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
  ...

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)
 

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

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)

include/sound/soc-dai.h

@@ -44,24 +44,6 @@ struct snd_pcm_substream;
 #define SND_SOC_DAIFMT_CONT		(0 << 4) /* continuous clock */
 #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.
  *
@@ -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;
 };

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

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

include/sound/wm9081.h

+/*
+ * 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

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);

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;

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"

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/

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.

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

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);
 
 

sound/soc/atmel/snd-soc-afeb9260.c

+/*
+ * 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");
+

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;
 

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);

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
 

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

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);

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);
 

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,
 };
 
 /*

sound/soc/codecs/spdif_transciever.c

+/*
+ * 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);
+

sound/soc/codecs/spdif_transciever.h

+/*
+ * 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 */

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)

sound/soc/codecs/stac9766.h

+/*
+ * 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

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;

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__ */

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

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);
 

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;

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]);

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;

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;
 	}
 

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);
 

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;
 }

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;
 	}

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);
 

sound/soc/codecs/wm8940.h

+/*
+ * 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 */
+

sound/soc/codecs/wm8960.h

+/*
+ * 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

sound/soc/codecs/wm8988.h

+/*
+ * 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

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;

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,
 	},

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;

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);

sound/soc/fsl/Kconfig

 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.
+

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
 

sound/soc/fsl/mpc5200_psc_ac97.h

+/*
+ * 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__ */

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,
 };