Merge branch 'topic/asoc' into for-linus

MAINTAINERS

@@ -533,6 +533,8 @@ L:	device-drivers-devel@blackfin.uclinux.org
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 W:	http://wiki.analog.com/
 S:	Supported
+F:	sound/soc/codecs/adau*
+F:	sound/soc/codecs/adav*
 F:	sound/soc/codecs/ad1*
 F:	sound/soc/codecs/ssm*
 

include/sound/soc-dai.h

@@ -209,6 +209,10 @@ struct snd_soc_dai_driver {
 	struct snd_soc_pcm_stream capture;
 	struct snd_soc_pcm_stream playback;
 	unsigned int symmetric_rates:1;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /*

include/sound/soc-dapm.h

@@ -348,6 +348,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+			     const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
@@ -429,6 +431,7 @@ struct snd_soc_dapm_path {
 	/* status */
 	u32 connect:1;	/* source and sink widgets are connected */
 	u32 walked:1;	/* path has been walked */
+	u32 weak:1;	/* path ignored for power management */
 
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink);
@@ -444,6 +447,7 @@ struct snd_soc_dapm_widget {
 	char *name;		/* widget name */
 	char *sname;	/* stream name */
 	struct snd_soc_codec *codec;
+	struct snd_soc_platform *platform;
 	struct list_head list;
 	struct snd_soc_dapm_context *dapm;
 
@@ -507,10 +511,11 @@ struct snd_soc_dapm_context {
 
 	struct device *dev; /* from parent - for debug */
 	struct snd_soc_codec *codec; /* parent codec */
+	struct snd_soc_platform *platform; /* parent platform */
 	struct snd_soc_card *card; /* parent card */
 
 	/* used during DAPM updates */
-	int dev_power;
+	enum snd_soc_bias_level target_bias_level;
 	struct list_head list;
 
 #ifdef CONFIG_DEBUG_FS

include/sound/soc.h

@@ -202,6 +202,16 @@
 #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
 	SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
 
+/*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST		-2
+#define SND_SOC_COMP_ORDER_EARLY		-1
+#define SND_SOC_COMP_ORDER_NORMAL		0
+#define SND_SOC_COMP_ORDER_LATE		1
+#define SND_SOC_COMP_ORDER_LAST		2
+
 /*
  * Bias levels
  *
@@ -214,10 +224,10 @@
  * @OFF:     Power Off. No restrictions on transition times.
  */
 enum snd_soc_bias_level {
-	SND_SOC_BIAS_OFF,
-	SND_SOC_BIAS_STANDBY,
-	SND_SOC_BIAS_PREPARE,
-	SND_SOC_BIAS_ON,
+	SND_SOC_BIAS_OFF = 0,
+	SND_SOC_BIAS_STANDBY = 1,
+	SND_SOC_BIAS_PREPARE = 2,
+	SND_SOC_BIAS_ON = 3,
 };
 
 struct snd_jack;
@@ -258,6 +268,11 @@ enum snd_soc_compress_type {
 	SND_SOC_RBTREE_COMPRESSION
 };
 
+enum snd_soc_pcm_subclass {
+	SND_SOC_PCM_CLASS_PCM	= 0,
+	SND_SOC_PCM_CLASS_BE	= 1,
+};
+
 int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			     unsigned int freq, int dir);
 int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -297,6 +312,10 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec,
 				      unsigned int reg);
 int snd_soc_default_writable_register(struct snd_soc_codec *codec,
 				      unsigned int reg);
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+					unsigned int reg);
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+					unsigned int reg, unsigned int val);
 
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
@@ -349,6 +368,8 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 				  const char *prefix);
 int snd_soc_add_controls(struct snd_soc_codec *codec,
 	const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
+	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@@ -612,6 +633,10 @@ struct snd_soc_codec_driver {
 
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /* SoC platform interface */
@@ -623,10 +648,17 @@ struct snd_soc_platform_driver {
 	int (*resume)(struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
-		struct snd_pcm *);
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
 	void (*pcm_free)(struct snd_pcm *);
 
+	/* Default control and setup, added after probe() is run */
+	const struct snd_kcontrol_new *controls;
+	int num_controls;
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	int num_dapm_routes;
+
 	/*
 	 * For platform caused delay reporting.
 	 * Optional.
@@ -636,6 +668,14 @@ struct snd_soc_platform_driver {
 
 	/* platform stream ops */
 	struct snd_pcm_ops *ops;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
+
+	/* platform IO - used for platform DAPM */
+	unsigned int (*read)(struct snd_soc_platform *, unsigned int);
+	int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
 };
 
 struct snd_soc_platform {
@@ -650,6 +690,8 @@ struct snd_soc_platform {
 	struct snd_soc_card *card;
 	struct list_head list;
 	struct list_head card_list;
+
+	struct snd_soc_dapm_context dapm;
 };
 
 struct snd_soc_dai_link {
@@ -725,8 +767,10 @@ struct snd_soc_card {
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_card *,
+			      struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 	int (*set_bias_level_post)(struct snd_soc_card *,
+				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level);
 
 	long pmdown_time;
@@ -789,6 +833,9 @@ struct snd_soc_pcm_runtime  {
 	struct device dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
+	struct mutex pcm_mutex;
+	enum snd_soc_pcm_subclass pcm_subclass;
+	struct snd_pcm_ops ops;
 
 	unsigned int complete:1;
 	unsigned int dev_registered:1;

include/trace/events/asoc.h

@@ -9,6 +9,7 @@
 
 struct snd_soc_jack;
 struct snd_soc_codec;
+struct snd_soc_platform;
 struct snd_soc_card;
 struct snd_soc_dapm_widget;
 
@@ -59,6 +60,50 @@ DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read,
 
 );
 
+DECLARE_EVENT_CLASS(snd_soc_preg,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val),
+
+	TP_STRUCT__entry(
+		__string(	name,		platform->name	)
+		__field(	int,		id		)
+		__field(	unsigned int,	reg		)
+		__field(	unsigned int,	val		)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, platform->name);
+		__entry->id = platform->id;
+		__entry->reg = reg;
+		__entry->val = val;
+	),
+
+	TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name),
+		  (int)__entry->id, (unsigned int)__entry->reg,
+		  (unsigned int)__entry->val)
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val)
+
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val)
+
+);
+
 DECLARE_EVENT_CLASS(snd_soc_card,
 
 	TP_PROTO(struct snd_soc_card *card, int val),

sound/soc/Makefile

 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
+snd-soc-core-objs += soc-pcm.o soc-io.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/

sound/soc/atmel/atmel-pcm.c

@@ -364,9 +364,11 @@ static struct snd_pcm_ops atmel_pcm_ops = {
 \*--------------------------------------------------------------------------*/
 static u64 atmel_pcm_dmamask = 0xffffffff;
 
-static int atmel_pcm_new(struct snd_card *card,
-	struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)
@@ -382,7 +384,7 @@ static int atmel_pcm_new(struct snd_card *card,
 	}
 
 	if (dai->driver->capture.channels_min) {
-		pr_debug("at32-pcm:"
+		pr_debug("atmel-pcm:"
 				"Allocating PCM capture DMA buffer\n");
 		ret = atmel_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);

sound/soc/atmel/atmel-pcm.h

@@ -60,7 +60,7 @@ struct atmel_ssc_mask {
  * This structure, shared between the PCM driver and the interface,
  * contains all information required by the PCM driver to perform the
  * PDC DMA operation.  All fields except dma_intr_handler() are initialized
- * by the interface.  The dms_intr_handler() pointer is set by the PCM
+ * by the interface.  The dma_intr_handler() pointer is set by the PCM
  * driver and called by the interface SSC interrupt handler if it is
  * non-NULL.
  */

sound/soc/atmel/atmel_ssc_dai.c

@@ -402,7 +402,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
 		&& bits > 16) {
 		printk(KERN_WARNING
-				"atmel_ssc_dai: sample size %d"
+				"atmel_ssc_dai: sample size %d "
 				"is too large for I2S\n", bits);
 		return -EINVAL;
 	}
@@ -838,10 +838,8 @@ int atmel_ssc_set_audio(int ssc_id)
 	}
 
 	ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id);
-	if (!ssc_pdev) {
-		ssc_free(ssc);
+	if (!ssc_pdev)
 		return -ENOMEM;
-	}
 
 	/* If we can grab the SSC briefly to parent the DAI device off it */
 	ssc = ssc_request(ssc_id);

sound/soc/atmel/sam9g20_wm8731.c

@@ -92,6 +92,7 @@ static struct snd_soc_ops at91sam9g20ek_ops = {
 };
 
 static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
 					enum snd_soc_bias_level level)
 {
 	static int mclk_on;

sound/soc/au1x/dbdma2.c

@@ -319,10 +319,11 @@ static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-static int au1xpsc_pcm_new(struct snd_card *card,
-			   struct snd_soc_dai *dai,
-			   struct snd_pcm *pcm)
+static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 		card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
 

sound/soc/blackfin/Kconfig

@@ -10,13 +10,36 @@ config SND_BF5XX_I2S
 
 config SND_BF5XX_SOC_SSM2602
 	tristate "SoC SSM2602 Audio support for BF52x ezkit"
-	depends on SND_BF5XX_I2S
+	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_SSM2602
-	select I2C
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
+config SND_SOC_BFIN_EVAL_ADAU1701
+	tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
+	depends on SND_BF5XX_I2S
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_ADAU1701
+	select I2C
+	help
+	  Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ
+	  board connected to one of the Blackfin evaluation boards like the
+	  BF5XX-STAMP or BF5XX-EZKIT.
+
+config SND_SOC_BFIN_EVAL_ADAV80X
+	tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
+	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_ADAV80X
+	help
+	  Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
+	  EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
+	  like the BF5XX-STAMP or BF5XX-EZKIT.
+
+	  Note: This driver assumes that the ADAV80X digital record and playback
+	  interfaces are connected to the first SPORT port on the BF5XX board.
+
 config SND_BF5XX_SOC_AD73311
 	tristate "SoC AD73311 Audio support for Blackfin"
 	depends on SND_BF5XX_I2S

sound/soc/blackfin/Makefile

@@ -21,9 +21,13 @@ snd-ad1980-objs := bf5xx-ad1980.o
 snd-ssm2602-objs := bf5xx-ssm2602.o
 snd-ad73311-objs := bf5xx-ad73311.o
 snd-ad193x-objs := bf5xx-ad193x.o
+snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
+snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
 
 obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
 obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o

sound/soc/blackfin/bf5xx-ac97-pcm.c

@@ -418,9 +418,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	pr_debug("%s enter\n", __func__);

sound/soc/blackfin/bf5xx-i2s-pcm.c

@@ -168,7 +168,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
 
 	snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware);
 
-	ret = snd_pcm_hw_constraint_integer(runtime, \
+	ret = snd_pcm_hw_constraint_integer(runtime,
 			SNDRV_PCM_HW_PARAM_PERIODS);
 	if (ret < 0)
 		goto out;
@@ -257,9 +257,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	pr_debug("%s enter\n", __func__);
@@ -304,8 +306,8 @@ static int __devexit bfin_i2s_soc_platform_remove(struct platform_device *pdev)
 
 static struct platform_driver bfin_i2s_pcm_driver = {
 	.driver = {
-			.name = "bfin-i2s-pcm-audio",
-			.owner = THIS_MODULE,
+		.name = "bfin-i2s-pcm-audio",
+		.owner = THIS_MODULE,
 	},
 
 	.probe = bfin_i2s_soc_platform_probe,

sound/soc/blackfin/bf5xx-tdm-pcm.c

@@ -283,9 +283,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)

sound/soc/blackfin/bfin-eval-adau1701.c

+/*
+ * Machine driver for EVAL-ADAU1701MINIZ on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/adau1701.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adau1701_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adau1701_dapm_routes[] = {
+	{ "Speaker", NULL, "OUT0" },
+	{ "Speaker", NULL, "OUT1" },
+	{ "Line Out", NULL, "OUT2" },
+	{ "Line Out", NULL, "OUT3" },
+
+	{ "IN0", NULL, "Line In" },
+	{ "IN1", NULL, "Line In" },
+};
+
+static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000,
+			SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+static struct snd_soc_ops bfin_eval_adau1701_ops = {
+	.hw_params = bfin_eval_adau1701_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = {
+	{
+		.name = "adau1701",
+		.stream_name = "adau1701",
+		.cpu_dai_name = "bfin-i2s.0",
+		.codec_dai_name = "adau1701",
+		.platform_name = "bfin-i2s-pcm-audio",
+		.codec_name = "adau1701.0-0034",
+		.ops = &bfin_eval_adau1701_ops,
+	},
+	{
+		.name = "adau1701",
+		.stream_name = "adau1701",
+		.cpu_dai_name = "bfin-i2s.1",
+		.codec_dai_name = "adau1701",
+		.platform_name = "bfin-i2s-pcm-audio",
+		.codec_name = "adau1701.0-0034",
+		.ops = &bfin_eval_adau1701_ops,
+	},
+};
+
+static struct snd_soc_card bfin_eval_adau1701 = {
+	.name = "bfin-eval-adau1701",
+	.dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM],
+	.num_links = 1,
+
+	.dapm_widgets		= bfin_eval_adau1701_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adau1701_dapm_widgets),
+	.dapm_routes		= bfin_eval_adau1701_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adau1701_dapm_routes),
+};
+
+static int bfin_eval_adau1701_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &bfin_eval_adau1701;
+
+	card->dev = &pdev->dev;
+
+	return snd_soc_register_card(&bfin_eval_adau1701);
+}
+
+static int __devexit bfin_eval_adau1701_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver bfin_eval_adau1701_driver = {
+	.driver = {
+		.name = "bfin-eval-adau1701",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bfin_eval_adau1701_probe,
+	.remove = __devexit_p(bfin_eval_adau1701_remove),
+};
+
+static int __init bfin_eval_adau1701_init(void)
+{
+	return platform_driver_register(&bfin_eval_adau1701_driver);
+}
+module_init(bfin_eval_adau1701_init);
+
+static void __exit bfin_eval_adau1701_exit(void)
+{
+	platform_driver_unregister(&bfin_eval_adau1701_driver);
+}
+module_exit(bfin_eval_adau1701_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bfin-eval-adau1701");

sound/soc/blackfin/bfin-eval-adav80x.c

+/*
+ * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include "../codecs/adav80x.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = {
+	{ "Line Out", NULL, "VOUTL" },
+	{ "Line Out", NULL, "VOUTR" },
+
+	{ "VINL", NULL, "Line In" },
+	{ "VINR", NULL, "Line In" },
+};
+
+static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL,
+			27000000, params_rate(params) * 256);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1,
+			params_rate(params) * 256, SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0,
+	    SND_SOC_CLOCK_OUT);
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0,
+	    SND_SOC_CLOCK_OUT);
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0,
+	    SND_SOC_CLOCK_OUT);
+
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0);
+
+	return 0;
+}
+
+static struct snd_soc_ops bfin_eval_adav80x_ops = {
+	.hw_params = bfin_eval_adav80x_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = {
+	{
+		.name = "adav80x",
+		.stream_name = "ADAV80x HiFi",
+		.cpu_dai_name = "bfin-i2s.0",
+		.codec_dai_name = "adav80x-hifi",
+		.platform_name = "bfin-i2s-pcm-audio",
+		.init = bfin_eval_adav80x_codec_init,
+		.ops = &bfin_eval_adav80x_ops,
+	},
+};
+
+static struct snd_soc_card bfin_eval_adav80x = {
+	.name = "bfin-eval-adav80x",
+	.dai_link = bfin_eval_adav80x_dais,
+	.num_links = ARRAY_SIZE(bfin_eval_adav80x_dais),
+
+	.dapm_widgets		= bfin_eval_adav80x_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets),
+	.dapm_routes		= bfin_eval_adav80x_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adav80x_dapm_routes),
+};
+
+enum bfin_eval_adav80x_type {
+	BFIN_EVAL_ADAV801,
+	BFIN_EVAL_ADAV803,
+};
+
+static int bfin_eval_adav80x_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &bfin_eval_adav80x;
+	const char *codec_name;
+
+	switch (platform_get_device_id(pdev)->driver_data) {
+	case BFIN_EVAL_ADAV801:
+		codec_name = "spi0.1";
+		break;
+	case BFIN_EVAL_ADAV803:
+		codec_name = "adav803.0-0034";
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	bfin_eval_adav80x_dais[0].codec_name = codec_name;
+
+	card->dev = &pdev->dev;
+
+	return snd_soc_register_card(&bfin_eval_adav80x);
+}
+
+static int __devexit bfin_eval_adav80x_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static const struct platform_device_id bfin_eval_adav80x_ids[] = {
+	{ "bfin-eval-adav801", BFIN_EVAL_ADAV801 },
+	{ "bfin-eval-adav803", BFIN_EVAL_ADAV803 },
+	{ },
+};
+MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids);
+
+static struct platform_driver bfin_eval_adav80x_driver = {
+	.driver = {
+		.name = "bfin-eval-adav80x",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bfin_eval_adav80x_probe,
+	.remove = __devexit_p(bfin_eval_adav80x_remove),
+	.id_table = bfin_eval_adav80x_ids,
+};
+
+static int __init bfin_eval_adav80x_init(void)
+{
+	return platform_driver_register(&bfin_eval_adav80x_driver);
+}
+module_init(bfin_eval_adav80x_init);
+
+static void __exit bfin_eval_adav80x_exit(void)
+{
+	platform_driver_unregister(&bfin_eval_adav80x_driver);
+}
+module_exit(bfin_eval_adav80x_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver");
+MODULE_LICENSE("GPL");

sound/soc/codecs/Kconfig

@@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311
+	select SND_SOC_ADAV80X
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
@@ -42,6 +43,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_SN95031 if INTEL_SCU_IPC
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_STA32X if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -71,6 +73,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8770 if SPI_MASTER
 	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8782
 	select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
@@ -84,6 +87,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8971 if I2C
 	select SND_SOC_WM8974 if I2C
 	select SND_SOC_WM8978 if I2C
+	select SND_SOC_WM8983 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8990 if I2C
@@ -130,7 +134,14 @@ config SND_SOC_AD1980
 
 config SND_SOC_AD73311
 	tristate
-	
+
+config SND_SOC_ADAU1701
+	select SIGMA
+	tristate
+
+config SND_SOC_ADAV80X
+	tristate
+
 config SND_SOC_ADS117X
 	tristate
 
@@ -216,6 +227,9 @@ config SND_SOC_SPDIF
 config SND_SOC_SSM2602
 	tristate
 
+config SND_SOC_STA32X
+	tristate
+
 config SND_SOC_STAC9766
 	tristate
 
@@ -299,6 +313,9 @@ config SND_SOC_WM8770
 config SND_SOC_WM8776
 	tristate
 
+config SND_SOC_WM8782
+	tristate
+
 config SND_SOC_WM8804
 	tristate
 
@@ -338,6 +355,9 @@ config SND_SOC_WM8974
 config SND_SOC_WM8978
 	tristate
 
+config SND_SOC_WM8983
+	tristate
+
 config SND_SOC_WM8985
 	tristate
 

sound/soc/codecs/Makefile

@@ -4,6 +4,8 @@ snd-soc-ad1836-objs := ad1836.o
 snd-soc-ad193x-objs := ad193x.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
+snd-soc-adau1701-objs := adau1701.o
+snd-soc-adav80x-objs := adav80x.o
 snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
@@ -28,6 +30,7 @@ snd-soc-alc5623-objs := alc5623.o
 snd-soc-sn95031-objs := sn95031.o
 snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-sta32x-objs := sta32x.o
 snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -55,6 +58,7 @@ snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8770-objs := wm8770.o
 snd-soc-wm8776-objs := wm8776.o
+snd-soc-wm8782-objs := wm8782.o
 snd-soc-wm8804-objs := wm8804.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
@@ -68,6 +72,7 @@ snd-soc-wm8962-objs := wm8962.o
 snd-soc-wm8971-objs := wm8971.o
 snd-soc-wm8974-objs := wm8974.o
 snd-soc-wm8978-objs := wm8978.o
+snd-soc-wm8983-objs := wm8983.o
 snd-soc-wm8985-objs := wm8985.o
 snd-soc-wm8988-objs := wm8988.o
 snd-soc-wm8990-objs := wm8990.o
@@ -95,6 +100,8 @@ obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU1701)  += snd-soc-adau1701.o
+obj-$(CONFIG_SND_SOC_ADAV80X)  += snd-soc-adav80x.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
@@ -120,6 +127,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
 obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
@@ -147,6 +155,7 @@ obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8770)	+= snd-soc-wm8770.o
 obj-$(CONFIG_SND_SOC_WM8776)	+= snd-soc-wm8776.o
+obj-$(CONFIG_SND_SOC_WM8782)	+= snd-soc-wm8782.o
 obj-$(CONFIG_SND_SOC_WM8804)	+= snd-soc-wm8804.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
@@ -160,6 +169,7 @@ obj-$(CONFIG_SND_SOC_WM8962)	+= snd-soc-wm8962.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8974)	+= snd-soc-wm8974.o
 obj-$(CONFIG_SND_SOC_WM8978)	+= snd-soc-wm8978.o
+obj-$(CONFIG_SND_SOC_WM8983)	+= snd-soc-wm8983.o
 obj-$(CONFIG_SND_SOC_WM8985)	+= snd-soc-wm8985.o
 obj-$(CONFIG_SND_SOC_WM8988)	+= snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o

sound/soc/codecs/ad1836.h

 /*
- * File:         sound/soc/codecs/ad1836.h
- * Based on:
- * Author:       Barry Song <Barry.Song@analog.com>
+ * Audio Codec driver supporting:
+ *  AD1835A, AD1836, AD1837A, AD1838A, AD1839A
  *
- * Created:      Aug 04, 2009
- * Description:  definitions for AD1836 registers
+ * Copyright 2009-2011 Analog Devices Inc.
  *
- * Modified:
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * Licensed under the GPL-2 or later.
  */
 
 #ifndef __AD1836_H__
@@ -21,39 +12,30 @@
 
 #define AD1836_DAC_CTRL1               0
 #define AD1836_DAC_POWERDOWN           2
-#define AD1836_DAC_SERFMT_MASK	       0xE0
+#define AD1836_DAC_SERFMT_MASK         0xE0
 #define AD1836_DAC_SERFMT_PCK256       (0x4 << 5)
 #define AD1836_DAC_SERFMT_PCK128       (0x5 << 5)
 #define AD1836_DAC_WORD_LEN_MASK       0x18
 #define AD1836_DAC_WORD_LEN_OFFSET     3
 
 #define AD1836_DAC_CTRL2               1
-#define AD1836_DACL1_MUTE              0
-#define AD1836_DACR1_MUTE              1
-#define AD1836_DACL2_MUTE              2
-#define AD1836_DACR2_MUTE              3
-#define AD1836_DACL3_MUTE              4
-#define AD1836_DACR3_MUTE              5
 
-#define AD1836_DAC_L1_VOL              2
-#define AD1836_DAC_R1_VOL              3
-#define AD1836_DAC_L2_VOL              4
-#define AD1836_DAC_R2_VOL              5
-#define AD1836_DAC_L3_VOL              6
-#define AD1836_DAC_R3_VOL              7
+/* These macros are one-based. So AD183X_MUTE_LEFT(1) will return the mute bit
+ * for the first ADC/DAC */
+#define AD1836_MUTE_LEFT(x) (((x) * 2) - 2)
+#define AD1836_MUTE_RIGHT(x) (((x) * 2) - 1)
+
+#define AD1836_DAC_L_VOL(x) ((x) * 2)
+#define AD1836_DAC_R_VOL(x) (1 + ((x) * 2))
 
 #define AD1836_ADC_CTRL1               12
 #define AD1836_ADC_POWERDOWN           7
 #define AD1836_ADC_HIGHPASS_FILTER     8
 
 #define AD1836_ADC_CTRL2               13
-#define AD1836_ADCL1_MUTE 		0
-#define AD1836_ADCR1_MUTE 		1
-#define AD1836_ADCL2_MUTE 		2
-#define AD1836_ADCR2_MUTE 		3
 #define AD1836_ADC_WORD_LEN_MASK       0x30
 #define AD1836_ADC_WORD_OFFSET         5
-#define AD1836_ADC_SERFMT_MASK	       (7 << 6)
+#define AD1836_ADC_SERFMT_MASK         (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
 #define AD1836_ADC_AUX                 (0x6 << 6)

sound/soc/codecs/adau1701.h

+/*
+ * header file for ADAU1701 SigmaDSP processor
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAU1701_H
+#define _ADAU1701_H
+
+enum adau1701_clk_src {
+	ADAU1701_CLK_SRC_OSC,
+	ADAU1701_CLK_SRC_MCLK,
+};
+
+#endif

sound/soc/codecs/adav80x.h

+/*
+ * header file for ADAV80X parts
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAV80X_H
+#define _ADAV80X_H
+
+enum adav80x_pll_src {
+	ADAV80X_PLL_SRC_XIN,
+	ADAV80X_PLL_SRC_XTAL,
+	ADAV80X_PLL_SRC_MCLKI,
+};
+
+enum adav80x_pll {
+	ADAV80X_PLL1 = 0,
+	ADAV80X_PLL2 = 1,
+};
+
+enum adav80x_clk_src {
+	ADAV80X_CLK_XIN = 0,
+	ADAV80X_CLK_MCLKI = 1,
+	ADAV80X_CLK_PLL1 = 2,
+	ADAV80X_CLK_PLL2 = 3,
+	ADAV80X_CLK_XTAL = 6,
+
+	ADAV80X_CLK_SYSCLK1 = 6,
+	ADAV80X_CLK_SYSCLK2 = 7,
+	ADAV80X_CLK_SYSCLK3 = 8,
+};
+
+#endif

sound/soc/codecs/ak4641.c

@@ -457,7 +457,7 @@ static struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
 	.set_sysclk   = ak4641_set_dai_sysclk,
 };
 
-struct snd_soc_dai_driver ak4641_dai[] = {
+static struct snd_soc_dai_driver ak4641_dai[] = {
 {
 	.name = "ak4641-hifi",
 	.id = 1,

sound/soc/codecs/cs4270.c

@@ -636,10 +636,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
 #endif /* CONFIG_PM */
 
 /*
- * ASoC codec device structure
- *
- * Assign this variable to the codec_dev field of the machine driver's
- * snd_soc_device structure.
+ * ASoC codec driver structure
  */
 static const struct snd_soc_codec_driver soc_codec_device_cs4270 = {
 	.probe =		cs4270_probe,

sound/soc/codecs/max98088.c

@@ -1397,8 +1397,6 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai,
        if (freq == max98088->sysclk)
                return 0;
 
-       max98088->sysclk = freq; /* remember current sysclk */
-
        /* Setup clocks for slave mode, and using the PLL
         * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
         *         0x02 (when master clk is 20MHz to 30MHz)..

sound/soc/codecs/max98095.c

@@ -1517,8 +1517,6 @@ static int max98095_dai_set_sysclk(struct snd_soc_dai *dai,
 	if (freq == max98095->sysclk)
 		return 0;
 
-	max98095->sysclk = freq; /* remember current sysclk */
-
 	/* Setup clocks for slave mode, and using the PLL
 	 * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
 	 *         0x02 (when master clk is 20MHz to 40MHz)..
@@ -2261,11 +2259,11 @@ static int max98095_probe(struct snd_soc_codec *codec)
 
 	ret = snd_soc_read(codec, M98095_0FF_REV_ID);
 	if (ret < 0) {
-		dev_err(codec->dev, "Failed to read device revision: %d\n",
+		dev_err(codec->dev, "Failure reading hardware revision: %d\n",
 			ret);
 		goto err_access;
 	}
-	dev_info(codec->dev, "revision %c\n", ret + 'A');
+	dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A');
 
 	snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV);
 
@@ -2342,8 +2340,8 @@ static int max98095_i2c_probe(struct i2c_client *i2c,
 	max98095->control_data = i2c;
 	max98095->pdata = i2c->dev.platform_data;
 
-	ret = snd_soc_register_codec(&i2c->dev,
-			&soc_codec_dev_max98095, &max98095_dai[0], 3);
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095,
+				     max98095_dai, ARRAY_SIZE(max98095_dai));
 	if (ret < 0)
 		kfree(max98095);
 	return ret;

sound/soc/codecs/sta32x.h

+/*
+ * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on code from:
+ *	Wolfson Microelectronics PLC.
+ *	Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+#ifndef _ASOC_STA_32X_H
+#define _ASOC_STA_32X_H
+
+/* STA326 register addresses */
+
+#define STA32X_REGISTER_COUNT	0x2d
+
+#define STA32X_CONFA	0x00
+#define STA32X_CONFB    0x01
+#define STA32X_CONFC    0x02
+#define STA32X_CONFD    0x03
+#define STA32X_CONFE    0x04
+#define STA32X_CONFF    0x05
+#define STA32X_MMUTE    0x06
+#define STA32X_MVOL     0x07
+#define STA32X_C1VOL    0x08
+#define STA32X_C2VOL    0x09
+#define STA32X_C3VOL    0x0a
+#define STA32X_AUTO1    0x0b
+#define STA32X_AUTO2    0x0c
+#define STA32X_AUTO3    0x0d
+#define STA32X_C1CFG    0x0e
+#define STA32X_C2CFG    0x0f
+#define STA32X_C3CFG    0x10
+#define STA32X_TONE     0x11
+#define STA32X_L1AR     0x12
+#define STA32X_L1ATRT   0x13
+#define STA32X_L2AR     0x14
+#define STA32X_L2ATRT   0x15
+#define STA32X_CFADDR2  0x16
+#define STA32X_B1CF1    0x17
+#define STA32X_B1CF2    0x18
+#define STA32X_B1CF3    0x19
+#define STA32X_B2CF1    0x1a
+#define STA32X_B2CF2    0x1b
+#define STA32X_B2CF3    0x1c
+#define STA32X_A1CF1    0x1d
+#define STA32X_A1CF2    0x1e
+#define STA32X_A1CF3    0x1f
+#define STA32X_A2CF1    0x20
+#define STA32X_A2CF2    0x21
+#define STA32X_A2CF3    0x22
+#define STA32X_B0CF1    0x23
+#define STA32X_B0CF2    0x24
+#define STA32X_B0CF3    0x25
+#define STA32X_CFUD     0x26
+#define STA32X_MPCC1    0x27
+#define STA32X_MPCC2    0x28
+/* Reserved 0x29 */
+/* Reserved 0x2a */
+#define STA32X_Reserved 0x2a
+#define STA32X_FDRC1    0x2b
+#define STA32X_FDRC2    0x2c
+/* Reserved 0x2d */
+
+
+/* STA326 register field definitions */
+
+/* 0x00 CONFA */
+#define STA32X_CONFA_MCS_MASK	0x03
+#define STA32X_CONFA_MCS_SHIFT	0
+#define STA32X_CONFA_IR_MASK	0x18
+#define STA32X_CONFA_IR_SHIFT	3
+#define STA32X_CONFA_TWRB	0x20
+#define STA32X_CONFA_TWAB	0x40
+#define STA32X_CONFA_FDRB	0x80
+
+/* 0x01 CONFB */
+#define STA32X_CONFB_SAI_MASK	0x0f
+#define STA32X_CONFB_SAI_SHIFT	0
+#define STA32X_CONFB_SAIFB	0x10
+#define STA32X_CONFB_DSCKE	0x20
+#define STA32X_CONFB_C1IM	0x40
+#define STA32X_CONFB_C2IM	0x80
+
+/* 0x02 CONFC */
+#define STA32X_CONFC_OM_MASK	0x03
+#define STA32X_CONFC_OM_SHIFT	0
+#define STA32X_CONFC_CSZ_MASK	0x7c
+#define STA32X_CONFC_CSZ_SHIFT	2
+
+/* 0x03 CONFD */
+#define STA32X_CONFD_HPB	0x01
+#define STA32X_CONFD_HPB_SHIFT	0
+#define STA32X_CONFD_DEMP	0x02
+#define STA32X_CONFD_DEMP_SHIFT	1
+#define STA32X_CONFD_DSPB	0x04
+#define STA32X_CONFD_DSPB_SHIFT	2
+#define STA32X_CONFD_PSL	0x08
+#define STA32X_CONFD_PSL_SHIFT	3
+#define STA32X_CONFD_BQL	0x10
+#define STA32X_CONFD_BQL_SHIFT	4
+#define STA32X_CONFD_DRC	0x20
+#define STA32X_CONFD_DRC_SHIFT	5
+#define STA32X_CONFD_ZDE	0x40
+#define STA32X_CONFD_ZDE_SHIFT	6
+#define STA32X_CONFD_MME	0x80
+#define STA32X_CONFD_MME_SHIFT	7
+
+/* 0x04 CONFE */
+#define STA32X_CONFE_MPCV	0x01
+#define STA32X_CONFE_MPCV_SHIFT	0
+#define STA32X_CONFE_MPC	0x02
+#define STA32X_CONFE_MPC_SHIFT	1
+#define STA32X_CONFE_AME	0x08
+#define STA32X_CONFE_AME_SHIFT	3
+#define STA32X_CONFE_PWMS	0x10
+#define STA32X_CONFE_PWMS_SHIFT	4
+#define STA32X_CONFE_ZCE	0x40
+#define STA32X_CONFE_ZCE_SHIFT	6
+#define STA32X_CONFE_SVE	0x80
+#define STA32X_CONFE_SVE_SHIFT	7
+
+/* 0x05 CONFF */
+#define STA32X_CONFF_OCFG_MASK	0x03
+#define STA32X_CONFF_OCFG_SHIFT	0
+#define STA32X_CONFF_IDE	0x04
+#define STA32X_CONFF_IDE_SHIFT	3
+#define STA32X_CONFF_BCLE	0x08
+#define STA32X_CONFF_ECLE	0x20
+#define STA32X_CONFF_PWDN	0x40
+#define STA32X_CONFF_EAPD	0x80
+
+/* 0x06 MMUTE */
+#define STA32X_MMUTE_MMUTE	0x01
+
+/* 0x0b AUTO1 */
+#define STA32X_AUTO1_AMEQ_MASK	0x03
+#define STA32X_AUTO1_AMEQ_SHIFT	0
+#define STA32X_AUTO1_AMV_MASK	0xc0
+#define STA32X_AUTO1_AMV_SHIFT	2
+#define STA32X_AUTO1_AMGC_MASK	0x30
+#define STA32X_AUTO1_AMGC_SHIFT	4
+#define STA32X_AUTO1_AMPS	0x80
+
+/* 0x0c AUTO2 */
+#define STA32X_AUTO2_AMAME	0x01
+#define STA32X_AUTO2_AMAM_MASK	0x0e
+#define STA32X_AUTO2_AMAM_SHIFT	1
+#define STA32X_AUTO2_XO_MASK	0xf0
+#define STA32X_AUTO2_XO_SHIFT	4
+
+/* 0x0d AUTO3 */
+#define STA32X_AUTO3_PEQ_MASK	0x1f
+#define STA32X_AUTO3_PEQ_SHIFT	0
+
+/* 0x0e 0x0f 0x10 CxCFG */
+#define STA32X_CxCFG_TCB	0x01	/* only C1 and C2 */
+#define STA32X_CxCFG_TCB_SHIFT	0
+#define STA32X_CxCFG_EQBP	0x02	/* only C1 and C2 */
+#define STA32X_CxCFG_EQBP_SHIFT	1
+#define STA32X_CxCFG_VBP	0x03
+#define STA32X_CxCFG_VBP_SHIFT	2
+#define STA32X_CxCFG_BO		0x04
+#define STA32X_CxCFG_LS_MASK	0x30
+#define STA32X_CxCFG_LS_SHIFT	4
+#define STA32X_CxCFG_OM_MASK	0xc0
+#define STA32X_CxCFG_OM_SHIFT	6
+
+/* 0x11 TONE */
+#define STA32X_TONE_BTC_SHIFT	0
+#define STA32X_TONE_TTC_SHIFT	4
+
+/* 0x12 0x13 0x14 0x15 limiter attack/release */
+#define STA32X_LxA_SHIFT	0
+#define STA32X_LxR_SHIFT	4
+
+/* 0x26 CFUD */
+#define STA32X_CFUD_W1		0x01
+#define STA32X_CFUD_WA		0x02
+#define STA32X_CFUD_R1		0x04
+#define STA32X_CFUD_RA		0x08
+
+
+/* biquad filter coefficient table offsets */
+#define STA32X_C1_BQ_BASE	0
+#define STA32X_C2_BQ_BASE	20
+#define STA32X_CH_BQ_NUM	4
+#define STA32X_BQ_NUM_COEF	5
+#define STA32X_XO_HP_BQ_BASE	40
+#define STA32X_XO_LP_BQ_BASE	45
+#define STA32X_C1_PRESCALE	50
+#define STA32X_C2_PRESCALE	51
+#define STA32X_C1_POSTSCALE	52
+#define STA32X_C2_POSTSCALE	53
+#define STA32X_C3_POSTSCALE	54
+#define STA32X_TW_POSTSCALE	55
+#define STA32X_C1_MIX1		56
+#define STA32X_C1_MIX2		57
+#define STA32X_C2_MIX1		58
+#define STA32X_C2_MIX2		59
+#define STA32X_C3_MIX1		60
+#define STA32X_C3_MIX2		61
+
+#endif /* _ASOC_STA_32X_H */

sound/soc/codecs/tlv320aic3x.c

@@ -226,11 +226,13 @@ static const char *aic3x_adc_hpf[] =
 #define RDAC_ENUM	1
 #define LHPCOM_ENUM	2
 #define RHPCOM_ENUM	3
-#define LINE1L_ENUM	4
-#define LINE1R_ENUM	5
-#define LINE2L_ENUM	6
-#define LINE2R_ENUM	7
-#define ADC_HPF_ENUM	8
+#define LINE1L_2_L_ENUM	4
+#define LINE1L_2_R_ENUM	5
+#define LINE1R_2_L_ENUM	6
+#define LINE1R_2_R_ENUM	7
+#define LINE2L_ENUM	8
+#define LINE2R_ENUM	9
+#define ADC_HPF_ENUM	10
 
 static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
@@ -238,6 +240,8 @@ static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux),
 	SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux),
 	SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+	SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+	SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
 	SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
 	SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
 	SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
@@ -490,12 +494,16 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
 };
 
 /* Left Line1 Mux */
-static const struct snd_kcontrol_new aic3x_left_line1_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_ENUM]);
+static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]);
+static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]);
 
 /* Right Line1 Mux */
-static const struct snd_kcontrol_new aic3x_right_line1_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_ENUM]);
+static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]);
+static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]);
 
 /* Left Line2 Mux */
 static const struct snd_kcontrol_new aic3x_left_line2_mux_controls =
@@ -535,9 +543,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 			   &aic3x_left_pga_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
 	SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
-			 &aic3x_left_line1_mux_controls),
+			 &aic3x_left_line1l_mux_controls),
 	SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0,
-			 &aic3x_left_line1_mux_controls),
+			 &aic3x_left_line1r_mux_controls),
 	SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_left_line2_mux_controls),
 
@@ -548,9 +556,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 			   &aic3x_right_pga_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
 	SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0,
-			 &aic3x_right_line1_mux_controls),
+			 &aic3x_right_line1l_mux_controls),
 	SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
-			 &aic3x_right_line1_mux_controls),
+			 &aic3x_right_line1r_mux_controls),
 	SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_right_line2_mux_controls),
 

sound/soc/codecs/twl6040.c

@@ -954,9 +954,9 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0);
 
 /*
  * MICGAIN volume control:
- * from -6 to 30 dB in 6 dB steps
+ * from 6 to 30 dB in 6 dB steps
  */
-static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0);
+static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0);
 
 /*
  * AFMGAIN volume control:

sound/soc/codecs/wm8782.c

+/*
+ * sound/soc/codecs/wm8782.c
+ * simple, strap-pin configured 24bit 2ch ADC
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on ad73311.c
+ * Copyright:	Analog Device Inc.
+ * Author:	Cliff Cai <cliff.cai@analog.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver wm8782_dai = {
+	.name = "wm8782",
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		/* For configurations with FSAMPEN=0 */
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			   SNDRV_PCM_FMTBIT_S20_3LE |
+			   SNDRV_PCM_FMTBIT_S24_LE,
+	},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8782;
+
+static __devinit int wm8782_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev,
+			&soc_codec_dev_wm8782, &wm8782_dai, 1);
+}
+
+static int __devexit wm8782_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver wm8782_codec_driver = {
+	.driver = {
+		.name = "wm8782",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8782_probe,
+	.remove = wm8782_remove,
+};
+
+static int __init wm8782_init(void)
+{
+	return platform_driver_register(&wm8782_codec_driver);
+}
+module_init(wm8782_init);
+
+static void __exit wm8782_exit(void)
+{
+	platform_driver_unregister(&wm8782_codec_driver);
+}
+module_exit(wm8782_exit);
+
+MODULE_DESCRIPTION("ASoC WM8782 driver");
+MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/wm8900.c

@@ -1167,6 +1167,7 @@ static int wm8900_resume(struct snd_soc_codec *codec)
 		ret = wm8900_set_fll(codec, 0, fll_in, fll_out);
 		if (ret != 0) {
 			dev_err(codec->dev, "Failed to restart FLL\n");
+			kfree(cache);
 			return ret;
 		}
 	}

sound/soc/codecs/wm8904.c

@@ -2560,6 +2560,7 @@ static __devexit int wm8904_i2c_remove(struct i2c_client *client)
 static const struct i2c_device_id wm8904_i2c_id[] = {
 	{ "wm8904", WM8904 },
 	{ "wm8912", WM8912 },
+	{ "wm8918", WM8904 },   /* Actually a subset, updates to follow */
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id);

sound/soc/codecs/wm8915.c

@@ -41,14 +41,12 @@
 #define HPOUT2L 4
 #define HPOUT2R 8
 
-#define WM8915_NUM_SUPPLIES 6
+#define WM8915_NUM_SUPPLIES 4
 static const char *wm8915_supply_names[WM8915_NUM_SUPPLIES] = {
-	"DCVDD",
 	"DBVDD",
 	"AVDD1",
 	"AVDD2",
 	"CPVDD",
-	"MICVDD",
 };
 
 struct wm8915_priv {
@@ -57,6 +55,7 @@ struct wm8915_priv {
 	int ldo1ena;
 
 	int sysclk;
+	int sysclk_src;
 
 	int fll_src;
 	int fll_fref;
@@ -76,6 +75,7 @@ struct wm8915_priv {
 	struct wm8915_pdata pdata;
 
 	int rx_rate[WM8915_AIFS];
+	int bclk_rate[WM8915_AIFS];
 
 	/* Platform dependant ReTune mobile configuration */
 	int num_retune_mobile_texts;
@@ -113,8 +113,6 @@ WM8915_REGULATOR_EVENT(0)
 WM8915_REGULATOR_EVENT(1)
 WM8915_REGULATOR_EVENT(2)
 WM8915_REGULATOR_EVENT(3)
-WM8915_REGULATOR_EVENT(4)
-WM8915_REGULATOR_EVENT(5)
 
 static const u16 wm8915_reg[WM8915_MAX_REGISTER] = {
 	[WM8915_SOFTWARE_RESET] = 0x8915,
@@ -1565,6 +1563,50 @@ static int wm8915_reset(struct snd_soc_codec *codec)
 	return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915);
 }
 
+static const int bclk_divs[] = {
+	1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
+};
+
+static void wm8915_update_bclk(struct snd_soc_codec *codec)
+{
+	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+	int aif, best, cur_val, bclk_rate, bclk_reg, i;
+
+	/* Don't bother if we're in a low frequency idle mode that
+	 * can't support audio.
+	 */
+	if (wm8915->sysclk < 64000)
+		return;
+
+	for (aif = 0; aif < WM8915_AIFS; aif++) {
+		switch (aif) {
+		case 0:
+			bclk_reg = WM8915_AIF1_BCLK;
+			break;
+		case 1:
+			bclk_reg = WM8915_AIF2_BCLK;
+			break;
+		}
+
+		bclk_rate = wm8915->bclk_rate[aif];
+
+		/* Pick a divisor for BCLK as close as we can get to ideal */
+		best = 0;
+		for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+			cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
+			if (cur_val < 0) /* BCLK table is sorted */
+				break;
+			best = i;
+		}
+		bclk_rate = wm8915->sysclk / bclk_divs[best];
+		dev_dbg(codec->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
+			bclk_divs[best], bclk_rate);
+
+		snd_soc_update_bits(codec, bclk_reg,
+				    WM8915_AIF1_BCLK_DIV_MASK, best);
+	}
+}
+
 static int wm8915_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
@@ -1717,10 +1759,6 @@ static int wm8915_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static const int bclk_divs[] = {
-	1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
-};
-
 static const int dsp_divs[] = {
 	48000, 32000, 16000, 8000
 };
@@ -1731,17 +1769,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
-	int bits, i, bclk_rate, best, cur_val;
+	int bits, i, bclk_rate;
 	int aifdata = 0;
-	int bclk = 0;
 	int lrclk = 0;
 	int dsp = 0;
-	int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift;
-
-	if (!wm8915->sysclk) {
-		dev_err(codec->dev, "SYSCLK not configured\n");
-		return -EINVAL;
-	}
+	int aifdata_reg, lrclk_reg, dsp_shift;
 
 	switch (dai->id) {
 	case 0:
@@ -1753,7 +1785,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream,
 			aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1;
 			lrclk_reg = WM8915_AIF1_TX_LRCLK_1;
 		}
-		bclk_reg = WM8915_AIF1_BCLK;
 		dsp_shift = 0;
 		break;
 	case 1:
@@ -1765,7 +1796,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream,
 			aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1;
 			lrclk_reg = WM8915_AIF2_TX_LRCLK_1;
 		}
-		bclk_reg = WM8915_AIF2_BCLK;
 		dsp_shift = WM8915_DSP2_DIV_SHIFT;
 		break;
 	default:
@@ -1779,6 +1809,9 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream,
 		return bclk_rate;
 	}
 
+	wm8915->bclk_rate[dai->id] = bclk_rate;
+	wm8915->rx_rate[dai->id] = params_rate(params);
+
 	/* Needs looking at for TDM */
 	bits = snd_pcm_format_width(params_format(params));
 	if (bits < 0)
@@ -1796,18 +1829,7 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream,
 	}
 	dsp |= i << dsp_shift;
 
-	/* Pick a divisor for BCLK as close as we can get to ideal */
-	best = 0;
-	for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
-		cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
-		if (cur_val < 0) /* BCLK table is sorted */
-			break;
-		best = i;
-	}
-	bclk_rate = wm8915->sysclk / bclk_divs[best];
-	dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
-		bclk_divs[best], bclk_rate);
-	bclk |= best;
+	wm8915_update_bclk(codec);
 
 	lrclk = bclk_rate / params_rate(params);
 	dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
@@ -1817,14 +1839,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream,
 			    WM8915_AIF1TX_WL_MASK |
 			    WM8915_AIF1TX_SLOT_LEN_MASK,
 			    aifdata);
-	snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk);
 	snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK,
 			    lrclk);
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2,
 			    WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp);
 
-	wm8915->rx_rate[dai->id] = params_rate(params);
-
 	return 0;
 }
 
@@ -1838,6 +1857,9 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
 	int src;
 	int old;
 
+	if (freq == wm8915->sysclk && clk_id == wm8915->sysclk_src)
+		return 0;
+
 	/* Disable SYSCLK while we reconfigure */
 	old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
@@ -1882,6 +1904,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
+	wm8915_update_bclk(codec);
+
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
 			    WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK,
 			    src << WM8915_SYSCLK_SRC_SHIFT | ratediv);
@@ -1889,6 +1913,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
 			    WM8915_SYSCLK_ENA, old);
 
+	wm8915->sysclk_src = clk_id;
+
 	return 0;
 }
 
@@ -2007,6 +2033,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 			  unsigned int Fref, unsigned int Fout)
 {
 	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+	struct i2c_client *i2c = to_i2c_client(codec->dev);
 	struct _fll_div fll_div;
 	unsigned long timeout;
 	int ret, reg;
@@ -2093,7 +2120,18 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	else
 		timeout = msecs_to_jiffies(2);
 
-	wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+	/* Allow substantially longer if we've actually got the IRQ */
+	if (i2c->irq)
+		timeout *= 1000;
+
+	ret = wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+
+	if (ret == 0 && i2c->irq) {
+		dev_err(codec->dev, "Timed out waiting for FLL\n");
+		ret = -ETIMEDOUT;
+	} else {
+		ret = 0;
+	}
 
 	dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
 
@@ -2101,7 +2139,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	wm8915->fll_fout = Fout;
 	wm8915->fll_src = source;
 
-	return 0;
+	return ret;
 }
 
 #ifdef CONFIG_GPIOLIB
@@ -2293,6 +2331,12 @@ static void wm8915_micd(struct snd_soc_codec *codec)
 				    SND_JACK_HEADSET | SND_JACK_BTN_0);
 		wm8915->jack_mic = true;
 		wm8915->detecting = false;
+
+		/* Increase poll rate to give better responsiveness
+		 * for buttons */
+		snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+				    WM8915_MICD_RATE_MASK,
+				    5 << WM8915_MICD_RATE_SHIFT);
 	}
 
 	/* If we detected a lower impedence during initial startup
@@ -2333,15 +2377,17 @@ static void wm8915_micd(struct snd_soc_codec *codec)
 					    SND_JACK_HEADPHONE,
 					    SND_JACK_HEADSET |
 					    SND_JACK_BTN_0);
+
+			/* Increase the detection rate a bit for
+			 * responsiveness.
+			 */
+			snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+					    WM8915_MICD_RATE_MASK,
+					    7 << WM8915_MICD_RATE_SHIFT);
+
 			wm8915->detecting = false;
 		}
 	}
-
-	/* Increase poll rate to give better responsiveness for buttons */
-	if (!wm8915->detecting)
-		snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
-				    WM8915_MICD_RATE_MASK,
-				    5 << WM8915_MICD_RATE_SHIFT);
 }
 
 static irqreturn_t wm8915_irq(int irq, void *data)
@@ -2383,6 +2429,20 @@ static irqreturn_t wm8915_irq(int irq, void *data)
 	}
 }
 
+static irqreturn_t wm8915_edge_irq(int irq, void *data)
+{
+	irqreturn_t ret = IRQ_NONE;
+	irqreturn_t val;
+
+	do {
+		val = wm8915_irq(irq, data);
+		if (val != IRQ_NONE)
+			ret = val;
+	} while (val != IRQ_NONE);
+
+	return ret;
+}
+
 static void wm8915_retune_mobile_pdata(struct snd_soc_codec *codec)
 {
 	struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
@@ -2482,8 +2542,6 @@ static int wm8915_probe(struct snd_soc_codec *codec)
 	wm8915->disable_nb[1].notifier_call = wm8915_regulator_event_1;
 	wm8915->disable_nb[2].notifier_call = wm8915_regulator_event_2;
 	wm8915->disable_nb[3].notifier_call = wm8915_regulator_event_3;
-	wm8915->disable_nb[4].notifier_call = wm8915_regulator_event_4;
-	wm8915->disable_nb[5].notifier_call = wm8915_regulator_event_5;
 
 	/* This should really be moved into the regulator core */
 	for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) {
@@ -2709,8 +2767,14 @@ static int wm8915_probe(struct snd_soc_codec *codec)
 
 		irq_flags |= IRQF_ONESHOT;
 
-		ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
-					   irq_flags, "wm8915", codec);
+		if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING))
+			ret = request_threaded_irq(i2c->irq, NULL,
+						   wm8915_edge_irq,
+						   irq_flags, "wm8915", codec);
+		else
+			ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
+						   irq_flags, "wm8915", codec);
+
 		if (ret == 0) {
 			/* Unmask the interrupt */
 			snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL,

sound/soc/codecs/wm8940.c

@@ -297,8 +297,6 @@ static int wm8940_add_widgets(struct snd_soc_codec *codec)
 	if (ret)
 		goto error_ret;
 	ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-	if (ret)
-		goto error_ret;
 
 error_ret:
 	return ret;
@@ -683,8 +681,6 @@ static int wm8940_resume(struct snd_soc_codec *codec)
 		}
 	}
 	ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	if (ret)
-		goto error_ret;
 
 error_ret:
 	return ret;
@@ -730,9 +726,6 @@ static int wm8940_probe(struct snd_soc_codec *codec)
 	if (ret)
 		return ret;
 	ret = wm8940_add_widgets(codec);
-	if (ret)
-		return ret;
-
 	return ret;
 }
 

sound/soc/codecs/wm8962.c

@@ -78,6 +78,8 @@ struct wm8962_priv {
 #ifdef CONFIG_GPIOLIB
 	struct gpio_chip gpio_chip;
 #endif
+
+	int irq;
 };
 
 /* We can't use the same notifier block for more than one supply and
@@ -1982,6 +1984,7 @@ static const unsigned int classd_tlv[] = {
 	0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
 	7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
 };
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 
 /* The VU bits for the headphones are in a different register to the mute
  * bits and only take effect on the PGA if it is actually powered.
@@ -2119,6 +2122,18 @@ SOC_SINGLE_TLV("HPMIXR MIXINR Volume", WM8962_HEADPHONE_MIXER_4,
 
 SOC_SINGLE_TLV("Speaker Boost Volume", WM8962_CLASS_D_CONTROL_2, 0, 7, 0,
 	       classd_tlv),
+
+SOC_SINGLE("EQ Switch", WM8962_EQ1, WM8962_EQ_ENA_SHIFT, 1, 0),
+SOC_DOUBLE_R_TLV("EQ1 Volume", WM8962_EQ2, WM8962_EQ22,
+		 WM8962_EQL_B1_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ2 Volume", WM8962_EQ2, WM8962_EQ22,
+		 WM8962_EQL_B2_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ3 Volume", WM8962_EQ2, WM8962_EQ22,
+		 WM8962_EQL_B3_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
+		 WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
+		 WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
 };
 
 static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2184,6 +2199,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
 			struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+	unsigned long timeout;
 	int src;
 	int fll;
 
@@ -2203,9 +2220,19 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		if (fll)
+		if (fll) {
 			snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 					    WM8962_FLL_ENA, WM8962_FLL_ENA);
+			if (wm8962->irq) {
+				timeout = msecs_to_jiffies(5);
+				timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+								      timeout);
+
+				if (timeout == 0)
+					dev_err(codec->dev,
+						"Timed out starting FLL\n");
+			}
+		}
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
@@ -2763,18 +2790,44 @@ static const int bclk_divs[] = {
 	1, -1, 2, 3, 4, -1, 6, 8, -1, 12, 16, 24, -1, 32, 32, 32
 };
 
+static const int sysclk_rates[] = {
+	64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
+};
+
 static void wm8962_configure_bclk(struct snd_soc_codec *codec)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	int dspclk, i;
 	int clocking2 = 0;
+	int clocking4 = 0;
 	int aif2 = 0;
 
-	if (!wm8962->bclk) {
-		dev_dbg(codec->dev, "No BCLK rate configured\n");
+	if (!wm8962->sysclk_rate) {
+		dev_dbg(codec->dev, "No SYSCLK configured\n");
+		return;
+	}
+
+	if (!wm8962->bclk || !wm8962->lrclk) {
+		dev_dbg(codec->dev, "No audio clocks configured\n");
 		return;
 	}
 
+	for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
+		if (sysclk_rates[i] == wm8962->sysclk_rate / wm8962->lrclk) {
+			clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
+			break;
+		}
+	}
+
+	if (i == ARRAY_SIZE(sysclk_rates)) {
+		dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
+			wm8962->sysclk_rate / wm8962->lrclk);
+		return;
+	}
+
+	snd_soc_update_bits(codec, WM8962_CLOCKING_4,
+			    WM8962_SYSCLK_RATE_MASK, clocking4);
+
 	dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
 	if (dspclk < 0) {
 		dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
@@ -2844,6 +2897,8 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 		/* VMID 2*50k */
 		snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
 				    WM8962_VMID_SEL_MASK, 0x80);
+
+		wm8962_configure_bclk(codec);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -2876,8 +2931,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 			snd_soc_update_bits(codec, WM8962_CLOCKING2,
 					    WM8962_CLKREG_OVD,
 					    WM8962_CLKREG_OVD);
-
-			wm8962_configure_bclk(codec);
 		}
 
 		/* VMID 2*250k */
@@ -2918,10 +2971,6 @@ static const struct {
 	{ 96000, 6 },
 };
 
-static const int sysclk_rates[] = {
-	64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
-};
-
 static int wm8962_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_pcm_hw_params *params,
 			    struct snd_soc_dai *dai)
@@ -2929,41 +2978,27 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	int rate = params_rate(params);
 	int i;
 	int aif0 = 0;
 	int adctl3 = 0;
-	int clocking4 = 0;
 
 	wm8962->bclk = snd_soc_params_to_bclk(params);
 	wm8962->lrclk = params_rate(params);
 
 	for (i = 0; i < ARRAY_SIZE(sr_vals); i++) {
-		if (sr_vals[i].rate == rate) {
+		if (sr_vals[i].rate == wm8962->lrclk) {
 			adctl3 |= sr_vals[i].reg;
 			break;
 		}
 	}
 	if (i == ARRAY_SIZE(sr_vals)) {
-		dev_err(codec->dev, "Unsupported rate %dHz\n", rate);
+		dev_err(codec->dev, "Unsupported rate %dHz\n", wm8962->lrclk);
 		return -EINVAL;
 	}
 
-	if (rate % 8000 == 0)
+	if (wm8962->lrclk % 8000 == 0)
 		adctl3 |= WM8962_SAMPLE_RATE_INT_MODE;
 
-	for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
-		if (sysclk_rates[i] == wm8962->sysclk_rate / rate) {
-			clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
-			break;
-		}
-	}
-	if (i == ARRAY_SIZE(sysclk_rates)) {
-		dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
-			wm8962->sysclk_rate / rate);
-		return -EINVAL;
-	}
-
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
@@ -2985,8 +3020,6 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_3,
 			    WM8962_SAMPLE_RATE_INT_MODE |
 			    WM8962_SAMPLE_RATE_MASK, adctl3);
-	snd_soc_update_bits(codec, WM8962_CLOCKING_4,
-			    WM8962_SYSCLK_RATE_MASK, clocking4);
 
 	wm8962_configure_bclk(codec);
 
@@ -3261,16 +3294,31 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 
 	dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
 
-	/* This should be a massive overestimate */
-	timeout = msecs_to_jiffies(1);
+	ret = 0;
+
+	if (fll1 & WM8962_FLL_ENA) {
+		/* This should be a massive overestimate but go even
+		 * higher if we'll error out
+		 */
+		if (wm8962->irq)
+			timeout = msecs_to_jiffies(5);
+		else
+			timeout = msecs_to_jiffies(1);
+
+		timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+						      timeout);
 
-	wait_for_completion_timeout(&wm8962->fll_lock, timeout);
+		if (timeout == 0 && wm8962->irq) {
+			dev_err(codec->dev, "FLL lock timed out");
+			ret = -ETIMEDOUT;
+		}
+	}
 
 	wm8962->fll_fref = Fref;
 	wm8962->fll_fout = Fout;
 	wm8962->fll_src = source;
 
-	return 0;
+	return ret;
 }
 
 static int wm8962_mute(struct snd_soc_dai *dai, int mute)
@@ -3731,8 +3779,6 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	int ret;
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
-	struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
-					      dev);
 	u16 *reg_cache = codec->reg_cache;
 	int i, trigger, irq_pol;
 	bool dmicclk, dmicdat;
@@ -3871,6 +3917,9 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME,
 			    WM8962_HPOUT_VU, WM8962_HPOUT_VU);
 
+	/* Stereo control for EQ */
+	snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0);
+
 	wm8962_add_widgets(codec);
 
 	/* Save boards having to disable DMIC when not in use */
@@ -3899,7 +3948,7 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	wm8962_init_beep(codec);
 	wm8962_init_gpio(codec);
 
-	if (i2c->irq) {
+	if (wm8962->irq) {
 		if (pdata && pdata->irq_active_low) {
 			trigger = IRQF_TRIGGER_LOW;
 			irq_pol = WM8962_IRQ_POL;
@@ -3911,12 +3960,13 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 		snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL,
 				    WM8962_IRQ_POL, irq_pol);
 
-		ret = request_threaded_irq(i2c->irq, NULL, wm8962_irq,
+		ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq,
 					   trigger | IRQF_ONESHOT,
 					   "wm8962", codec);
 		if (ret != 0) {
 			dev_err(codec->dev, "Failed to request IRQ %d: %d\n",
-				i2c->irq, ret);
+				wm8962->irq, ret);
+			wm8962->irq = 0;
 			/* Non-fatal */
 		} else {
 			/* Enable some IRQs by default */
@@ -3941,12 +3991,10 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 static int wm8962_remove(struct snd_soc_codec *codec)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
-					      dev);
 	int i;
 
-	if (i2c->irq)
-		free_irq(i2c->irq, codec);
+	if (wm8962->irq)
+		free_irq(wm8962->irq, codec);
 
 	cancel_delayed_work_sync(&wm8962->mic_work);
 
@@ -3986,6 +4034,8 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, wm8962);
 
+	wm8962->irq = i2c->irq;
+
 	ret = snd_soc_register_codec(&i2c->dev,
 				     &soc_codec_dev_wm8962, &wm8962_dai, 1);
 	if (ret < 0)

sound/soc/codecs/wm8993.c

@@ -876,7 +876,7 @@ SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0,
 		   left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
 SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0,
 		   right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-
+SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
 };
 
 static const struct snd_soc_dapm_route routes[] = {
@@ -1434,6 +1434,7 @@ static int wm8993_probe(struct snd_soc_codec *codec)
 
 	wm8993->hubs_data.hp_startup_mode = 1;
 	wm8993->hubs_data.dcs_codes = -2;
+	wm8993->hubs_data.series_startup = 1;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
 	if (ret != 0) {

sound/soc/codecs/wm8994.h

@@ -11,6 +11,7 @@
 
 #include <sound/soc.h>
 #include <linux/firmware.h>
+#include <linux/completion.h>
 
 #include "wm_hubs.h"
 
@@ -79,6 +80,8 @@ struct wm8994_priv {
 	int mclk[2];
 	int aifclk[2];
 	struct wm8994_fll_config fll[2], fll_suspend[2];
+	struct completion fll_locked[2];
+	bool fll_locked_irq;
 
 	int dac_rates[2];
 	int lrclk_shared[2];

sound/soc/codecs/wm9081.c

@@ -727,7 +727,7 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
 SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
 
 SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
 
 SND_SOC_DAPM_OUTPUT("LINEOUT"),
 SND_SOC_DAPM_OUTPUT("SPKN"),

sound/soc/codecs/wm_hubs.h

@@ -14,6 +14,9 @@
 #ifndef _WM_HUBS_H
 #define _WM_HUBS_H
 
+#include <linux/completion.h>
+#include <linux/interrupt.h>
+
 struct snd_soc_codec;
 
 extern const unsigned int wm_hubs_spkmix_tlv[];
@@ -23,9 +26,14 @@ struct wm_hubs_data {
 	int dcs_codes;
 	int dcs_readback_mode;
 	int hp_startup_mode;
+	int series_startup;
+	int no_series_update;
 
 	bool class_w;
 	u16 class_w_dcs;
+
+	bool dcs_done_irq;
+	struct completion dcs_done;
 };
 
 extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
@@ -36,4 +44,6 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *,
 					 int jd_scthr, int jd_thr,
 					 int micbias1_lvl, int micbias2_lvl);
 
+extern irqreturn_t wm_hubs_dcs_done(int irq, void *data);
+
 #endif

sound/soc/ep93xx/ep93xx-pcm.c

@@ -266,9 +266,11 @@ static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 ep93xx_pcm_dmamask = 0xffffffff;
 
-static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-			  struct snd_pcm *pcm)
+static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	int ret = 0;
 
 	if (!card->dev->dma_mask)

sound/soc/fsl/fsl_dma.c

@@ -294,9 +294,11 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
  * Regardless of where the memory is actually allocated, since the device can
  * technically DMA to any 36-bit address, we do need to set the DMA mask to 36.
  */
-static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
-	struct snd_pcm *pcm)
+static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
 	static u64 fsl_dma_dmamask = DMA_BIT_MASK(36);
 	int ret;
 
@@ -939,7 +941,7 @@ static int __devinit fsl_soc_dma_probe(struct platform_device *pdev)
 
 	iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
 	if (iprop)
-		dma->ssi_fifo_depth = *iprop;
+		dma->ssi_fifo_depth = be32_to_cpup(iprop);
 	else
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		dma->ssi_fifo_depth = 8;

sound/soc/fsl/fsl_ssi.c

@@ -678,7 +678,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 		kfree(ssi_private);
 		return ret;
 	}
-	ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start);
+	ssi_private->ssi = of_iomap(np, 0);
+	if (!ssi_private->ssi) {
+		dev_err(&pdev->dev, "could not map device resources\n");
+		kfree(ssi_private);
+		return -ENOMEM;
+	}
 	ssi_private->ssi_phys = res.start;
 	ssi_private->irq = irq_of_parse_and_map(np, 0);
 
@@ -691,7 +696,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 	/* Determine the FIFO depth. */
 	iprop = of_get_property(np, "fsl,fifo-depth", NULL);
 	if (iprop)
-		ssi_private->fifo_depth = *iprop;
+		ssi_private->fifo_depth = be32_to_cpup(iprop);
 	else
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		ssi_private->fifo_depth = 8;