psc_i2s_stream *s;
-	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
-	u16 imr;
-	u8 psc_cmd;
-	unsigned long flags;
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
-		" stream_id=%i\n",
-		substream, cmd, substream->pstr->stream);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		s->period_bytes = frames_to_bytes(runtime,
-						  runtime->period_size);
-		s->period_start = virt_to_phys(runtime->dma_area);
-		s->period_end = s->period_start +
-				(s->period_bytes * runtime->periods);
-		s->period_next_pt = s->period_start;
-		s->period_current_pt = s->period_start;
-		s->active = 1;
-
-		/* First; reset everything */
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
-			out_8(&regs->command, MPC52xx_PSC_RST_RX);
-			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
-		} else {
-			out_8(&regs->command, MPC52xx_PSC_RST_TX);
-			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
-		}
-
-		/* Next, fill up the bestcomm bd queue and enable DMA.
-		 * This will begin filling the PSC's fifo. */
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-			bcom_gen_bd_rx_reset(s->bcom_task);
-		else
-			bcom_gen_bd_tx_reset(s->bcom_task);
-		while (!bcom_queue_full(s->bcom_task))
-			psc_i2s_bcom_enqueue_next_buffer(s);
-		bcom_enable(s->bcom_task);
-
-		/* Due to errata in the i2s mode; need to line up enabling
-		 * the transmitter with a transition on the frame sync
-		 * line */
-
-		spin_lock_irqsave(&psc_i2s->lock, flags);
-		/* first make sure it is low */
-		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
-			;
-		/* then wait for the transition to high */
-		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
-			;
-		/* Finally, enable the PSC.
-		 * Receiver must always be enabled; even when we only want
-		 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
-		psc_cmd = MPC52xx_PSC_RX_ENABLE;
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
-		out_8(&regs->command, psc_cmd);
-		spin_unlock_irqrestore(&psc_i2s->lock, flags);
-
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-		/* Turn off the PSC */
-		s->active = 0;
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
-			if (!psc_i2s->playback.active) {
-				out_8(&regs->command, 2 << 4);	/* reset rx */
-				out_8(&regs->command, 3 << 4);	/* reset tx */
-				out_8(&regs->command, 4 << 4);	/* reset err */
-			}
-		} else {
-			out_8(&regs->command, 3 << 4);	/* reset tx */
-			out_8(&regs->command, 4 << 4);	/* reset err */
-			if (!psc_i2s->capture.active)
-				out_8(&regs->command, 2 << 4);	/* reset rx */
-		}
-
-		bcom_disable(s->bcom_task);
-		while (!bcom_queue_empty(s->bcom_task))
-			bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-
-		break;
-
-	default:
-		dev_dbg(psc_i2s->dev, "invalid command\n");
+		dev_dbg(psc_dma->dev, "invalid format\n");
 		return -EINVAL;
 	}
-
-	/* Update interrupt enable settings */
-	imr = 0;
-	if (psc_i2s->playback.active)
-		imr |= MPC52xx_PSC_IMR_TXEMP;
-	if (psc_i2s->capture.active)
-		imr |= MPC52xx_PSC_IMR_ORERR;
-	out_be16(&regs->isr_imr.imr, imr);
+	out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);
 
 	return 0;
 }
 
 /**
- * psc_i2s_shutdown: shutdown the data transfer on a stream
- *
- * Shutdown the PSC if there are no other substreams open.
- */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
-
-	/*
-	 * If this is the last active substream, disable the PSC and release
-	 * the IRQ.
-	 */
-	if (!psc_i2s->playback.active &&
-	    !psc_i2s->capture.active) {
-
-		/* Disable all interrupts and reset the PSC */
-		out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
-		out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
-		out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
-		out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
-		out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
-
-		/* Release irqs */
-		free_irq(psc_i2s->irq, psc_i2s);
-		free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
-		free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
-	}
-}
-
-/**
  * psc_i2s_set_sysclk: set the clock frequency and direction
  *
  * This function is called by the machine driver to tell us what the clock
@@ -433,8 +89,8 @@ static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
 static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 			      int clk_id, unsigned int freq, int dir)
 {
-	struct psc_i2s *psc_i2s = cpu_dai->private_data;
-	dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+	struct psc_dma *psc_dma = cpu_dai->private_data;
+	dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
 				cpu_dai, dir);
 	return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
 }
@@ -452,8 +108,8 @@ static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
  */
 static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 {
-	struct psc_i2s *psc_i2s = cpu_dai->private_data;
-	dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+	struct psc_dma *psc_dma = cpu_dai->private_data;
+	dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
 				cpu_dai, format);
 	return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
 }
@@ -469,16 +125,13 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
 static struct snd_soc_dai_ops psc_i2s_dai_ops = {
-	.startup	= psc_i2s_startup,
 	.hw_params	= psc_i2s_hw_params,
-	.hw_free	= psc_i2s_hw_free,
-	.shutdown	= psc_i2s_shutdown,
-	.trigger	= psc_i2s_trigger,
 	.set_sysclk	= psc_i2s_set_sysclk,
 	.set_fmt	= psc_i2s_set_fmt,
 };
 
-static struct snd_soc_dai psc_i2s_dai_template = {
+struct snd_soc_dai psc_i2s_dai[] = {{
+	.name   = "I2S",
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
@@ -492,223 +145,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.formats = PSC_I2S_FORMATS,
 	},
 	.ops = &psc_i2s_dai_ops,
-};
-
-/* ---------------------------------------------------------------------
- * The PSC I2S 'ASoC platform' driver
- *
- * Can be referenced by an 'ASoC machine' driver
- * This driver only deals with the audio bus; it doesn't have any
- * interaction with the attached codec
- */
-
-static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
-	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_BATCH,
-	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
-		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
-	.rate_min = 8000,
-	.rate_max = 48000,
-	.channels_min = 2,
-	.channels_max = 2,
-	.period_bytes_max	= 1024 * 1024,
-	.period_bytes_min	= 32,
-	.periods_min		= 2,
-	.periods_max		= 256,
-	.buffer_bytes_max	= 2 * 1024 * 1024,
-	.fifo_size		= 0,
-};
-
-static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
-
-	s->stream = substream;
-	return 0;
-}
-
-static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	s->stream = NULL;
-	return 0;
-}
-
-static snd_pcm_uframes_t
-psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
-	dma_addr_t count;
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	count = s->period_current_pt - s->period_start;
-
-	return bytes_to_frames(substream->runtime, count);
-}
-
-static struct snd_pcm_ops psc_i2s_pcm_ops = {
-	.open		= psc_i2s_pcm_open,
-	.close		= psc_i2s_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.pointer	= psc_i2s_pcm_pointer,
-};
-
-static u64 psc_i2s_pcm_dmamask = 0xffffffff;
-static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-			   struct snd_pcm *pcm)
-{
-	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
-	int rc = 0;
-
-	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
-		card, dai, pcm);
-
-	if (!card->dev->dma_mask)
-		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
-	if (!card->dev->coherent_dma_mask)
-		card->dev->coherent_dma_mask = 0xffffffff;
-
-	if (pcm->streams[0].substream) {
-		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
-					&pcm->streams[0].substream->dma_buffer);
-		if (rc)
-			goto playback_alloc_err;
-	}
-
-	if (pcm->streams[1].substream) {
-		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
-					&pcm->streams[1].substream->dma_buffer);
-		if (rc)
-			goto capture_alloc_err;
-	}
-
-	return 0;
-
- capture_alloc_err:
-	if (pcm->streams[0].substream)
-		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
- playback_alloc_err:
-	dev_err(card->dev, "Cannot allocate buffer(s)\n");
-	return -ENOMEM;
-}
-
-static void psc_i2s_pcm_free(struct snd_pcm *pcm)
-{
-	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	struct snd_pcm_substream *substream;
-	int stream;
-
-	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
-
-	for (stream = 0; stream < 2; stream++) {
-		substream = pcm->streams[stream].substream;
-		if (substream) {
-			snd_dma_free_pages(&substream->dma_buffer);
-			substream->dma_buffer.area = NULL;
-			substream->dma_buffer.addr = 0;
-		}
-	}
-}
-
-struct snd_soc_platform psc_i2s_pcm_soc_platform = {
-	.name		= "mpc5200-psc-audio",
-	.pcm_ops	= &psc_i2s_pcm_ops,
-	.pcm_new	= &psc_i2s_pcm_new,
-	.pcm_free	= &psc_i2s_pcm_free,
-};
-
-/* ---------------------------------------------------------------------
- * Sysfs attributes for debugging
- */
-
-static ssize_t psc_i2s_status_show(struct device *dev,
-			   struct device_attribute *attr, char *buf)
-{
-	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-
-	return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x "
-			"tfnum=%i tfstat=0x%.4x\n",
-			in_be16(&psc_i2s->psc_regs->sr_csr.status),
-			in_be32(&psc_i2s->psc_regs->sicr),
-			in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
-			in_be16(&psc_i2s->fifo_regs->rfstat),
-			in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
-			in_be16(&psc_i2s->fifo_regs->tfstat));
-}
-
-static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name)
-{
-	if (strcmp(name, "playback_underrun") == 0)
-		return &psc_i2s->stats.underrun_count;
-	if (strcmp(name, "capture_overrun") == 0)
-		return &psc_i2s->stats.overrun_count;
-
-	return NULL;
-}
-
-static ssize_t psc_i2s_stat_show(struct device *dev,
-				 struct device_attribute *attr, char *buf)
-{
-	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-	int *attrib;
-
-	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
-	if (!attrib)
-		return 0;
-
-	return sprintf(buf, "%i\n", *attrib);
-}
-
-static ssize_t psc_i2s_stat_store(struct device *dev,
-				  struct device_attribute *attr,
-				  const char *buf,
-				  size_t count)
-{
-	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-	int *attrib;
-
-	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
-	if (!attrib)
-		return 0;
-
-	*attrib = simple_strtoul(buf, NULL, 0);
-	return count;
-}
-
-static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
-static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,
-			psc_i2s_stat_store);
-static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
-			psc_i2s_stat_store);
+} };
+EXPORT_SYMBOL_GPL(psc_i2s_dai);
 
 /* ---------------------------------------------------------------------
  * OF platform bus binding code:
@@ -718,150 +156,65 @@ static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
 static int __devinit psc_i2s_of_probe(struct of_device *op,
 				      const struct of_device_id *match)
 {
-	phys_addr_t fifo;
-	struct psc_i2s *psc_i2s;
-	struct resource res;
-	int size, psc_id, irq, rc;
-	const __be32 *prop;
-	void __iomem *regs;
-
-	dev_dbg(&op->dev, "probing psc i2s device\n");
-
-	/* Get the PSC ID */
-	prop = of_get_property(op->node, "cell-index", &size);
-	if (!prop || size < sizeof *prop)
-		return -ENODEV;
-	psc_id = be32_to_cpu(*prop);
-
-	/* Fetch the registers and IRQ of the PSC */
-	irq = irq_of_parse_and_map(op->node, 0);
-	if (of_address_to_resource(op->node, 0, &res)) {
-		dev_err(&op->dev, "Missing reg property\n");
-		return -ENODEV;
-	}
-	regs = ioremap(res.start, 1 + res.end - res.start);
-	if (!regs) {
-		dev_err(&op->dev, "Could not map registers\n");
-		return -ENODEV;
-	}
+	int rc;
+	struct psc_dma *psc_dma;
+	struct mpc52xx_psc __iomem *regs;
 
-	/* Allocate and initialize the driver private data */
-	psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
-	if (!psc_i2s) {
-		iounmap(regs);
-		return -ENOMEM;
-	}
-	spin_lock_init(&psc_i2s->lock);
-	psc_i2s->irq = irq;
-	psc_i2s->psc_regs = regs;
-	psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
-	psc_i2s->dev = &op->dev;
-	psc_i2s->playback.psc_i2s = psc_i2s;
-	psc_i2s->capture.psc_i2s = psc_i2s;
-	snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
-
-	/* Fill out the CPU DAI structure */
-	memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
-	psc_i2s->dai.private_data = psc_i2s;
-	psc_i2s->dai.name = psc_i2s->name;
-	psc_i2s->dai.id = psc_id;
-
-	/* Find the address of the fifo data registers and setup the
-	 * DMA tasks */
-	fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
-	psc_i2s->capture.bcom_task =
-		bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
-	psc_i2s->playback.bcom_task =
-		bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
-	if (!psc_i2s->capture.bcom_task ||
-	    !psc_i2s->playback.bcom_task) {
-		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
-		iounmap(regs);
-		kfree(psc_i2s);
-		return -ENODEV;
+	rc = mpc5200_audio_dma_create(op);
+	if (rc != 0)
+		return rc;
+
+	rc = snd_soc_register_dais(psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
+	if (rc != 0) {
+		pr_err("Failed to register DAI\n");
+		return 0;
 	}
 
-	/* Disable all interrupts and reset the PSC */
-	out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
-	out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */
-	out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */
-	out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
-	out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+	psc_dma = dev_get_drvdata(&op->dev);
+	regs = psc_dma->psc_regs;
 
 	/* Configure the serial interface mode; defaulting to CODEC8 mode */
-	psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+	psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
 			MPC52xx_PSC_SICR_CLKPOL;
-	if (of_get_property(op->node, "fsl,cellslave", NULL))
-		psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE |
-				 MPC52xx_PSC_SICR_GENCLK;
-	out_be32(&psc_i2s->psc_regs->sicr,
-		 psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
+	out_be32(&psc_dma->psc_regs->sicr,
+		 psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
 
 	/* Check for the codec handle.  If it is not present then we
 	 * are done */
 	if (!of_get_property(op->node, "codec-handle", NULL))
 		return 0;
 
-	/* Set up mode register;
-	 * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
-	 * Second write: register Normal mode for non loopback
-	 */
-	out_8(&psc_i2s->psc_regs->mode, 0);
-	out_8(&psc_i2s->psc_regs->mode, 0);
-
-	/* Set the TX and RX fifo alarm thresholds */
-	out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100);
-	out_8(&psc_i2s->fifo_regs->rfcntl, 0x4);
-	out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100);
-	out_8(&psc_i2s->fifo_regs->tfcntl, 0x7);
-
-	/* Lookup the IRQ numbers */
-	psc_i2s->playback.irq =
-		bcom_get_task_irq(psc_i2s->playback.bcom_task);
-	psc_i2s->capture.irq =
-		bcom_get_task_irq(psc_i2s->capture.bcom_task);
-
-	/* Save what we've done so it can be found again later */
-	dev_set_drvdata(&op->dev, psc_i2s);
-
-	/* Register the SYSFS files */
-	rc = device_create_file(psc_i2s->dev, &dev_attr_status);
-	rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
-	rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
-	if (rc)
-		dev_info(psc_i2s->dev, "error creating sysfs files\n");
-
-	snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
-
-	/* Tell the ASoC OF helpers about it */
-	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
-				     &psc_i2s->dai);
+	/* Due to errata in the dma mode; need to line up enabling
+	 * the transmitter with a transition on the frame sync
+	 * line */
+
+	/* first make sure it is low */
+	while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
+		;
+	/* then wait for the transition to high */
+	while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
+		;
+	/* Finally, enable the PSC.
+	 * Receiver must always be enabled; even when we only want
+	 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
+
+	/* Go */
+	out_8(&psc_dma->psc_regs->command,
+			MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
 
 	return 0;
+
 }
 
 static int __devexit psc_i2s_of_remove(struct of_device *op)
 {
-	struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
-
-	dev_dbg(&op->dev, "psc_i2s_remove()\n");
-
-	snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
-
-	bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
-	bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
-
-	iounmap(psc_i2s->psc_regs);
-	iounmap(psc_i2s->fifo_regs);
-	kfree(psc_i2s);
-	dev_set_drvdata(&op->dev, NULL);
-
-	return 0;
+	return mpc5200_audio_dma_destroy(op);
 }
 
 /* Match table for of_platform binding */
 static struct of_device_id psc_i2s_match[] __devinitdata = {
 	{ .compatible = "fsl,mpc5200-psc-i2s", },
+	{ .compatible = "fsl,mpc5200b-psc-i2s", },
 	{}
 };
 MODULE_DEVICE_TABLE(of, psc_i2s_match);
@@ -892,4 +245,7 @@ static void __exit psc_i2s_exit(void)
 }
 module_exit(psc_i2s_exit);
 
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
+MODULE_LICENSE("GPL");
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h
new file mode 100644
index 00000000000..ce55e070fdf
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_i2s.h
@@ -0,0 +1,12 @@
+/*
+ * Freescale MPC5200 PSC in I2S mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
+
+extern struct snd_soc_dai psc_i2s_dai[];
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
new file mode 100644
index 00000000000..8766f7a3893
--- /dev/null
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Phytec pcm030 driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/wm9712.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link pcm030_fabric_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 Analog",
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+	.name = "AC97",
+	.stream_name = "AC97 IEC958",
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int pcm030_fabric_init(void)
+{
+	struct platform_device *pdev;
+	int rc;
+
+	if (!machine_is_compatible("phytec,pcm030"))
+		return -ENODEV;
+
+	card.platform = &mpc5200_audio_dma_platform;
+	card.name = "pcm030";
+	card.dai_link = pcm030_fabric_dai;
+	card.num_links = ARRAY_SIZE(pcm030_fabric_dai);
+
+	device.card = &card;
+	device.codec_dev = &soc_codec_dev_wm9712;
+
+	pdev = platform_device_alloc("soc-audio", 1);
+	if (!pdev) {
+		pr_err("pcm030_fabric_init: platform_device_alloc() failed\n");
+		return -ENODEV;
+	}
+
+	platform_set_drvdata(pdev, &device);
+	device.dev = &pdev->dev;
+
+	rc = platform_device_add(pdev);
+	if (rc) {
+		pr_err("pcm030_fabric_init: platform_device_add() failed\n");
+		return -ENODEV;
+	}
+	return 0;
+}
+
+module_init(pcm030_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 675732e724d..b771238662b 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -39,6 +39,14 @@ config SND_OMAP_SOC_OMAP2EVM
 	help
 	  Say Y if you want to add support for SoC audio on the omap2evm board.
 
+config SND_OMAP_SOC_OMAP3EVM
+	tristate "SoC Audio support for OMAP3EVM board"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the omap3evm board.
+
 config SND_OMAP_SOC_SDP3430
 	tristate "SoC Audio support for Texas Instruments SDP3430"
 	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 0c9e4ac3766..a37f4986238 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -10,6 +10,7 @@ snd-soc-n810-objs := n810.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
@@ -18,6 +19,7 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 91ef17992de..b60b1dfbc43 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -383,10 +383,9 @@ static int __init n810_soc_init(void)
 	clk_set_parent(sys_clkout2_src, func96m_clk);
 	clk_set_rate(sys_clkout2, 12000000);
 
-	if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
-		BUG();
-	if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
-		BUG();
+	BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+	       (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
+
 	gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
 	gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
 
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 91261428384..a5d46a7b196 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -215,8 +215,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
-	int wlen, channels;
+	int wlen, channels, wpf;
 	unsigned long port;
+	unsigned int format;
 
 	if (cpu_class_is_omap1()) {
 		dma = omap1_dma_reqs[bus_id][substream->stream];
@@ -244,18 +245,24 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		return 0;
 	}
 
-	channels = params_channels(params);
+	format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+	wpf = channels = params_channels(params);
 	switch (channels) {
 	case 2:
-		/* Use dual-phase frames */
-		regs->rcr2	|= RPHASE;
-		regs->xcr2	|= XPHASE;
+		if (format == SND_SOC_DAIFMT_I2S) {
+			/* Use dual-phase frames */
+			regs->rcr2	|= RPHASE;
+			regs->xcr2	|= XPHASE;
+			/* Set 1 word per (McBSP) frame for phase1 and phase2 */
+			wpf--;
+			regs->rcr2	|= RFRLEN2(wpf - 1);
+			regs->xcr2	|= XFRLEN2(wpf - 1);
+		}
 	case 1:
-		/* Set 1 word per (McBSP) frame */
-		regs->rcr2	|= RFRLEN2(1 - 1);
-		regs->rcr1	|= RFRLEN1(1 - 1);
-		regs->xcr2	|= XFRLEN2(1 - 1);
-		regs->xcr1	|= XFRLEN1(1 - 1);
+	case 4:
+		/* Set word per (McBSP) frame for phase1 */
+		regs->rcr1	|= RFRLEN1(wpf - 1);
+		regs->xcr1	|= XFRLEN1(wpf - 1);
 		break;
 	default:
 		/* Unsupported number of channels */
@@ -277,11 +284,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* Set FS period and length in terms of bit clock periods */
-	switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	switch (format) {
 	case SND_SOC_DAIFMT_I2S:
-		regs->srgr2	|= FPER(wlen * 2 - 1);
+		regs->srgr2	|= FPER(wlen * channels - 1);
 		regs->srgr1	|= FWID(wlen - 1);
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
 	case SND_SOC_DAIFMT_DSP_B:
 		regs->srgr2	|= FPER(wlen * channels - 1);
 		regs->srgr1	|= FWID(0);
@@ -326,6 +334,13 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		regs->rcr2	|= RDATDLY(1);
 		regs->xcr2	|= XDATDLY(1);
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		/* 1-bit data delay */
+		regs->rcr2      |= RDATDLY(1);
+		regs->xcr2      |= XDATDLY(1);
+		/* Invert FS polarity configuration */
+		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		/* 0-bit data delay */
 		regs->rcr2      |= RDATDLY(0);
@@ -492,13 +507,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
 	.id = (link_id),					\
 	.playback = {						\
 		.channels_min = 1,				\
-		.channels_max = 2,				\
+		.channels_max = 4,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
 	.capture = {						\
 		.channels_min = 1,				\
-		.channels_max = 2,				\
+		.channels_max = 4,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 07cf7f46b58..6454e15f7d2 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -87,8 +87,10 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
 	int err = 0;
 
+	/* return if this is a bufferless transfer e.g.
+	 * codec <--> BT codec or GSM modem -- lg FIXME */
 	if (!dma_data)
-		return -ENODEV;
+		return 0;
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = params_buffer_bytes(params);
@@ -134,6 +136,11 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	struct omap_pcm_dma_data *dma_data = prtd->dma_data;
 	struct omap_dma_channel_params dma_params;
 
+	/* return if this is a bufferless transfer e.g.
+	 * codec <--> BT codec or GSM modem -- lg FIXME */
+	if (!prtd->dma_data)
+		return 0;
+
 	memset(&dma_params, 0, sizeof(dma_params));
 	/*
 	 * Note: Regardless of interface data formats supported by OMAP McBSP
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index 0c2322dcf02..027e1a40f8a 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = {
 	.name = "TWL4030",
 	.stream_name = "TWL4030",
 	.cpu_dai = &omap_mcbsp_dai[0],
-	.codec_dai = &twl4030_dai,
+	.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
 	.ops = &omap2evm_ops,
 };
 
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index fd24a4acd2f..b0cff9f33b7 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -41,23 +41,33 @@ static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int fmt;
 	int ret;
 
+	switch (params_channels(params)) {
+	case 2: /* Stereo I2S mode */
+		fmt =	SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	case 4: /* Four channel TDM mode */
+		fmt =	SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_IB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	/* Set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai,
-				  SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
 	if (ret < 0) {
 		printk(KERN_ERR "can't set codec DAI configuration\n");
 		return ret;
 	}
 
 	/* Set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
 	if (ret < 0) {
 		printk(KERN_ERR "can't set cpu DAI configuration\n");
 		return ret;
@@ -83,7 +93,7 @@ static struct snd_soc_dai_link omap3beagle_dai = {
 	.name = "TWL4030",
 	.stream_name = "TWL4030",
 	.cpu_dai = &omap_mcbsp_dai[0],
-	.codec_dai = &twl4030_dai,
+	.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
 	.ops = &omap3beagle_ops,
 };
 
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
new file mode 100644
index 00000000000..9114c263077
--- /dev/null
+++ b/sound/soc/omap/omap3evm.c
@@ -0,0 +1,147 @@
+/*
+ * omap3evm.c  -- ALSA SoC support for OMAP3 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2008 Texas Instruments, Incorporated
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3evm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap3evm_ops = {
+	.hw_params = omap3evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3evm_dai = {
+	.name 		= "TWL4030",
+	.stream_name 	= "TWL4030",
+	.cpu_dai 	= &omap_mcbsp_dai[0],
+	.codec_dai 	= &twl4030_dai[TWL4030_DAI_HIFI],
+	.ops 		= &omap3evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3evm = {
+	.name = "omap3evm",
+	.platform = &omap_soc_platform,
+	.dai_link = &omap3evm_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3evm_snd_devdata = {
+	.card = &snd_soc_omap3evm,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3evm_snd_device;
+
+static int __init omap3evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3evm()) {
+		pr_err("Not OMAP3 EVM!\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3 EVM SoC init\n");
+
+	omap3evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap3evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata);
+	omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev;
+	*(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1;
+
+	ret = platform_device_add(omap3evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap3evm_snd_device);
+
+	return ret;
+}
+
+static void __exit omap3evm_soc_exit(void)
+{
+	platform_device_unregister(omap3evm_snd_device);
+}
+
+module_init(omap3evm_soc_init);
+module_exit(omap3evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index fe282d4ef42..ad219aaf7cb 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
 		.name = "PCM1773",
 		.stream_name = "HiFi Out",
 		.cpu_dai = &omap_mcbsp_dai[0],
-		.codec_dai = &twl4030_dai,
+		.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
 		.ops = &omap3pandora_out_ops,
 		.init = omap3pandora_out_init,
 	}, {
 		.name = "TWL4030",
 		.stream_name = "Line/Mic In",
 		.cpu_dai = &omap_mcbsp_dai[1],
-		.codec_dai = &twl4030_dai,
+		.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
 		.ops = &omap3pandora_in_ops,
 		.init = omap3pandora_in_init,
 	}
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index a72dc4e159e..ec4f8fd8b3a 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = {
 	.name = "TWL4030",
 	.stream_name = "TWL4030",
 	.cpu_dai = &omap_mcbsp_dai[0],
-	.codec_dai = &twl4030_dai,
+	.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
 	.ops = &overo_ops,
 };
 
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 10f1c867f11..b719e5db4f5 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -84,6 +84,49 @@ static struct snd_soc_ops sdp3430_ops = {
 	.hw_params = sdp3430_hw_params,
 };
 
+static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBS_CFM);
+	if (ret) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops sdp3430_voice_ops = {
+	.hw_params = sdp3430_hw_voice_params,
+};
+
 /* Headset jack */
 static struct snd_soc_jack hs_jack;
 
@@ -192,28 +235,58 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec)
 	return ret;
 }
 
+static int sdp3430_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+	unsigned short reg;
+
+	/* Enable voice interface */
+	reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+	reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+	codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+	return 0;
+}
+
+
 /* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sdp3430_dai = {
-	.name = "TWL4030",
-	.stream_name = "TWL4030",
-	.cpu_dai = &omap_mcbsp_dai[0],
-	.codec_dai = &twl4030_dai,
-	.init = sdp3430_twl4030_init,
-	.ops = &sdp3430_ops,
+static struct snd_soc_dai_link sdp3430_dai[] = {
+	{
+		.name = "TWL4030 I2S",
+		.stream_name = "TWL4030 Audio",
+		.cpu_dai = &omap_mcbsp_dai[0],
+		.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+		.init = sdp3430_twl4030_init,
+		.ops = &sdp3430_ops,
+	},
+	{
+		.name = "TWL4030 PCM",
+		.stream_name = "TWL4030 Voice",
+		.cpu_dai = &omap_mcbsp_dai[1],
+		.codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+		.init = sdp3430_twl4030_voice_init,
+		.ops = &sdp3430_voice_ops,
+	},
 };
 
 /* Audio machine driver */
 static struct snd_soc_card snd_soc_sdp3430 = {
 	.name = "SDP3430",
 	.platform = &omap_soc_platform,
-	.dai_link = &sdp3430_dai,
-	.num_links = 1,
+	.dai_link = sdp3430_dai,
+	.num_links = ARRAY_SIZE(sdp3430_dai),
+};
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+	.ramp_delay_value = 3,
+	.sysclk = 26000,
 };
 
 /* Audio subsystem */
 static struct snd_soc_device sdp3430_snd_devdata = {
 	.card = &snd_soc_sdp3430,
 	.codec_dev = &soc_codec_dev_twl4030,
+	.codec_data = &twl4030_setup,
 };
 
 static struct platform_device *sdp3430_snd_device;
@@ -236,7 +309,8 @@ static int __init sdp3430_soc_init(void)
 
 	platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
 	sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
-	*(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+	*(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+	*(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
 
 	ret = platform_device_add(sdp3430_snd_device);
 	if (ret)
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ad8a10fe629..dcd163a4ee9 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800
 	  Toshiba e800 PDA
 
 config SND_PXA2XX_SOC_EM_X270
-	tristate "SoC Audio support for CompuLab EM-x270"
+	tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
 	depends on SND_PXA2XX_SOC && MACH_EM_X270
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
 	  Say Y if you want to add support for SoC audio on
-	  CompuLab EM-x270.
+	  CompuLab EM-x270, eXeda and CM-X300 machines.
 
 config SND_PXA2XX_SOC_PALM27X
 	bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
@@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701
         help
           Say Y if you want to add support for SoC audio on the
           MIO A701.
+
+config SND_PXA2XX_SOC_IMOTE2
+       tristate "SoC Audio support for IMote 2"
+       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+       select SND_PXA2XX_SOC_I2S
+       select SND_SOC_WM8940
+       help
+         Say Y if you want to add support for SoC audio on the
+	 IMote 2.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 4b90c3ccae4..6e096b48033 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o
 snd-soc-zylonite-objs := zylonite.o
 snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
+snd-soc-imote2-objs := imote2.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -35,3 +36,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
+obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 949be9c2a01..f4756e4025f 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -1,7 +1,7 @@
 /*
- * em-x270.c  --  SoC audio for EM-X270
+ * SoC audio driver for EM-X270, eXeda and CM-X300
  *
- * Copyright 2007 CompuLab, Ltd.
+ * Copyright 2007, 2009 CompuLab, Ltd.
  *
  * Author: Mike Rapoport <mike@compulab.co.il>
  *
@@ -68,7 +68,8 @@ static int __init em_x270_init(void)
 {
 	int ret;
 
-	if (!machine_is_em_x270())
+	if (!(machine_is_em_x270() || machine_is_exeda()
+	      || machine_is_cm_x300()))
 		return -ENODEV;
 
 	em_x270_snd_device = platform_device_alloc("soc-audio", -1);
@@ -95,5 +96,5 @@ module_exit(em_x270_exit);
 
 /* Module information */
 MODULE_AUTHOR("Mike Rapoport");
-MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
new file mode 100644
index 00000000000..405587a0116
--- /dev/null
+++ b/sound/soc/pxa/imote2.c
@@ -0,0 +1,114 @@
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm8940.h"
+#include "pxa2xx-i2s.h"
+#include "pxa2xx-pcm.h"
+
+static int imote2_asoc_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int clk = 0;
+	int ret;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+				  | SND_SOC_DAIFMT_NB_NF
+				  | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* CPU should be clock master */
+	ret = snd_soc_dai_set_fmt(cpu_dai,  SND_SOC_DAIFMT_I2S
+				  | SND_SOC_DAIFMT_NB_NF
+				  | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk,
+		SND_SOC_CLOCK_OUT);
+
+	return ret;
+}
+
+static struct snd_soc_ops imote2_asoc_ops = {
+	.hw_params = imote2_asoc_hw_params,
+};
+
+static struct snd_soc_dai_link imote2_dai = {
+	.name = "WM8940",
+	.stream_name = "WM8940",
+	.cpu_dai = &pxa_i2s_dai,
+	.codec_dai = &wm8940_dai,
+	.ops = &imote2_asoc_ops,
+};
+
+static struct snd_soc_card snd_soc_imote2 = {
+	.name = "Imote2",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = &imote2_dai,
+	.num_links = 1,
+};
+
+static struct snd_soc_device imote2_snd_devdata = {
+	.card = &snd_soc_imote2,
+	.codec_dev = &soc_codec_dev_wm8940,
+};
+
+static struct platform_device *imote2_snd_device;
+
+static int __init imote2_asoc_init(void)
+{
+	int ret;
+
+	if (!machine_is_intelmote2())
+		return -ENODEV;
+	imote2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!imote2_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata);
+	imote2_snd_devdata.dev = &imote2_snd_device->dev;
+	ret = platform_device_add(imote2_snd_device);
+	if (ret)
+		platform_device_put(imote2_snd_device);
+
+	return ret;
+}
+module_init(imote2_asoc_init);
+
+static void __exit imote2_asoc_exit(void)
+{
+	platform_device_unregister(imote2_snd_device);
+}
+module_exit(imote2_asoc_exit);
+
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_DESCRIPTION("ALSA SoC Imote 2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 0625c342a1c..c89a3cdf31e 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -106,7 +106,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_16;
 			break;
-		case 32:
+		default: /* 32 */
 			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_8;
 		}
@@ -118,7 +118,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_4;
 			break;
-		case 32:
+		default: /* 32 */
 			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_2;
 		}
@@ -130,7 +130,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_2;
 			break;
-		case 32:
+		default: /* 32 */
 			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_1;
 		}
@@ -142,7 +142,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_2;
 			break;
-		case 32:
+		default: /* 32 */
 			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_1;
 		}
@@ -154,19 +154,20 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_2;
 			break;
-		case 32:
+		default: /* 32 */
 			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_1;
 		}
 		break;
 	case 96000:
+	default:
 		acps = 12235000;
 		switch (width) {
 		case 16:
 			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_1;
 			break;
-		case 32:
+		default: /* 32 */
 			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
 			acds = PXA_SSP_CLK_AUDIO_DIV_2;
 			div4 = PXA_SSP_CLK_SCDB_1;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 286be31545d..19c45409d94 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -50,139 +50,6 @@ struct ssp_priv {
 #endif
 };
 
-#define PXA2xx_SSP1_BASE	0x41000000
-#define PXA27x_SSP2_BASE	0x41700000
-#define PXA27x_SSP3_BASE	0x41900000
-#define PXA3xx_SSP4_BASE	0x41a00000
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
-	.name			= "SSP1 PCM Mono out",
-	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
-	.drcmr			= &DRCMR(14),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
-	.name			= "SSP1 PCM Mono in",
-	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
-	.drcmr			= &DRCMR(13),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
-	.name			= "SSP1 PCM Stereo out",
-	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
-	.drcmr			= &DRCMR(14),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
-	.name			= "SSP1 PCM Stereo in",
-	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
-	.drcmr			= &DRCMR(13),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
-	.name			= "SSP2 PCM Mono out",
-	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
-	.drcmr			= &DRCMR(16),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
-	.name			= "SSP2 PCM Mono in",
-	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
-	.drcmr			= &DRCMR(15),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
-	.name			= "SSP2 PCM Stereo out",
-	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
-	.drcmr			= &DRCMR(16),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
-	.name			= "SSP2 PCM Stereo in",
-	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
-	.drcmr			= &DRCMR(15),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
-	.name			= "SSP3 PCM Mono out",
-	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
-	.drcmr			= &DRCMR(67),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
-	.name			= "SSP3 PCM Mono in",
-	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
-	.drcmr			= &DRCMR(66),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
-	.name			= "SSP3 PCM Stereo out",
-	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
-	.drcmr			= &DRCMR(67),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
-	.name			= "SSP3 PCM Stereo in",
-	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
-	.drcmr			= &DRCMR(66),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
-	.name			= "SSP4 PCM Mono out",
-	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
-	.drcmr			= &DRCMR(67),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
-	.name			= "SSP4 PCM Mono in",
-	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
-	.drcmr			= &DRCMR(66),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
-	.name			= "SSP4 PCM Stereo out",
-	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
-	.drcmr			= &DRCMR(67),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
-	.name			= "SSP4 PCM Stereo in",
-	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
-	.drcmr			= &DRCMR(66),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
 static void dump_registers(struct ssp_device *ssp)
 {
 	dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
@@ -194,25 +61,33 @@ static void dump_registers(struct ssp_device *ssp)
 		 ssp_read_reg(ssp, SSACD));
 }
 
-static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
-	{
-		&pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
-		&pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
-	},
-	{
-		&pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
-		&pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
-	},
-	{
-		&pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
-		&pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
-	},
-	{
-		&pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
-		&pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
-	},
+struct pxa2xx_pcm_dma_data {
+	struct pxa2xx_pcm_dma_params params;
+	char name[20];
 };
 
+static struct pxa2xx_pcm_dma_params *
+ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
+{
+	struct pxa2xx_pcm_dma_data *dma;
+
+	dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+	if (dma == NULL)
+		return NULL;
+
+	snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
+			width4 ? "32-bit" : "16-bit", out ? "out" : "in");
+
+	dma->params.name = dma->name;
+	dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx);
+	dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) :
+				  (DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
+			(width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
+	dma->params.dev_addr = ssp->phys_base + SSDR;
+
+	return &dma->params;
+}
+
 static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
@@ -227,6 +102,11 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 		clk_enable(priv->dev.ssp->clk);
 		ssp_disable(&priv->dev);
 	}
+
+	if (cpu_dai->dma_data) {
+		kfree(cpu_dai->dma_data);
+		cpu_dai->dma_data = NULL;
+	}
 	return ret;
 }
 
@@ -241,6 +121,11 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 		ssp_disable(&priv->dev);
 		clk_disable(priv->dev.ssp->clk);
 	}
+
+	if (cpu_dai->dma_data) {
+		kfree(cpu_dai->dma_data);
+		cpu_dai->dma_data = NULL;
+	}
 }
 
 #ifdef CONFIG_PM
@@ -323,7 +208,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 
 	dev_dbg(&ssp->pdev->dev,
-		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
+		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
 		cpu_dai->id, clk_id, freq);
 
 	switch (clk_id) {
@@ -472,7 +357,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
 			ssacd |= (0x6 << 4);
 
 			dev_dbg(&ssp->pdev->dev,
-				"Using SSACDD %x to supply %dHz\n",
+				"Using SSACDD %x to supply %uHz\n",
 				val, freq_out);
 			break;
 		}
@@ -589,7 +474,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		case SND_SOC_DAIFMT_NB_IF:
 			break;
 		case SND_SOC_DAIFMT_IB_IF:
-			sspsp |= SSPSP_SCMODE(3);
+			sspsp |= SSPSP_SCMODE(2);
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
 			break;
 		default:
 			return -EINVAL;
@@ -606,7 +494,13 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		case SND_SOC_DAIFMT_NB_NF:
 			sspsp |= SSPSP_SFRMP;
 			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			break;
 		case SND_SOC_DAIFMT_IB_IF:
+			sspsp |= SSPSP_SCMODE(2);
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
 			break;
 		default:
 			return -EINVAL;
@@ -644,25 +538,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct ssp_priv *priv = cpu_dai->private_data;
 	struct ssp_device *ssp = priv->dev.ssp;
-	int dma = 0, chn = params_channels(params);
+	int chn = params_channels(params);
 	u32 sscr0;
 	u32 sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
 	int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
 
-	/* select correct DMA params */
-	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
-		dma = 1; /* capture DMA offset is 1,3 */
+	/* generate correct DMA params */
+	if (cpu_dai->dma_data)
+		kfree(cpu_dai->dma_data);
+
 	/* Network mode with one active slot (ttsa == 1) can be used
 	 * to force 16-bit frame width on the wire (for S16_LE), even
 	 * with two channels. Use 16-bit DMA transfers for this case.
 	 */
-	if (((chn == 2) && (ttsa != 1)) || (width == 32))
-		dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
-
-	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
-
-	dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+	cpu_dai->dma_data = ssp_get_dma_params(ssp,
+			((chn == 2) && (ttsa != 1)) || (width == 32),
+			substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
 	/* we can only change the settings if the port is not in use */
 	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 2f4b6e489b7..4743e262895 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -106,10 +106,8 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
 	if (IS_ERR(clk_i2s))
 		return PTR_ERR(clk_i2s);
 
-	if (!cpu_dai->active) {
-		SACR0 |= SACR0_RST;
+	if (!cpu_dai->active)
 		SACR0 = 0;
-	}
 
 	return 0;
 }
@@ -178,9 +176,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	/* is port used by another stream */
 	if (!(SACR0 & SACR0_ENB)) {
-
 		SACR0 = 0;
-		SACR1 = 0;
 		if (pxa_i2s.master)
 			SACR0 |= SACR0_BCKD;
 
@@ -226,6 +222,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			SACR1 &= ~SACR1_DRPL;
+		else
+			SACR1 &= ~SACR1_DREC;
 		SACR0 |= SACR0_ENB;
 		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
@@ -252,21 +252,16 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
 		SAIMR &= ~SAIMR_RFS;
 	}
 
-	if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
+	if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
 		SACR0 &= ~SACR0_ENB;
 		pxa_i2s_wait();
 		clk_disable(clk_i2s);
 	}
-
-	clk_put(clk_i2s);
 }
 
 #ifdef CONFIG_PM
 static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
-	if (!dai->active)
-		return 0;
-
 	/* store registers */
 	pxa_i2s.sacr0 = SACR0;
 	pxa_i2s.sacr1 = SACR1;
@@ -281,16 +276,14 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 
 static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
-	if (!dai->active)
-		return 0;
-
 	pxa_i2s_wait();
 
-	SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
+	SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB;
 	SACR1 = pxa_i2s.sacr1;
 	SAIMR = pxa_i2s.saimr;
 	SADIV = pxa_i2s.sadiv;
-	SACR0 |= SACR0_ENB;
+
+	SACR0 = pxa_i2s.sacr0;
 
 	return 0;
 }
@@ -329,6 +322,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.rates = PXA2XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = &pxa_i2s_dai_ops,
+	.symmetric_rates = 1,
 };
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);
@@ -346,6 +340,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev)
 	if (ret != 0)
 		clk_put(clk_i2s);
 
+	/*
+	 * PXA Developer's Manual:
+	 * If SACR0[ENB] is toggled in the middle of a normal operation,
+	 * the SACR0[RST] bit must also be set and cleared to reset all
+	 * I2S controller registers.
+	 */
+	SACR0 = SACR0_RST;
+	SACR0 = 0;
+	/* Make sure RPL and REC are disabled */
+	SACR1 = SACR1_DRPL | SACR1_DREC;
+	/* Along with FIFO servicing */
+	SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
+
 	return ret;
 }
 
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index ab680aac3fc..1a283170ca9 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -37,6 +37,20 @@
 
 #include "s3c-i2s-v2.h"
 
+#undef S3C_IIS_V2_SUPPORTED
+
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifndef S3C_IIS_V2_SUPPORTED
+#error Unsupported CPU model
+#endif
+
 #define S3C2412_I2S_DEBUG_CON 0
 
 static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
@@ -75,7 +89,7 @@ static inline void dbg_showcon(const char *fn, u32 con)
 
 
 /* Turn on or off the transmission path. */
-void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 {
 	void __iomem *regs = i2s->regs;
 	u32 fic, con, mod;
@@ -105,7 +119,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
+			break;
 		}
 
 		writel(con, regs + S3C2412_IISCON);
@@ -132,7 +148,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
+			break;
 		}
 
 		writel(mod, regs + S3C2412_IISMOD);
@@ -143,9 +161,8 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 	dbg_showcon(__func__, con);
 	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
 }
-EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl);
 
-void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 {
 	void __iomem *regs = i2s->regs;
 	u32 fic, con, mod;
@@ -175,7 +192,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
 		}
 
 		writel(mod, regs + S3C2412_IISMOD);
@@ -199,7 +217,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
 		}
 
 		writel(con, regs + S3C2412_IISCON);
@@ -209,7 +228,6 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 	fic = readl(regs + S3C2412_IISFIC);
 	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
 }
-EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl);
 
 /*
  * Wait for the LR signal to allow synchronisation to the L/R clock
@@ -266,7 +284,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
  */
 #define IISMOD_MASTER_MASK (1 << 11)
 #define IISMOD_SLAVE (1 << 11)
-#define IISMOD_MASTER (0x0)
+#define IISMOD_MASTER (0 << 11)
 #endif
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -281,7 +299,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		iismod |= IISMOD_MASTER;
 		break;
 	default:
-		pr_debug("unknwon master/slave format\n");
+		pr_err("unknwon master/slave format\n");
 		return -EINVAL;
 	}
 
@@ -298,7 +316,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		iismod |= S3C2412_IISMOD_SDF_IIS;
 		break;
 	default:
-		pr_debug("Unknown data format\n");
+		pr_err("Unknown data format\n");
 		return -EINVAL;
 	}
 
@@ -327,6 +345,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	iismod = readl(i2s->regs + S3C2412_IISMOD);
 	pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
 
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		iismod |= S3C2412_IISMOD_8BIT;
@@ -335,6 +354,25 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 		iismod &= ~S3C2412_IISMOD_8BIT;
 		break;
 	}
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+	iismod &= ~0x606;
+	/* Sample size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		/* 8 bit sample, 16fs BCLK */
+		iismod |= 0x2004;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		/* 16 bit sample, 32fs BCLK */
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		/* 24 bit sample, 48fs BCLK */
+		iismod |= 0x4002;
+		break;
+	}
+#endif
 
 	writel(iismod, i2s->regs + S3C2412_IISMOD);
 	pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
@@ -489,6 +527,8 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
 	unsigned int best_rate = 0;
 	unsigned int best_deviation = INT_MAX;
 
+	pr_debug("Input clock rate %ldHz\n", clkrate);
+
 	if (fstab == NULL)
 		fstab = iis_fs_tab;
 
@@ -507,7 +547,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
 		actual = clkrate / (fsdiv * div);
 		deviation = actual - rate;
 
-		printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
+		printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n",
 		       fsdiv, div, actual, deviation);
 
 		deviation = abs(deviation);
@@ -523,7 +563,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
 			break;
 	}
 
-	printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
+	printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n",
 	       best_fs, best_div, best_rate);
 
 	info->fs_div = best_fs;
@@ -539,12 +579,31 @@ int s3c_i2sv2_probe(struct platform_device *pdev,
 		    unsigned long base)
 {
 	struct device *dev = &pdev->dev;
+	unsigned int iismod;
 
 	i2s->dev = dev;
 
 	/* record our i2s structure for later use in the callbacks */
 	dai->private_data = i2s;
 
+	if (!base) {
+		struct resource *res = platform_get_resource(pdev,
+							     IORESOURCE_MEM,
+							     0);
+		if (!res) {
+			dev_err(dev, "Unable to get register resource\n");
+			return -ENXIO;
+		}
+
+		if (!request_mem_region(res->start, resource_size(res),
+					"s3c64xx-i2s-v4")) {
+			dev_err(dev, "Unable to request register region\n");
+			return -EBUSY;
+		}
+
+		base = res->start;
+	}
+
 	i2s->regs = ioremap(base, 0x100);
 	if (i2s->regs == NULL) {
 		dev_err(dev, "cannot ioremap registers\n");
@@ -560,12 +619,16 @@ int s3c_i2sv2_probe(struct platform_device *pdev,
 
 	clk_enable(i2s->iis_pclk);
 
+	/* Mark ourselves as in TXRX mode so we can run through our cleanup
+	 * process without warnings. */
+	iismod = readl(i2s->regs + S3C2412_IISMOD);
+	iismod |= S3C2412_IISMOD_MODE_TXRX;
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
 	s3c2412_snd_txctrl(i2s, 0);
 	s3c2412_snd_rxctrl(i2s, 0);
 
 	return 0;
 }
-
 EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
 
 #ifdef CONFIG_PM
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index b7e0b3f0bfc..168a088ba76 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -120,7 +120,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 
 	s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
 	if (s3c2412_i2s.iis_cclk == NULL) {
-		pr_debug("failed to get i2sclk clock\n");
+		pr_err("failed to get i2sclk clock\n");
 		iounmap(s3c2412_i2s.regs);
 		return -ENODEV;
 	}
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 33c5de7e255..3c06c401d0f 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -108,48 +108,19 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
-
-unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai)
+struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
 {
 	struct s3c_i2sv2_info *i2s = to_info(dai);
 
-	return clk_get_rate(i2s->iis_cclk);
+	return i2s->iis_cclk;
 }
-EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
 
 static int s3c64xx_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	struct device *dev = &pdev->dev;
-	struct s3c_i2sv2_info *i2s;
-	int ret;
-
-	dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id);
-
-	if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) {
-		dev_err(dev, "id %d out of range\n", pdev->id);
-		return -EINVAL;
-	}
-
-	i2s = &s3c64xx_i2s[pdev->id];
-
-	ret = s3c_i2sv2_probe(pdev, dai, i2s,
-			      pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
-	if (ret)
-		return ret;
-
-	i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
-	i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
-
-	i2s->iis_cclk = clk_get(dev, "audio-bus");
-	if (IS_ERR(i2s->iis_cclk)) {
-		dev_err(dev, "failed to get audio-bus");
-		iounmap(i2s->regs);
-		return -ENODEV;
-	}
-
 	/* configure GPIO for i2s port */
-	switch (pdev->id) {
+	switch (dai->id) {
 	case 0:
 		s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK);
 		s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK);
@@ -175,41 +146,122 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev,
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define S3C64XX_I2S_FMTS \
-	(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
+	(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
+	 SNDRV_PCM_FMTBIT_S24_LE)
 
 static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
 	.set_sysclk	= s3c64xx_i2s_set_sysclk,	
 };
 
-struct snd_soc_dai s3c64xx_i2s_dai = {
-	.name		= "s3c64xx-i2s",
-	.id		= 0,
-	.probe		= s3c64xx_i2s_probe,
-	.playback = {
-		.channels_min	= 2,
-		.channels_max	= 2,
-		.rates		= S3C64XX_I2S_RATES,
-		.formats	= S3C64XX_I2S_FMTS,
+struct snd_soc_dai s3c64xx_i2s_dai[] = {
+	{
+		.name		= "s3c64xx-i2s",
+		.id		= 0,
+		.probe		= s3c64xx_i2s_probe,
+		.playback = {
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= S3C64XX_I2S_RATES,
+			.formats	= S3C64XX_I2S_FMTS,
+		},
+		.capture = {
+			 .channels_min	= 2,
+			 .channels_max	= 2,
+			 .rates		= S3C64XX_I2S_RATES,
+			 .formats	= S3C64XX_I2S_FMTS,
+		 },
+		.ops = &s3c64xx_i2s_dai_ops,
+		.symmetric_rates = 1,
 	},
-	.capture = {
-		.channels_min	= 2,
-		.channels_max	= 2,
-		.rates		= S3C64XX_I2S_RATES,
-		.formats	= S3C64XX_I2S_FMTS,
+	{
+		.name		= "s3c64xx-i2s",
+		.id		= 1,
+		.probe		= s3c64xx_i2s_probe,
+		.playback = {
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= S3C64XX_I2S_RATES,
+			.formats	= S3C64XX_I2S_FMTS,
+		},
+		.capture = {
+			 .channels_min	= 2,
+			 .channels_max	= 2,
+			 .rates		= S3C64XX_I2S_RATES,
+			 .formats	= S3C64XX_I2S_FMTS,
+		 },
+		.ops = &s3c64xx_i2s_dai_ops,
+		.symmetric_rates = 1,
 	},
-	.ops = &s3c64xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
 
+static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
+{
+	struct s3c_i2sv2_info *i2s;
+	struct snd_soc_dai *dai;
+	int ret;
+
+	if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) {
+		dev_err(&pdev->dev, "id %d out of range\n", pdev->id);
+		return -EINVAL;
+	}
+
+	i2s = &s3c64xx_i2s[pdev->id];
+	dai = &s3c64xx_i2s_dai[pdev->id];
+	dai->dev = &pdev->dev;
+
+	i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
+	i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
+
+	i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus");
+	if (IS_ERR(i2s->iis_cclk)) {
+		dev_err(&pdev->dev, "failed to get audio-bus\n");
+		ret = PTR_ERR(i2s->iis_cclk);
+		goto err;
+	}
+
+	ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
+	if (ret)
+		goto err_clk;
+
+	ret = s3c_i2sv2_register_dai(dai);
+	if (ret != 0)
+		goto err_i2sv2;
+
+	return 0;
+
+err_i2sv2:
+	/* Not implemented for I2Sv2 core yet */
+err_clk:
+	clk_put(i2s->iis_cclk);
+err:
+	return ret;
+}
+
+static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev)
+{
+	dev_err(&pdev->dev, "Device removal not yet supported\n");
+	return 0;
+}
+
+static struct platform_driver s3c64xx_iis_driver = {
+	.probe  = s3c64xx_iis_dev_probe,
+	.remove = s3c64xx_iis_dev_remove,
+	.driver = {
+		.name = "s3c64xx-iis",
+		.owner = THIS_MODULE,
+	},
+};
+
 static int __init s3c64xx_i2s_init(void)
 {
-	return  s3c_i2sv2_register_dai(&s3c64xx_i2s_dai);
+	return platform_driver_register(&s3c64xx_iis_driver);
 }
 module_init(s3c64xx_i2s_init);
 
 static void __exit s3c64xx_i2s_exit(void)
 {
-	snd_soc_unregister_dai(&s3c64xx_i2s_dai);
+	platform_driver_unregister(&s3c64xx_iis_driver);
 }
 module_exit(s3c64xx_i2s_exit);
 
@@ -217,6 +269,3 @@ module_exit(s3c64xx_i2s_exit);
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C64XX I2S SoC Interface");
 MODULE_LICENSE("GPL");
-
-
-
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index b7ffe3c38b6..02148cee261 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -15,6 +15,8 @@
 #ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H
 #define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__
 
+struct clk;
+
 #include "s3c-i2s-v2.h"
 
 #define S3C64XX_DIV_BCLK	S3C_I2SV2_DIV_BCLK
@@ -24,8 +26,8 @@
 #define S3C64XX_CLKSRC_PCLK	(0)
 #define S3C64XX_CLKSRC_MUX	(1)
 
-extern struct snd_soc_dai s3c64xx_i2s_dai;
+extern struct snd_soc_dai s3c64xx_i2s_dai[];
 
-extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
+extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai);
 
 #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig
new file mode 100644
index 00000000000..c74eb3d4a47
--- /dev/null
+++ b/sound/soc/s6000/Kconfig
@@ -0,0 +1,19 @@
+config SND_S6000_SOC
+	tristate "SoC Audio for the Stretch s6000 family"
+	depends on XTENSA_VARIANT_S6000
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  s6000 family chips. You will also need to select the platform
+	  to support below.
+
+config SND_S6000_SOC_I2S
+	tristate
+
+config SND_S6000_SOC_S6IPCAM
+	tristate "SoC Audio support for Stretch 6105 IP Camera"
+	depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105
+	select SND_S6000_SOC_I2S
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Stretch s6105 IP Camera Reference Design.
diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile
new file mode 100644
index 00000000000..7a613612e01
--- /dev/null
+++ b/sound/soc/s6000/Makefile
@@ -0,0 +1,11 @@
+# s6000 Platform Support
+snd-soc-s6000-objs := s6000-pcm.o
+snd-soc-s6000-i2s-objs := s6000-i2s.o
+
+obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o
+obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o
+
+# s6105 Machine Support
+snd-soc-s6ipcam-objs := s6105-ipcam.o
+
+obj-$(CONFIG_SND_S6000_SOC_S6IPCAM) += snd-soc-s6ipcam.o
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
new file mode 100644
index 00000000000..c5cda187eca
--- /dev/null
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -0,0 +1,629 @@
+/*
+ * ALSA SoC I2S Audio Layer for the Stretch S6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "s6000-i2s.h"
+#include "s6000-pcm.h"
+
+struct s6000_i2s_dev {
+	dma_addr_t sifbase;
+	u8 __iomem *scbbase;
+	unsigned int wide;
+	unsigned int channel_in;
+	unsigned int channel_out;
+	unsigned int lines_in;
+	unsigned int lines_out;
+	struct s6000_pcm_dma_params dma_params;
+};
+
+#define S6_I2S_INTERRUPT_STATUS	0x00
+#define   S6_I2S_INT_OVERRUN	1
+#define   S6_I2S_INT_UNDERRUN	2
+#define   S6_I2S_INT_ALIGNMENT	4
+#define S6_I2S_INTERRUPT_ENABLE	0x04
+#define S6_I2S_INTERRUPT_RAW	0x08
+#define S6_I2S_INTERRUPT_CLEAR	0x0C
+#define S6_I2S_INTERRUPT_SET	0x10
+#define S6_I2S_MODE		0x20
+#define   S6_I2S_DUAL		0
+#define   S6_I2S_WIDE		1
+#define S6_I2S_TX_DEFAULT	0x24
+#define S6_I2S_DATA_CFG(c)	(0x40 + 0x10 * (c))
+#define   S6_I2S_IN		0
+#define   S6_I2S_OUT		1
+#define   S6_I2S_UNUSED		2
+#define S6_I2S_INTERFACE_CFG(c)	(0x44 + 0x10 * (c))
+#define   S6_I2S_DIV_MASK	0x001fff
+#define   S6_I2S_16BIT		0x000000
+#define   S6_I2S_20BIT		0x002000
+#define   S6_I2S_24BIT		0x004000
+#define   S6_I2S_32BIT		0x006000
+#define   S6_I2S_BITS_MASK	0x006000
+#define   S6_I2S_MEM_16BIT	0x000000
+#define   S6_I2S_MEM_32BIT	0x008000
+#define   S6_I2S_MEM_MASK	0x008000
+#define   S6_I2S_CHANNELS_SHIFT	16
+#define   S6_I2S_CHANNELS_MASK	0x030000
+#define   S6_I2S_SCK_IN		0x000000
+#define   S6_I2S_SCK_OUT	0x040000
+#define   S6_I2S_SCK_DIR	0x040000
+#define   S6_I2S_WS_IN		0x000000
+#define   S6_I2S_WS_OUT		0x080000
+#define   S6_I2S_WS_DIR		0x080000
+#define   S6_I2S_LEFT_FIRST	0x000000
+#define   S6_I2S_RIGHT_FIRST	0x100000
+#define   S6_I2S_FIRST		0x100000
+#define   S6_I2S_CUR_SCK	0x200000
+#define   S6_I2S_CUR_WS		0x400000
+#define S6_I2S_ENABLE(c)	(0x48 + 0x10 * (c))
+#define   S6_I2S_DISABLE_IF	0x02
+#define   S6_I2S_ENABLE_IF	0x03
+#define   S6_I2S_IS_BUSY	0x04
+#define   S6_I2S_DMA_ACTIVE	0x08
+#define   S6_I2S_IS_ENABLED	0x10
+
+#define S6_I2S_NUM_LINES	4
+
+#define S6_I2S_SIF_PORT0	0x0000000
+#define S6_I2S_SIF_PORT1	0x0000080 /* docs say 0x0000010 */
+
+static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val)
+{
+	writel(val, dev->scbbase + reg);
+}
+
+static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg)
+{
+	return readl(dev->scbbase + reg);
+}
+
+static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg,
+				  u32 mask, u32 val)
+{
+	val ^= s6_i2s_read_reg(dev, reg) & ~mask;
+	s6_i2s_write_reg(dev, reg, val);
+}
+
+static void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, int channel)
+{
+	int i, j, cur, prev;
+
+	/*
+	 * Wait for WCLK to toggle 5 times before enabling the channel
+	 * s6000 Family Datasheet 3.6.4:
+	 *   "At least two cycles of WS must occur between commands
+	 *    to disable or enable the interface"
+	 */
+	j = 0;
+	prev = ~S6_I2S_CUR_WS;
+	for (i = 1000000; --i && j < 6; ) {
+		cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel))
+		       & S6_I2S_CUR_WS;
+		if (prev != cur) {
+			prev = cur;
+			j++;
+		}
+	}
+	if (j < 6)
+		printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n");
+
+	s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF);
+}
+
+static void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, int channel)
+{
+	s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF);
+}
+
+static void s6000_i2s_start(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data;
+	int channel;
+
+	channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+			dev->channel_out : dev->channel_in;
+
+	s6000_i2s_start_channel(dev, channel);
+}
+
+static void s6000_i2s_stop(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data;
+	int channel;
+
+	channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+			dev->channel_out : dev->channel_in;
+
+	s6000_i2s_stop_channel(dev, channel);
+}
+
+static int s6000_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			     int after)
+{
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ^ !after)
+			s6000_i2s_start(substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!after)
+			s6000_i2s_stop(substream);
+	}
+	return 0;
+}
+
+static unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev)
+{
+	unsigned int pending;
+	pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW);
+	pending &= S6_I2S_INT_ALIGNMENT |
+		   S6_I2S_INT_UNDERRUN |
+		   S6_I2S_INT_OVERRUN;
+	s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending);
+
+	return pending;
+}
+
+static unsigned int s6000_i2s_check_xrun(struct snd_soc_dai *cpu_dai)
+{
+	struct s6000_i2s_dev *dev = cpu_dai->private_data;
+	unsigned int errors;
+	unsigned int ret;
+
+	errors = s6000_i2s_int_sources(dev);
+	if (likely(!errors))
+		return 0;
+
+	ret = 0;
+	if (errors & S6_I2S_INT_ALIGNMENT)
+		printk(KERN_ERR "s6000-i2s: WCLK misaligned\n");
+	if (errors & S6_I2S_INT_UNDERRUN)
+		ret |= 1 << SNDRV_PCM_STREAM_PLAYBACK;
+	if (errors & S6_I2S_INT_OVERRUN)
+		ret |= 1 << SNDRV_PCM_STREAM_CAPTURE;
+	return ret;
+}
+
+static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev)
+{
+	int channel;
+	int n = 50;
+	for (channel = 0; channel < 2; channel++) {
+		while (--n >= 0) {
+			int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel));
+			if ((v & S6_I2S_IS_ENABLED)
+			    || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY)))
+				break;
+			udelay(20);
+		}
+	}
+	if (n < 0)
+		printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces");
+}
+
+static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+				   unsigned int fmt)
+{
+	struct s6000_i2s_dev *dev = cpu_dai->private_data;
+	u32 w;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		w = S6_I2S_SCK_IN | S6_I2S_WS_IN;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		w = S6_I2S_SCK_OUT | S6_I2S_WS_IN;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		w = S6_I2S_SCK_IN | S6_I2S_WS_OUT;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		w |= S6_I2S_LEFT_FIRST;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		w |= S6_I2S_RIGHT_FIRST;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0),
+		       S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w);
+	s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1),
+		       S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w);
+
+	return 0;
+}
+
+static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+{
+	struct s6000_i2s_dev *dev = dai->private_data;
+
+	if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2)
+		return -EINVAL;
+
+	s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id),
+		       S6_I2S_DIV_MASK, div / 2 - 1);
+	return 0;
+}
+
+static int s6000_i2s_hw_params(struct snd_pcm_substream *substream,
+			       struct snd_pcm_hw_params *params,
+			       struct snd_soc_dai *dai)
+{
+	struct s6000_i2s_dev *dev = dai->private_data;
+	int interf;
+	u32 w = 0;
+
+	if (dev->wide)
+		interf = 0;
+	else {
+		w |= (((params_channels(params) - 2) / 2)
+		      << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK;
+		interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+				? dev->channel_out : dev->channel_in;
+	}
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT;
+		break;
+	default:
+		printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n",
+		       params_format(params));
+		return -EINVAL;
+	}
+
+	if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf))
+	     & S6_I2S_IS_ENABLED) {
+		printk(KERN_ERR "s6000-i2s: interface already enabled\n");
+		return -EBUSY;
+	}
+
+	s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf),
+		       S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK,
+		       w);
+
+	return 0;
+}
+
+static int s6000_i2s_dai_probe(struct platform_device *pdev,
+			       struct snd_soc_dai *dai)
+{
+	struct s6000_i2s_dev *dev = dai->private_data;
+	struct s6000_snd_platform_data *pdata = pdev->dev.platform_data;
+
+	if (!pdata)
+		return -EINVAL;
+
+	dev->wide = pdata->wide;
+	dev->channel_in = pdata->channel_in;
+	dev->channel_out = pdata->channel_out;
+	dev->lines_in = pdata->lines_in;
+	dev->lines_out = pdata->lines_out;
+
+	s6_i2s_write_reg(dev, S6_I2S_MODE,
+			 dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL);
+
+	if (dev->wide) {
+		int i;
+
+		if (dev->lines_in + dev->lines_out > S6_I2S_NUM_LINES)
+			return -EINVAL;
+
+		dev->channel_in = 0;
+		dev->channel_out = 1;
+		dai->capture.channels_min = 2 * dev->lines_in;
+		dai->capture.channels_max = dai->capture.channels_min;
+		dai->playback.channels_min = 2 * dev->lines_out;
+		dai->playback.channels_max = dai->playback.channels_min;
+
+		for (i = 0; i < dev->lines_out; i++)
+			s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT);
+
+		for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++)
+			s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i),
+					 S6_I2S_UNUSED);
+
+		for (; i < S6_I2S_NUM_LINES; i++)
+			s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN);
+	} else {
+		unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED};
+
+		if (dev->lines_in > 1 || dev->lines_out > 1)
+			return -EINVAL;
+
+		dai->capture.channels_min = 2 * dev->lines_in;
+		dai->capture.channels_max = 8 * dev->lines_in;
+		dai->playback.channels_min = 2 * dev->lines_out;
+		dai->playback.channels_max = 8 * dev->lines_out;
+
+		if (dev->lines_in)
+			cfg[dev->channel_in] = S6_I2S_IN;
+		if (dev->lines_out)
+			cfg[dev->channel_out] = S6_I2S_OUT;
+
+		s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]);
+		s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]);
+	}
+
+	if (dev->lines_out) {
+		if (dev->lines_in) {
+			if (!dev->dma_params.dma_out)
+				return -ENODEV;
+		} else {
+			dev->dma_params.dma_out = dev->dma_params.dma_in;
+			dev->dma_params.dma_in = 0;
+		}
+	}
+	dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ?
+					S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0);
+	dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ?
+					S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0);
+	dev->dma_params.same_rate = pdata->same_rate | pdata->wide;
+	return 0;
+}
+
+#define S6000_I2S_RATES	(SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
+			 SNDRV_PCM_RATE_8000_192000)
+#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops s6000_i2s_dai_ops = {
+	.set_fmt = s6000_i2s_set_dai_fmt,
+	.set_clkdiv = s6000_i2s_set_clkdiv,
+	.hw_params = s6000_i2s_hw_params,
+};
+
+struct snd_soc_dai s6000_i2s_dai = {
+	.name = "s6000-i2s",
+	.id = 0,
+	.probe = s6000_i2s_dai_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 8,
+		.formats = S6000_I2S_FORMATS,
+		.rates = S6000_I2S_RATES,
+		.rate_min = 0,
+		.rate_max = 1562500,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 8,
+		.formats = S6000_I2S_FORMATS,
+		.rates = S6000_I2S_RATES,
+		.rate_min = 0,
+		.rate_max = 1562500,
+	},
+	.ops = &s6000_i2s_dai_ops,
+}
+EXPORT_SYMBOL_GPL(s6000_i2s_dai);
+
+static int __devinit s6000_i2s_probe(struct platform_device *pdev)
+{
+	struct s6000_i2s_dev *dev;
+	struct resource *scbmem, *sifmem, *region, *dma1, *dma2;
+	u8 __iomem *mmio;
+	int ret;
+
+	scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!scbmem) {
+		dev_err(&pdev->dev, "no mem resource?\n");
+		ret = -ENODEV;
+		goto err_release_none;
+	}
+
+	region = request_mem_region(scbmem->start,
+				    scbmem->end - scbmem->start + 1,
+				    pdev->name);
+	if (!region) {
+		dev_err(&pdev->dev, "I2S SCB region already claimed\n");
+		ret = -EBUSY;
+		goto err_release_none;
+	}
+
+	mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1);
+	if (!mmio) {
+		dev_err(&pdev->dev, "can't ioremap SCB region\n");
+		ret = -ENOMEM;
+		goto err_release_scb;
+	}
+
+	sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1);
+	if (!sifmem) {
+		dev_err(&pdev->dev, "no second mem resource?\n");
+		ret = -ENODEV;
+		goto err_release_map;
+	}
+
+	region = request_mem_region(sifmem->start,
+				    sifmem->end - sifmem->start + 1,
+				    pdev->name);
+	if (!region) {
+		dev_err(&pdev->dev, "I2S SIF region already claimed\n");
+		ret = -EBUSY;
+		goto err_release_map;
+	}
+
+	dma1 = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dma1) {
+		dev_err(&pdev->dev, "no dma resource?\n");
+		ret = -ENODEV;
+		goto err_release_sif;
+	}
+
+	region = request_mem_region(dma1->start, dma1->end - dma1->start + 1,
+				    pdev->name);
+	if (!region) {
+		dev_err(&pdev->dev, "I2S DMA region already claimed\n");
+		ret = -EBUSY;
+		goto err_release_sif;
+	}
+
+	dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (dma2) {
+		region = request_mem_region(dma2->start,
+					    dma2->end - dma2->start + 1,
+					    pdev->name);
+		if (!region) {
+			dev_err(&pdev->dev,
+				"I2S DMA region already claimed\n");
+			ret = -EBUSY;
+			goto err_release_dma1;
+		}
+	}
+
+	dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL);
+	if (!dev) {
+		ret = -ENOMEM;
+		goto err_release_dma2;
+	}
+
+	s6000_i2s_dai.dev = &pdev->dev;
+	s6000_i2s_dai.private_data = dev;
+	s6000_i2s_dai.dma_data = &dev->dma_params;
+
+	dev->sifbase = sifmem->start;
+	dev->scbbase = mmio;
+
+	s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0);
+	s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR,
+			 S6_I2S_INT_ALIGNMENT |
+			 S6_I2S_INT_UNDERRUN |
+			 S6_I2S_INT_OVERRUN);
+
+	s6000_i2s_stop_channel(dev, 0);
+	s6000_i2s_stop_channel(dev, 1);
+	s6000_i2s_wait_disabled(dev);
+
+	dev->dma_params.check_xrun = s6000_i2s_check_xrun;
+	dev->dma_params.trigger = s6000_i2s_trigger;
+	dev->dma_params.dma_in = dma1->start;
+	dev->dma_params.dma_out = dma2 ? dma2->start : 0;
+	dev->dma_params.irq = platform_get_irq(pdev, 0);
+	if (dev->dma_params.irq < 0) {
+		dev_err(&pdev->dev, "no irq resource?\n");
+		ret = -ENODEV;
+		goto err_release_dev;
+	}
+
+	s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE,
+			 S6_I2S_INT_ALIGNMENT |
+			 S6_I2S_INT_UNDERRUN |
+			 S6_I2S_INT_OVERRUN);
+
+	ret = snd_soc_register_dai(&s6000_i2s_dai);
+	if (ret)
+		goto err_release_dev;
+
+	return 0;
+
+err_release_dev:
+	kfree(dev);
+err_release_dma2:
+	if (dma2)
+		release_mem_region(dma2->start, dma2->end - dma2->start + 1);
+err_release_dma1:
+	release_mem_region(dma1->start, dma1->end - dma1->start + 1);
+err_release_sif:
+	release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1);
+err_release_map:
+	iounmap(mmio);
+err_release_scb:
+	release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1);
+err_release_none:
+	return ret;
+}
+
+static void __devexit s6000_i2s_remove(struct platform_device *pdev)
+{
+	struct s6000_i2s_dev *dev = s6000_i2s_dai.private_data;
+	struct resource *region;
+	void __iomem *mmio = dev->scbbase;
+
+	snd_soc_unregister_dai(&s6000_i2s_dai);
+
+	s6000_i2s_stop_channel(dev, 0);
+	s6000_i2s_stop_channel(dev, 1);
+
+	s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0);
+	s6000_i2s_dai.private_data = 0;
+	kfree(dev);
+
+	region = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	release_mem_region(region->start, region->end - region->start + 1);
+
+	region = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (region)
+		release_mem_region(region->start,
+				   region->end - region->start + 1);
+
+	region = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	release_mem_region(region->start, (region->end - region->start) + 1);
+
+	iounmap(mmio);
+	region = platform_get_resource(pdev, IORESOURCE_IO, 0);
+	release_mem_region(region->start, (region->end - region->start) + 1);
+}
+
+static struct platform_driver s6000_i2s_driver = {
+	.probe  = s6000_i2s_probe,
+	.remove = __devexit_p(s6000_i2s_remove),
+	.driver = {
+		.name   = "s6000-i2s",
+		.owner  = THIS_MODULE,
+	},
+};
+
+static int __init s6000_i2s_init(void)
+{
+	return platform_driver_register(&s6000_i2s_driver);
+}
+module_init(s6000_i2s_init);
+
+static void __exit s6000_i2s_exit(void)
+{
+	platform_driver_unregister(&s6000_i2s_driver);
+}
+module_exit(s6000_i2s_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h
new file mode 100644
index 00000000000..2375fdfe6db
--- /dev/null
+++ b/sound/soc/s6000/s6000-i2s.h
@@ -0,0 +1,25 @@
+/*
+ * ALSA SoC I2S Audio Layer for the Stretch s6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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 _S6000_I2S_H
+#define _S6000_I2S_H
+
+extern struct snd_soc_dai s6000_i2s_dai;
+
+struct s6000_snd_platform_data {
+	int lines_in;
+	int lines_out;
+	int channel_in;
+	int channel_out;
+	int wide;
+	int same_rate;
+};
+#endif
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
new file mode 100644
index 00000000000..83b8028e209
--- /dev/null
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -0,0 +1,497 @@
+/*
+ * ALSA PCM interface for the Stetch s6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+#include <variant/dmac.h>
+
+#include "s6000-pcm.h"
+
+#define S6_PCM_PREALLOCATE_SIZE (96 * 1024)
+#define S6_PCM_PREALLOCATE_MAX  (2048 * 1024)
+
+static struct snd_pcm_hardware s6000_pcm_hardware = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX),
+	.formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE),
+	.rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
+		  SNDRV_PCM_RATE_8000_192000),
+	.rate_min = 0,
+	.rate_max = 1562500,
+	.channels_min = 2,
+	.channels_max = 8,
+	.buffer_bytes_max = 0x7ffffff0,
+	.period_bytes_min = 16,
+	.period_bytes_max = 0xfffff0,
+	.periods_min = 2,
+	.periods_max = 1024, /* no limit */
+	.fifo_size = 0,
+};
+
+struct s6000_runtime_data {
+	spinlock_t lock;
+	int period;		/* current DMA period */
+};
+
+static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct s6000_runtime_data *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	int channel;
+	unsigned int period_size;
+	unsigned int dma_offset;
+	dma_addr_t dma_pos;
+	dma_addr_t src, dst;
+
+	period_size = snd_pcm_lib_period_bytes(substream);
+	dma_offset = prtd->period * period_size;
+	dma_pos = runtime->dma_addr + dma_offset;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		src = dma_pos;
+		dst = par->sif_out;
+		channel = par->dma_out;
+	} else {
+		src = par->sif_in;
+		dst = dma_pos;
+		channel = par->dma_in;
+	}
+
+	if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel),
+				    DMA_INDEX_CHNL(channel)))
+		return;
+
+	if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) {
+		printk(KERN_ERR "s6000-pcm: fifo full\n");
+		return;
+	}
+
+	BUG_ON(period_size & 15);
+	s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel),
+			src, dst, period_size);
+
+	prtd->period++;
+	if (unlikely(prtd->period >= runtime->periods))
+		prtd->period = 0;
+}
+
+static irqreturn_t s6000_pcm_irq(int irq, void *data)
+{
+	struct snd_pcm *pcm = data;
+	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	struct s6000_runtime_data *prtd;
+	unsigned int has_xrun;
+	int i, ret = IRQ_NONE;
+	u32 channel[2] = {
+		[SNDRV_PCM_STREAM_PLAYBACK] = params->dma_out,
+		[SNDRV_PCM_STREAM_CAPTURE] = params->dma_in
+	};
+
+	has_xrun = params->check_xrun(runtime->dai->cpu_dai);
+
+	for (i = 0; i < ARRAY_SIZE(channel); ++i) {
+		struct snd_pcm_substream *substream = pcm->streams[i].substream;
+		unsigned int pending;
+
+		if (!channel[i])
+			continue;
+
+		if (unlikely(has_xrun & (1 << i)) &&
+		    substream->runtime &&
+		    snd_pcm_running(substream)) {
+			dev_dbg(pcm->dev, "xrun\n");
+			snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+			ret = IRQ_HANDLED;
+		}
+
+		pending = s6dmac_int_sources(DMA_MASK_DMAC(channel[i]),
+					     DMA_INDEX_CHNL(channel[i]));
+
+		if (pending & 1) {
+			ret = IRQ_HANDLED;
+			if (likely(substream->runtime &&
+				   snd_pcm_running(substream))) {
+				snd_pcm_period_elapsed(substream);
+				dev_dbg(pcm->dev, "period elapsed %x %x\n",
+				       s6dmac_cur_src(DMA_MASK_DMAC(channel[i]),
+						   DMA_INDEX_CHNL(channel[i])),
+				       s6dmac_cur_dst(DMA_MASK_DMAC(channel[i]),
+						   DMA_INDEX_CHNL(channel[i])));
+				prtd = substream->runtime->private_data;
+				spin_lock(&prtd->lock);
+				s6000_pcm_enqueue_dma(substream);
+				spin_unlock(&prtd->lock);
+			}
+		}
+
+		if (unlikely(pending & ~7)) {
+			if (pending & (1 << 3))
+				printk(KERN_WARNING
+				       "s6000-pcm: DMA %x Underflow\n",
+				       channel[i]);
+			if (pending & (1 << 4))
+				printk(KERN_WARNING
+				       "s6000-pcm: DMA %x Overflow\n",
+				       channel[i]);
+			if (pending & 0x1e0)
+				printk(KERN_WARNING
+				       "s6000-pcm: DMA %x Master Error "
+				       "(mask %x)\n",
+				       channel[i], pending >> 5);
+
+		}
+	}
+
+	return ret;
+}
+
+static int s6000_pcm_start(struct snd_pcm_substream *substream)
+{
+	struct s6000_runtime_data *prtd = substream->runtime->private_data;
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	unsigned long flags;
+	int srcinc;
+	u32 dma;
+
+	spin_lock_irqsave(&prtd->lock, flags);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		srcinc = 1;
+		dma = par->dma_out;
+	} else {
+		srcinc = 0;
+		dma = par->dma_in;
+	}
+	s6dmac_enable_chan(DMA_MASK_DMAC(dma), DMA_INDEX_CHNL(dma),
+			   1 /* priority 1 (0 is max) */,
+			   0 /* peripheral requests w/o xfer length mode */,
+			   srcinc /* source address increment */,
+			   srcinc^1 /* destination address increment */,
+			   0 /* chunksize 0 (skip impossible on this dma) */,
+			   0 /* source skip after chunk (impossible) */,
+			   0 /* destination skip after chunk (impossible) */,
+			   4 /* 16 byte burst size */,
+			   -1 /* don't conserve bandwidth */,
+			   0 /* low watermark irq descriptor theshold */,
+			   0 /* disable hardware timestamps */,
+			   1 /* enable channel */);
+
+	s6000_pcm_enqueue_dma(substream);
+	s6000_pcm_enqueue_dma(substream);
+
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	return 0;
+}
+
+static int s6000_pcm_stop(struct snd_pcm_substream *substream)
+{
+	struct s6000_runtime_data *prtd = substream->runtime->private_data;
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	unsigned long flags;
+	u32 channel;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		channel = par->dma_out;
+	else
+		channel = par->dma_in;
+
+	s6dmac_set_terminal_count(DMA_MASK_DMAC(channel),
+				  DMA_INDEX_CHNL(channel), 0);
+
+	spin_lock_irqsave(&prtd->lock, flags);
+
+	s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel));
+
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	return 0;
+}
+
+static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	int ret;
+
+	ret = par->trigger(substream, cmd, 0);
+	if (ret < 0)
+		return ret;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ret = s6000_pcm_start(substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ret = s6000_pcm_stop(substream);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	if (ret < 0)
+		return ret;
+
+	return par->trigger(substream, cmd, 1);
+}
+
+static int s6000_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct s6000_runtime_data *prtd = substream->runtime->private_data;
+
+	prtd->period = 0;
+
+	return 0;
+}
+
+static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct s6000_runtime_data *prtd = runtime->private_data;
+	unsigned long flags;
+	unsigned int offset;
+	dma_addr_t count;
+
+	spin_lock_irqsave(&prtd->lock, flags);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out),
+				       DMA_INDEX_CHNL(par->dma_out));
+	else
+		count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in),
+				       DMA_INDEX_CHNL(par->dma_in));
+
+	count -= runtime->dma_addr;
+
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	offset = bytes_to_frames(runtime, count);
+	if (unlikely(offset >= runtime->buffer_size))
+		offset = 0;
+
+	return offset;
+}
+
+static int s6000_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct s6000_runtime_data *prtd;
+	int ret;
+
+	snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware);
+
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16);
+	if (ret < 0)
+		return ret;
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16);
+	if (ret < 0)
+		return ret;
+	ret = snd_pcm_hw_constraint_integer(runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		return ret;
+
+	if (par->same_rate) {
+		int rate;
+		spin_lock(&par->lock); /* needed? */
+		rate = par->rate;
+		spin_unlock(&par->lock);
+		if (rate != -1) {
+			ret = snd_pcm_hw_constraint_minmax(runtime,
+							SNDRV_PCM_HW_PARAM_RATE,
+							rate, rate);
+			if (ret < 0)
+				return ret;
+		}
+	}
+
+	prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL);
+	if (prtd == NULL)
+		return -ENOMEM;
+
+	spin_lock_init(&prtd->lock);
+
+	runtime->private_data = prtd;
+
+	return 0;
+}
+
+static int s6000_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct s6000_runtime_data *prtd = runtime->private_data;
+
+	kfree(prtd);
+
+	return 0;
+}
+
+static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	int ret;
+	ret = snd_pcm_lib_malloc_pages(substream,
+				       params_buffer_bytes(hw_params));
+	if (ret < 0) {
+		printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n");
+		return ret;
+	}
+
+	if (par->same_rate) {
+		spin_lock(&par->lock);
+		if (par->rate == -1 ||
+		    !(par->in_use & ~(1 << substream->stream))) {
+			par->rate = params_rate(hw_params);
+			par->in_use |= 1 << substream->stream;
+		} else if (params_rate(hw_params) != par->rate) {
+			snd_pcm_lib_free_pages(substream);
+			par->in_use &= ~(1 << substream->stream);
+			ret = -EBUSY;
+		}
+		spin_unlock(&par->lock);
+	}
+	return ret;
+}
+
+static int s6000_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+
+	spin_lock(&par->lock);
+	par->in_use &= ~(1 << substream->stream);
+	if (!par->in_use)
+		par->rate = -1;
+	spin_unlock(&par->lock);
+
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_pcm_ops s6000_pcm_ops = {
+	.open = 	s6000_pcm_open,
+	.close = 	s6000_pcm_close,
+	.ioctl = 	snd_pcm_lib_ioctl,
+	.hw_params = 	s6000_pcm_hw_params,
+	.hw_free = 	s6000_pcm_hw_free,
+	.trigger =	s6000_pcm_trigger,
+	.prepare = 	s6000_pcm_prepare,
+	.pointer = 	s6000_pcm_pointer,
+};
+
+static void s6000_pcm_free(struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+
+	free_irq(params->irq, pcm);
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static u64 s6000_pcm_dmamask = DMA_32BIT_MASK;
+
+static int s6000_pcm_new(struct snd_card *card,
+			 struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	int res;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &s6000_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+	if (params->dma_in) {
+		s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in),
+				    DMA_INDEX_CHNL(params->dma_in));
+		s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in),
+				   DMA_INDEX_CHNL(params->dma_in));
+	}
+
+	if (params->dma_out) {
+		s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out),
+				    DMA_INDEX_CHNL(params->dma_out));
+		s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out),
+				   DMA_INDEX_CHNL(params->dma_out));
+	}
+
+	res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED,
+			  s6000_soc_platform.name, pcm);
+	if (res) {
+		printk(KERN_ERR "s6000-pcm couldn't get IRQ\n");
+		return res;
+	}
+
+	res = snd_pcm_lib_preallocate_pages_for_all(pcm,
+						    SNDRV_DMA_TYPE_DEV,
+						    card->dev,
+						    S6_PCM_PREALLOCATE_SIZE,
+						    S6_PCM_PREALLOCATE_MAX);
+	if (res)
+		printk(KERN_WARNING "s6000-pcm: preallocation failed\n");
+
+	spin_lock_init(&params->lock);
+	params->in_use = 0;
+	params->rate = -1;
+	return 0;
+}
+
+struct snd_soc_platform s6000_soc_platform = {
+	.name = 	"s6000-audio",
+	.pcm_ops = 	&s6000_pcm_ops,
+	.pcm_new = 	s6000_pcm_new,
+	.pcm_free = 	s6000_pcm_free,
+};
+EXPORT_SYMBOL_GPL(s6000_soc_platform);
+
+static int __init s6000_pcm_init(void)
+{
+	return snd_soc_register_platform(&s6000_soc_platform);
+}
+module_init(s6000_pcm_init);
+
+static void __exit s6000_pcm_exit(void)
+{
+	snd_soc_unregister_platform(&s6000_soc_platform);
+}
+module_exit(s6000_pcm_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h
new file mode 100644
index 00000000000..96f23f6f52b
--- /dev/null
+++ b/sound/soc/s6000/s6000-pcm.h
@@ -0,0 +1,35 @@
+/*
+ * ALSA PCM interface for the Stretch s6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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 _S6000_PCM_H
+#define _S6000_PCM_H
+
+struct snd_soc_dai;
+struct snd_pcm_substream;
+
+struct s6000_pcm_dma_params {
+	unsigned int (*check_xrun)(struct snd_soc_dai *cpu_dai);
+	int (*trigger)(struct snd_pcm_substream *substream, int cmd, int after);
+	dma_addr_t sif_in;
+	dma_addr_t sif_out;
+	u32 dma_in;
+	u32 dma_out;
+	int irq;
+	int same_rate;
+
+	spinlock_t lock;
+	int in_use;
+	int rate;
+};
+
+extern struct snd_soc_platform s6000_soc_platform;
+
+#endif
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
new file mode 100644
index 00000000000..b5f95f9781c
--- /dev/null
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -0,0 +1,244 @@
+/*
+ * ASoC driver for Stretch s6105 IP camera platform
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <variant/dmac.h>
+
+#include "../codecs/tlv320aic3x.h"
+#include "s6000-pcm.h"
+#include "s6000-i2s.h"
+
+#define S6105_CAM_CODEC_CLOCK 12288000
+
+static int s6105_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret = 0;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+					     SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
+					   SND_SOC_DAIFMT_NB_NF);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, S6105_CAM_CODEC_CLOCK,
+					    SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops s6105_ops = {
+	.hw_params = s6105_hw_params,
+};
+
+/* s6105 machine dapm widgets */
+static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Audio Out Differential", NULL),
+	SND_SOC_DAPM_LINE("Audio Out Stereo", NULL),
+	SND_SOC_DAPM_LINE("Audio In", NULL),
+};
+
+/* s6105 machine audio_mapnections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Audio Out connected to HPLOUT, HPLCOM, HPROUT */
+	{"Audio Out Differential", NULL, "HPLOUT"},
+	{"Audio Out Differential", NULL, "HPLCOM"},
+	{"Audio Out Stereo", NULL, "HPLOUT"},
+	{"Audio Out Stereo", NULL, "HPROUT"},
+
+	/* Audio In connected to LINE1L, LINE1R */
+	{"LINE1L", NULL, "Audio In"},
+	{"LINE1R", NULL, "Audio In"},
+};
+
+static int output_type_info(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 2;
+	if (uinfo->value.enumerated.item) {
+		uinfo->value.enumerated.item = 1;
+		strcpy(uinfo->value.enumerated.name, "HPLOUT/HPROUT");
+	} else {
+		strcpy(uinfo->value.enumerated.name, "HPLOUT/HPLCOM");
+	}
+	return 0;
+}
+
+static int output_type_get(struct snd_kcontrol *kcontrol,
+			   struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.enumerated.item[0] = kcontrol->private_value;
+	return 0;
+}
+
+static int output_type_put(struct snd_kcontrol *kcontrol,
+			   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	unsigned int val = (ucontrol->value.enumerated.item[0] != 0);
+	char *differential = "Audio Out Differential";
+	char *stereo = "Audio Out Stereo";
+
+	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);
+
+	return 1;
+}
+
+static const struct snd_kcontrol_new audio_out_mux = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Master Output Mux",
+	.index = 0,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = output_type_info,
+	.get = output_type_get,
+	.put = output_type_put,
+	.private_value = 1 /* default to stereo */
+};
+
+/* Logic for a aic3x as connected on the s6105 ip camera ref design */
+static int s6105_aic3x_init(struct snd_soc_codec *codec)
+{
+	/* Add s6105 specific widgets */
+	snd_soc_dapm_new_controls(codec, 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));
+
+	/* not present */
+	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(codec, "LINE2L");
+	snd_soc_dapm_nc_pin(codec, "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");
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(codec, "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_sync(codec);
+
+	snd_ctl_add(codec->card, snd_ctl_new1(&audio_out_mux, codec));
+
+	return 0;
+}
+
+/* s6105 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link s6105_dai = {
+	.name = "TLV320AIC31",
+	.stream_name = "AIC31",
+	.cpu_dai = &s6000_i2s_dai,
+	.codec_dai = &aic3x_dai,
+	.init = s6105_aic3x_init,
+	.ops = &s6105_ops,
+};
+
+/* s6105 audio machine driver */
+static struct snd_soc_card snd_soc_card_s6105 = {
+	.name = "Stretch IP Camera",
+	.platform = &s6000_soc_platform,
+	.dai_link = &s6105_dai,
+	.num_links = 1,
+};
+
+/* s6105 audio private data */
+static struct aic3x_setup_data s6105_aic3x_setup = {
+	.i2c_bus = 0,
+	.i2c_address = 0x18,
+};
+
+/* s6105 audio subsystem */
+static struct snd_soc_device s6105_snd_devdata = {
+	.card = &snd_soc_card_s6105,
+	.codec_dev = &soc_codec_dev_aic3x,
+	.codec_data = &s6105_aic3x_setup,
+};
+
+static struct s6000_snd_platform_data __initdata s6105_snd_data = {
+	.wide		= 0,
+	.channel_in	= 0,
+	.channel_out	= 1,
+	.lines_in	= 1,
+	.lines_out	= 1,
+	.same_rate	= 1,
+};
+
+static struct platform_device *s6105_snd_device;
+
+static int __init s6105_init(void)
+{
+	int ret;
+
+	s6105_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!s6105_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(s6105_snd_device, &s6105_snd_devdata);
+	s6105_snd_devdata.dev = &s6105_snd_device->dev;
+	platform_device_add_data(s6105_snd_device, &s6105_snd_data,
+				 sizeof(s6105_snd_data));
+
+	ret = platform_device_add(s6105_snd_device);
+	if (ret)
+		platform_device_put(s6105_snd_device);
+
+	return ret;
+}
+
+static void __exit s6105_exit(void)
+{
+	platform_device_unregister(s6105_snd_device);
+}
+
+module_init(s6105_init);
+module_exit(s6105_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6105 IP camera ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 56fa0872abb..b378096cadb 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -145,7 +145,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream,
 	recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1;
 
 	pr_debug("ssi_hw_params() enter\nssicr was    %08lx\n", ssicr);
-	pr_debug("bits: %d channels: %d\n", bits, channels);
+	pr_debug("bits: %u channels: %u\n", bits, channels);
 
 	ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA |
 		   CR_SWL_MASK);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1cd149b9ce6..3f44150d8e3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -113,6 +113,35 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
+static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
+	int ret;
+
+	if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates ||
+	    machine->symmetric_rates) {
+		dev_dbg(card->dev, "Symmetry forces %dHz rate\n", 
+			machine->rate);
+
+		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+						   SNDRV_PCM_HW_PARAM_RATE,
+						   machine->rate,
+						   machine->rate);
+		if (ret < 0) {
+			dev_err(card->dev,
+				"Unable to apply rate symmetry constraint: %d\n", ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
 /*
  * Called by ALSA when a PCM substream is opened, the runtime->hw record is
  * then initialized and any private data can be allocated. This also calls
@@ -221,6 +250,13 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		goto machine_err;
 	}
 
+	/* Symmetry only applies if we've already got an active stream. */
+	if (cpu_dai->active || codec_dai->active) {
+		ret = soc_pcm_apply_symmetry(substream);
+		if (ret != 0)
+			goto machine_err;
+	}
+
 	pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
 	pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
 	pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
@@ -263,7 +299,6 @@ static void close_delayed_work(struct work_struct *work)
 {
 	struct snd_soc_card *card = container_of(work, struct snd_soc_card,
 						 delayed_work.work);
-	struct snd_soc_device *socdev = card->socdev;
 	struct snd_soc_codec *codec = card->codec;
 	struct snd_soc_dai *codec_dai;
 	int i;
@@ -279,27 +314,10 @@ static void close_delayed_work(struct work_struct *work)
 
 		/* are we waiting on this codec DAI stream */
 		if (codec_dai->pop_wait == 1) {
-
-			/* Reduce power if no longer active */
-			if (codec->active == 0) {
-				pr_debug("pop wq D1 %s %s\n", codec->name,
-					 codec_dai->playback.stream_name);
-				snd_soc_dapm_set_bias_level(socdev,
-					SND_SOC_BIAS_PREPARE);
-			}
-
 			codec_dai->pop_wait = 0;
 			snd_soc_dapm_stream_event(codec,
 				codec_dai->playback.stream_name,
 				SND_SOC_DAPM_STREAM_STOP);
-
-			/* Fall into standby if no longer active */
-			if (codec->active == 0) {
-				pr_debug("pop wq D3 %s %s\n", codec->name,
-					 codec_dai->playback.stream_name);
-				snd_soc_dapm_set_bias_level(socdev,
-					SND_SOC_BIAS_STANDBY);
-			}
 		}
 	}
 	mutex_unlock(&pcm_mutex);
@@ -363,10 +381,6 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 		snd_soc_dapm_stream_event(codec,
 			codec_dai->capture.stream_name,
 			SND_SOC_DAPM_STREAM_STOP);
-
-		if (codec->active == 0 && codec_dai->pop_wait == 0)
-			snd_soc_dapm_set_bias_level(socdev,
-						SND_SOC_BIAS_STANDBY);
 	}
 
 	mutex_unlock(&pcm_mutex);
@@ -431,36 +445,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		cancel_delayed_work(&card->delayed_work);
 	}
 
-	/* do we need to power up codec */
-	if (codec->bias_level != SND_SOC_BIAS_ON) {
-		snd_soc_dapm_set_bias_level(socdev,
-					    SND_SOC_BIAS_PREPARE);
-
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			snd_soc_dapm_stream_event(codec,
-					codec_dai->playback.stream_name,
-					SND_SOC_DAPM_STREAM_START);
-		else
-			snd_soc_dapm_stream_event(codec,
-					codec_dai->capture.stream_name,
-					SND_SOC_DAPM_STREAM_START);
-
-		snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
-		snd_soc_dai_digital_mute(codec_dai, 0);
-
-	} else {
-		/* codec already powered - power on widgets */
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			snd_soc_dapm_stream_event(codec,
-					codec_dai->playback.stream_name,
-					SND_SOC_DAPM_STREAM_START);
-		else
-			snd_soc_dapm_stream_event(codec,
-					codec_dai->capture.stream_name,
-					SND_SOC_DAPM_STREAM_START);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_dapm_stream_event(codec,
+					  codec_dai->playback.stream_name,
+					  SND_SOC_DAPM_STREAM_START);
+	else
+		snd_soc_dapm_stream_event(codec,
+					  codec_dai->capture.stream_name,
+					  SND_SOC_DAPM_STREAM_START);
 
-		snd_soc_dai_digital_mute(codec_dai, 0);
-	}
+	snd_soc_dai_digital_mute(codec_dai, 0);
 
 out:
 	mutex_unlock(&pcm_mutex);
@@ -521,6 +515,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
+	machine->rate = params_rate(params);
+
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -632,6 +628,12 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_codec *codec = card->codec;
 	int i;
 
+	/* If the initialization of this soc device failed, there is no codec
+	 * associated with it. Just bail out in this case.
+	 */
+	if (!codec)
+		return 0;
+
 	/* Due to the resume being scheduled into a workqueue we could
 	* suspend before that's finished - wait for it to complete.
 	 */
@@ -1334,6 +1336,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 		return ret;
 	}
 
+	codec->socdev = socdev;
 	codec->card->dev = socdev->dev;
 	codec->card->private_data = codec;
 	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
@@ -1744,7 +1747,7 @@ int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
 {
 	int max = kcontrol->private_value;
 
-	if (max == 1)
+	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
 	else
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -1774,7 +1777,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
 	unsigned int shift = mc->shift;
 	unsigned int rshift = mc->rshift;
 
-	if (max == 1)
+	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
 	else
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -1881,7 +1884,7 @@ int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	int max = mc->max;
 
-	if (max == 1)
+	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
 	else
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -2065,7 +2068,7 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->ops->set_sysclk)
+	if (dai->ops && dai->ops->set_sysclk)
 		return dai->ops->set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
@@ -2085,7 +2088,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->ops->set_clkdiv)
+	if (dai->ops && dai->ops->set_clkdiv)
 		return dai->ops->set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
@@ -2104,7 +2107,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->ops->set_pll)
+	if (dai->ops && dai->ops->set_pll)
 		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
@@ -2120,7 +2123,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->ops->set_fmt)
+	if (dai->ops && dai->ops->set_fmt)
 		return dai->ops->set_fmt(dai, fmt);
 	else
 		return -EINVAL;
@@ -2139,7 +2142,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->ops->set_sysclk)
+	if (dai->ops && dai->ops->set_tdm_slot)
 		return dai->ops->set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
@@ -2155,7 +2158,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->ops->set_sysclk)
+	if (dai->ops && dai->ops->set_tristate)
 		return dai->ops->set_tristate(dai, tristate);
 	else
 		return -EINVAL;
@@ -2171,7 +2174,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->ops->digital_mute)
+	if (dai->ops && dai->ops->digital_mute)
 		return dai->ops->digital_mute(dai, mute);
 	else
 		return -EINVAL;
@@ -2352,6 +2355,39 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform)
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
 
+static u64 codec_format_map[] = {
+	SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
+	SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
+	SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
+	SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
+	SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
+	SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+	SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+	SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
+	SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
+	SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
+	SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
+	SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
+	SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
+	SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
+	| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+};
+
+/* Fix up the DAI formats for endianness: codecs don't actually see
+ * the endianness of the data but we're using the CPU format
+ * definitions which do need to include endianness so we ensure that
+ * codec DAIs always have both big and little endian variants set.
+ */
+static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
+		if (stream->formats & codec_format_map[i])
+			stream->formats |= codec_format_map[i];
+}
+
 /**
  * snd_soc_register_codec - Register a codec with the ASoC core
  *
@@ -2359,6 +2395,8 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
  */
 int snd_soc_register_codec(struct snd_soc_codec *codec)
 {
+	int i;
+
 	if (!codec->name)
 		return -EINVAL;
 
@@ -2368,6 +2406,11 @@ int snd_soc_register_codec(struct snd_soc_codec *codec)
 
 	INIT_LIST_HEAD(&codec->list);
 
+	for (i = 0; i < codec->num_dai; i++) {
+		fixup_codec_formats(&codec->dai[i].playback);
+		fixup_codec_formats(&codec->dai[i].capture);
+	}
+
 	mutex_lock(&client_mutex);
 	list_add(&codec->list, &codec_list);
 	snd_soc_instantiate_cards();
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 735903a7467..21c69074aa1 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -12,7 +12,7 @@
  *  Features:
  *    o Changes power status of internal codec blocks depending on the
  *      dynamic configuration of codec internal audio paths and active
- *      DAC's/ADC's.
+ *      DACs/ADCs.
  *    o Platform power domain - can support external components i.e. amps and
  *      mic/meadphone insertion events.
  *    o Automatic Mic Bias support
@@ -52,23 +52,21 @@
 
 /* dapm power sequences - make this per codec in the future */
 static int dapm_up_seq[] = {
-	snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
-	snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac,
-	snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga,
-	snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
+	snd_soc_dapm_pre, snd_soc_dapm_supply, snd_soc_dapm_micbias,
+	snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
+	snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl,
+	snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
+	snd_soc_dapm_post
 };
 
 static int dapm_down_seq[] = {
 	snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
 	snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
 	snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
-	snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post
+	snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_supply,
+	snd_soc_dapm_post
 };
 
-static int dapm_status = 1;
-module_param(dapm_status, int, 0);
-MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
-
 static void pop_wait(u32 pop_time)
 {
 	if (pop_time)
@@ -96,6 +94,48 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
 	return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
 }
 
+/**
+ * snd_soc_dapm_set_bias_level - set the bias level for the system
+ * @socdev: audio device
+ * @level: level to configure
+ *
+ * Configure the bias (power) levels for the SoC audio device.
+ *
+ * Returns 0 for success else error.
+ */
+static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
+				       enum snd_soc_bias_level level)
+{
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		dev_dbg(socdev->dev, "Setting full bias\n");
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		dev_dbg(socdev->dev, "Setting bias prepare\n");
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		dev_dbg(socdev->dev, "Setting standby bias\n");
+		break;
+	case SND_SOC_BIAS_OFF:
+		dev_dbg(socdev->dev, "Setting bias off\n");
+		break;
+	default:
+		dev_err(socdev->dev, "Setting invalid bias %d\n", level);
+		return -EINVAL;
+	}
+
+	if (card->set_bias_level)
+		ret = card->set_bias_level(card, level);
+	if (ret == 0 && codec->set_bias_level)
+		ret = codec->set_bias_level(codec, level);
+
+	return ret;
+}
+
 /* set up initial codec paths */
 static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	struct snd_soc_dapm_path *p, int i)
@@ -165,6 +205,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_dac:
 	case snd_soc_dapm_micbias:
 	case snd_soc_dapm_vmid:
+	case snd_soc_dapm_supply:
 		p->connect = 1;
 	break;
 	/* does effect routing - dynamically connected */
@@ -179,7 +220,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	}
 }
 
-/* connect mux widget to it's interconnecting audio paths */
+/* connect mux widget to its interconnecting audio paths */
 static int dapm_connect_mux(struct snd_soc_codec *codec,
 	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
 	struct snd_soc_dapm_path *path, const char *control_name,
@@ -202,7 +243,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec,
 	return -ENODEV;
 }
 
-/* connect mixer widget to it's interconnecting audio paths */
+/* connect mixer widget to its interconnecting audio paths */
 static int dapm_connect_mixer(struct snd_soc_codec *codec,
 	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
 	struct snd_soc_dapm_path *path, const char *control_name)
@@ -357,8 +398,9 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
 				path->long_name);
 			ret = snd_ctl_add(codec->card, path->kcontrol);
 			if (ret < 0) {
-				printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
-						path->long_name);
+				printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n",
+				       path->long_name,
+				       ret);
 				kfree(path->long_name);
 				path->long_name = NULL;
 				return ret;
@@ -434,6 +476,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
 	struct snd_soc_dapm_path *path;
 	int con = 0;
 
+	if (widget->id == snd_soc_dapm_supply)
+		return 0;
+
 	if (widget->id == snd_soc_dapm_adc && widget->active)
 		return 1;
 
@@ -470,6 +515,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 	struct snd_soc_dapm_path *path;
 	int con = 0;
 
+	if (widget->id == snd_soc_dapm_supply)
+		return 0;
+
 	/* active stream ? */
 	if (widget->id == snd_soc_dapm_dac && widget->active)
 		return 1;
@@ -521,84 +569,12 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 }
 EXPORT_SYMBOL_GPL(dapm_reg_event);
 
-/*
- * Scan a single DAPM widget for a complete audio path and update the
- * power status appropriately.
+/* Standard power change method, used to apply power changes to most
+ * widgets.
  */
-static int dapm_power_widget(struct snd_soc_codec *codec, int event,
-			     struct snd_soc_dapm_widget *w)
+static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w)
 {
-	int in, out, power_change, power, ret;
-
-	/* vmid - no action */
-	if (w->id == snd_soc_dapm_vmid)
-		return 0;
-
-	/* active ADC */
-	if (w->id == snd_soc_dapm_adc && w->active) {
-		in = is_connected_input_ep(w);
-		dapm_clear_walk(w->codec);
-		w->power = (in != 0) ? 1 : 0;
-		dapm_update_bits(w);
-		return 0;
-	}
-
-	/* active DAC */
-	if (w->id == snd_soc_dapm_dac && w->active) {
-		out = is_connected_output_ep(w);
-		dapm_clear_walk(w->codec);
-		w->power = (out != 0) ? 1 : 0;
-		dapm_update_bits(w);
-		return 0;
-	}
-
-	/* pre and post event widgets */
-	if (w->id == snd_soc_dapm_pre) {
-		if (!w->event)
-			return 0;
-
-		if (event == SND_SOC_DAPM_STREAM_START) {
-			ret = w->event(w,
-				       NULL, SND_SOC_DAPM_PRE_PMU);
-			if (ret < 0)
-				return ret;
-		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
-			ret = w->event(w,
-				       NULL, SND_SOC_DAPM_PRE_PMD);
-			if (ret < 0)
-				return ret;
-		}
-		return 0;
-	}
-	if (w->id == snd_soc_dapm_post) {
-		if (!w->event)
-			return 0;
-
-		if (event == SND_SOC_DAPM_STREAM_START) {
-			ret = w->event(w,
-				       NULL, SND_SOC_DAPM_POST_PMU);
-			if (ret < 0)
-				return ret;
-		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
-			ret = w->event(w,
-				       NULL, SND_SOC_DAPM_POST_PMD);
-			if (ret < 0)
-				return ret;
-		}
-		return 0;
-	}
-
-	/* all other widgets */
-	in = is_connected_input_ep(w);
-	dapm_clear_walk(w->codec);
-	out = is_connected_output_ep(w);
-	dapm_clear_walk(w->codec);
-	power = (out != 0 && in != 0) ? 1 : 0;
-	power_change = (w->power == power) ? 0 : 1;
-	w->power = power;
-
-	if (!power_change)
-		return 0;
+	int ret;
 
 	/* call any power change event handlers */
 	if (w->event)
@@ -607,7 +583,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
 			 w->name, w->event_flags);
 
 	/* power up pre event */
-	if (power && w->event &&
+	if (w->power && w->event &&
 	    (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
 		ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
 		if (ret < 0)
@@ -615,7 +591,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
 	}
 
 	/* power down pre event */
-	if (!power && w->event &&
+	if (!w->power && w->event &&
 	    (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
 		ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
 		if (ret < 0)
@@ -623,17 +599,17 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
 	}
 
 	/* Lower PGA volume to reduce pops */
-	if (w->id == snd_soc_dapm_pga && !power)
-		dapm_set_pga(w, power);
+	if (w->id == snd_soc_dapm_pga && !w->power)
+		dapm_set_pga(w, w->power);
 
 	dapm_update_bits(w);
 
 	/* Raise PGA volume to reduce pops */
-	if (w->id == snd_soc_dapm_pga && power)
-		dapm_set_pga(w, power);
+	if (w->id == snd_soc_dapm_pga && w->power)
+		dapm_set_pga(w, w->power);
 
 	/* power up post event */
-	if (power && w->event &&
+	if (w->power && w->event &&
 	    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
 		ret = w->event(w,
 			       NULL, SND_SOC_DAPM_POST_PMU);
@@ -642,7 +618,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
 	}
 
 	/* power down post event */
-	if (!power && w->event &&
+	if (!w->power && w->event &&
 	    (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
 		ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
 		if (ret < 0)
@@ -652,6 +628,116 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
 	return 0;
 }
 
+/* Generic check to see if a widget should be powered.
+ */
+static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
+{
+	int in, out;
+
+	in = is_connected_input_ep(w);
+	dapm_clear_walk(w->codec);
+	out = is_connected_output_ep(w);
+	dapm_clear_walk(w->codec);
+	return out != 0 && in != 0;
+}
+
+/* Check to see if an ADC has power */
+static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
+{
+	int in;
+
+	if (w->active) {
+		in = is_connected_input_ep(w);
+		dapm_clear_walk(w->codec);
+		return in != 0;
+	} else {
+		return dapm_generic_check_power(w);
+	}
+}
+
+/* Check to see if a DAC has power */
+static int dapm_dac_check_power(struct snd_soc_dapm_widget *w)
+{
+	int out;
+
+	if (w->active) {
+		out = is_connected_output_ep(w);
+		dapm_clear_walk(w->codec);
+		return out != 0;
+	} else {
+		return dapm_generic_check_power(w);
+	}
+}
+
+/* Check to see if a power supply is needed */
+static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *path;
+	int power = 0;
+
+	/* Check if one of our outputs is connected */
+	list_for_each_entry(path, &w->sinks, list_source) {
+		if (path->sink && path->sink->power_check &&
+		    path->sink->power_check(path->sink)) {
+			power = 1;
+			break;
+		}
+	}
+
+	dapm_clear_walk(w->codec);
+
+	return power;
+}
+
+/*
+ * Scan a single DAPM widget for a complete audio path and update the
+ * power status appropriately.
+ */
+static int dapm_power_widget(struct snd_soc_codec *codec, int event,
+			     struct snd_soc_dapm_widget *w)
+{
+	int ret;
+
+	switch (w->id) {
+	case snd_soc_dapm_pre:
+		if (!w->event)
+			return 0;
+
+		if (event == SND_SOC_DAPM_STREAM_START) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_PRE_PMU);
+			if (ret < 0)
+				return ret;
+		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_PRE_PMD);
+			if (ret < 0)
+				return ret;
+		}
+		return 0;
+
+	case snd_soc_dapm_post:
+		if (!w->event)
+			return 0;
+
+		if (event == SND_SOC_DAPM_STREAM_START) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_POST_PMU);
+			if (ret < 0)
+				return ret;
+		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_POST_PMD);
+			if (ret < 0)
+				return ret;
+		}
+		return 0;
+
+	default:
+		return dapm_generic_apply_power(w);
+	}
+}
+
 /*
  * Scan each dapm widget for complete audio path.
  * A complete path is a route that has valid endpoints i.e.:-
@@ -663,31 +749,102 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
  */
 static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 {
+	struct snd_soc_device *socdev = codec->socdev;
 	struct snd_soc_dapm_widget *w;
-	int i, c = 1, *seq = NULL, ret = 0;
-
-	/* do we have a sequenced stream event */
-	if (event == SND_SOC_DAPM_STREAM_START) {
-		c = ARRAY_SIZE(dapm_up_seq);
-		seq = dapm_up_seq;
-	} else if (event == SND_SOC_DAPM_STREAM_STOP) {
-		c = ARRAY_SIZE(dapm_down_seq);
-		seq = dapm_down_seq;
+	int ret = 0;
+	int i, power;
+	int sys_power = 0;
+
+	INIT_LIST_HEAD(&codec->up_list);
+	INIT_LIST_HEAD(&codec->down_list);
+
+	/* 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) {
+		switch (w->id) {
+		case snd_soc_dapm_pre:
+			list_add_tail(&codec->down_list, &w->power_list);
+			break;
+		case snd_soc_dapm_post:
+			list_add_tail(&codec->up_list, &w->power_list);
+			break;
+
+		default:
+			if (!w->power_check)
+				continue;
+
+			power = w->power_check(w);
+			if (power)
+				sys_power = 1;
+
+			if (w->power == power)
+				continue;
+
+			if (power)
+				list_add_tail(&w->power_list, &codec->up_list);
+			else
+				list_add_tail(&w->power_list,
+					      &codec->down_list);
+
+			w->power = power;
+			break;
+		}
 	}
 
-	for (i = 0; i < c; i++) {
-		list_for_each_entry(w, &codec->dapm_widgets, list) {
+	/* 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(socdev,
+						  SND_SOC_BIAS_PREPARE);
+		if (ret != 0)
+			pr_err("Failed to prepare bias: %d\n", ret);
+	}
 
+	/* Power down widgets first; try to avoid amplifying pops. */
+	for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) {
+		list_for_each_entry(w, &codec->down_list, power_list) {
 			/* is widget in stream order */
-			if (seq && seq[i] && w->id != seq[i])
+			if (w->id != dapm_down_seq[i])
 				continue;
 
 			ret = dapm_power_widget(codec, event, w);
 			if (ret != 0)
-				return ret;
+				pr_err("Failed to power down %s: %d\n",
+				       w->name, ret);
 		}
 	}
 
+	/* Now power up. */
+	for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) {
+		list_for_each_entry(w, &codec->up_list, power_list) {
+			/* is widget in stream order */
+			if (w->id != dapm_up_seq[i])
+				continue;
+
+			ret = dapm_power_widget(codec, event, w);
+			if (ret != 0)
+				pr_err("Failed to power up %s: %d\n",
+				       w->name, ret);
+		}
+	}
+
+	/* 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(socdev,
+						  SND_SOC_BIAS_STANDBY);
+		if (ret != 0)
+			pr_err("Failed to apply standby 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(socdev,
+						  SND_SOC_BIAS_ON);
+		if (ret != 0)
+			pr_err("Failed to apply active bias: %d\n", ret);
+	}
+
 	return 0;
 }
 
@@ -723,6 +880,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_mixer:
 		case snd_soc_dapm_mixer_named_ctl:
+		case snd_soc_dapm_supply:
 			if (w->name) {
 				in = is_connected_input_ep(w);
 				dapm_clear_walk(w->codec);
@@ -851,6 +1009,7 @@ static ssize_t dapm_widget_show(struct device *dev,
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_mixer:
 		case snd_soc_dapm_mixer_named_ctl:
+		case snd_soc_dapm_supply:
 			if (w->name)
 				count += sprintf(buf + count, "%s: %s\n",
 					w->name, w->power ? "On":"Off");
@@ -883,16 +1042,12 @@ static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
 
 int snd_soc_dapm_sys_add(struct device *dev)
 {
-	if (!dapm_status)
-		return 0;
 	return device_create_file(dev, &dev_attr_dapm_widget);
 }
 
 static void snd_soc_dapm_sys_remove(struct device *dev)
 {
-	if (dapm_status) {
-		device_remove_file(dev, &dev_attr_dapm_widget);
-	}
+	device_remove_file(dev, &dev_attr_dapm_widget);
 }
 
 /* free all dapm widgets and resources */
@@ -1015,6 +1170,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
 	case snd_soc_dapm_vmid:
 	case snd_soc_dapm_pre:
 	case snd_soc_dapm_post:
+	case snd_soc_dapm_supply:
 		list_add(&path->list, &codec->dapm_paths);
 		list_add(&path->list_sink, &wsink->sources);
 		list_add(&path->list_source, &wsource->sinks);
@@ -1108,15 +1264,22 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
 		case snd_soc_dapm_switch:
 		case snd_soc_dapm_mixer:
 		case snd_soc_dapm_mixer_named_ctl:
+			w->power_check = dapm_generic_check_power;
 			dapm_new_mixer(codec, 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);
 			break;
 		case snd_soc_dapm_adc:
+			w->power_check = dapm_adc_check_power;
+			break;
 		case snd_soc_dapm_dac:
+			w->power_check = dapm_dac_check_power;
+			break;
 		case snd_soc_dapm_pga:
+			w->power_check = dapm_generic_check_power;
 			dapm_new_pga(codec, w);
 			break;
 		case snd_soc_dapm_input:
@@ -1126,6 +1289,10 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
 		case snd_soc_dapm_hp:
 		case snd_soc_dapm_mic:
 		case snd_soc_dapm_line:
+			w->power_check = dapm_generic_check_power;
+			break;
+		case snd_soc_dapm_supply:
+			w->power_check = dapm_supply_check_power;
 		case snd_soc_dapm_vmid:
 		case snd_soc_dapm_pre:
 		case snd_soc_dapm_post:
@@ -1626,35 +1793,11 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 
 /**
- * snd_soc_dapm_set_bias_level - set the bias level for the system
- * @socdev: audio device
- * @level: level to configure
- *
- * Configure the bias (power) levels for the SoC audio device.
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
-				enum snd_soc_bias_level level)
-{
-	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret = 0;
-
-	if (card->set_bias_level)
-		ret = card->set_bias_level(card, level);
-	if (ret == 0 && codec->set_bias_level)
-		ret = codec->set_bias_level(codec, level);
-
-	return ret;
-}
-
-/**
  * snd_soc_dapm_enable_pin - enable pin.
  * @codec: SoC codec
  * @pin: pin name
  *
- * Enables input/output pin and it's parents or children widgets iff there is
+ * Enables input/output pin and its parents or children widgets iff there is
  * a valid audio route and active audio stream.
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
@@ -1670,7 +1813,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
  * @codec: SoC codec
  * @pin: pin name
  *
- * Disables input/output pin and it's parents or children widgets.
+ * 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.
  */
diff --git a/sound/soc/txx9/Kconfig b/sound/soc/txx9/Kconfig
new file mode 100644
index 00000000000..ebc9327eae7
--- /dev/null
+++ b/sound/soc/txx9/Kconfig
@@ -0,0 +1,29 @@
+##
+## TXx9 ACLC
+##
+config SND_SOC_TXX9ACLC
+	tristate "SoC Audio for TXx9"
+	depends on HAS_TXX9_ACLC && TXX9_DMAC
+	help
+	  This option enables support for the AC Link Controllers in TXx9 SoC.
+
+config HAS_TXX9_ACLC
+	bool
+
+config SND_SOC_TXX9ACLC_AC97
+	tristate
+	select AC97_BUS
+	select SND_AC97_CODEC
+	select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_TXX9ACLC_GENERIC
+	tristate "Generic TXx9 ACLC sound machine"
+	depends on SND_SOC_TXX9ACLC
+	select SND_SOC_TXX9ACLC_AC97
+	select SND_SOC_AC97_CODEC
+	help
+	  This is a generic AC97 sound machine for use in TXx9 based systems.
diff --git a/sound/soc/txx9/Makefile b/sound/soc/txx9/Makefile
new file mode 100644
index 00000000000..551f16c0c4f
--- /dev/null
+++ b/sound/soc/txx9/Makefile
@@ -0,0 +1,11 @@
+# Platform
+snd-soc-txx9aclc-objs := txx9aclc.o
+snd-soc-txx9aclc-ac97-objs := txx9aclc-ac97.o
+
+obj-$(CONFIG_SND_SOC_TXX9ACLC) += snd-soc-txx9aclc.o
+obj-$(CONFIG_SND_SOC_TXX9ACLC_AC97) += snd-soc-txx9aclc-ac97.o
+
+# Machine
+snd-soc-txx9aclc-generic-objs := txx9aclc-generic.o
+
+obj-$(CONFIG_SND_SOC_TXX9ACLC_GENERIC) += snd-soc-txx9aclc-generic.o
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
new file mode 100644
index 00000000000..0f83bdb9b16
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -0,0 +1,255 @@
+/*
+ * TXx9 ACLC AC97 driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include "txx9aclc.h"
+
+#define AC97_DIR	\
+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AC97_RATES	\
+	SNDRV_PCM_RATE_8000_48000
+
+#ifdef __BIG_ENDIAN
+#define AC97_FMTS	SNDRV_PCM_FMTBIT_S16_BE
+#else
+#define AC97_FMTS	SNDRV_PCM_FMTBIT_S16_LE
+#endif
+
+static DECLARE_WAIT_QUEUE_HEAD(ac97_waitq);
+
+/* REVISIT: How to find txx9aclc_soc_device from snd_ac97? */
+static struct txx9aclc_soc_device *txx9aclc_soc_dev;
+
+static int txx9aclc_regready(struct txx9aclc_soc_device *dev)
+{
+	struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+
+	return __raw_readl(drvdata->base + ACINTSTS) & ACINT_REGACCRDY;
+}
+
+/* AC97 controller reads codec register */
+static unsigned short txx9aclc_ac97_read(struct snd_ac97 *ac97,
+					 unsigned short reg)
+{
+	struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+	struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+	void __iomem *base = drvdata->base;
+	u32 dat;
+
+	if (!(__raw_readl(base + ACINTSTS) & ACINT_CODECRDY(ac97->num)))
+		return 0xffff;
+	reg |= ac97->num << 7;
+	dat = (reg << ACREGACC_REG_SHIFT) | ACREGACC_READ;
+	__raw_writel(dat, base + ACREGACC);
+	__raw_writel(ACINT_REGACCRDY, base + ACINTEN);
+	if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) {
+		__raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+		dev_err(dev->soc_dev.dev, "ac97 read timeout (reg %#x)\n", reg);
+		dat = 0xffff;
+		goto done;
+	}
+	dat = __raw_readl(base + ACREGACC);
+	if (((dat >> ACREGACC_REG_SHIFT) & 0xff) != reg) {
+		dev_err(dev->soc_dev.dev, "reg mismatch %x with %x\n",
+			dat, reg);
+		dat = 0xffff;
+		goto done;
+	}
+	dat = (dat >> ACREGACC_DAT_SHIFT) & 0xffff;
+done:
+	__raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+	return dat;
+}
+
+/* AC97 controller writes to codec register */
+static void txx9aclc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+				unsigned short val)
+{
+	struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+	struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+	void __iomem *base = drvdata->base;
+
+	__raw_writel(((reg | (ac97->num << 7)) << ACREGACC_REG_SHIFT) |
+		     (val << ACREGACC_DAT_SHIFT),
+		     base + ACREGACC);
+	__raw_writel(ACINT_REGACCRDY, base + ACINTEN);
+	if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) {
+		dev_err(dev->soc_dev.dev,
+			"ac97 write timeout (reg %#x)\n", reg);
+	}
+	__raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+}
+
+static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+	struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+	struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+	void __iomem *base = drvdata->base;
+	u32 ready = ACINT_CODECRDY(ac97->num) | ACINT_REGACCRDY;
+
+	__raw_writel(ACCTL_ENLINK, base + ACCTLDIS);
+	mmiowb();
+	udelay(1);
+	__raw_writel(ACCTL_ENLINK, base + ACCTLEN);
+	/* wait for primary codec ready status */
+	__raw_writel(ready, base + ACINTEN);
+	if (!wait_event_timeout(ac97_waitq,
+				(__raw_readl(base + ACINTSTS) & ready) == ready,
+				HZ)) {
+		dev_err(&ac97->dev, "primary codec is not ready "
+			"(status %#x)\n",
+			__raw_readl(base + ACINTSTS));
+	}
+	__raw_writel(ACINT_REGACCRDY, base + ACINTSTS);
+	__raw_writel(ready, base + ACINTDIS);
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+	.read		= txx9aclc_ac97_read,
+	.write		= txx9aclc_ac97_write,
+	.reset		= txx9aclc_ac97_cold_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id)
+{
+	struct txx9aclc_plat_drvdata *drvdata = dev_id;
+	void __iomem *base = drvdata->base;
+
+	__raw_writel(__raw_readl(base + ACINTMSTS), base + ACINTDIS);
+	wake_up(&ac97_waitq);
+	return IRQ_HANDLED;
+}
+
+static int txx9aclc_ac97_probe(struct platform_device *pdev,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct txx9aclc_soc_device *dev =
+		container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+
+	dev->aclc_pdev = to_platform_device(dai->dev);
+	txx9aclc_soc_dev = dev;
+	return 0;
+}
+
+static void txx9aclc_ac97_remove(struct platform_device *pdev,
+				 struct snd_soc_dai *dai)
+{
+	struct platform_device *aclc_pdev = to_platform_device(dai->dev);
+	struct txx9aclc_plat_drvdata *drvdata = platform_get_drvdata(aclc_pdev);
+
+	/* disable AC-link */
+	__raw_writel(ACCTL_ENLINK, drvdata->base + ACCTLDIS);
+	txx9aclc_soc_dev = NULL;
+}
+
+struct snd_soc_dai txx9aclc_ac97_dai = {
+	.name			= "txx9aclc_ac97",
+	.ac97_control		= 1,
+	.probe			= txx9aclc_ac97_probe,
+	.remove			= txx9aclc_ac97_remove,
+	.playback = {
+		.rates		= AC97_RATES,
+		.formats	= AC97_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+	.capture = {
+		.rates		= AC97_RATES,
+		.formats	= AC97_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+};
+EXPORT_SYMBOL_GPL(txx9aclc_ac97_dai);
+
+static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev)
+{
+	struct txx9aclc_plat_drvdata *drvdata;
+	struct resource *r;
+	int err;
+	int irq;
+
+	irq = platform_get_irq(pdev, 0);
+	if (irq < 0)
+		return irq;
+	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!r)
+		return -EBUSY;
+
+	if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r),
+				     dev_name(&pdev->dev)))
+		return -EBUSY;
+
+	drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+	if (!drvdata)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, drvdata);
+	drvdata->physbase = r->start;
+	if (sizeof(drvdata->physbase) > sizeof(r->start) &&
+	    r->start >= TXX9_DIRECTMAP_BASE &&
+	    r->start < TXX9_DIRECTMAP_BASE + 0x400000)
+		drvdata->physbase |= 0xf00000000ull;
+	drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r));
+	if (!drvdata->base)
+		return -EBUSY;
+	err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq,
+			       IRQF_DISABLED, dev_name(&pdev->dev), drvdata);
+	if (err < 0)
+		return err;
+
+	txx9aclc_ac97_dai.dev = &pdev->dev;
+	return snd_soc_register_dai(&txx9aclc_ac97_dai);
+}
+
+static int __devexit txx9aclc_ac97_dev_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&txx9aclc_ac97_dai);
+	return 0;
+}
+
+static struct platform_driver txx9aclc_ac97_driver = {
+	.probe		= txx9aclc_ac97_dev_probe,
+	.remove		= __devexit_p(txx9aclc_ac97_dev_remove),
+	.driver		= {
+		.name	= "txx9aclc-ac97",
+		.owner	= THIS_MODULE,
+	},
+};
+
+static int __init txx9aclc_ac97_init(void)
+{
+	return platform_driver_register(&txx9aclc_ac97_driver);
+}
+
+static void __exit txx9aclc_ac97_exit(void)
+{
+	platform_driver_unregister(&txx9aclc_ac97_driver);
+}
+
+module_init(txx9aclc_ac97_init);
+module_exit(txx9aclc_ac97_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("TXx9 ACLC AC97 driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c
new file mode 100644
index 00000000000..3175de9a92c
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc-generic.c
@@ -0,0 +1,98 @@
+/*
+ * Generic TXx9 ACLC machine driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This is a very generic AC97 sound machine driver for boards which
+ * have (AC97) audio at ACLC (e.g. RBTX49XX boards).
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include "../codecs/ac97.h"
+#include "txx9aclc.h"
+
+static struct snd_soc_dai_link txx9aclc_generic_dai = {
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &txx9aclc_ac97_dai,
+	.codec_dai = &ac97_dai,
+};
+
+static struct snd_soc_card txx9aclc_generic_card = {
+	.name		= "Generic TXx9 ACLC Audio",
+	.platform	= &txx9aclc_soc_platform,
+	.dai_link	= &txx9aclc_generic_dai,
+	.num_links	= 1,
+};
+
+static struct txx9aclc_soc_device txx9aclc_generic_soc_device = {
+	.soc_dev = {
+		.card		= &txx9aclc_generic_card,
+		.codec_dev	= &soc_codec_dev_ac97,
+	},
+};
+
+static int __init txx9aclc_generic_probe(struct platform_device *pdev)
+{
+	struct txx9aclc_soc_device *dev = &txx9aclc_generic_soc_device;
+	struct platform_device *soc_pdev;
+	int ret;
+
+	soc_pdev = platform_device_alloc("soc-audio", -1);
+	if (!soc_pdev)
+		return -ENOMEM;
+	platform_set_drvdata(soc_pdev, &dev->soc_dev);
+	dev->soc_dev.dev = &soc_pdev->dev;
+	ret = platform_device_add(soc_pdev);
+	if (ret) {
+		platform_device_put(soc_pdev);
+		return ret;
+	}
+	platform_set_drvdata(pdev, soc_pdev);
+	return 0;
+}
+
+static int __exit txx9aclc_generic_remove(struct platform_device *pdev)
+{
+	struct platform_device *soc_pdev = platform_get_drvdata(pdev);
+
+	platform_device_unregister(soc_pdev);
+	return 0;
+}
+
+static struct platform_driver txx9aclc_generic_driver = {
+	.remove = txx9aclc_generic_remove,
+	.driver = {
+		.name = "txx9aclc-generic",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init txx9aclc_generic_init(void)
+{
+	return platform_driver_probe(&txx9aclc_generic_driver,
+				     txx9aclc_generic_probe);
+}
+
+static void __exit txx9aclc_generic_exit(void)
+{
+	platform_driver_unregister(&txx9aclc_generic_driver);
+}
+
+module_init(txx9aclc_generic_init);
+module_exit(txx9aclc_generic_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
new file mode 100644
index 00000000000..fa336616152
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc.c
@@ -0,0 +1,430 @@
+/*
+ * Generic TXx9 ACLC platform driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * 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/init.h>
+#include <linux/platform_device.h>
+#include <linux/scatterlist.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "txx9aclc.h"
+
+static const struct snd_pcm_hardware txx9aclc_pcm_hardware = {
+	/*
+	 * REVISIT: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID
+	 * needs more works for noncoherent MIPS.
+	 */
+	.info		  = SNDRV_PCM_INFO_INTERLEAVED |
+			    SNDRV_PCM_INFO_BATCH |
+			    SNDRV_PCM_INFO_PAUSE,
+#ifdef __BIG_ENDIAN
+	.formats	  = SNDRV_PCM_FMTBIT_S16_BE,
+#else
+	.formats	  = SNDRV_PCM_FMTBIT_S16_LE,
+#endif
+	.period_bytes_min = 1024,
+	.period_bytes_max = 8 * 1024,
+	.periods_min	  = 2,
+	.periods_max	  = 4096,
+	.buffer_bytes_max = 32 * 1024,
+};
+
+static int txx9aclc_pcm_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct txx9aclc_dmadata *dmadata = runtime->private_data;
+	int ret;
+
+	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	if (ret < 0)
+		return ret;
+
+	dev_dbg(socdev->dev,
+		"runtime->dma_area = %#lx dma_addr = %#lx dma_bytes = %zd "
+		"runtime->min_align %ld\n",
+		(unsigned long)runtime->dma_area,
+		(unsigned long)runtime->dma_addr, runtime->dma_bytes,
+		runtime->min_align);
+	dev_dbg(socdev->dev,
+		"periods %d period_bytes %d stream %d\n",
+		params_periods(params), params_period_bytes(params),
+		substream->stream);
+
+	dmadata->substream = substream;
+	dmadata->pos = 0;
+	return 0;
+}
+
+static int txx9aclc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int txx9aclc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct txx9aclc_dmadata *dmadata = runtime->private_data;
+
+	dmadata->dma_addr = runtime->dma_addr;
+	dmadata->buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+	dmadata->period_bytes = snd_pcm_lib_period_bytes(substream);
+
+	if (dmadata->buffer_bytes == dmadata->period_bytes) {
+		dmadata->frag_bytes = dmadata->period_bytes >> 1;
+		dmadata->frags = 2;
+	} else {
+		dmadata->frag_bytes = dmadata->period_bytes;
+		dmadata->frags = dmadata->buffer_bytes / dmadata->period_bytes;
+	}
+	dmadata->frag_count = 0;
+	dmadata->pos = 0;
+	return 0;
+}
+
+static void txx9aclc_dma_complete(void *arg)
+{
+	struct txx9aclc_dmadata *dmadata = arg;
+	unsigned long flags;
+
+	/* dma completion handler cannot submit new operations */
+	spin_lock_irqsave(&dmadata->dma_lock, flags);
+	if (dmadata->frag_count >= 0) {
+		dmadata->dmacount--;
+		BUG_ON(dmadata->dmacount < 0);
+		tasklet_schedule(&dmadata->tasklet);
+	}
+	spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+}
+
+static struct dma_async_tx_descriptor *
+txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
+{
+	struct dma_chan *chan = dmadata->dma_chan;
+	struct dma_async_tx_descriptor *desc;
+	struct scatterlist sg;
+
+	sg_init_table(&sg, 1);
+	sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)),
+		    dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1));
+	sg_dma_address(&sg) = buf_dma_addr;
+	desc = chan->device->device_prep_slave_sg(chan, &sg, 1,
+		dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+		DMA_TO_DEVICE : DMA_FROM_DEVICE,
+		DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+	if (!desc) {
+		dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+		return NULL;
+	}
+	desc->callback = txx9aclc_dma_complete;
+	desc->callback_param = dmadata;
+	desc->tx_submit(desc);
+	return desc;
+}
+
+#define NR_DMA_CHAIN		2
+
+static void txx9aclc_dma_tasklet(unsigned long data)
+{
+	struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data;
+	struct dma_chan *chan = dmadata->dma_chan;
+	struct dma_async_tx_descriptor *desc;
+	struct snd_pcm_substream *substream = dmadata->substream;
+	u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+		ACCTL_AUDODMA : ACCTL_AUDIDMA;
+	int i;
+	unsigned long flags;
+
+	spin_lock_irqsave(&dmadata->dma_lock, flags);
+	if (dmadata->frag_count < 0) {
+		struct txx9aclc_soc_device *dev =
+			container_of(dmadata, struct txx9aclc_soc_device,
+				     dmadata[substream->stream]);
+		struct txx9aclc_plat_drvdata *drvdata =
+			txx9aclc_get_plat_drvdata(dev);
+		void __iomem *base = drvdata->base;
+
+		spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+		chan->device->device_terminate_all(chan);
+		/* first time */
+		for (i = 0; i < NR_DMA_CHAIN; i++) {
+			desc = txx9aclc_dma_submit(dmadata,
+				dmadata->dma_addr + i * dmadata->frag_bytes);
+			if (!desc)
+				return;
+		}
+		dmadata->dmacount = NR_DMA_CHAIN;
+		chan->device->device_issue_pending(chan);
+		spin_lock_irqsave(&dmadata->dma_lock, flags);
+		__raw_writel(ctlbit, base + ACCTLEN);
+		dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags;
+		spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+		return;
+	}
+	BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN);
+	while (dmadata->dmacount < NR_DMA_CHAIN) {
+		dmadata->dmacount++;
+		spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+		desc = txx9aclc_dma_submit(dmadata,
+			dmadata->dma_addr +
+			dmadata->frag_count * dmadata->frag_bytes);
+		if (!desc)
+			return;
+		chan->device->device_issue_pending(chan);
+
+		spin_lock_irqsave(&dmadata->dma_lock, flags);
+		dmadata->frag_count++;
+		dmadata->frag_count %= dmadata->frags;
+		dmadata->pos += dmadata->frag_bytes;
+		dmadata->pos %= dmadata->buffer_bytes;
+		if ((dmadata->frag_count * dmadata->frag_bytes) %
+		    dmadata->period_bytes == 0)
+			snd_pcm_period_elapsed(substream);
+	}
+	spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+}
+
+static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct txx9aclc_soc_device *dev =
+		container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev);
+	struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+	void __iomem *base = drvdata->base;
+	unsigned long flags;
+	int ret = 0;
+	u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+		ACCTL_AUDODMA : ACCTL_AUDIDMA;
+
+	spin_lock_irqsave(&dmadata->dma_lock, flags);
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		dmadata->frag_count = -1;
+		tasklet_schedule(&dmadata->tasklet);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		__raw_writel(ctlbit, base + ACCTLDIS);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		__raw_writel(ctlbit, base + ACCTLEN);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+	return ret;
+}
+
+static snd_pcm_uframes_t
+txx9aclc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+
+	return bytes_to_frames(substream->runtime, dmadata->pos);
+}
+
+static int txx9aclc_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct txx9aclc_soc_device *dev =
+		container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev);
+	struct txx9aclc_dmadata *dmadata = &dev->dmadata[substream->stream];
+	int ret;
+
+	ret = snd_soc_set_runtime_hwparams(substream, &txx9aclc_pcm_hardware);
+	if (ret)
+		return ret;
+	/* ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(substream->runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		return ret;
+	substream->runtime->private_data = dmadata;
+	return 0;
+}
+
+static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+	struct dma_chan *chan = dmadata->dma_chan;
+
+	dmadata->frag_count = -1;
+	chan->device->device_terminate_all(chan);
+	return 0;
+}
+
+static struct snd_pcm_ops txx9aclc_pcm_ops = {
+	.open		= txx9aclc_pcm_open,
+	.close		= txx9aclc_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= txx9aclc_pcm_hw_params,
+	.hw_free	= txx9aclc_pcm_hw_free,
+	.prepare	= txx9aclc_pcm_prepare,
+	.trigger	= txx9aclc_pcm_trigger,
+	.pointer	= txx9aclc_pcm_pointer,
+};
+
+static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+			    struct snd_pcm *pcm)
+{
+	return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+		card->dev, 64 * 1024, 4 * 1024 * 1024);
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+	struct txx9aclc_dmadata *dmadata = param;
+	char devname[BUS_ID_SIZE + 2];
+
+	sprintf(devname, "%s.%d", dmadata->dma_res->name,
+		(int)dmadata->dma_res->start);
+	if (strcmp(dev_name(chan->device->dev), devname) == 0) {
+		chan->private = &dmadata->dma_slave;
+		return true;
+	}
+	return false;
+}
+
+static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev,
+			     struct txx9aclc_dmadata *dmadata)
+{
+	struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+	struct txx9dmac_slave *ds = &dmadata->dma_slave;
+	dma_cap_mask_t mask;
+
+	spin_lock_init(&dmadata->dma_lock);
+
+	ds->reg_width = sizeof(u32);
+	if (dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ds->tx_reg = drvdata->physbase + ACAUDODAT;
+		ds->rx_reg = 0;
+	} else {
+		ds->tx_reg = 0;
+		ds->rx_reg = drvdata->physbase + ACAUDIDAT;
+	}
+
+	/* Try to grab a DMA channel */
+	dma_cap_zero(mask);
+	dma_cap_set(DMA_SLAVE, mask);
+	dmadata->dma_chan = dma_request_channel(mask, filter, dmadata);
+	if (!dmadata->dma_chan) {
+		dev_err(dev->soc_dev.dev,
+			"DMA channel for %s is not available\n",
+			dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+			"playback" : "capture");
+		return -EBUSY;
+	}
+	tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet,
+		     (unsigned long)dmadata);
+	return 0;
+}
+
+static int txx9aclc_pcm_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct txx9aclc_soc_device *dev =
+		container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+	struct resource *r;
+	int i;
+	int ret;
+
+	dev->dmadata[0].stream = SNDRV_PCM_STREAM_PLAYBACK;
+	dev->dmadata[1].stream = SNDRV_PCM_STREAM_CAPTURE;
+	for (i = 0; i < 2; i++) {
+		r = platform_get_resource(dev->aclc_pdev, IORESOURCE_DMA, i);
+		if (!r) {
+			ret = -EBUSY;
+			goto exit;
+		}
+		dev->dmadata[i].dma_res = r;
+		ret = txx9aclc_dma_init(dev, &dev->dmadata[i]);
+		if (ret)
+			goto exit;
+	}
+	return 0;
+
+exit:
+	for (i = 0; i < 2; i++) {
+		if (dev->dmadata[i].dma_chan)
+			dma_release_channel(dev->dmadata[i].dma_chan);
+		dev->dmadata[i].dma_chan = NULL;
+	}
+	return ret;
+}
+
+static int txx9aclc_pcm_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct txx9aclc_soc_device *dev =
+		container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+	struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+	void __iomem *base = drvdata->base;
+	int i;
+
+	/* disable all FIFO DMAs */
+	__raw_writel(ACCTL_AUDODMA | ACCTL_AUDIDMA, base + ACCTLDIS);
+	/* dummy R/W to clear pending DMAREQ if any */
+	__raw_writel(__raw_readl(base + ACAUDIDAT), base + ACAUDODAT);
+
+	for (i = 0; i < 2; i++) {
+		struct txx9aclc_dmadata *dmadata = &dev->dmadata[i];
+		struct dma_chan *chan = dmadata->dma_chan;
+		if (chan) {
+			dmadata->frag_count = -1;
+			chan->device->device_terminate_all(chan);
+			dma_release_channel(chan);
+		}
+		dev->dmadata[i].dma_chan = NULL;
+	}
+	return 0;
+}
+
+struct snd_soc_platform txx9aclc_soc_platform = {
+	.name		= "txx9aclc-audio",
+	.probe		= txx9aclc_pcm_probe,
+	.remove		= txx9aclc_pcm_remove,
+	.pcm_ops 	= &txx9aclc_pcm_ops,
+	.pcm_new	= txx9aclc_pcm_new,
+	.pcm_free	= txx9aclc_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(txx9aclc_soc_platform);
+
+static int __init txx9aclc_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&txx9aclc_soc_platform);
+}
+
+static void __exit txx9aclc_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&txx9aclc_soc_platform);
+}
+
+module_init(txx9aclc_soc_platform_init);
+module_exit(txx9aclc_soc_platform_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc.h b/sound/soc/txx9/txx9aclc.h
new file mode 100644
index 00000000000..6769aab41b3
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc.h
@@ -0,0 +1,83 @@
+/*
+ * TXx9 SoC AC Link Controller
+ *
+ * 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 __TXX9ACLC_H
+#define __TXX9ACLC_H
+
+#include <linux/interrupt.h>
+#include <asm/txx9/dmac.h>
+
+#define ACCTLEN			0x00	/* control enable */
+#define ACCTLDIS		0x04	/* control disable */
+#define   ACCTL_ENLINK		0x00000001	/* enable/disable AC-link */
+#define   ACCTL_AUDODMA		0x00000100	/* AUDODMA enable/disable */
+#define   ACCTL_AUDIDMA		0x00001000	/* AUDIDMA enable/disable */
+#define   ACCTL_AUDOEHLT	0x00010000	/* AUDO error halt
+						   enable/disable */
+#define   ACCTL_AUDIEHLT	0x00100000	/* AUDI error halt
+						   enable/disable */
+#define ACREGACC		0x08	/* codec register access */
+#define   ACREGACC_DAT_SHIFT	0	/* data field */
+#define   ACREGACC_REG_SHIFT	16	/* address field */
+#define   ACREGACC_CODECID_SHIFT	24	/* CODEC ID field */
+#define   ACREGACC_READ		0x80000000	/* CODEC read */
+#define   ACREGACC_WRITE	0x00000000	/* CODEC write */
+#define ACINTSTS		0x10	/* interrupt status */
+#define ACINTMSTS		0x14	/* interrupt masked status */
+#define ACINTEN			0x18	/* interrupt enable */
+#define ACINTDIS		0x1c	/* interrupt disable */
+#define   ACINT_CODECRDY(n)	(0x00000001 << (n))	/* CODECn ready */
+#define   ACINT_REGACCRDY	0x00000010	/* ACREGACC ready */
+#define   ACINT_AUDOERR		0x00000100	/* AUDO underrun error */
+#define   ACINT_AUDIERR		0x00001000	/* AUDI overrun error */
+#define ACDMASTS		0x80	/* DMA request status */
+#define   ACDMA_AUDO		0x00000001	/* AUDODMA pending */
+#define   ACDMA_AUDI		0x00000010	/* AUDIDMA pending */
+#define ACAUDODAT		0xa0	/* audio out data */
+#define ACAUDIDAT		0xb0	/* audio in data */
+#define ACREVID			0xfc	/* revision ID */
+
+struct txx9aclc_dmadata {
+	struct resource *dma_res;
+	struct txx9dmac_slave dma_slave;
+	struct dma_chan *dma_chan;
+	struct tasklet_struct tasklet;
+	spinlock_t dma_lock;
+	int stream; /* SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE */
+	struct snd_pcm_substream *substream;
+	unsigned long pos;
+	dma_addr_t dma_addr;
+	unsigned long buffer_bytes;
+	unsigned long period_bytes;
+	unsigned long frag_bytes;
+	int frags;
+	int frag_count;
+	int dmacount;
+};
+
+struct txx9aclc_plat_drvdata {
+	void __iomem *base;
+	u64 physbase;
+};
+
+struct txx9aclc_soc_device {
+	struct snd_soc_device soc_dev;
+	struct platform_device *aclc_pdev;	/* for ioresources, drvdata */
+	struct txx9aclc_dmadata dmadata[2];
+};
+
+static inline struct txx9aclc_plat_drvdata *txx9aclc_get_plat_drvdata(
+	struct txx9aclc_soc_device *sdev)
+{
+	return platform_get_drvdata(sdev->aclc_pdev);
+}
+
+extern struct snd_soc_platform txx9aclc_soc_platform;
+extern struct snd_soc_dai txx9aclc_ac97_dai;
+
+#endif /* __TXX9ACLC_H */