Merge branch 'fix/hda' into topic/hda

125 files changed, 2650 insertions, 1163 deletions

diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c
index f84f3e50578..fa8ab2815a9 100644
--- a/sound/aoa/soundbus/core.c
+++ b/sound/aoa/soundbus/core.c
@@ -176,7 +176,7 @@ int soundbus_add_one(struct soundbus_dev *dev)
 		return -EINVAL;
 	}
 
-	snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount);
+	dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount);
 	dev->ofdev.dev.bus = &soundbus_bus_type;
 	return of_device_register(&dev->ofdev);
 }
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
index e6beb92c693..b4590df0746 100644
--- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
@@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	struct i2sbus_dev *dev;
 	struct device_node *child = NULL, *sound = NULL;
 	struct resource *r;
-	int i, layout = 0, rlen;
+	int i, layout = 0, rlen, ok = force;
 	static const char *rnames[] = { "i2sbus: %s (control)",
 					"i2sbus: %s (tx)",
 					"i2sbus: %s (rx)" };
@@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 			layout = *layout_id;
 			snprintf(dev->sound.modalias, 32,
 				 "sound-layout-%d", layout);
-			force = 1;
+			ok = 1;
 		}
 	}
 	/* for the time being, until we can handle non-layout-id
@@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	 * When there are two i2s busses and only one has a layout-id,
 	 * then this depends on the order, but that isn't important
 	 * either as the second one in that case is just a modem. */
-	if (!force) {
+	if (!ok) {
 		kfree(dev);
 		return -ENODEV;
 	}
diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h
index 622cd37a011..a0f223c13f6 100644
--- a/sound/aoa/soundbus/soundbus.h
+++ b/sound/aoa/soundbus/soundbus.h
@@ -8,7 +8,7 @@
 #ifndef __SOUNDBUS_H
 #define __SOUNDBUS_H
 
-#include <asm/of_device.h>
+#include <linux/of_device.h>
 #include <sound/pcm.h>
 #include <linux/list.h>
 
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 99026dfb81e..34c1d94f921 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -50,7 +50,7 @@ unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	mutex_lock(&car_mutex);
 
 	/* set up primary or secondary codec space */
-	if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS)
+	if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS)
 		reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
 	else
 		reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
@@ -90,7 +90,7 @@ void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	mutex_lock(&car_mutex);
 
 	/* set up primary or secondary codec space */
-	if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS)
+	if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS)
 		reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
 	else
 		reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
@@ -200,7 +200,7 @@ static inline void pxa_ac97_cold_pxa3xx(void)
 bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 {
 #ifdef CONFIG_PXA25x
-	if (cpu_is_pxa21x() || cpu_is_pxa25x())
+	if (cpu_is_pxa25x())
 		pxa_ac97_warm_pxa25x();
 	else
 #endif
@@ -230,7 +230,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
 bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 {
 #ifdef CONFIG_PXA25x
-	if (cpu_is_pxa21x() || cpu_is_pxa25x())
+	if (cpu_is_pxa25x())
 		pxa_ac97_cold_pxa25x();
 	else
 #endif
@@ -301,7 +301,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend);
 
 int pxa2xx_ac97_hw_resume(void)
 {
-	if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) {
+	if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
 		pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
 		pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
 		pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
@@ -325,7 +325,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
 	if (ret < 0)
 		goto err;
 
-	if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) {
+	if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
 		pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
 		pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
 		pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index cba71d86754..c2635beb4c8 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -44,7 +44,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = {
 	.name			= "AC97 PCM out",
 	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMRTXPCDR,
+	.drcmr			= &DRCMR(12),
 	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
@@ -52,7 +52,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = {
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = {
 	.name			= "AC97 PCM in",
 	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMRRXPCDR,
+	.drcmr			= &DRCMR(11),
 	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 1c93eb77cb9..75a0d746fb6 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 		goto out;
 
 	ret = -ENOMEM;
-	rtd = kmalloc(sizeof(*rtd), GFP_KERNEL);
+	rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
 	if (!rtd)
 		goto out;
 	rtd->dma_desc_array =
diff --git a/sound/core/control.c b/sound/core/control.c
index 6d71f9a7ccb..636b3b52ef8 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -113,7 +113,6 @@ static int snd_ctl_release(struct inode *inode, struct file *file)
 	unsigned int idx;
 
 	ctl = file->private_data;
-	fasync_helper(-1, file, 0, &ctl->fasync);
 	file->private_data = NULL;
 	card = ctl->card;
 	write_lock_irqsave(&card->ctl_files_rwlock, flags);
@@ -225,8 +224,13 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol,
 	kctl.id.iface = ncontrol->iface;
 	kctl.id.device = ncontrol->device;
 	kctl.id.subdevice = ncontrol->subdevice;
-	if (ncontrol->name)
+	if (ncontrol->name) {
 		strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name));
+		if (strcmp(ncontrol->name, kctl.id.name) != 0)
+			snd_printk(KERN_WARNING
+				   "Control name '%s' truncated to '%s'\n",
+				   ncontrol->name, kctl.id.name);
+	}
 	kctl.id.index = ncontrol->index;
 	kctl.count = ncontrol->count ? ncontrol->count : 1;
 	access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
diff --git a/sound/core/init.c b/sound/core/init.c
index 8af467df924..b47ff8b44be 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -264,8 +264,11 @@ static int snd_disconnect_release(struct inode *inode, struct file *file)
 	}
 	spin_unlock(&shutdown_lock);
 
-	if (likely(df))
+	if (likely(df)) {
+		if ((file->f_flags & FASYNC) && df->disconnected_f_op->fasync)
+			df->disconnected_f_op->fasync(-1, file, 0);
 		return df->disconnected_f_op->release(inode, file);
+	}
 
 	panic("%s(%p, %p) failed!", __func__, inode, file);
 }
@@ -549,9 +552,9 @@ int snd_card_register(struct snd_card *card)
 		return -EINVAL;
 #ifndef CONFIG_SYSFS_DEPRECATED
 	if (!card->card_dev) {
-		card->card_dev = device_create_drvdata(sound_class, card->dev,
-						       MKDEV(0, 0), NULL,
-						       "card%i", card->number);
+		card->card_dev = device_create(sound_class, card->dev,
+					       MKDEV(0, 0), NULL,
+					       "card%i", card->number);
 		if (IS_ERR(card->card_dev))
 			card->card_dev = NULL;
 	}
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 438445f77d6..284432f427f 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -151,6 +151,9 @@ EXPORT_SYMBOL(snd_jack_set_parent);
  */
 void snd_jack_report(struct snd_jack *jack, int status)
 {
+	if (!jack)
+		return;
+
 	if (jack->type & SND_JACK_HEADPHONE)
 		input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT,
 				    status & SND_JACK_HEADPHONE);
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index a7b46ec72f3..1b3534d6768 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -33,9 +33,6 @@
 #include <linux/moduleparam.h>
 #include <linux/mutex.h>
 #include <sound/memalloc.h>
-#ifdef CONFIG_SBUS
-#include <asm/sbus.h>
-#endif
 
 
 MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>");
@@ -162,39 +159,6 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
 }
 #endif /* CONFIG_HAS_DMA */
 
-#ifdef CONFIG_SBUS
-
-static void *snd_malloc_sbus_pages(struct device *dev, size_t size,
-				   dma_addr_t *dma_addr)
-{
-	struct sbus_dev *sdev = (struct sbus_dev *)dev;
-	int pg;
-	void *res;
-
-	if (WARN_ON(!dma_addr))
-		return NULL;
-	pg = get_order(size);
-	res = sbus_alloc_consistent(sdev, PAGE_SIZE * (1 << pg), dma_addr);
-	if (res != NULL)
-		inc_snd_pages(pg);
-	return res;
-}
-
-static void snd_free_sbus_pages(struct device *dev, size_t size,
-				void *ptr, dma_addr_t dma_addr)
-{
-	struct sbus_dev *sdev = (struct sbus_dev *)dev;
-	int pg;
-
-	if (ptr == NULL)
-		return;
-	pg = get_order(size);
-	dec_snd_pages(pg);
-	sbus_free_consistent(sdev, PAGE_SIZE * (1 << pg), ptr, dma_addr);
-}
-
-#endif /* CONFIG_SBUS */
-
 /*
  *
  *  ALSA generic memory management
@@ -231,11 +195,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 		dmab->area = snd_malloc_pages(size, (unsigned long)device);
 		dmab->addr = 0;
 		break;
-#ifdef CONFIG_SBUS
-	case SNDRV_DMA_TYPE_SBUS:
-		dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr);
-		break;
-#endif
 #ifdef CONFIG_HAS_DMA
 	case SNDRV_DMA_TYPE_DEV:
 		dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
@@ -306,11 +265,6 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
 	case SNDRV_DMA_TYPE_CONTINUOUS:
 		snd_free_pages(dmab->area, dmab->bytes);
 		break;
-#ifdef CONFIG_SBUS
-	case SNDRV_DMA_TYPE_SBUS:
-		snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
-		break;
-#endif
 #ifdef CONFIG_HAS_DMA
 	case SNDRV_DMA_TYPE_DEV:
 		snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
@@ -419,7 +373,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset)
 	long pages = snd_allocated_pages >> (PAGE_SHIFT-12);
 	struct snd_mem_list *mem;
 	int devno;
-	static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "SBUS" };
+	static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" };
 
 	mutex_lock(&list_mutex);
 	seq_printf(seq, "pages  : %li bytes (%li pages per %likB)\n",
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 1af62b8b86c..e17836680f4 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -2283,7 +2283,7 @@ static int snd_pcm_oss_open_file(struct file *file,
 	int idx, err;
 	struct snd_pcm_oss_file *pcm_oss_file;
 	struct snd_pcm_substream *substream;
-	unsigned int f_mode = file->f_mode;
+	fmode_t f_mode = file->f_mode;
 
 	if (rpcm_oss_file)
 		*rpcm_oss_file = NULL;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 6ea5cfb8399..921691080f3 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -908,12 +908,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond,
 EXPORT_SYMBOL(snd_pcm_hw_rule_add);
 
 /**
- * snd_pcm_hw_constraint_mask
+ * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the mask
  * @mask: the bitmap mask
  *
- * Apply the constraint of the given bitmap mask to a mask parameter.
+ * Apply the constraint of the given bitmap mask to a 32-bit mask parameter.
  */
 int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
 			       u_int32_t mask)
@@ -928,12 +928,12 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param
 }
 
 /**
- * snd_pcm_hw_constraint_mask64
+ * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the mask
  * @mask: the 64bit bitmap mask
  *
- * Apply the constraint of the given bitmap mask to a mask parameter.
+ * Apply the constraint of the given bitmap mask to a 64-bit mask parameter.
  */
 int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
 				 u_int64_t mask)
@@ -949,7 +949,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par
 }
 
 /**
- * snd_pcm_hw_constraint_integer
+ * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the integer constraint
  *
@@ -964,7 +964,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa
 EXPORT_SYMBOL(snd_pcm_hw_constraint_integer);
 
 /**
- * snd_pcm_hw_constraint_minmax
+ * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the range
  * @min: the minimal value
@@ -995,7 +995,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params,
 
 
 /**
- * snd_pcm_hw_constraint_list
+ * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the list constraint
@@ -1031,7 +1031,7 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_ratnums
+ * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the ratnums constraint
@@ -1064,7 +1064,7 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_ratdens
+ * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the ratdens constraint
@@ -1095,7 +1095,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_msbits
+ * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @width: sample bits width
@@ -1123,7 +1123,7 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_step
+ * snd_pcm_hw_constraint_step - add a hw constraint step rule
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the step constraint
@@ -1154,7 +1154,7 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm
 }		
 
 /**
- * snd_pcm_hw_constraint_pow2
+ * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the power-of-2 constraint
@@ -1202,13 +1202,13 @@ void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params)
 EXPORT_SYMBOL(_snd_pcm_hw_params_any);
 
 /**
- * snd_pcm_hw_param_value
+ * snd_pcm_hw_param_value - return @params field @var value
  * @params: the hw_params instance
  * @var: parameter to retrieve
- * @dir: pointer to the direction (-1,0,1) or NULL
+ * @dir: pointer to the direction (-1,0,1) or %NULL
  *
- * Return the value for field PAR if it's fixed in configuration space 
- *  defined by PARAMS. Return -EINVAL otherwise
+ * Return the value for field @var if it's fixed in configuration space
+ * defined by @params. Return -%EINVAL otherwise.
  */
 int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params,
 			   snd_pcm_hw_param_t var, int *dir)
@@ -1271,13 +1271,13 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params,
 
 
 /**
- * snd_pcm_hw_param_first
+ * snd_pcm_hw_param_first - refine config space and return minimum value
  * @pcm: PCM instance
  * @params: the hw_params instance
  * @var: parameter to retrieve
- * @dir: pointer to the direction (-1,0,1) or NULL
+ * @dir: pointer to the direction (-1,0,1) or %NULL
  *
- * Inside configuration space defined by PARAMS remove from PAR all 
+ * Inside configuration space defined by @params remove from @var all
  * values > minimum. Reduce configuration space accordingly.
  * Return the minimum.
  */
@@ -1317,13 +1317,13 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params,
 
 
 /**
- * snd_pcm_hw_param_last
+ * snd_pcm_hw_param_last - refine config space and return maximum value
  * @pcm: PCM instance
  * @params: the hw_params instance
  * @var: parameter to retrieve
- * @dir: pointer to the direction (-1,0,1) or NULL
+ * @dir: pointer to the direction (-1,0,1) or %NULL
  *
- * Inside configuration space defined by PARAMS remove from PAR all 
+ * Inside configuration space defined by @params remove from @var all
  * values < maximum. Reduce configuration space accordingly.
  * Return the maximum.
  */
@@ -1345,11 +1345,11 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm,
 EXPORT_SYMBOL(snd_pcm_hw_param_last);
 
 /**
- * snd_pcm_hw_param_choose
+ * snd_pcm_hw_param_choose - choose a configuration defined by @params
  * @pcm: PCM instance
  * @params: the hw_params instance
  *
- * Choose one configuration from configuration space defined by PARAMS
+ * Choose one configuration from configuration space defined by @params.
  * The configuration chosen is that obtained fixing in this order:
  * first access, first format, first subformat, min channels,
  * min rate, min period time, max buffer size, min tick time
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 89b7f549beb..ea2bf82c937 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -319,6 +319,7 @@ EXPORT_SYMBOL(snd_pcm_format_physical_width);
 /**
  * snd_pcm_format_size - return the byte size of samples on the given format
  * @format: the format to check
+ * @samples: sampling rate
  *
  * Returns the byte size of the given samples for the format, or a
  * negative error code if unknown format.
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index e61e12506de..a789efc9df3 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -875,10 +875,8 @@ static struct action_ops snd_pcm_action_start = {
 };
 
 /**
- * snd_pcm_start
+ * snd_pcm_start - start all linked streams
  * @substream: the PCM substream instance
- *
- * Start all linked streams.
  */
 int snd_pcm_start(struct snd_pcm_substream *substream)
 {
@@ -926,12 +924,11 @@ static struct action_ops snd_pcm_action_stop = {
 };
 
 /**
- * snd_pcm_stop
+ * snd_pcm_stop - try to stop all running streams in the substream group
  * @substream: the PCM substream instance
  * @state: PCM state after stopping the stream
  *
- * Try to stop all running streams in the substream group.
- * The state of each stream is changed to the given value after that unconditionally.
+ * The state of each stream is then changed to the given state unconditionally.
  */
 int snd_pcm_stop(struct snd_pcm_substream *substream, int state)
 {
@@ -941,11 +938,10 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, int state)
 EXPORT_SYMBOL(snd_pcm_stop);
 
 /**
- * snd_pcm_drain_done
+ * snd_pcm_drain_done - stop the DMA only when the given stream is playback
  * @substream: the PCM substream
  *
- * Stop the DMA only when the given stream is playback.
- * The state is changed to SETUP.
+ * After stopping, the state is changed to SETUP.
  * Unlike snd_pcm_stop(), this affects only the given stream.
  */
 int snd_pcm_drain_done(struct snd_pcm_substream *substream)
@@ -1065,10 +1061,9 @@ static struct action_ops snd_pcm_action_suspend = {
 };
 
 /**
- * snd_pcm_suspend
+ * snd_pcm_suspend - trigger SUSPEND to all linked streams
  * @substream: the PCM substream
  *
- * Trigger SUSPEND to all linked streams.
  * After this call, all streams are changed to SUSPENDED state.
  */
 int snd_pcm_suspend(struct snd_pcm_substream *substream)
@@ -1088,10 +1083,9 @@ int snd_pcm_suspend(struct snd_pcm_substream *substream)
 EXPORT_SYMBOL(snd_pcm_suspend);
 
 /**
- * snd_pcm_suspend_all
+ * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm
  * @pcm: the PCM instance
  *
- * Trigger SUSPEND to all substreams in the given pcm.
  * After this call, all streams are changed to SUSPENDED state.
  */
 int snd_pcm_suspend_all(struct snd_pcm *pcm)
@@ -1313,11 +1307,9 @@ static struct action_ops snd_pcm_action_prepare = {
 };
 
 /**
- * snd_pcm_prepare
+ * snd_pcm_prepare - prepare the PCM substream to be triggerable
  * @substream: the PCM substream instance
  * @file: file to refer f_flags
- *
- * Prepare the PCM substream to be triggerable.
  */
 static int snd_pcm_prepare(struct snd_pcm_substream *substream,
 			   struct file *file)
@@ -2177,7 +2169,6 @@ static int snd_pcm_release(struct inode *inode, struct file *file)
 	if (snd_BUG_ON(!substream))
 		return -ENXIO;
 	pcm = substream->pcm;
-	fasync_helper(-1, file, 0, &substream->runtime->fasync);
 	mutex_lock(&pcm->open_mutex);
 	snd_pcm_release_substream(substream);
 	kfree(pcm_file);
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index c4995c9f573..39672f68ce5 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -148,6 +148,8 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream)
 
 static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up)
 {
+	if (!substream->opened)
+		return;
 	if (up) {
 		tasklet_hi_schedule(&substream->runtime->tasklet);
 	} else {
@@ -158,6 +160,8 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs
 
 static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
 {
+	if (!substream->opened)
+		return;
 	substream->ops->trigger(substream, up);
 	if (!up && substream->runtime->event)
 		tasklet_kill(&substream->runtime->tasklet);
@@ -857,6 +861,8 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
 	int result = 0, count1;
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
 
+	if (!substream->opened)
+		return -EBADFD;
 	if (runtime->buffer == NULL) {
 		snd_printd("snd_rawmidi_receive: input is not active!!!\n");
 		return -EINVAL;
@@ -1126,6 +1132,8 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count)
 int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
 			 unsigned char *buffer, int count)
 {
+	if (!substream->opened)
+		return -EBADFD;
 	count = snd_rawmidi_transmit_peek(substream, buffer, count);
 	if (count < 0)
 		return count;
diff --git a/sound/core/sound.c b/sound/core/sound.c
index c0685e2f0af..44a69bb8d4f 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -274,9 +274,8 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev,
 		return minor;
 	}
 	snd_minors[minor] = preg;
-	preg->dev = device_create_drvdata(sound_class, device,
-					  MKDEV(major, minor),
-					  private_data, "%s", name);
+	preg->dev = device_create(sound_class, device, MKDEV(major, minor),
+				  private_data, "%s", name);
 	if (IS_ERR(preg->dev)) {
 		snd_minors[minor] = NULL;
 		mutex_unlock(&sound_mutex);
diff --git a/sound/core/timer.c b/sound/core/timer.c
index e582face89d..c584408c9f1 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1263,7 +1263,6 @@ static int snd_timer_user_release(struct inode *inode, struct file *file)
 	if (file->private_data) {
 		tu = file->private_data;
 		file->private_data = NULL;
-		fasync_helper(-1, file, 0, &tu->fasync);
 		if (tu->timeri)
 			snd_timer_close(tu->timeri);
 		kfree(tu->queue);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index e5e749f3e0e..73be7e14a60 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -51,7 +51,7 @@ static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime)
 	if (err < 0)
 		return err;
 	err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX);
-	if (err) < 0)
+	if (err < 0)
 		return err;
 	return 0;
 }
diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
index ecdbeb6d360..7783843ca9a 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
 	/* get irq */
 	irq = platform_get_irq(pfdev, 0);
 	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
-			pfdev->dev.bus_id, (void *)ml403_ac97cr)) {
+			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
 			   irq);
@@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
 		   ml403_ac97cr->irq);
 	irq = platform_get_irq(pfdev, 1);
 	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
-			pfdev->dev.bus_id, (void *)ml403_ac97cr)) {
+			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
 			   irq);
diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c
index cd9b83e7f7d..0444cdeb4be 100644
--- a/sound/drivers/pcsp/pcsp_input.c
+++ b/sound/drivers/pcsp/pcsp_input.c
@@ -24,13 +24,13 @@ static void pcspkr_do_sound(unsigned int count)
 	spin_lock_irqsave(&i8253_lock, flags);
 
 	if (count) {
-		/* enable counter 2 */
-		outb_p(inb_p(0x61) | 3, 0x61);
 		/* set command for counter 2, 2 byte write */
 		outb_p(0xB6, 0x43);
 		/* select desired HZ */
 		outb_p(count & 0xff, 0x42);
 		outb((count >> 8) & 0xff, 0x42);
+		/* enable counter 2 */
+		outb_p(inb_p(0x61) | 3, 0x61);
 	} else {
 		/* disable counter 2 */
 		outb(inb_p(0x61) & 0xFC, 0x61);
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index e341f3f83b6..1f42e406311 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -34,7 +34,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 		chip->thalf = 0;
 		if (!atomic_read(&chip->timer_active))
 			return HRTIMER_NORESTART;
-		hrtimer_forward(&chip->timer, chip->timer.expires,
+		hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
 				ktime_set(0, chip->ns_rem));
 		return HRTIMER_RESTART;
 	}
@@ -118,7 +118,8 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 	chip->ns_rem = PCSP_PERIOD_NS();
 	ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem);
 	chip->ns_rem -= ns;
-	hrtimer_forward(&chip->timer, chip->timer.expires, ktime_set(0, ns));
+	hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
+							ktime_set(0, ns));
 	return HRTIMER_RESTART;
 
 exit_nr_unlock2:
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index 83e90057270..c13a178383b 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -87,8 +87,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea)
 static int snd_tea575x_ioctl(struct inode *inode, struct file *file,
 			     unsigned int cmd, unsigned long data)
 {
-	struct video_device *dev = video_devdata(file);
-	struct snd_tea575x *tea = video_get_drvdata(dev);
+	struct snd_tea575x *tea = video_drvdata(file);
 	void __user *arg = (void __user *)data;
 	
 	switch(cmd) {
@@ -175,6 +174,21 @@ static void snd_tea575x_release(struct video_device *vfd)
 {
 }
 
+static int snd_tea575x_exclusive_open(struct inode *inode, struct file *file)
+{
+	struct snd_tea575x *tea = video_drvdata(file);
+
+	return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0;
+}
+
+static int snd_tea575x_exclusive_release(struct inode *inode, struct file *file)
+{
+	struct snd_tea575x *tea = video_drvdata(file);
+
+	clear_bit(0, &tea->in_use);
+	return 0;
+}
+
 /*
  * initialize all the tea575x chips
  */
@@ -193,9 +207,10 @@ void snd_tea575x_init(struct snd_tea575x *tea)
 	tea->vd.release = snd_tea575x_release;
 	video_set_drvdata(&tea->vd, tea);
 	tea->vd.fops = &tea->fops;
+	tea->in_use = 0;
 	tea->fops.owner = tea->card->module;
-	tea->fops.open = video_exclusive_open;
-	tea->fops.release = video_exclusive_release;
+	tea->fops.open = snd_tea575x_exclusive_open;
+	tea->fops.release = snd_tea575x_exclusive_release;
 	tea->fops.ioctl = snd_tea575x_ioctl;
 	if (video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->dev_nr - 1) < 0) {
 		snd_printk(KERN_ERR "unable to register tea575x tuner\n");
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 660beb41f76..ce0aa044e27 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -211,7 +211,7 @@ config SND_GUSCLASSIC
 
 config SND_GUSEXTREME
 	tristate "Gravis UltraSound Extreme"
-	select SND_HWDEP
+	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_PCM
 	help
diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c
index b68d20edc20..223a6c03881 100644
--- a/sound/isa/ad1848/ad1848.c
+++ b/sound/isa/ad1848/ad1848.c
@@ -70,15 +70,15 @@ static int __devinit snd_ad1848_match(struct device *dev, unsigned int n)
 		return 0;
 
 	if (port[n] == SNDRV_AUTO_PORT) {
-		snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
+		dev_err(dev, "please specify port\n");
 		return 0;
 	}
 	if (irq[n] == SNDRV_AUTO_IRQ) {
-		snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id);
+		dev_err(dev, "please specify irq\n");
 		return 0;	
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
-		snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id);
+		dev_err(dev, "please specify dma1\n");
 		return 0;
 	}
 	return 1;
diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c
index efa8c80d05b..374b7177e11 100644
--- a/sound/isa/adlib.c
+++ b/sound/isa/adlib.c
@@ -36,7 +36,7 @@ static int __devinit snd_adlib_match(struct device *dev, unsigned int n)
 		return 0;
 
 	if (port[n] == SNDRV_AUTO_PORT) {
-		snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
+		dev_err(dev, "please specify port\n");
 		return 0;
 	}
 	return 1;
@@ -55,13 +55,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
 
 	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
 	if (!card) {
-		snd_printk(KERN_ERR "%s: could not create card\n", dev->bus_id);
+		dev_err(dev, "could not create card\n");
 		return -EINVAL;
 	}
 
 	card->private_data = request_region(port[n], 4, CRD_NAME);
 	if (!card->private_data) {
-		snd_printk(KERN_ERR "%s: could not grab ports\n", dev->bus_id);
+		dev_err(dev, "could not grab ports\n");
 		error = -EBUSY;
 		goto out;
 	}
@@ -73,13 +73,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
 
 	error = snd_opl3_create(card, port[n], port[n] + 2, OPL3_HW_AUTO, 1, &opl3);
 	if (error < 0) {
-		snd_printk(KERN_ERR "%s: could not create OPL\n", dev->bus_id);
+		dev_err(dev, "could not create OPL\n");
 		goto out;
 	}
 
 	error = snd_opl3_hwdep_new(opl3, 0, 0, NULL);
 	if (error < 0) {
-		snd_printk(KERN_ERR "%s: could not create FM\n", dev->bus_id);
+		dev_err(dev, "could not create FM\n");
 		goto out;
 	}
 
@@ -87,7 +87,7 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
 
 	error = snd_card_register(card);
 	if (error < 0) {
-		snd_printk(KERN_ERR "%s: could not register card\n", dev->bus_id);
+		dev_err(dev, "could not register card\n");
 		goto out;
 	}
 
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
index ddd289120aa..f019d449e2d 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -74,15 +74,15 @@ static int __devinit snd_cs4231_match(struct device *dev, unsigned int n)
 		return 0;
 
 	if (port[n] == SNDRV_AUTO_PORT) {
-		snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
+		dev_err(dev, "please specify port\n");
 		return 0;
 	}
 	if (irq[n] == SNDRV_AUTO_IRQ) {
-		snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id);
+		dev_err(dev, "please specify irq\n");
 		return 0;
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
-		snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id);
+		dev_err(dev, "please specify dma1\n");
 		return 0;
 	}
 	return 1;
@@ -133,7 +133,7 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n)
 					mpu_port[n], 0, mpu_irq[n],
 					mpu_irq[n] >= 0 ? IRQF_DISABLED : 0,
 					NULL) < 0)
-			printk(KERN_WARNING "%s: MPU401 not detected\n", dev->bus_id);
+			dev_warn(dev, "MPU401 not detected\n");
 	}
 
 	snd_card_set_dev(card, dev);
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 91f9c15d3e3..019c9401663 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -488,19 +488,19 @@ static int __devinit snd_cs423x_isa_match(struct device *pdev,
 		return 0;
 
 	if (port[dev] == SNDRV_AUTO_PORT) {
-		snd_printk(KERN_ERR "%s: please specify port\n", pdev->bus_id);
+		dev_err(pdev, "please specify port\n");
 		return 0;
 	}
 	if (cport[dev] == SNDRV_AUTO_PORT) {
-		snd_printk(KERN_ERR "%s: please specify cport\n", pdev->bus_id);
+		dev_err(pdev, "please specify cport\n");
 		return 0;
 	}
 	if (irq[dev] == SNDRV_AUTO_IRQ) {
-		snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id);
+		dev_err(pdev, "please specify irq\n");
 		return 0;
 	}
 	if (dma1[dev] == SNDRV_AUTO_DMA) {
-		snd_printk(KERN_ERR "%s: please specify dma1\n", pdev->bus_id);
+		dev_err(pdev, "please specify dma1\n");
 		return 0;
 	}
 	return 1;
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index f88639ea64b..b46377139cf 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -88,16 +88,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card,
 	if (irq[n] == SNDRV_AUTO_IRQ) {
 		irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (irq[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free IRQ\n",
-				dev->bus_id);
+			dev_err(dev, "unable to find a free IRQ\n");
 			return -EBUSY;
 		}
 	}
 	if (dma8[n] == SNDRV_AUTO_DMA) {
 		dma8[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma8[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free DMA\n",
-				dev->bus_id);
+			dev_err(dev, "unable to find a free DMA\n");
 			return -EBUSY;
 		}
 	}
@@ -147,8 +145,7 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n)
 
 	if (snd_opl3_create(card, chip->port, chip->port + 2,
 			OPL3_HW_OPL3, 0, &opl3) < 0)
-		printk(KERN_WARNING "%s: opl3 not detected at 0x%lx\n",
-			dev->bus_id, chip->port);
+		dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port);
 	else {
 		error =	snd_opl3_hwdep_new(opl3, 0, 1, NULL);
 		if (error < 0)
diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c
index 8f914b37bf8..426532a4d73 100644
--- a/sound/isa/gus/gusclassic.c
+++ b/sound/isa/gus/gusclassic.c
@@ -90,24 +90,21 @@ static int __devinit snd_gusclassic_create(struct snd_card *card,
 	if (irq[n] == SNDRV_AUTO_IRQ) {
 		irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (irq[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free IRQ\n",
-				dev->bus_id);
+			dev_err(dev, "unable to find a free IRQ\n");
 			return -EBUSY;
 		}
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
 		dma1[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma1[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free DMA1\n",
-				dev->bus_id);
+			dev_err(dev, "unable to find a free DMA1\n");
 			return -EBUSY;
 		}
 	}
 	if (dma2[n] == SNDRV_AUTO_DMA) {
 		dma2[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma2[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free DMA2\n",
-				dev->bus_id);
+			dev_err(dev, "unable to find a free DMA2\n");
 			return -EBUSY;
 		}
 	}
@@ -174,8 +171,8 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n)
 
 	error = -ENODEV;
 	if (gus->max_flag || gus->ess_flag) {
-		snd_printk(KERN_ERR "%s: GUS Classic or ACE soundcard was "
-			"not detected at 0x%lx\n", dev->bus_id, gus->gf1.port);
+		dev_err(dev, "GUS Classic or ACE soundcard was "
+			"not detected at 0x%lx\n", gus->gf1.port);
 		goto out;
 	}
 
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index da13185eb0a..7ad4c3b41a8 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -106,16 +106,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card,
 	if (irq[n] == SNDRV_AUTO_IRQ) {
 		irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (irq[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free IRQ "
-				"for ES1688\n", dev->bus_id);
+			dev_err(dev, "unable to find a free IRQ for ES1688\n");
 			return -EBUSY;
 		}
 	}
 	if (dma8[n] == SNDRV_AUTO_DMA) {
 		dma8[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma8[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free DMA "
-				"for ES1688\n", dev->bus_id);
+			dev_err(dev, "unable to find a free DMA for ES1688\n");
 			return -EBUSY;
 		}
 	}
@@ -143,16 +141,14 @@ static int __devinit snd_gusextreme_gus_card_create(struct snd_card *card,
 	if (gf1_irq[n] == SNDRV_AUTO_IRQ) {
 		gf1_irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (gf1_irq[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free IRQ "
-				"for GF1\n", dev->bus_id);
+			dev_err(dev, "unable to find a free IRQ for GF1\n");
 			return -EBUSY;
 		}
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
 		dma1[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma1[n] < 0) {
-			snd_printk(KERN_ERR "%s: unable to find a free DMA "
-				"for GF1\n", dev->bus_id);
+			dev_err(dev, "unable to find a free DMA for GF1\n");
 			return -EBUSY;
 		}
 	}
@@ -278,8 +274,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
 
 	error = -ENODEV;
 	if (!gus->ess_flag) {
-		snd_printk(KERN_ERR "%s: GUS Extreme soundcard was not "
-			"detected at 0x%lx\n", dev->bus_id, gus->gf1.port);
+		dev_err(dev, "GUS Extreme soundcard was not "
+			"detected at 0x%lx\n", gus->gf1.port);
 		goto out;
 	}
 	gus->codec_flag = 1;
@@ -310,8 +306,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
 
 	if (snd_opl3_create(card, es1688->port, es1688->port + 2,
 			OPL3_HW_OPL3, 0, &opl3) < 0)
-		printk(KERN_ERR "%s: opl3 not detected at 0x%lx\n",
-			dev->bus_id, es1688->port);
+		dev_warn(dev, "opl3 not detected at 0x%lx\n", es1688->port);
 	else {
 		error = snd_opl3_hwdep_new(opl3, 0, 2, NULL);
 		if (error < 0)
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index 336a3427790..667eccc676a 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -85,11 +85,11 @@ static int __devinit snd_sb8_match(struct device *pdev, unsigned int dev)
 	if (!enable[dev])
 		return 0;
 	if (irq[dev] == SNDRV_AUTO_IRQ) {
-		snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id);
+		dev_err(pdev, "please specify irq\n");
 		return 0;
 	}
 	if (dma8[dev] == SNDRV_AUTO_DMA) {
-		snd_printk(KERN_ERR "%s: please specify dma8\n", pdev->bus_id);
+		dev_err(pdev, "please specify dma8\n");
 		return 0;
 	}
 	return 1;
diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c
index b63839e8f9b..456a1b4d783 100644
--- a/sound/oss/ac97_codec.c
+++ b/sound/oss/ac97_codec.c
@@ -30,7 +30,7 @@
  **************************************************************************
  *
  * History
- * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
  *	Removed non existant WM9700
  *	Added support for WM9705, WM9708, WM9709, WM9710, WM9711
  *	WM9712 and WM9717
diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c
index 23018a7c063..81e1f443d09 100644
--- a/sound/oss/au1550_ac97.c
+++ b/sound/oss/au1550_ac97.c
@@ -93,7 +93,7 @@ static struct au1550_state {
 	spinlock_t      lock;
 	struct mutex open_mutex;
 	struct mutex sem;
-	mode_t          open_mode;
+	fmode_t          open_mode;
 	wait_queue_head_t open_wait;
 
 	struct dmabuf {
diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h
index d978b009656..1308d8d3418 100644
--- a/sound/oss/dmasound/dmasound.h
+++ b/sound/oss/dmasound/dmasound.h
@@ -129,7 +129,7 @@ typedef struct {
     int (*mixer_ioctl)(u_int, u_long);	/* optional */
     int (*write_sq_setup)(void);	/* optional */
     int (*read_sq_setup)(void);		/* optional */
-    int (*sq_open)(mode_t);		/* optional */
+    int (*sq_open)(fmode_t);		/* optional */
     int (*state_info)(char *, size_t);	/* optional */
     void (*abort_read)(void);		/* optional */
     int min_dsp_speed;
@@ -235,7 +235,7 @@ struct sound_queue {
      */
     int active;
     wait_queue_head_t action_queue, open_queue, sync_queue;
-    int open_mode;
+    int non_blocking;
     int busy, syncing, xruns, died;
 };
 
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 285239d64b8..4d45bd63718 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -143,7 +143,7 @@ static int AtaMixerIoctl(u_int cmd, u_long arg);
 static int TTMixerIoctl(u_int cmd, u_long arg);
 static int FalconMixerIoctl(u_int cmd, u_long arg);
 static int AtaWriteSqSetup(void);
-static int AtaSqOpen(mode_t mode);
+static int AtaSqOpen(fmode_t mode);
 static int TTStateInfo(char *buffer, size_t space);
 static int FalconStateInfo(char *buffer, size_t space);
 
@@ -1461,7 +1461,7 @@ static int AtaWriteSqSetup(void)
 	return 0 ;
 }
 
-static int AtaSqOpen(mode_t mode)
+static int AtaSqOpen(fmode_t mode)
 {
 	write_sq_ignore_int = 1;
 	return 0 ;
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index 95fc5c68175..793b7f47843 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -212,7 +212,7 @@ static int irq_installed;
 #endif /* MODULE */
 
 /* control over who can modify resources shared between play/record */
-static mode_t shared_resource_owner;
+static fmode_t shared_resource_owner;
 static int shared_resources_initialised;
 
     /*
@@ -603,7 +603,7 @@ static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft,
 	while (uLeft) {
 		while (write_sq.count >= write_sq.max_active) {
 			sq_play();
-			if (write_sq.open_mode & O_NONBLOCK)
+			if (write_sq.non_blocking)
 				return uWritten > 0 ? uWritten : -EAGAIN;
 			SLEEP(write_sq.action_queue);
 			if (signal_pending(current))
@@ -668,7 +668,7 @@ static inline void sq_init_waitqueue(struct sound_queue *sq)
 
 #if 0 /* blocking open() */
 static inline void sq_wake_up(struct sound_queue *sq, struct file *file,
-			      mode_t mode)
+			      fmode_t mode)
 {
 	if (file->f_mode & mode) {
 		sq->busy = 0; /* CHECK: IS THIS OK??? */
@@ -677,7 +677,7 @@ static inline void sq_wake_up(struct sound_queue *sq, struct file *file,
 }
 #endif
 
-static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode,
+static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode,
 		    int numbufs, int bufsize)
 {
 	int rc = 0;
@@ -718,7 +718,7 @@ static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode,
 			return rc;
 		}
 
-		sq->open_mode = file->f_mode;
+		sq->non_blocking = file->f_flags & O_NONBLOCK;
 	}
 	return rc;
 }
@@ -891,10 +891,10 @@ static int sq_release(struct inode *inode, struct file *file)
    is the owner - if we have problems.
 */
 
-static int shared_resources_are_mine(mode_t md)
+static int shared_resources_are_mine(fmode_t md)
 {
 	if (shared_resource_owner)
-		return (shared_resource_owner & md ) ;
+		return (shared_resource_owner & md) != 0;
 	else {
 		shared_resource_owner = md ;
 		return 1 ;
diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c
index eb9bc365530..c180598f171 100644
--- a/sound/oss/kahlua.c
+++ b/sound/oss/kahlua.c
@@ -1,7 +1,7 @@
 /*
  *	Initialisation code for Cyrix/NatSemi VSA1 softaudio
  *
- *	(C) Copyright 2003 Red Hat Inc <alan@redhat.com>
+ *	(C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
  *
  * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
  * The older version (VSA1) provides fairly good soundblaster emulation
diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h
index 61b3955481c..c8be47ec2b7 100644
--- a/sound/oss/msnd.h
+++ b/sound/oss/msnd.h
@@ -211,7 +211,7 @@ typedef struct multisound_dev {
 
 	/* State variables */
 	enum { msndClassic, msndPinnacle } type;
-	mode_t mode;
+	fmode_t mode;
 	unsigned long flags;
 #define F_RESETTING			0
 #define F_HAVEDIGITAL			1
diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c
index b493660deb3..e5d42399491 100644
--- a/sound/oss/sh_dac_audio.c
+++ b/sound/oss/sh_dac_audio.c
@@ -26,7 +26,7 @@
 #include <asm/cpu/dac.h>
 #include <asm/cpu/timer.h>
 #include <asm/machvec.h>
-#include <asm/hp6xx.h>
+#include <mach/hp6xx.h>
 #include <asm/hd64461.h>
 
 #define MODNAME "sh_dac_audio"
diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h
index 1a00a321061..55271fbe7f4 100644
--- a/sound/oss/sound_config.h
+++ b/sound/oss/sound_config.h
@@ -110,24 +110,16 @@ struct channel_info {
 #define OPEN_WRITE	PCM_ENABLE_OUTPUT
 #define OPEN_READWRITE	(OPEN_READ|OPEN_WRITE)
 
-#if OPEN_READ == FMODE_READ && OPEN_WRITE == FMODE_WRITE
-
-static inline int translate_mode(struct file *file)
-{
-	return file->f_mode;
-}
-
-#else
-
 static inline int translate_mode(struct file *file)
 {
-	return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) |
-		((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0);
+	if (OPEN_READ == (__force int)FMODE_READ &&
+	    OPEN_WRITE == (__force int)FMODE_WRITE)
+		return (__force int)(file->f_mode & (FMODE_READ | FMODE_WRITE));
+	else
+		return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) |
+			((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0);
 }
 
-#endif
-
-
 #include "sound_calls.h"
 #include "dev_table.h"
 
diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c
index 7d89c081a08..61aaedae6b7 100644
--- a/sound/oss/soundcard.c
+++ b/sound/oss/soundcard.c
@@ -560,19 +560,18 @@ static int __init oss_init(void)
 	sound_dmap_flag = (dmabuf > 0 ? 1 : 0);
 
 	for (i = 0; i < ARRAY_SIZE(dev_list); i++) {
-		device_create_drvdata(sound_class, NULL,
-				      MKDEV(SOUND_MAJOR, dev_list[i].minor),
-				      NULL, "%s", dev_list[i].name);
+		device_create(sound_class, NULL,
+			      MKDEV(SOUND_MAJOR, dev_list[i].minor), NULL,
+			      "%s", dev_list[i].name);
 
 		if (!dev_list[i].num)
 			continue;
 
 		for (j = 1; j < *dev_list[i].num; j++)
-			device_create_drvdata(sound_class, NULL,
-					      MKDEV(SOUND_MAJOR,
-						    dev_list[i].minor + (j*0x10)),
-					      NULL,
-					      "%s%d", dev_list[i].name, j);
+			device_create(sound_class, NULL,
+				      MKDEV(SOUND_MAJOR,
+					    dev_list[i].minor + (j*0x10)),
+				      NULL, "%s%d", dev_list[i].name, j);
 	}
 
 	if (sound_nblocks >= 1024)
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 044453a4ee5..41562ecde5b 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -295,7 +295,7 @@ struct cs4297a_state {
 	struct mutex open_mutex;
 	struct mutex open_sem_adc;
 	struct mutex open_sem_dac;
-	mode_t open_mode;
+	fmode_t open_mode;
 	wait_queue_head_t open_wait;
 	wait_queue_head_t open_wait_adc;
 	wait_queue_head_t open_wait_dac;
diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c
index dcbb3f739e6..78b8acc7c3b 100644
--- a/sound/oss/vwsnd.c
+++ b/sound/oss/vwsnd.c
@@ -1509,7 +1509,7 @@ typedef struct vwsnd_dev {
 	struct mutex open_mutex;
 	struct mutex io_mutex;
 	struct mutex mix_mutex;
-	mode_t		open_mode;
+	fmode_t		open_mode;
 	wait_queue_head_t open_wait;
 
 	lithium_t	lith;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 6704acbca8c..bd510eceff1 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1927,9 +1927,9 @@ static int snd_ac97_dev_register(struct snd_device *device)
 	ac97->dev.bus = &ac97_bus_type;
 	ac97->dev.parent = ac97->bus->card->dev;
 	ac97->dev.release = ac97_device_release;
-	snprintf(ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
-		 ac97->bus->card->number, ac97->num,
-		 snd_ac97_get_short_name(ac97));
+	dev_set_name(&ac97->dev, "%d-%d:%s",
+		     ac97->bus->card->number, ac97->num,
+		     snd_ac97_get_short_name(ac97));
 	if ((err = device_register(&ac97->dev)) < 0) {
 		snd_printk(KERN_ERR "Can't register ac97 bus\n");
 		ac97->dev.bus = NULL;
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 6ce3cbe98a6..6e831aff1bd 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
 }
 
 /*
- * May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
  *  removed broken wolfson00 patch.
  *  added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
  */
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 92f3a976ef2..a7f38e63303 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card,
 		goto free_and_ret;
 
 	chip->bar = pci_resource_start(pci, 0);
-	chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0));
+	chip->iobase = pci_ioremap_bar(pci, 0);
 	if (chip->iobase == NULL) {
 		printk(KERN_ERR PFX "unable to reserve region.\n");
 		err = -EBUSY;
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 085a52b8c80..226fe8237d3 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
 		return err;
 	}
 	chip->addr = pci_resource_start(pci, 0);
-	chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0));
+	chip->remap_addr = pci_ioremap_bar(pci, 0);
 	if (chip->remap_addr == NULL) {
 		snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
 		snd_atiixp_free(chip);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 2f106306c7f..0e6e5cc1c50 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
 		return err;
 	}
 	chip->addr = pci_resource_start(pci, 0);
-	chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0));
+	chip->remap_addr = pci_ioremap_bar(pci, 0);
 	if (chip->remap_addr == NULL) {
 		snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
 		snd_atiixp_free(chip);
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 68368e49007..a36d4d1fd41 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
 	if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0)
 		goto regions_out;
 
-	chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
-	                             pci_resource_len(pci, 0));
+	chip->mmio = pci_ioremap_bar(pci, 0);
 	if (!chip->mmio) {
 		printk(KERN_ERR "MMIO area remap failed.\n");
 		err = -ENOMEM;
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 3aa8d973540..1aa1c040254 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card,
 		pci_disable_device(pci);
 		return err;
 	}
-	chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
-				     pci_resource_len(pci, 0));
+	chip->mmio = pci_ioremap_bar(pci, 0);
 	if (!chip->mmio) {
 		snd_printk(KERN_ERR "cannot remap io memory\n");
 		err = -ENOMEM;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index a7d89662acf..88fbf285d2b 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -759,7 +759,6 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
 			       SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
 			       SPCS_GENERATIONSTATUS | 0x00001200 |
 			       0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT );
-	}
 #endif
 
 	return 0;
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index ef9308f7c45..192e7842e18 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card,
 	chip->ba0_addr = pci_resource_start(pci, 0);
 	chip->ba1_addr = pci_resource_start(pci, 1);
 
-	chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0));
-	chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1));
+	chip->ba0 = pci_ioremap_bar(pci, 0);
+	chip->ba1 = pci_ioremap_bar(pci, 1);
 	if (!chip->ba0 || !chip->ba1) {
 		snd_cs4281_free(chip);
 		return -ENOMEM;
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index 7ff8b68e997..6dea5b5cc77 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -2,7 +2,7 @@
  * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio
  *
  * 	(C) Copyright 2007 Ash Willis <ashwillis@programmer.net>
- *	(C) Copyright 2003 Red Hat Inc <alan@redhat.com>
+ *	(C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
  *
  * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did
  * mess with it a bit. The chip seems to have to have trouble with full duplex
@@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card,
 	}
 	chip->pci_base = pci_resource_start(pci, 0);
 
-	mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0));
+	mem = pci_ioremap_bar(pci, 0);
 	if (mem == NULL) {
 		kfree(chip);
 		pci_disable_device(pci);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 2f283ea6ad9..de5ee8f097f 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1464,6 +1464,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
+	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102,
 	 .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", 
@@ -1473,6 +1474,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
+	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102,
 	 .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", 
@@ -1482,6 +1484,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
+	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	/* Audigy 2 */
 	/* Tested by James@superbug.co.uk 3rd July 2005 */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f2337e4eddd..a26ae8c4cf7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2182,7 +2182,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	chip->addr = pci_resource_start(pci, 0);
-	chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0));
+	chip->remap_addr = pci_ioremap_bar(pci, 0);
 	if (chip->remap_addr == NULL) {
 		snd_printk(KERN_ERR SFX "ioremap error\n");
 		err = -ENXIO;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4c851fd5556..71c3ccfcde1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -69,6 +69,7 @@ enum {
 };
 
 enum {
+	STAC_92HD73XX_NO_JD, /* no jack-detection */
 	STAC_92HD73XX_REF,
 	STAC_DELL_M6_AMIC,
 	STAC_DELL_M6_DMIC,
@@ -127,6 +128,7 @@ enum {
 };
 
 enum {
+	STAC_D965_REF_NO_JD, /* no jack-detection */
 	STAC_D965_REF,
 	STAC_D965_3ST,
 	STAC_D965_5ST,
@@ -1664,6 +1666,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 };
 
 static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
+	[STAC_92HD73XX_NO_JD] = "no-jd",
 	[STAC_92HD73XX_REF] = "ref",
 	[STAC_DELL_M6_AMIC] = "dell-m6-amic",
 	[STAC_DELL_M6_DMIC] = "dell-m6-dmic",
@@ -1693,6 +1696,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"unknown Dell", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x029f,
 				"Dell Studio 1537", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0,
+				"Dell Studio 17", STAC_DELL_M6_DMIC),
 	{} /* terminator */
 };
 
@@ -2080,6 +2085,7 @@ static unsigned int dell_3st_pin_configs[14] = {
 };
 
 static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
+	[STAC_D965_REF_NO_JD] = ref927x_pin_configs,
 	[STAC_D965_REF]  = ref927x_pin_configs,
 	[STAC_D965_3ST]  = d965_3st_pin_configs,
 	[STAC_D965_5ST]  = d965_5st_pin_configs,
@@ -2088,6 +2094,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 };
 
 static const char *stac927x_models[STAC_927X_MODELS] = {
+	[STAC_D965_REF_NO_JD]	= "ref-no-jd",
 	[STAC_D965_REF]		= "ref",
 	[STAC_D965_3ST]		= "3stack",
 	[STAC_D965_5ST]		= "5stack",
@@ -4545,14 +4552,17 @@ again:
 
 	switch (spec->multiout.num_dacs) {
 	case 0x3: /* 6 Channel */
+		spec->multiout.hp_nid = 0x17;
 		spec->mixer = stac92hd73xx_6ch_mixer;
 		spec->init = stac92hd73xx_6ch_core_init;
 		break;
 	case 0x4: /* 8 Channel */
+		spec->multiout.hp_nid = 0x18;
 		spec->mixer = stac92hd73xx_8ch_mixer;
 		spec->init = stac92hd73xx_8ch_core_init;
 		break;
 	case 0x5: /* 10 Channel */
+		spec->multiout.hp_nid = 0x19;
 		spec->mixer = stac92hd73xx_10ch_mixer;
 		spec->init = stac92hd73xx_10ch_core_init;
 	};
@@ -4642,6 +4652,9 @@ again:
 		return err;
 	}
 
+	if (spec->board_config == STAC_92HD73XX_NO_JD)
+		spec->hp_detect = 0;
+
 	codec->patch_ops = stac92xx_patch_ops;
 
 	codec->proc_widget_hook = stac92hd7x_proc_hook;
@@ -5138,6 +5151,10 @@ static int patch_stac927x(struct hda_codec *codec)
 	 */
 	codec->bus->needs_damn_long_delay = 1;
 
+	/* no jack detecion for ref-no-jd model */
+	if (spec->board_config == STAC_D965_REF_NO_JD)
+		spec->hp_detect = 0;
+
 	return 0;
 }
 
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 5b442383fcd..58d7cda03de 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2688,12 +2688,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 		return err;
 	}
 
-	if (ice_has_con_ac97(ice))
+	if (ice_has_con_ac97(ice)) {
 		err = snd_ice1712_pcm(ice, pcm_dev++, NULL);
 		if (err < 0) {
 			snd_card_free(card);
 			return err;
 		}
+	}
 
 	err = snd_ice1712_ac97_mixer(ice);
 	if (err < 0) {
@@ -2715,12 +2716,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 		}
 	}
 
-	if (ice_has_con_ac97(ice))
+	if (ice_has_con_ac97(ice)) {
 		err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL);
 		if (err < 0) {
 			snd_card_free(card);
 			return err;
 		}
+	}
 
 	if (!c->no_mpu401) {
 		err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index c88d1eace1c..19d3391e229 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2702,6 +2702,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000),
 	SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100),
 	SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000),
+	SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),
 	SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000),
 	{ }	/* terminator */
 };
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 2d0dce649a6..ae7601f353a 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci,
 	}
 	for (i = 0; i < 2; i++) {
 		mgr->mem[i].phys = pci_resource_start(pci, i);
-		mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys,
-						   pci_resource_len(pci, i));
+		mgr->mem[i].virt = pci_ioremap_bar(pci, i);
 		if (!mgr->mem[i].virt) {
 		        printk(KERN_ERR "unable to remap resource 0x%lx\n",
 			       mgr->mem[i].phys);
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 0e06c6c9fcc..73de6e989b3 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1229,8 +1229,11 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 		return -ENOMEM;
 	}
 
-	if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST))
+	if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) {
+		kfree(mgr);
+		pci_disable_device(pci);
 		return -ENODEV;
+	}
 	card_name = pcxhr_board_params[pci_id->driver_data].board_name;
 	mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips;
 	mgr->capture_chips  = pcxhr_board_params[pci_id->driver_data].capture_chips;
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index d723543bead..736246f98ac 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -4548,11 +4548,20 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 {
 	struct hdsp *hdsp = (struct hdsp *)hw->private_data;	
 	void __user *argp = (void __user *)arg;
+	int err;
 
 	switch (cmd) {
 	case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: {
 		struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg;
 
+		err = hdsp_check_for_iobox(hdsp);
+		if (err < 0)
+			return err;
+
+		err = hdsp_check_for_firmware(hdsp, 1);
+		if (err < 0)
+			return err;
+
 		if (!(hdsp->state & HDSP_FirmwareLoaded)) {
 			snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n");
 			return -EINVAL;
@@ -4572,10 +4581,14 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 		unsigned long flags;
 		int i;
 		
-		if (!(hdsp->state & HDSP_FirmwareLoaded)) {
-			snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n");	
-			return -EINVAL;
-		}
+		err = hdsp_check_for_iobox(hdsp);
+		if (err < 0)
+			return err;
+
+		err = hdsp_check_for_firmware(hdsp, 1);
+		if (err < 0)
+			return err;
+
 		spin_lock_irqsave(&hdsp->lock, flags);
 		info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp);
 		info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp);
@@ -5045,6 +5058,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 		/* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
 		ssleep(2);
 
+		err = hdsp_check_for_iobox(hdsp);
+		if (err < 0)
+			return err;
+
 		if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
 #ifdef HDSP_FW_LOADER
 			if ((err = hdsp_request_fw_loader(hdsp)) < 0)
@@ -5057,7 +5074,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 				/* init is complete, we return */
 				return 0;
 #endif
-			/* no iobox connected, we defer initialization */
+			/* we defer initialization */
 			snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n");
 			if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
 				return err;
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 20d0e328288..8f9e3859c37 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -666,6 +666,7 @@ static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card)
 	card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
 	card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM;
 	card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL;
+	memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8);
 
 	ret = snd_ps3_change_avsetting(card);
 
@@ -685,6 +686,7 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream)
 {
 	struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
 	struct snd_ps3_avsetting_info avs;
+	int ret;
 
 	avs = card->avs;
 
@@ -729,19 +731,92 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream)
 		return 1;
 	}
 
-	if ((card->avs.avs_audio_width != avs.avs_audio_width) ||
-	    (card->avs.avs_audio_rate != avs.avs_audio_rate)) {
-		card->avs = avs;
-		snd_ps3_change_avsetting(card);
+	memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8);
 
+	if (memcmp(&card->avs, &avs, sizeof(avs))) {
 		pr_debug("%s: after freq=%d width=%d\n", __func__,
 			 card->avs.avs_audio_rate, card->avs.avs_audio_width);
 
-		return 0;
+		card->avs = avs;
+		snd_ps3_change_avsetting(card);
+		ret = 0;
 	} else
+		ret = 1;
+
+	/* check CS non-audio bit and mute accordingly */
+	if (avs.avs_cs_info[0] & 0x02)
+		ps3av_audio_mute_analog(1); /* mute if non-audio */
+	else
+		ps3av_audio_mute_analog(0);
+
+	return ret;
+}
+
+/*
+ * SPDIF status bits controls
+ */
+static int snd_ps3_spdif_mask_info(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+	return 0;
+}
+
+/* FIXME: ps3av_set_audio_mode() assumes only consumer mode */
+static int snd_ps3_spdif_cmask_get(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	memset(ucontrol->value.iec958.status, 0xff, 8);
+	return 0;
+}
+
+static int snd_ps3_spdif_pmask_get(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	return 0;
+}
+
+static int snd_ps3_spdif_default_get(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	memcpy(ucontrol->value.iec958.status, ps3av_mode_cs_info, 8);
+	return 0;
+}
+
+static int snd_ps3_spdif_default_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	if (memcmp(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8)) {
+		memcpy(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8);
 		return 1;
+	}
+	return 0;
 }
 
+static struct snd_kcontrol_new spdif_ctls[] = {
+	{
+		.access = SNDRV_CTL_ELEM_ACCESS_READ,
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+		.info = snd_ps3_spdif_mask_info,
+		.get = snd_ps3_spdif_cmask_get,
+	},
+	{
+		.access = SNDRV_CTL_ELEM_ACCESS_READ,
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+		.info = snd_ps3_spdif_mask_info,
+		.get = snd_ps3_spdif_pmask_get,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+		.info = snd_ps3_spdif_mask_info,
+		.get = snd_ps3_spdif_default_get,
+		.put = snd_ps3_spdif_default_put,
+	},
+};
 
 
 static int snd_ps3_map_mmio(void)
@@ -842,7 +917,7 @@ static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start)
 
 static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
 {
-	int ret;
+	int i, ret;
 	u64 lpar_addr, lpar_size;
 
 	BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1));
@@ -903,6 +978,15 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
 	strcpy(the_card.card->driver, "PS3");
 	strcpy(the_card.card->shortname, "PS3");
 	strcpy(the_card.card->longname, "PS3 sound");
+
+	/* create control elements */
+	for (i = 0; i < ARRAY_SIZE(spdif_ctls); i++) {
+		ret = snd_ctl_add(the_card.card,
+				  snd_ctl_new1(&spdif_ctls[i], &the_card));
+		if (ret < 0)
+			goto clean_card;
+	}
+
 	/* create PCM devices instance */
 	/* NOTE:this driver works assuming pcm:substream = 1:1 */
 	ret = snd_pcm_new(the_card.card,
diff --git a/sound/ppc/snd_ps3.h b/sound/ppc/snd_ps3.h
index 4b7e6fbbe50..326fb29e82d 100644
--- a/sound/ppc/snd_ps3.h
+++ b/sound/ppc/snd_ps3.h
@@ -51,6 +51,7 @@ struct snd_ps3_avsetting_info {
 	uint32_t avs_audio_width;
 	uint32_t avs_audio_format; /* fixed */
 	uint32_t avs_audio_source; /* fixed */
+	unsigned char avs_cs_info[8];
 };
 /*
  * PS3 audio 'card' instance
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
index 98a2d5826a8..b1966e4dfcd 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -304,7 +304,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
 
 
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 	/* speaker connected to SPKOUT */
 	{"Ext Spk", NULL, "SPKOUTP"},
 	{"Ext Spk", NULL, "SPKOUTN"},
@@ -312,9 +312,6 @@ static const char *intercon[][3] = {
 	{"Mic Bias", NULL, "Int Mic"},
 	{"MICN", NULL, "Mic Bias"},
 	{"MICP", NULL, "Mic Bias"},
-
-	/* Terminator */
-	{NULL, NULL, NULL},
 };
 
 
@@ -334,11 +331,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec)
 	/*
 	 * Setup audio path interconnects
 	 */
-	for (i = 0; intercon[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec,
-					   intercon[i][0],
-					   intercon[i][1], intercon[i][2]);
-	}
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
 
 
 	/* always connected pins */
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
index 905186502e0..85a883299c2 100644
--- a/sound/soc/at91/Kconfig
+++ b/sound/soc/at91/Kconfig
@@ -8,20 +8,3 @@ config SND_AT91_SOC
 
 config SND_AT91_SOC_SSC
 	tristate
-
-config SND_AT91_SOC_ETI_B1_WM8731
-	tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards"
-	depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
-	select SND_AT91_SOC_SSC
-	select SND_SOC_WM8731
-	help
-	  Say Y if you want to add support for SoC audio on WM8731-based
-	  Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
-
-config SND_AT91_SOC_ETI_SLAVE
-	bool "Run codec in slave Mode on Endrelia boards"
-	depends on SND_AT91_SOC_ETI_B1_WM8731
-	default n
-	help
-	  Say Y if you want to run with the AT91 SSC generating the BCLK
-	  and LRC signals on Endrelia boards.
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
index f23da17cc32..b817f11df28 100644
--- a/sound/soc/at91/Makefile
+++ b/sound/soc/at91/Makefile
@@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o
 
 obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
 obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
-
-# AT91 Machine Support
-snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
-
-obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index a5b1a79ebff..1b61cc46126 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -5,7 +5,7 @@
  *         Endrelia Technologies Inc.
  *
  * Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
deleted file mode 100644
index 684781e4088..00000000000
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ /dev/null
@@ -1,349 +0,0 @@
-/*
- * eti_b1_wm8731  --  SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
- *
- * Author:	Frank Mandarino <fmandarino@endrelia.com>
- *		Endrelia Technologies Inc.
- * Created:	Mar 29, 2006
- *
- * Based on corgi.c by:
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
- *          Richard Purdie <richard@openedhand.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/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/clk.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-
-#include "../codecs/wm8731.h"
-#include "at91-pcm.h"
-#include "at91-ssc.h"
-
-#if 0
-#define	DBG(x...)	printk(KERN_INFO "eti_b1_wm8731: " x)
-#else
-#define	DBG(x...)
-#endif
-
-static struct clk *pck1_clk;
-static struct clk *pllb_clk;
-
-
-static int eti_b1_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	int ret;
-
-	/* cpu clock is the AT91 master clock sent to the SSC */
-	ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
-		60000000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* codec system clock is supplied by PCK1, set to 12MHz */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
-		12000000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* Start PCK1 clock. */
-	clk_enable(pck1_clk);
-	DBG("pck1 started\n");
-
-	return 0;
-}
-
-static void eti_b1_shutdown(struct snd_pcm_substream *substream)
-{
-	/* Stop PCK1 clock. */
-	clk_disable(pck1_clk);
-	DBG("pck1 stopped\n");
-}
-
-static int eti_b1_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	int ret;
-
-#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
-	unsigned int rate;
-	int cmr_div, period;
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/*
-	 * The SSC clock dividers depend on the sample rate.  The CMR.DIV
-	 * field divides the system master clock MCK to drive the SSC TK
-	 * signal which provides the codec BCLK.  The TCMR.PERIOD and
-	 * RCMR.PERIOD fields further divide the BCLK signal to drive
-	 * the SSC TF and RF signals which provide the codec DACLRC and
-	 * ADCLRC clocks.
-	 *
-	 * The dividers were determined through trial and error, where a
-	 * CMR.DIV value is chosen such that the resulting BCLK value is
-	 * divisible, or almost divisible, by (2 * sample rate), and then
-	 * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
-	 */
-	rate = params_rate(params);
-
-	switch (rate) {
-	case 8000:
-		cmr_div = 25;	/* BCLK = 60MHz/(2*25) = 1.2MHz */
-		period = 74;	/* LRC = BCLK/(2*(74+1)) = 8000Hz */
-		break;
-	case 32000:
-		cmr_div = 7;	/* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
-		period = 66;	/* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
-		break;
-	case 48000:
-		cmr_div = 13;	/* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
-		period = 23;	/* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
-		break;
-	default:
-		printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
-		return -EINVAL;
-	}
-
-	/* set the MCK divider for BCLK */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
-	if (ret < 0)
-		return ret;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		/* set the BCLK divider for DACLRC */
-		ret = snd_soc_dai_set_clkdiv(cpu_dai,
-						AT91SSC_TCMR_PERIOD, period);
-	} else {
-		/* set the BCLK divider for ADCLRC */
-		ret = snd_soc_dai_set_clkdiv(cpu_dai,
-						AT91SSC_RCMR_PERIOD, period);
-	}
-	if (ret < 0)
-		return ret;
-
-#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
-	/*
-	 * Codec in Master Mode.
-	 */
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
-
-	return 0;
-}
-
-static struct snd_soc_ops eti_b1_ops = {
-	.startup = eti_b1_startup,
-	.hw_params = eti_b1_hw_params,
-	.shutdown = eti_b1_shutdown,
-};
-
-
-static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Int Mic", NULL),
-	SND_SOC_DAPM_SPK("Ext Spk", NULL),
-};
-
-static const struct snd_soc_dapm_route intercon[] = {
-
-	/* speaker connected to LHPOUT */
-	{"Ext Spk", NULL, "LHPOUT"},
-
-	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
-	{"MICIN", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Int Mic"},
-};
-
-/*
- * Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
- */
-static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
-{
-	DBG("eti_b1_wm8731_init() called\n");
-
-	/* Add specific widgets */
-	snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets,
-				  ARRAY_SIZE(eti_b1_dapm_widgets));
-
-	/* Set up specific audio path interconnects */
-	snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
-
-	/* not connected */
-	snd_soc_dapm_disable_pin(codec, "RLINEIN");
-	snd_soc_dapm_disable_pin(codec, "LLINEIN");
-
-	/* always connected */
-	snd_soc_dapm_enable_pin(codec, "Int Mic");
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
-
-	snd_soc_dapm_sync(codec);
-
-	return 0;
-}
-
-static struct snd_soc_dai_link eti_b1_dai = {
-	.name = "WM8731",
-	.stream_name = "WM8731 PCM",
-	.cpu_dai = &at91_ssc_dai[1],
-	.codec_dai = &wm8731_dai,
-	.init = eti_b1_wm8731_init,
-	.ops = &eti_b1_ops,
-};
-
-static struct snd_soc_machine snd_soc_machine_eti_b1 = {
-	.name = "ETI_B1_WM8731",
-	.dai_link = &eti_b1_dai,
-	.num_links = 1,
-};
-
-static struct wm8731_setup_data eti_b1_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1a,
-};
-
-static struct snd_soc_device eti_b1_snd_devdata = {
-	.machine = &snd_soc_machine_eti_b1,
-	.platform = &at91_soc_platform,
-	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &eti_b1_wm8731_setup,
-};
-
-static struct platform_device *eti_b1_snd_device;
-
-static int __init eti_b1_init(void)
-{
-	int ret;
-	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
-
-	if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
-		DBG("SSC1 memory region is busy\n");
-		return -EBUSY;
-	}
-
-	ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
-	if (!ssc->base) {
-		DBG("SSC1 memory ioremap failed\n");
-		ret = -ENOMEM;
-		goto fail_release_mem;
-	}
-
-	ssc->pid = AT91RM9200_ID_SSC1;
-
-	eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!eti_b1_snd_device) {
-		DBG("platform device allocation failed\n");
-		ret = -ENOMEM;
-		goto fail_io_unmap;
-	}
-
-	platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
-	eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
-
-	ret = platform_device_add(eti_b1_snd_device);
-	if (ret) {
-		DBG("platform device add failed\n");
-		platform_device_put(eti_b1_snd_device);
-		goto fail_io_unmap;
-	}
-
-	at91_set_A_periph(AT91_PIN_PB6, 0);	/* TF1 */
-	at91_set_A_periph(AT91_PIN_PB7, 0);	/* TK1 */
-	at91_set_A_periph(AT91_PIN_PB8, 0);	/* TD1 */
-	at91_set_A_periph(AT91_PIN_PB9, 0);	/* RD1 */
-/*	at91_set_A_periph(AT91_PIN_PB10, 0);*/	/* RK1 */
-	at91_set_A_periph(AT91_PIN_PB11, 0);	/* RF1 */
-
-	/*
-	 * Set PCK1 parent to PLLB and its rate to 12 Mhz.
-	 */
-	pllb_clk = clk_get(NULL, "pllb");
-	pck1_clk = clk_get(NULL, "pck1");
-
-	clk_set_parent(pck1_clk, pllb_clk);
-	clk_set_rate(pck1_clk, 12000000);
-
-	DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
-
-	/* assign the GPIO pin to PCK1 */
-	at91_set_B_periph(AT91_PIN_PA24, 0);
-
-#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
-	printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n");
-#else
-	printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n");
-#endif
-	return ret;
-
-fail_io_unmap:
-	iounmap(ssc->base);
-fail_release_mem:
-	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
-	return ret;
-}
-
-static void __exit eti_b1_exit(void)
-{
-	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
-
-	clk_put(pck1_clk);
-	clk_put(pllb_clk);
-
-	platform_device_unregister(eti_b1_snd_device);
-
-	iounmap(ssc->base);
-	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
-}
-
-module_init(eti_b1_init);
-module_exit(eti_b1_exit);
-
-/* Module information */
-MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
-MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index f98331d099e..dc006206f62 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
+config SND_BF5XX_SOC_AD73311
+	tristate "SoC AD73311 Audio support for Blackfin"
+	depends on SND_BF5XX_I2S
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_AD73311
+	help
+	  Say Y if you want to add support for AD73311 codec on Blackfin.
+
+config SND_BFIN_AD73311_SE
+	int "PF pin for AD73311L Chip Select"
+	depends on SND_BF5XX_SOC_AD73311
+	default 4
+	help
+	  Enter the GPIO used to control AD73311's SE pin. Acceptable
+	  values are 0 to 7
+
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
 	depends on BLACKFIN && SND_SOC
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 9ea8bd9e0ba..97bb37a6359 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
 # Blackfin Machine Support
 snd-ad1980-objs := bf5xx-ad1980.o
 snd-ssm2602-objs := bf5xx-ssm2602.o
-
+snd-ad73311-objs := bf5xx-ad73311.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
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 51f4907c483..25e50d2ea1e 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
 		sport->tx_pos += runtime->period_size;
 		if (sport->tx_pos >= runtime->buffer_size)
 			sport->tx_pos %= runtime->buffer_size;
+		sport->tx_delay_pos = sport->tx_pos;
 	} else {
 		bf5xx_ac97_to_pcm(
 			(struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
@@ -72,7 +73,15 @@ static void bf5xx_dma_irq(void *data)
 	struct snd_pcm_substream *pcm = data;
 #if defined(CONFIG_SND_MMAP_SUPPORT)
 	struct snd_pcm_runtime *runtime = pcm->runtime;
+	struct sport_device *sport = runtime->private_data;
 	bf5xx_mmap_copy(pcm, runtime->period_size);
+	if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (sport->once == 0) {
+			snd_pcm_period_elapsed(pcm);
+			bf5xx_mmap_copy(pcm, runtime->period_size);
+			sport->once = 1;
+		}
+	}
 #endif
 	snd_pcm_period_elapsed(pcm);
 }
@@ -114,6 +123,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
 
 static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+	memset(runtime->dma_area, 0, runtime->buffer_size);
 	snd_pcm_lib_free_pages(substream);
 	return 0;
 }
@@ -127,16 +140,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
 	 * SPORT working in TMD mode(include AC97).
 	 */
 #if defined(CONFIG_SND_MMAP_SUPPORT)
-	size_t size = bf5xx_pcm_hardware.buffer_bytes_max
-			* sizeof(struct ac97_frame) / 4;
-	/*clean up intermediate buffer*/
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		memset(sport->tx_dma_buf, 0, size);
 		sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
 		sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
 			runtime->period_size * sizeof(struct ac97_frame));
 	} else {
-		memset(sport->rx_dma_buf, 0, size);
 		sport_set_rx_callback(sport, bf5xx_dma_irq, substream);
 		sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods,
 			runtime->period_size * sizeof(struct ac97_frame));
@@ -164,8 +172,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	pr_debug("%s enter\n", __func__);
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			bf5xx_mmap_copy(substream, runtime->period_size);
+			snd_pcm_period_elapsed(substream);
+			sport->tx_delay_pos = 0;
 			sport_tx_start(sport);
+		}
 		else
 			sport_rx_start(sport);
 		break;
@@ -198,7 +210,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 
 #if defined(CONFIG_SND_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		curr = sport->tx_pos;
+		curr = sport->tx_delay_pos;
 	else
 		curr = sport->rx_pos;
 #else
@@ -237,6 +249,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
 	return ret;
 }
 
+static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct sport_device *sport = runtime->private_data;
+
+	pr_debug("%s enter\n", __func__);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		sport->once = 0;
+		memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
+	} else
+		memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
+
+	return 0;
+}
+
 #ifdef CONFIG_SND_MMAP_SUPPORT
 static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma)
@@ -272,6 +299,7 @@ static	int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 
 struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
 	.open		= bf5xx_pcm_open,
+	.close		= bf5xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
 	.hw_free	= bf5xx_pcm_hw_free,
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index c782e311fd5..5e5aafb6485 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -129,7 +129,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data)
 	struct ac97_frame *nextwrite;
 
 	sport_incfrag(sport, &nextfrag, 1);
-	sport_incfrag(sport, &nextfrag, 1);
 
 	nextwrite = (struct ac97_frame *)(sport->tx_buf + \
 			nextfrag * sport->tx_fragsize);
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
new file mode 100644
index 00000000000..622c9b90953
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -0,0 +1,240 @@
+/*
+ * File:         sound/soc/blackfin/bf5xx-ad73311.c
+ * Author:       Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created:      Thur Sep 25 2008
+ * Description:  Board driver for ad73311 sound chip
+ *
+ * Modified:
+ *               Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs:         Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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, see the file COPYING, or 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/device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+
+#include <asm/blackfin.h>
+#include <asm/cacheflush.h>
+#include <asm/irq.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "../codecs/ad73311.h"
+#include "bf5xx-sport.h"
+#include "bf5xx-i2s-pcm.h"
+#include "bf5xx-i2s.h"
+
+#if CONFIG_SND_BF5XX_SPORT_NUM == 0
+#define bfin_write_SPORT_TCR1	bfin_write_SPORT0_TCR1
+#define bfin_read_SPORT_TCR1	bfin_read_SPORT0_TCR1
+#define bfin_write_SPORT_TCR2	bfin_write_SPORT0_TCR2
+#define bfin_write_SPORT_TX16	bfin_write_SPORT0_TX16
+#define bfin_read_SPORT_STAT	bfin_read_SPORT0_STAT
+#else
+#define bfin_write_SPORT_TCR1	bfin_write_SPORT1_TCR1
+#define bfin_read_SPORT_TCR1	bfin_read_SPORT1_TCR1
+#define bfin_write_SPORT_TCR2	bfin_write_SPORT1_TCR2
+#define bfin_write_SPORT_TX16	bfin_write_SPORT1_TX16
+#define bfin_read_SPORT_STAT	bfin_read_SPORT1_STAT
+#endif
+
+#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
+
+static struct snd_soc_machine bf5xx_ad73311;
+
+static int snd_ad73311_startup(void)
+{
+	pr_debug("%s enter\n", __func__);
+
+	/* Pull up SE pin on AD73311L */
+	gpio_set_value(GPIO_SE, 1);
+	return 0;
+}
+
+static int snd_ad73311_configure(void)
+{
+	unsigned short ctrl_regs[6];
+	unsigned short status = 0;
+	int count = 0;
+
+	/* DMCLK = MCLK = 16.384 MHz
+	 * SCLK = DMCLK/8 = 2.048 MHz
+	 * Sample Rate = DMCLK/2048  = 8 KHz
+	 */
+	ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \
+			REGB_SCDIV(0) | REGB_DIRATE(0);
+	ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \
+			REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ;
+	ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \
+			REGD_IGS(2);
+	ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f);
+	ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ;
+	ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA;
+
+	local_irq_disable();
+	snd_ad73311_startup();
+	udelay(1);
+
+	bfin_write_SPORT_TCR1(TFSR);
+	bfin_write_SPORT_TCR2(0xF);
+	SSYNC();
+
+	/* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to
+	 * FIFO before enable SPORT to transfer the data
+	 */
+	for (count = 0; count < 6; count++)
+		bfin_write_SPORT_TX16(ctrl_regs[count]);
+	SSYNC();
+	bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN);
+	SSYNC();
+
+	/* When TUVF is set, the data is already send out */
+	while (!(status & TUVF) && count++ < 10000) {
+		udelay(1);
+		status = bfin_read_SPORT_STAT();
+		SSYNC();
+	}
+	bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN);
+	SSYNC();
+	local_irq_enable();
+
+	if (count == 10000) {
+		printk(KERN_ERR "ad73311: failed to configure codec\n");
+		return -1;
+	}
+	return 0;
+}
+
+static int bf5xx_probe(struct platform_device *pdev)
+{
+	int err;
+	if (gpio_request(GPIO_SE, "AD73311_SE")) {
+		printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE);
+		return -EBUSY;
+	}
+
+	gpio_direction_output(GPIO_SE, 0);
+
+	err = snd_ad73311_configure();
+	if (err < 0)
+		return -EFAULT;
+
+	return 0;
+}
+
+static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	pr_debug("%s enter\n", __func__);
+	cpu_dai->private_data = sport_handle;
+	return 0;
+}
+
+static int bf5xx_ad73311_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->dai->cpu_dai;
+	int ret = 0;
+
+	pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
+		params_format(params));
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+
+static struct snd_soc_ops bf5xx_ad73311_ops = {
+	.startup = bf5xx_ad73311_startup,
+	.hw_params = bf5xx_ad73311_hw_params,
+};
+
+static struct snd_soc_dai_link bf5xx_ad73311_dai = {
+	.name = "ad73311",
+	.stream_name = "AD73311",
+	.cpu_dai = &bf5xx_i2s_dai,
+	.codec_dai = &ad73311_dai,
+	.ops = &bf5xx_ad73311_ops,
+};
+
+static struct snd_soc_machine bf5xx_ad73311 = {
+	.name = "bf5xx_ad73311",
+	.probe = bf5xx_probe,
+	.dai_link = &bf5xx_ad73311_dai,
+	.num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
+	.machine = &bf5xx_ad73311,
+	.platform = &bf5xx_i2s_soc_platform,
+	.codec_dev = &soc_codec_dev_ad73311,
+};
+
+static struct platform_device *bf52x_ad73311_snd_device;
+
+static int __init bf5xx_ad73311_init(void)
+{
+	int ret;
+
+	pr_debug("%s enter\n", __func__);
+	bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!bf52x_ad73311_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
+	bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev;
+	ret = platform_device_add(bf52x_ad73311_snd_device);
+
+	if (ret)
+		platform_device_put(bf52x_ad73311_snd_device);
+
+	return ret;
+}
+
+static void __exit bf5xx_ad73311_exit(void)
+{
+	pr_debug("%s enter\n", __func__);
+	platform_device_unregister(bf52x_ad73311_snd_device);
+}
+
+module_init(bf5xx_ad73311_init);
+module_exit(bf5xx_ad73311_exit);
+
+/* Module information */
+MODULE_AUTHOR("Cliff Cai");
+MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 43a4092eeb8..e020c160ee4 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -70,6 +70,25 @@ static struct sport_param sport_params[2] = {
 	}
 };
 
+/*
+ * Setting the TFS pin selector for SPORT 0 based on whether the selected
+ * port id F or G. If the port is F then no conflict should exist for the
+ * TFS. When Port G is selected and EMAC then there is a conflict between
+ * the PHY interrupt line and TFS.  Current settings prevent the conflict
+ * by ignoring the TFS pin when Port G is selected. This allows both
+ * ssm2602 using Port G and EMAC concurrently.
+ */
+#ifdef CONFIG_BF527_SPORT0_PORTF
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
+#else
+#define LOCAL_SPORT0_TFS (0)
+#endif
+
+static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+		P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0},
+		{P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI,
+		P_SPORT1_RSCLK, P_SPORT1_TFS, 0} };
+
 static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		unsigned int fmt)
 {
@@ -78,28 +97,34 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	/* interface format:support I2S,slave mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
+		bf5xx_i2s.tcr1 |= TFSR | TCKFE;
+		bf5xx_i2s.rcr1 |= RFSR | RCKFE;
+		bf5xx_i2s.tcr2 |= TSFSE;
+		bf5xx_i2s.rcr2 |= RSFSE;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		bf5xx_i2s.tcr1 |= TFSR;
+		bf5xx_i2s.rcr1 |= RFSR;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
 		ret = -EINVAL;
 		break;
 	default:
+		printk(KERN_ERR "%s: Unknown DAI format type\n", __func__);
 		ret = -EINVAL;
 		break;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBS_CFS:
-		ret = -EINVAL;
-		break;
-	case SND_SOC_DAIFMT_CBM_CFS:
-		ret = -EINVAL;
-		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBM_CFS:
 	case SND_SOC_DAIFMT_CBS_CFM:
 		ret = -EINVAL;
 		break;
 	default:
+		printk(KERN_ERR "%s: Unknown DAI master type\n", __func__);
 		ret = -EINVAL;
 		break;
 	}
@@ -127,14 +152,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bf5xx_i2s.tcr2 |= 15;
 		bf5xx_i2s.rcr2 |= 15;
+		sport_handle->wdsize = 2;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		bf5xx_i2s.tcr2 |= 23;
 		bf5xx_i2s.rcr2 |= 23;
+		sport_handle->wdsize = 3;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		bf5xx_i2s.tcr2 |= 31;
 		bf5xx_i2s.rcr2 |= 31;
+		sport_handle->wdsize = 4;
 		break;
 	}
 
@@ -145,17 +173,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 		 * need to configure both of them at the time when the first
 		 * stream is opened.
 		 *
-		 * CPU DAI format:I2S, slave mode.
+		 * CPU DAI:slave mode.
 		 */
-		ret = sport_config_rx(sport_handle, RFSR | RCKFE,
-				      RSFSE|bf5xx_i2s.rcr2, 0, 0);
+		ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
+				      bf5xx_i2s.rcr2, 0, 0);
 		if (ret) {
 			pr_err("SPORT is busy!\n");
 			return -EBUSY;
 		}
 
-		ret = sport_config_tx(sport_handle, TFSR | TCKFE,
-				      TSFSE|bf5xx_i2s.tcr2, 0, 0);
+		ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
+				      bf5xx_i2s.tcr2, 0, 0);
 		if (ret) {
 			pr_err("SPORT is busy!\n");
 			return -EBUSY;
@@ -174,13 +202,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
 static int bf5xx_i2s_probe(struct platform_device *pdev,
 			   struct snd_soc_dai *dai)
 {
-	u16 sport_req[][7] = {
-		{ P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
-		  P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
-		{ P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
-		  P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
-	};
-
 	pr_debug("%s enter\n", __func__);
 	if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
 		pr_err("Requesting Peripherals failed\n");
@@ -198,6 +219,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
+static void bf5xx_i2s_remove(struct platform_device *pdev,
+			   struct snd_soc_dai *dai)
+{
+	pr_debug("%s enter\n", __func__);
+	peripheral_free_list(&sport_req[sport_num][0]);
+}
+
 #ifdef CONFIG_PM
 static int bf5xx_i2s_suspend(struct platform_device *dev,
 			     struct snd_soc_dai *dai)
@@ -263,15 +291,16 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 	.id = 0,
 	.type = SND_SOC_DAI_I2S,
 	.probe = bf5xx_i2s_probe,
+	.remove = bf5xx_i2s_remove,
 	.suspend = bf5xx_i2s_suspend,
 	.resume = bf5xx_i2s_resume,
 	.playback = {
-		.channels_min = 2,
+		.channels_min = 1,
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
 	.capture = {
-		.channels_min = 2,
+		.channels_min = 1,
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h
index 4c163454bbf..fcadcc081f7 100644
--- a/sound/soc/blackfin/bf5xx-sport.h
+++ b/sound/soc/blackfin/bf5xx-sport.h
@@ -123,6 +123,8 @@ struct sport_device {
 	int rx_pos;
 	unsigned int tx_buffer_size;
 	unsigned int rx_buffer_size;
+	int tx_delay_pos;
+	int once;
 #endif
 	void *private_data;
 };
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e0b9869df0f..38a0e3b620a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -3,9 +3,11 @@ config SND_SOC_ALL_CODECS
 	depends on I2C
 	select SPI
 	select SPI_MASTER
+	select SND_SOC_AD73311
 	select SND_SOC_AK4535
 	select SND_SOC_CS4270
 	select SND_SOC_SSM2602
+	select SND_SOC_TLV320AIC23
 	select SND_SOC_TLV320AIC26
 	select SND_SOC_TLV320AIC3X
 	select SND_SOC_UDA1380
@@ -34,6 +36,9 @@ config SND_SOC_AC97_CODEC
 config SND_SOC_AD1980
 	tristate
 
+config SND_SOC_AD73311
+	tristate
+
 config SND_SOC_AK4535
 	tristate
 
@@ -58,9 +63,13 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_SSM2602
 	tristate
 
+config SND_SOC_TLV320AIC23
+	tristate
+	depends on I2C
+
 config SND_SOC_TLV320AIC26
-	tristate "TI TLV320AIC26 Codec support"
-	depends on SND_SOC && SPI
+	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
+	depends on SPI
 
 config SND_SOC_TLV320AIC3X
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index f977978a340..90f0a585fc7 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,8 +1,10 @@
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1980-objs := ad1980.o
+snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-uda1380-objs := uda1380.o
@@ -20,9 +22,11 @@ snd-soc-wm9713-objs := wm9713.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
+obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 61fd96ca7bc..bd1ebdc6c86 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -2,8 +2,7 @@
  * ac97.c  --  ALSA Soc AC97 codec support
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 4e09c1f2c06..1397b8e06c0 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -13,7 +13,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
new file mode 100644
index 00000000000..37af8607b00
--- /dev/null
+++ b/sound/soc/codecs/ad73311.c
@@ -0,0 +1,107 @@
+/*
+ * ad73311.c  --  ALSA Soc AD73311 codec support
+ *
+ * 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.
+ *
+ *  Revision history
+ *    25th Sep 2008   Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.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>
+
+#include "ad73311.h"
+
+struct snd_soc_dai ad73311_dai = {
+	.name = "AD73311",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+};
+EXPORT_SYMBOL_GPL(ad73311_dai);
+
+static int ad73311_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+	mutex_init(&codec->mutex);
+	codec->name = "AD73311";
+	codec->owner = THIS_MODULE;
+	codec->dai = &ad73311_dai;
+	codec->num_dai = 1;
+	socdev->codec = codec;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "ad73311: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "ad73311: failed to register card\n");
+		goto register_err;
+	}
+
+	return ret;
+
+register_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	kfree(socdev->codec);
+	socdev->codec = NULL;
+	return ret;
+}
+
+static int ad73311_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec == NULL)
+		return 0;
+	snd_soc_free_pcms(socdev);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad73311 = {
+	.probe = 	ad73311_soc_probe,
+	.remove = 	ad73311_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
+
+MODULE_DESCRIPTION("ASoC ad73311 driver");
+MODULE_AUTHOR("Cliff Cai ");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
new file mode 100644
index 00000000000..507ce0c30ed
--- /dev/null
+++ b/sound/soc/codecs/ad73311.h
@@ -0,0 +1,90 @@
+/*
+ * File:         sound/soc/codec/ad73311.h
+ * Based on:
+ * Author:       Cliff Cai <cliff.cai@analog.com>
+ *
+ * Created:      Thur Sep 25, 2008
+ * Description:  definitions for AD73311 registers
+ *
+ *
+ * Modified:
+ *               Copyright 2006 Analog Devices Inc.
+ *
+ * Bugs:         Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+#ifndef __AD73311_H__
+#define __AD73311_H__
+
+#define AD_CONTROL	0x8000
+#define AD_DATA		0x0000
+#define AD_READ		0x4000
+#define AD_WRITE	0x0000
+
+/* Control register A */
+#define CTRL_REG_A	(0 << 8)
+
+#define REGA_MODE_PRO	0x00
+#define REGA_MODE_DATA	0x01
+#define REGA_MODE_MIXED	0x03
+#define REGA_DLB		0x04
+#define REGA_SLB		0x08
+#define REGA_DEVC(x)		((x & 0x7) << 4)
+#define REGA_RESET		0x80
+
+/* Control register B */
+#define CTRL_REG_B	(1 << 8)
+
+#define REGB_DIRATE(x)	(x & 0x3)
+#define REGB_SCDIV(x)	((x & 0x3) << 2)
+#define REGB_MCDIV(x)	((x & 0x7) << 4)
+#define REGB_CEE		(1 << 7)
+
+/* Control register C */
+#define CTRL_REG_C	(2 << 8)
+
+#define REGC_PUDEV		(1 << 0)
+#define REGC_PUADC		(1 << 3)
+#define REGC_PUDAC		(1 << 4)
+#define REGC_PUREF		(1 << 5)
+#define REGC_REFUSE		(1 << 6)
+
+/* Control register D */
+#define CTRL_REG_D	(3 << 8)
+
+#define REGD_IGS(x)		(x & 0x7)
+#define REGD_RMOD		(1 << 3)
+#define REGD_OGS(x)		((x & 0x7) << 4)
+#define REGD_MUTE		(x << 7)
+
+/* Control register E */
+#define CTRL_REG_E	(4 << 8)
+
+#define REGE_DA(x)		(x & 0x1f)
+#define REGE_IBYP		(1 << 5)
+
+/* Control register F */
+#define CTRL_REG_F	(5 << 8)
+
+#define REGF_SEEN		(1 << 5)
+#define REGF_INV		(1 << 6)
+#define REGF_ALB		(1 << 7)
+
+extern struct snd_soc_dai ad73311_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad73311;
+#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 088cf992772..2a89b5888e1 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -28,7 +28,6 @@
 
 #include "ak4535.h"
 
-#define AUDIO_NAME "ak4535"
 #define AK4535_VERSION "0.3"
 
 struct snd_soc_codec_device soc_codec_dev_ak4535;
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 940ce1c3522..44ef0dacd56 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -42,7 +42,6 @@
 
 #include "ssm2602.h"
 
-#define AUDIO_NAME "ssm2602"
 #define SSM2602_VERSION "0.1"
 
 struct snd_soc_codec_device soc_codec_dev_ssm2602;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
new file mode 100644
index 00000000000..44308dac9e1
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -0,0 +1,714 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author:      Arun KS, <arunks@mistralsolutions.com>
+ * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * 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.
+ *
+ * Notes:
+ *  The AIC23 is a driver for a low power stereo audio
+ *  codec tlv320aic23
+ *
+ *  The machine layer should disable unsupported inputs/outputs by
+ *  snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+
+#include "tlv320aic23.h"
+
+#define AIC23_VERSION "0.1"
+
+struct tlv320aic23_srate_reg_info {
+	u32 sample_rate;
+	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
+	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
+};
+
+/*
+ * AIC23 register cache
+ */
+static const u16 tlv320aic23_reg[] = {
+	0x0097, 0x0097, 0x00F9, 0x00F9,	/* 0 */
+	0x001A, 0x0004, 0x0007, 0x0001,	/* 4 */
+	0x0020, 0x0000, 0x0000, 0x0000,	/* 8 */
+	0x0000, 0x0000, 0x0000, 0x0000,	/* 12 */
+};
+
+/*
+ * read tlv320aic23 register cache
+ */
+static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
+						      *codec, unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write tlv320aic23 register cache
+ */
+static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
+					       u8 reg, u16 value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the tlv320aic23 register space
+ */
+static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
+			     unsigned int value)
+{
+
+	u8 data[2];
+
+	/* TLV320AIC23 has 7 bit address and 9 bits of data
+	 * so we need to switch one data bit into reg and rest
+	 * of data into val
+	 */
+
+	if ((reg < 0 || reg > 9) && (reg != 15)) {
+		printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+		return -1;
+	}
+
+	data[0] = (reg << 1) | (value >> 8 & 0x01);
+	data[1] = value & 0xff;
+
+	tlv320aic23_write_reg_cache(codec, reg, value);
+
+	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__,
+	       value, reg);
+
+	return -EIO;
+}
+
+static const char *rec_src_text[] = { "Line", "Mic" };
+static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum rec_src_enum =
+	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Input Select", rec_src_enum);
+
+static const struct soc_enum tlv320aic23_rec_src =
+	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+static const struct soc_enum tlv320aic23_deemph =
+	SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
+static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
+
+static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u16 val, reg;
+
+	val = (ucontrol->value.integer.value[0] & 0x07);
+
+	/* linear conversion to userspace
+	* 000	=	-6db
+	* 001	=	-9db
+	* 010	=	-12db
+	* 011	=	-18db (Min)
+	* 100	=	0db (Max)
+	*/
+	val = (val >= 4) ? 4  : (3 - val);
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
+	tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
+
+	return 0;
+}
+
+static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u16 val;
+
+	val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
+	val = val >> 6;
+	val = (val >= 4) ? 4  : (3 -  val);
+	ucontrol->value.integer.value[0] = val;
+	return 0;
+
+}
+
+#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
+	.put = snd_soc_tlv320aic23_put_volsw, \
+	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
+			 TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
+	SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
+	SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
+		     TLV320AIC23_RINVOL, 7, 1, 0),
+	SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
+			 TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
+	SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
+	SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
+	SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
+				  6, 4, 0, sidetone_vol_tlv),
+	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
+};
+
+/* add non dapm controls */
+static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
+{
+
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&tlv320aic23_snd_controls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+
+}
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
+	SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
+	SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+			 &tlv320aic23_rec_src_mux_controls),
+	SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
+			   &tlv320aic23_output_mixer_controls[0],
+			   ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LHPOUT"),
+	SND_SOC_DAPM_OUTPUT("RHPOUT"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+
+	SND_SOC_DAPM_INPUT("LLINEIN"),
+	SND_SOC_DAPM_INPUT("RLINEIN"),
+
+	SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* Output Mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
+
+	/* Outputs */
+	{"RHPOUT", NULL, "Output Mixer"},
+	{"LHPOUT", NULL, "Output Mixer"},
+	{"LOUT", NULL, "Output Mixer"},
+	{"ROUT", NULL, "Output Mixer"},
+
+	/* Inputs */
+	{"Line Input", "NULL", "LLINEIN"},
+	{"Line Input", "NULL", "RLINEIN"},
+
+	{"Mic Input", "NULL", "MICIN"},
+
+	/* input mux */
+	{"Capture Source", "Line", "Line Input"},
+	{"Capture Source", "Mic", "Mic Input"},
+	{"ADC", NULL, "Capture Source"},
+
+};
+
+/* tlv320aic23 related */
+static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
+	{4000, 0x06, 1},	/*  4000 */
+	{8000, 0x06, 0},	/*  8000 */
+	{16000, 0x0C, 1},	/* 16000 */
+	{22050, 0x11, 1},	/* 22050 */
+	{24000, 0x00, 1},	/* 24000 */
+	{32000, 0x0C, 0},	/* 32000 */
+	{44100, 0x11, 0},	/* 44100 */
+	{48000, 0x00, 0},	/* 48000 */
+	{88200, 0x1F, 0},	/* 88200 */
+	{96000, 0x0E, 0},	/* 96000 */
+};
+
+static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* set up audio path interconnects */
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static int tlv320aic23_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_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 iface_reg, data;
+	u8 count = 0;
+
+	iface_reg =
+	    tlv320aic23_read_reg_cache(codec,
+				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
+
+	/* Search for the right sample rate */
+	/* Verify what happens if the rate is not supported
+	 * now it goes to 96Khz */
+	while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
+	       (count < ARRAY_SIZE(srate_reg_info))) {
+		count++;
+	}
+
+	data =  (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
+		(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
+		TLV320AIC23_USB_CLK_ON;
+
+	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface_reg |= (0x01 << 2);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface_reg |= (0x02 << 2);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface_reg |= (0x03 << 2);
+		break;
+	}
+	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* set active */
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+
+	return 0;
+}
+
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* deactivate */
+	if (!codec->active) {
+		udelay(50);
+		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+	}
+}
+
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 reg;
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
+	if (mute)
+		reg |= TLV320AIC23_DACM_MUTE;
+
+	else
+		reg &= ~TLV320AIC23_DACM_MUTE;
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
+
+	return 0;
+}
+
+static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
+				   unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface_reg;
+
+	iface_reg =
+	    tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface_reg |= TLV320AIC23_MS_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface_reg |= TLV320AIC23_FOR_I2S;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface_reg |= TLV320AIC23_FOR_DSP;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg |= TLV320AIC23_FOR_LJUST;
+		break;
+	default:
+		return -EINVAL;
+
+	}
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				      int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	switch (freq) {
+	case 12000000:
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
+				      enum snd_soc_bias_level level)
+{
+	u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* vref/mid, osc on, dac unmute */
+		tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* everything off except vref/vmid, */
+		tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+		tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define AIC23_RATES	SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai tlv320aic23_dai = {
+	.name = "tlv320aic23",
+	.playback = {
+		     .stream_name = "Playback",
+		     .channels_min = 2,
+		     .channels_max = 2,
+		     .rates = AIC23_RATES,
+		     .formats = AIC23_FORMATS,},
+	.capture = {
+		    .stream_name = "Capture",
+		    .channels_min = 2,
+		    .channels_max = 2,
+		    .rates = AIC23_RATES,
+		    .formats = AIC23_FORMATS,},
+	.ops = {
+		.prepare = tlv320aic23_pcm_prepare,
+		.hw_params = tlv320aic23_hw_params,
+		.shutdown = tlv320aic23_shutdown,
+		},
+	.dai_ops = {
+		    .digital_mute = tlv320aic23_mute,
+		    .set_fmt = tlv320aic23_set_dai_fmt,
+		    .set_sysclk = tlv320aic23_set_dai_sysclk,
+		    }
+};
+EXPORT_SYMBOL_GPL(tlv320aic23_dai);
+
+static int tlv320aic23_suspend(struct platform_device *pdev,
+			       pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int tlv320aic23_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u16 reg;
+
+	/* Sync reg_cache with the hardware */
+	for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+		u16 val = tlv320aic23_read_reg_cache(codec, reg);
+		tlv320aic23_write(codec, reg, val);
+	}
+
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+
+/*
+ * initialise the AIC23 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int tlv320aic23_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+	u16 reg;
+
+	codec->name = "tlv320aic23";
+	codec->owner = THIS_MODULE;
+	codec->read = tlv320aic23_read_reg_cache;
+	codec->write = tlv320aic23_write;
+	codec->set_bias_level = tlv320aic23_set_bias_level;
+	codec->dai = &tlv320aic23_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
+	codec->reg_cache =
+	    kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	/* Reset codec */
+	tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+
+	/* Unmute input */
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
+	tlv320aic23_write(codec, TLV320AIC23_LINVOL,
+			  (reg & (~TLV320AIC23_LIM_MUTED)) |
+			  (TLV320AIC23_LRS_ENABLED));
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
+	tlv320aic23_write(codec, TLV320AIC23_RINVOL,
+			  (reg & (~TLV320AIC23_LIM_MUTED)) |
+			  TLV320AIC23_LRS_ENABLED);
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
+	tlv320aic23_write(codec, TLV320AIC23_ANLG,
+			 (reg) & (~TLV320AIC23_BYPASS_ON) &
+			 (~TLV320AIC23_MICM_MUTED));
+
+	/* Default output volume */
+	tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
+			  TLV320AIC23_DEFAULT_OUT_VOL &
+			  TLV320AIC23_OUT_VOL_MASK);
+	tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
+			  TLV320AIC23_DEFAULT_OUT_VOL &
+			  TLV320AIC23_OUT_VOL_MASK);
+
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
+
+	tlv320aic23_add_controls(codec);
+	tlv320aic23_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+static struct snd_soc_device *tlv320aic23_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * If the i2c layer weren't so broken, we could pass this kind of data
+ * around
+ */
+static int tlv320aic23_codec_probe(struct i2c_client *i2c,
+				   const struct i2c_device_id *i2c_id)
+{
+	struct snd_soc_device *socdev = tlv320aic23_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+		return -EINVAL;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = tlv320aic23_init(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
+{
+	put_device(&i2c->dev);
+	return 0;
+}
+
+static const struct i2c_device_id tlv320aic23_id[] = {
+	{"tlv320aic23", 0},
+	{}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
+
+static struct i2c_driver tlv320aic23_i2c_driver = {
+	.driver = {
+		   .name = "tlv320aic23",
+		   },
+	.probe = tlv320aic23_codec_probe,
+	.remove = __exit_p(tlv320aic23_i2c_remove),
+	.id_table = tlv320aic23_id,
+};
+
+#endif
+
+static int tlv320aic23_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	tlv320aic23_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	codec->hw_write = (hw_write_t) i2c_master_send;
+	codec->hw_read = NULL;
+	ret = i2c_add_driver(&tlv320aic23_i2c_driver);
+	if (ret != 0)
+		printk(KERN_ERR "can't add i2c driver");
+#endif
+	return ret;
+}
+
+static int tlv320aic23_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec->control_data)
+		tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&tlv320aic23_i2c_driver);
+#endif
+	kfree(codec->reg_cache);
+	kfree(codec);
+
+	return 0;
+}
+struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
+	.probe = tlv320aic23_probe,
+	.remove = tlv320aic23_remove,
+	.suspend = tlv320aic23_suspend,
+	.resume = tlv320aic23_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
new file mode 100644
index 00000000000..79d1faf8e57
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -0,0 +1,122 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author:      Arun KS, <arunks@mistralsolutions.com>
+ * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd
+ *
+ * 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 _TLV320AIC23_H
+#define _TLV320AIC23_H
+
+/* Codec TLV320AIC23 */
+#define TLV320AIC23_LINVOL		0x00
+#define TLV320AIC23_RINVOL		0x01
+#define TLV320AIC23_LCHNVOL		0x02
+#define TLV320AIC23_RCHNVOL		0x03
+#define TLV320AIC23_ANLG		0x04
+#define TLV320AIC23_DIGT		0x05
+#define TLV320AIC23_PWR			0x06
+#define TLV320AIC23_DIGT_FMT		0x07
+#define TLV320AIC23_SRATE		0x08
+#define TLV320AIC23_ACTIVE		0x09
+#define TLV320AIC23_RESET		0x0F
+
+/* Left (right) line input volume control register */
+#define TLV320AIC23_LRS_ENABLED		0x0100
+#define TLV320AIC23_LIM_MUTED		0x0080
+#define TLV320AIC23_LIV_DEFAULT		0x0017
+#define TLV320AIC23_LIV_MAX		0x001f
+#define TLV320AIC23_LIV_MIN		0x0000
+
+/* Left (right) channel headphone volume control register */
+#define TLV320AIC23_LZC_ON		0x0080
+#define TLV320AIC23_LHV_DEFAULT		0x0079
+#define TLV320AIC23_LHV_MAX		0x007f
+#define TLV320AIC23_LHV_MIN		0x0000
+
+/* Analog audio path control register */
+#define TLV320AIC23_STA_REG(x)		((x)<<6)
+#define TLV320AIC23_STE_ENABLED		0x0020
+#define TLV320AIC23_DAC_SELECTED	0x0010
+#define TLV320AIC23_BYPASS_ON		0x0008
+#define TLV320AIC23_INSEL_MIC		0x0004
+#define TLV320AIC23_MICM_MUTED		0x0002
+#define TLV320AIC23_MICB_20DB		0x0001
+
+/* Digital audio path control register */
+#define TLV320AIC23_DACM_MUTE		0x0008
+#define TLV320AIC23_DEEMP_32K		0x0002
+#define TLV320AIC23_DEEMP_44K		0x0004
+#define TLV320AIC23_DEEMP_48K		0x0006
+#define TLV320AIC23_ADCHP_ON		0x0001
+
+/* Power control down register */
+#define TLV320AIC23_DEVICE_PWR_OFF  	0x0080
+#define TLV320AIC23_CLK_OFF		0x0040
+#define TLV320AIC23_OSC_OFF		0x0020
+#define TLV320AIC23_OUT_OFF		0x0010
+#define TLV320AIC23_DAC_OFF		0x0008
+#define TLV320AIC23_ADC_OFF		0x0004
+#define TLV320AIC23_MIC_OFF		0x0002
+#define TLV320AIC23_LINE_OFF		0x0001
+
+/* Digital audio interface register */
+#define TLV320AIC23_MS_MASTER		0x0040
+#define TLV320AIC23_LRSWAP_ON		0x0020
+#define TLV320AIC23_LRP_ON		0x0010
+#define TLV320AIC23_IWL_16		0x0000
+#define TLV320AIC23_IWL_20		0x0004
+#define TLV320AIC23_IWL_24		0x0008
+#define TLV320AIC23_IWL_32		0x000C
+#define TLV320AIC23_FOR_I2S		0x0002
+#define TLV320AIC23_FOR_DSP		0x0003
+#define TLV320AIC23_FOR_LJUST		0x0001
+
+/* Sample rate control register */
+#define TLV320AIC23_CLKOUT_HALF		0x0080
+#define TLV320AIC23_CLKIN_HALF		0x0040
+#define TLV320AIC23_BOSR_384fs		0x0002	/* BOSR_272fs in USB mode */
+#define TLV320AIC23_USB_CLK_ON		0x0001
+#define TLV320AIC23_SR_MASK             0xf
+#define TLV320AIC23_CLKOUT_SHIFT        7
+#define TLV320AIC23_CLKIN_SHIFT         6
+#define TLV320AIC23_SR_SHIFT            2
+#define TLV320AIC23_BOSR_SHIFT          1
+
+/* Digital interface register */
+#define TLV320AIC23_ACT_ON		0x0001
+
+/*
+ * AUDIO related MACROS
+ */
+
+#define TLV320AIC23_DEFAULT_OUT_VOL	0x70
+#define TLV320AIC23_DEFAULT_IN_VOLUME	0x10
+
+#define TLV320AIC23_OUT_VOL_MIN		TLV320AIC23_LHV_MIN
+#define TLV320AIC23_OUT_VOL_MAX		TLV320AIC23_LHV_MAX
+#define TLV320AIC23_OUT_VO_RANGE	(TLV320AIC23_OUT_VOL_MAX - \
+					TLV320AIC23_OUT_VOL_MIN)
+#define TLV320AIC23_OUT_VOL_MASK	TLV320AIC23_OUT_VOL_MAX
+
+#define TLV320AIC23_IN_VOL_MIN		TLV320AIC23_LIV_MIN
+#define TLV320AIC23_IN_VOL_MAX		TLV320AIC23_LIV_MAX
+#define TLV320AIC23_IN_VOL_RANGE	(TLV320AIC23_IN_VOL_MAX - \
+					TLV320AIC23_IN_VOL_MIN)
+#define TLV320AIC23_IN_VOL_MASK		TLV320AIC23_IN_VOL_MAX
+
+#define TLV320AIC23_SIDETONE_MASK	0x1c0
+#define TLV320AIC23_SIDETONE_0		0x100
+#define TLV320AIC23_SIDETONE_6		0x000
+#define TLV320AIC23_SIDETONE_9		0x040
+#define TLV320AIC23_SIDETONE_12		0x080
+#define TLV320AIC23_SIDETONE_18		0x0c0
+
+extern struct snd_soc_dai tlv320aic23_dai;
+extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
+
+#endif /* _TLV320AIC23_H */
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 566a427c928..cff276ee261 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -48,7 +48,6 @@
 
 #include "tlv320aic3x.h"
 
-#define AUDIO_NAME "aic3x"
 #define AIC3X_VERSION "0.2"
 
 /* codec private data */
@@ -864,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	/* interface format */
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_I2S:
+	/*
+	 * match both interface format and signal polarities since they
+	 * are fixed
+	 */
+	switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+		       SND_SOC_DAIFMT_INV_MASK)) {
+	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
 		iface_breg |= (0x01 << 6);
 		break;
-	case SND_SOC_DAIFMT_RIGHT_J:
+	case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x02 << 6);
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
+	case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x03 << 6);
 		break;
 	default:
@@ -991,7 +994,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 struct snd_soc_dai aic3x_dai = {
-	.name = "aic3x",
+	.name = "tlv320aic3x",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
@@ -1055,7 +1058,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
-	codec->name = "aic3x";
+	codec->name = "tlv320aic3x";
 	codec->owner = THIS_MODULE;
 	codec->read = aic3x_read_reg_cache;
 	codec->write = aic3x_write;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index d206d7f892b..a69ee72a7af 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -36,7 +36,6 @@
 #include "uda1380.h"
 
 #define UDA1380_VERSION "0.6"
-#define AUDIO_NAME "uda1380"
 
 /*
  * uda1380 register cache
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 9a37c8d95ed..d8ca2da8d63 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -3,7 +3,7 @@
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
  *
- * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
@@ -18,6 +18,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -27,7 +28,6 @@
 
 #include "wm8510.h"
 
-#define AUDIO_NAME "wm8510"
 #define WM8510_VERSION "0.6"
 
 struct snd_soc_codec_device soc_codec_dev_wm8510;
@@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
 	0x0001,
 };
 
+#define WM8510_POWER1_BIASEN  0x08
+#define WM8510_POWER1_BUFIOEN 0x10
+
 /*
  * read wm8510 register cache
  */
@@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
 SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
 SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0,
-		 &wm8510_micpga_controls[0],
-		 ARRAY_SIZE(wm8510_micpga_controls)),
+SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0,
+		   &wm8510_micpga_controls[0],
+		   ARRAY_SIZE(wm8510_micpga_controls)),
 SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0,
 	&wm8510_boost_controls[0],
 	ARRAY_SIZE(wm8510_boost_controls)),
@@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute)
 static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
+	u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		wm8510_write(codec, WM8510_POWER1, 0x1ff);
-		wm8510_write(codec, WM8510_POWER2, 0x1ff);
-		wm8510_write(codec, WM8510_POWER3, 0x1ff);
-		break;
 	case SND_SOC_BIAS_PREPARE:
+		power1 |= 0x1;  /* VMID 50k */
+		wm8510_write(codec, WM8510_POWER1, power1);
+		break;
+
 	case SND_SOC_BIAS_STANDBY:
+		power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
+
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Initial cap charge at VMID 5k */
+			wm8510_write(codec, WM8510_POWER1, power1 | 0x3);
+			mdelay(100);
+		}
+
+		power1 |= 0x2;  /* VMID 500k */
+		wm8510_write(codec, WM8510_POWER1, power1);
 		break;
+
 	case SND_SOC_BIAS_OFF:
-		/* everything off, dac mute, inactive */
-		wm8510_write(codec, WM8510_POWER1, 0x0);
-		wm8510_write(codec, WM8510_POWER2, 0x0);
-		wm8510_write(codec, WM8510_POWER3, 0x0);
+		wm8510_write(codec, WM8510_POWER1, 0);
+		wm8510_write(codec, WM8510_POWER2, 0);
+		wm8510_write(codec, WM8510_POWER3, 0);
 		break;
 	}
+
 	codec->bias_level = level;
 	return 0;
 }
@@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	}
 
 	/* power on device */
+	codec->bias_level = SND_SOC_BIAS_OFF;
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm8510_add_controls(codec);
 	wm8510_add_widgets(codec);
@@ -747,6 +763,62 @@ err_driver:
 }
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8510_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8510_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8510_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8510\n");
+
+	return ret;
+}
+
+static int __devexit wm8510_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8510_spi_driver = {
+	.driver = {
+		.name	= "wm8510",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8510_spi_probe,
+	.remove		= __devexit_p(wm8510_spi_remove),
+};
+
+static int wm8510_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
 static int wm8510_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -772,8 +844,14 @@ static int wm8510_probe(struct platform_device *pdev)
 		codec->hw_write = (hw_write_t)i2c_master_send;
 		ret = wm8510_add_i2c_device(pdev, setup);
 	}
-#else
-	/* Add other interfaces here */
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	if (setup->spi) {
+		codec->hw_write = (hw_write_t)wm8510_spi_write;
+		ret = spi_register_driver(&wm8510_spi_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add spi driver");
+	}
 #endif
 
 	if (ret != 0)
@@ -796,6 +874,9 @@ static int wm8510_remove(struct platform_device *pdev)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8510_i2c_driver);
 #endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8510_spi_driver);
+#endif
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
index c5368396045..bdefcf5c69f 100644
--- a/sound/soc/codecs/wm8510.h
+++ b/sound/soc/codecs/wm8510.h
@@ -94,6 +94,7 @@
 #define WM8510_MCLKDIV_12	(7 << 5)
 
 struct wm8510_setup_data {
+	int spi;
 	int i2c_bus;
 	unsigned short i2c_address;
 };
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index df1ffbe305b..627ebfb4209 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -18,7 +18,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
@@ -36,7 +35,6 @@
 
 #include "wm8580.h"
 
-#define AUDIO_NAME "wm8580"
 #define WM8580_VERSION "0.1"
 
 struct pll_state {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7b64d9a7ff7..7f8a7e36b33 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -29,7 +29,6 @@
 
 #include "wm8731.h"
 
-#define AUDIO_NAME "wm8731"
 #define WM8731_VERSION "0.13"
 
 struct snd_soc_codec_device soc_codec_dev_wm8731;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 4892e398a59..9b7296ee5b0 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -29,7 +29,6 @@
 
 #include "wm8750.h"
 
-#define AUDIO_NAME "WM8750"
 #define WM8750_VERSION "0.12"
 
 /* codec private data */
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8c4df44f334..d426eaa2218 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -2,8 +2,7 @@
  * wm8753.c  --  WM8753 ALSA Soc Audio driver
  *
  * Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -40,6 +39,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -51,7 +51,6 @@
 
 #include "wm8753.h"
 
-#define AUDIO_NAME "wm8753"
 #define WM8753_VERSION "0.16"
 
 static int caps_charge = 2000;
@@ -1719,6 +1718,63 @@ err_driver:
 }
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8753_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8753_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8753_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8753\n");
+
+	return ret;
+}
+
+static int __devexit wm8753_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8753_spi_driver = {
+	.driver = {
+		.name	= "wm8753",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8753_spi_probe,
+	.remove		= __devexit_p(wm8753_spi_remove),
+};
+
+static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+#endif
+
+
 static int wm8753_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -1753,8 +1809,14 @@ static int wm8753_probe(struct platform_device *pdev)
 		codec->hw_write = (hw_write_t)i2c_master_send;
 		ret = wm8753_add_i2c_device(pdev, setup);
 	}
-#else
-		/* Add other interfaces here */
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	if (setup->spi) {
+		codec->hw_write = (hw_write_t)wm8753_spi_write;
+		ret = spi_register_driver(&wm8753_spi_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add spi driver");
+	}
 #endif
 
 	if (ret != 0) {
@@ -1798,6 +1860,9 @@ static int wm8753_remove(struct platform_device *pdev)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8753_i2c_driver);
 #endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8753_spi_driver);
+#endif
 	kfree(codec->private_data);
 	kfree(codec);
 
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index 7defde069f1..f55704ce931 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -2,8 +2,7 @@
  * wm8753.h  --  audio driver for WM8753
  *
  * Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -79,6 +78,7 @@
 #define WM8753_ADCTL2		0x3f
 
 struct wm8753_setup_data {
+	int spi;
 	int i2c_bus;
 	unsigned short i2c_address;
 };
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 0b8c6d38b48..3b326c9b558 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -18,7 +18,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index a3f54ec4226..ce40d787760 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = {
 
 /* Input PGAs - No TLV since the scale depends on PGA mode */
 SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0,
-	   7, 1, 0),
+	   7, 1, 1),
 SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0,
 	   0, 31, 0),
 SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1,
 	   6, 1, 0),
 
 SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0,
-	   7, 1, 0),
+	   7, 1, 1),
 SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0,
 	   0, 31, 0),
 SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 974a4cd0f3f..f41a578ddd4 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -29,7 +29,6 @@
 
 #include "wm8971.h"
 
-#define AUDIO_NAME "wm8971"
 #define WM8971_VERSION "0.9"
 
 #define	WM8971_REG_COUNT		43
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 63410d7b5ef..572d22b0880 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -30,7 +30,6 @@
 
 #include "wm8990.h"
 
-#define AUDIO_NAME "wm8990"
 #define WM8990_VERSION "0.2"
 
 /* codec private data */
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 2f1c91b1d55..ffb471e420e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -2,8 +2,7 @@
  * wm9712.c  --  ALSA Soc WM9712 codec support
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 441d0580db1..945b32ed988 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -2,8 +2,7 @@
  * wm9713.c  --  ALSA Soc WM9713 codec support
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -141,7 +140,7 @@ SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
 
 SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
 SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
-SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
 SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
 SOC_ENUM("ALC Function", wm9713_enum[6]),
 SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index bba9546ba5f..8d73edc5610 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -20,7 +20,8 @@ config SND_SOC_MPC8610_HPCD
 
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
+	depends on SND_SOC && PPC_MPC52xx && PPC_BESTCOMM
 	select SND_SOC_OF_SIMPLE
-	depends on SND_SOC && PPC_MPC52xx
+	select PPC_BESTCOMM_GEN_BD
 	help
 	  Say Y here to support the MPC5200 PSCs in I2S mode.
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 86923299bc1..94a02eaa482 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -277,7 +277,7 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
 	u16 imr;
 	u8 psc_cmd;
-	long flags;
+	unsigned long flags;
 
 	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
 		s = &psc_i2s->capture;
@@ -699,9 +699,11 @@ static ssize_t psc_i2s_stat_store(struct device *dev,
 	return count;
 }
 
-DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
-DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
-DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
+static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
+static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,
+			psc_i2s_stat_store);
+static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
+			psc_i2s_stat_store);
 
 /* ---------------------------------------------------------------------
  * OF platform bus binding code:
@@ -819,8 +821,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 
 	/* Register the SYSFS files */
 	rc = device_create_file(psc_i2s->dev, &dev_attr_status);
-	rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
-	rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
+	rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
+	rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
 	if (rc)
 		dev_info(psc_i2s->dev, "error creating sysfs files\n");
 
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index aea27e70043..8b7766b998d 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
+
+config SND_OMAP_SOC_OSK5912
+	tristate "SoC Audio support for omap osk5912"
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y if you want to add support for SoC audio on osk5912.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index d8d8d58075e..e09d1f297f6 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
+snd-soc-osk5912-objs := osk5912.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index d166b6b2a60..fae3ad36e0b 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* Not connected */
-	snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
-	snd_soc_dapm_disable_pin(codec, "HPLCOM");
-	snd_soc_dapm_disable_pin(codec, "HPRCOM");
+	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(codec, "HPLCOM");
+	snd_soc_dapm_nc_pin(codec, "HPRCOM");
 
 	/* Add N810 specific controls */
 	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 35310e16d7f..8485a8a9d0f 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -43,6 +43,7 @@
 struct omap_mcbsp_data {
 	unsigned int			bus_id;
 	struct omap_mcbsp_reg_cfg	regs;
+	unsigned int			fmt;
 	/*
 	 * Flags indicating is the bus already activated and configured by
 	 * another substream
@@ -59,12 +60,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
  * Stream DMA parameters. DMA request line and port address are set runtime
  * since they are different between OMAP1 and later OMAPs
  */
-static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = {
-{
-	{ .name		= "I2S PCM Stereo out", },
-	{ .name		= "I2S PCM Stereo in", },
-},
-};
+static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2];
 
 #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
 static const int omap1_dma_reqs[][2] = {
@@ -84,11 +80,22 @@ static const unsigned long omap1_mcbsp_port[][2] = {
 static const int omap1_dma_reqs[][2] = {};
 static const unsigned long omap1_mcbsp_port[][2] = {};
 #endif
-#if defined(CONFIG_ARCH_OMAP2420)
-static const int omap2420_dma_reqs[][2] = {
+
+#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
+static const int omap24xx_dma_reqs[][2] = {
 	{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
 	{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+	{ OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
+	{ OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
+	{ OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
+#endif
 };
+#else
+static const int omap24xx_dma_reqs[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP2420)
 static const unsigned long omap2420_mcbsp_port[][2] = {
 	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
 	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
@@ -96,10 +103,43 @@ static const unsigned long omap2420_mcbsp_port[][2] = {
 	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
 };
 #else
-static const int omap2420_dma_reqs[][2] = {};
 static const unsigned long omap2420_mcbsp_port[][2] = {};
 #endif
 
+#if defined(CONFIG_ARCH_OMAP2430)
+static const unsigned long omap2430_mcbsp_port[][2] = {
+	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap2430_mcbsp_port[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP34XX)
+static const unsigned long omap34xx_mcbsp_port[][2] = {
+	{ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap34xx_mcbsp_port[][2] = {};
+#endif
+
 static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -161,20 +201,26 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+	int wlen;
 	unsigned long port;
 
 	if (cpu_class_is_omap1()) {
 		dma = omap1_dma_reqs[bus_id][substream->stream];
 		port = omap1_mcbsp_port[bus_id][substream->stream];
 	} else if (cpu_is_omap2420()) {
-		dma = omap2420_dma_reqs[bus_id][substream->stream];
+		dma = omap24xx_dma_reqs[bus_id][substream->stream];
 		port = omap2420_mcbsp_port[bus_id][substream->stream];
+	} else if (cpu_is_omap2430()) {
+		dma = omap24xx_dma_reqs[bus_id][substream->stream];
+		port = omap2430_mcbsp_port[bus_id][substream->stream];
+	} else if (cpu_is_omap343x()) {
+		dma = omap24xx_dma_reqs[bus_id][substream->stream];
+		port = omap34xx_mcbsp_port[bus_id][substream->stream];
 	} else {
-		/*
-		 * TODO: Add support for 2430 and 3430
-		 */
 		return -ENODEV;
 	}
+	omap_mcbsp_dai_dma_params[id][substream->stream].name =
+		substream->stream ? "Audio Capture" : "Audio Playback";
 	omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
 	omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
 	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
@@ -200,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		/* Set word lengths */
+		wlen = 16;
 		regs->rcr2	|= RWDLEN2(OMAP_MCBSP_WORD_16);
 		regs->rcr1	|= RWDLEN1(OMAP_MCBSP_WORD_16);
 		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_16);
 		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_16);
-		/* Set FS period and length in terms of bit clock periods */
-		regs->srgr2	|= FPER(16 * 2 - 1);
-		regs->srgr1	|= FWID(16 - 1);
 		break;
 	default:
 		/* Unsupported PCM format */
 		return -EINVAL;
 	}
 
+	/* Set FS period and length in terms of bit clock periods */
+	switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		regs->srgr2	|= FPER(wlen * 2 - 1);
+		regs->srgr1	|= FWID(wlen - 1);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		regs->srgr2	|= FPER(wlen * 2 - 1);
+		regs->srgr1	|= FWID(wlen * 2 - 2);
+		break;
+	}
+
 	omap_mcbsp_config(bus_id, &mcbsp_data->regs);
 	mcbsp_data->configured = 1;
 
@@ -232,6 +288,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	if (mcbsp_data->configured)
 		return 0;
 
+	mcbsp_data->fmt = fmt;
 	memset(regs, 0, sizeof(*regs));
 	/* Generic McBSP register settings */
 	regs->spcr2	|= XINTM(3) | FREE;
@@ -245,6 +302,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		regs->rcr2	|= RDATDLY(1);
 		regs->xcr2	|= XDATDLY(1);
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		/* 0-bit data delay */
+		regs->rcr2      |= RDATDLY(0);
+		regs->xcr2      |= XDATDLY(0);
+		break;
 	default:
 		/* Unsupported data format */
 		return -EINVAL;
@@ -310,7 +372,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 				       int clk_id)
 {
 	int sel_bit;
-	u16 reg;
+	u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1;
 
 	if (cpu_class_is_omap1()) {
 		/* OMAP1's can use only external source clock */
@@ -320,6 +382,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 			return 0;
 	}
 
+	if (cpu_is_omap2420() && mcbsp_data->bus_id > 1)
+		return -EINVAL;
+
+	if (cpu_is_omap343x())
+		reg_devconf1 = OMAP343X_CONTROL_DEVCONF1;
+
 	switch (mcbsp_data->bus_id) {
 	case 0:
 		reg = OMAP2_CONTROL_DEVCONF0;
@@ -329,20 +397,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 		reg = OMAP2_CONTROL_DEVCONF0;
 		sel_bit = 6;
 		break;
-	/* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */
+	case 2:
+		reg = reg_devconf1;
+		sel_bit = 0;
+		break;
+	case 3:
+		reg = reg_devconf1;
+		sel_bit = 2;
+		break;
+	case 4:
+		reg = reg_devconf1;
+		sel_bit = 4;
+		break;
 	default:
 		return -EINVAL;
 	}
 
-	if (cpu_class_is_omap2()) {
-		if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) {
-			omap_ctrl_writel(omap_ctrl_readl(reg) &
-					 ~(1 << sel_bit), reg);
-		} else {
-			omap_ctrl_writel(omap_ctrl_readl(reg) |
-					 (1 << sel_bit), reg);
-		}
-	}
+	if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK)
+		omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
+	else
+		omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
 
 	return 0;
 }
@@ -376,37 +450,49 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
-struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = {
-{
-	.name = "omap-mcbsp-dai",
-	.id = 0,
-	.type = SND_SOC_DAI_I2S,
-	.playback = {
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = OMAP_MCBSP_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-	.capture = {
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = OMAP_MCBSP_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-	.ops = {
-		.startup = omap_mcbsp_dai_startup,
-		.shutdown = omap_mcbsp_dai_shutdown,
-		.trigger = omap_mcbsp_dai_trigger,
-		.hw_params = omap_mcbsp_dai_hw_params,
-	},
-	.dai_ops = {
-		.set_fmt = omap_mcbsp_dai_set_dai_fmt,
-		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,
-		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
-	},
-	.private_data = &mcbsp_data[0].bus_id,
-},
+#define OMAP_MCBSP_DAI_BUILDER(link_id)				\
+{								\
+	.name = "omap-mcbsp-dai-(link_id)",			\
+	.id = (link_id),					\
+	.type = SND_SOC_DAI_I2S,				\
+	.playback = {						\
+		.channels_min = 2,				\
+		.channels_max = 2,				\
+		.rates = OMAP_MCBSP_RATES,			\
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
+	},							\
+	.capture = {						\
+		.channels_min = 2,				\
+		.channels_max = 2,				\
+		.rates = OMAP_MCBSP_RATES,			\
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
+	},							\
+	.ops = {						\
+		.startup = omap_mcbsp_dai_startup,		\
+		.shutdown = omap_mcbsp_dai_shutdown,		\
+		.trigger = omap_mcbsp_dai_trigger,		\
+		.hw_params = omap_mcbsp_dai_hw_params,		\
+	},							\
+	.dai_ops = {						\
+		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
+		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
+		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
+	},							\
+	.private_data = &mcbsp_data[(link_id)].bus_id,		\
+}
+
+struct snd_soc_dai omap_mcbsp_dai[] = {
+	OMAP_MCBSP_DAI_BUILDER(0),
+	OMAP_MCBSP_DAI_BUILDER(1),
+#if NUM_LINKS >= 3
+	OMAP_MCBSP_DAI_BUILDER(2),
+#endif
+#if NUM_LINKS == 5
+	OMAP_MCBSP_DAI_BUILDER(3),
+	OMAP_MCBSP_DAI_BUILDER(4),
+#endif
 };
+
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index ed8afb55067..df7ad13ba73 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -38,11 +38,17 @@ enum omap_mcbsp_div {
 	OMAP_MCBSP_CLKGDV,		/* Sample rate generator divider */
 };
 
-/*
- * REVISIT: Preparation for the ASoC v2. Let the number of available links to
- * be same than number of McBSP ports found in OMAP(s) we are compiling for.
- */
-#define NUM_LINKS	1
+#if defined(CONFIG_ARCH_OMAP2420)
+#define NUM_LINKS	2
+#endif
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+#undef  NUM_LINKS
+#define NUM_LINKS	3
+#endif
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#undef  NUM_LINKS
+#define NUM_LINKS	5
+#endif
 
 extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 690bfeaec4a..e9084fdd208 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	prtd->dma_data = dma_data;
 	err = omap_request_dma(dma_data->dma_req, dma_data->name,
 			       omap_pcm_dma_irq, substream, &prtd->dma_ch);
-	if (!cpu_is_omap1510()) {
+	if (!err & !cpu_is_omap1510()) {
 		/*
 		 * Link channel with itself so DMA doesn't need any
 		 * reprogramming while looping the buffer
@@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 		dma_params.src_or_dst_synch	= OMAP_DMA_DST_SYNC;
 		dma_params.src_start		= runtime->dma_addr;
 		dma_params.dst_start		= dma_data->port_addr;
+		dma_params.dst_port		= OMAP_DMA_PORT_MPUI;
 	} else {
 		dma_params.src_amode		= OMAP_DMA_AMODE_CONSTANT;
 		dma_params.dst_amode		= OMAP_DMA_AMODE_POST_INC;
 		dma_params.src_or_dst_synch	= OMAP_DMA_SRC_SYNC;
 		dma_params.src_start		= dma_data->port_addr;
 		dma_params.dst_start		= runtime->dma_addr;
+		dma_params.src_port		= OMAP_DMA_PORT_MPUI;
 	}
 	/*
 	 * Set DMA transfer frame size equal to ALSA period size and frame
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
new file mode 100644
index 00000000000..0fe73379689
--- /dev/null
+++ b/sound/soc/omap/osk5912.c
@@ -0,0 +1,232 @@
+/*
+ * osk5912.c  --  SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS  <arunks@mistralsolutions.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static struct clk *tlv320aic23_mclk;
+
+static int osk_startup(struct snd_pcm_substream *substream)
+{
+	return clk_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable(tlv320aic23_mclk);
+}
+
+static int osk_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_DSP_A |
+				  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_DSP_A |
+				  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 osk_ops = {
+	.startup = osk_startup,
+	.hw_params = osk_hw_params,
+	.shutdown = osk_shutdown,
+};
+
+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 osk_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+	/* Add osk5912 specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up osk5912 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_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 osk_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = osk_tlv320aic23_init,
+	.ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_osk = {
+	.name = "OSK5912",
+	.dai_link = &osk_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device osk_snd_devdata = {
+	.machine = &snd_soc_machine_osk,
+	.platform = &omap_soc_platform,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *osk_snd_device;
+
+static int __init osk_soc_init(void)
+{
+	int err;
+	u32 curRate;
+	struct device *dev;
+
+	if (!(machine_is_omap_osk()))
+		return -ENODEV;
+
+	osk_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!osk_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
+	osk_snd_devdata.dev = &osk_snd_device->dev;
+	*(unsigned int *)osk_dai.cpu_dai->private_data = 0;	/* McBSP1 */
+	err = platform_device_add(osk_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &osk_snd_device->dev;
+
+	tlv320aic23_mclk = clk_get(dev, "mclk");
+	if (IS_ERR(tlv320aic23_mclk)) {
+		printk(KERN_ERR "Could not get mclk clock\n");
+		return -ENODEV;
+	}
+
+	if (clk_get_usecount(tlv320aic23_mclk) > 0) {
+		/* MCLK is already in use */
+		printk(KERN_WARNING
+		       "MCLK in use at %d Hz. We change it to %d Hz\n",
+		       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+	}
+
+	/*
+	 * Configure 12 MHz output on MCLK.
+	 */
+	curRate = (uint) clk_get_rate(tlv320aic23_mclk);
+	if (curRate != CODEC_CLOCK) {
+		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
+			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
+			err = -ECANCELED;
+			goto err1;
+		}
+	}
+
+	printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
+	       clk_get_usecount(tlv320aic23_mclk));
+
+	return 0;
+err1:
+	clk_put(tlv320aic23_mclk);
+	platform_device_del(osk_snd_device);
+	platform_device_put(osk_snd_device);
+
+	return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+	platform_device_unregister(osk_snd_device);
+}
+
+module_init(osk_soc_init);
+module_exit(osk_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC OSK 5912");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 72b7a5140bf..2718eaf7895 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -18,13 +18,13 @@
 #include <linux/timer.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
+#include <linux/gpio.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/hardware/scoop.h>
 #include <mach/pxa-regs.h>
 #include <mach/hardware.h>
 #include <mach/corgi.h>
@@ -54,8 +54,8 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
 	switch (corgi_jack_func) {
 	case CORGI_HP:
 		/* set = unmute headphone */
-		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
-		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 1);
 		snd_soc_dapm_disable_pin(codec, "Mic Jack");
 		snd_soc_dapm_disable_pin(codec, "Line Jack");
 		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
@@ -63,24 +63,24 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
 		break;
 	case CORGI_MIC:
 		/* reset = mute headphone */
-		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
-		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 0);
 		snd_soc_dapm_enable_pin(codec, "Mic Jack");
 		snd_soc_dapm_disable_pin(codec, "Line Jack");
 		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
 		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		break;
 	case CORGI_LINE:
-		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
-		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 0);
 		snd_soc_dapm_disable_pin(codec, "Mic Jack");
 		snd_soc_dapm_enable_pin(codec, "Line Jack");
 		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
 		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		break;
 	case CORGI_HEADSET:
-		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
-		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 1);
 		snd_soc_dapm_enable_pin(codec, "Mic Jack");
 		snd_soc_dapm_disable_pin(codec, "Line Jack");
 		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
@@ -114,8 +114,8 @@ static int corgi_shutdown(struct snd_pcm_substream *substream)
 	struct snd_soc_codec *codec = rtd->socdev->codec;
 
 	/* set = unmute headphone */
-	set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
-	set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+	gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+	gpio_set_value(CORGI_GPIO_MUTE_R, 1);
 	return 0;
 }
 
@@ -218,22 +218,14 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol,
 static int corgi_amp_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *k, int event)
 {
-	if (SND_SOC_DAPM_EVENT_ON(event))
-		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
-	else
-		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
-
+	gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
 	return 0;
 }
 
 static int corgi_mic_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *k, int event)
 {
-	if (SND_SOC_DAPM_EVENT_ON(event))
-		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
-	else
-		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
-
+	gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
 	return 0;
 }
 
@@ -289,8 +281,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_disable_pin(codec, "LLINEIN");
-	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_nc_pin(codec, "LLINEIN");
+	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 
 	/* Add corgi specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index d9c3f7b28be..e6ff6929ab4 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -9,7 +9,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index f84f7d8db09..4d9930c5278 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_disable_pin(codec, "LLINEIN");
-	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_nc_pin(codec, "LLINEIN");
+	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 	snd_soc_dapm_enable_pin(codec, "MICIN");
 
 	/* Add poodle specific controls */
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a80ae074b09..a7a3a9c5c6f 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
 	.name			= "AC97 PCM Stereo out",
 	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMRTXPCDR,
+	.drcmr			= &DRCMR(12),
 	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
@@ -57,7 +57,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
 	.name			= "AC97 PCM Stereo in",
 	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMRRXPCDR,
+	.drcmr			= &DRCMR(11),
 	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
@@ -65,7 +65,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
 	.name			= "AC97 Aux PCM (Slot 5) Mono out",
 	.dev_addr		= __PREG(MODR),
-	.drcmr			= &DRCMRTXMODR,
+	.drcmr			= &DRCMR(10),
 	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
 				  DCMD_BURST16 | DCMD_WIDTH2,
 };
@@ -73,7 +73,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
 	.name			= "AC97 Aux PCM (Slot 5) Mono in",
 	.dev_addr		= __PREG(MODR),
-	.drcmr			= &DRCMRRXMODR,
+	.drcmr			= &DRCMR(9),
 	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
 				  DCMD_BURST16 | DCMD_WIDTH2,
 };
@@ -81,7 +81,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
 	.name			= "AC97 Mic PCM (Slot 6) Mono in",
 	.dev_addr		= __PREG(MCDR),
-	.drcmr			= &DRCMRRXMCDR,
+	.drcmr			= &DRCMR(8),
 	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
 				  DCMD_BURST16 | DCMD_WIDTH2,
 };
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 39d19212f6d..e758034db5c 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -3,7 +3,7 @@
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *         lrg@slimlogic.co.uk
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -39,6 +39,45 @@ struct pxa2xx_gpio {
 	u32 frm;
 };
 
+/*
+ * I2S Controller Register and Bit Definitions
+ */
+#define SACR0		__REG(0x40400000)  /* Global Control Register */
+#define SACR1		__REG(0x40400004)  /* Serial Audio I 2 S/MSB-Justified Control Register */
+#define SASR0		__REG(0x4040000C)  /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */
+#define SAIMR		__REG(0x40400014)  /* Serial Audio Interrupt Mask Register */
+#define SAICR		__REG(0x40400018)  /* Serial Audio Interrupt Clear Register */
+#define SADIV		__REG(0x40400060)  /* Audio Clock Divider Register. */
+#define SADR		__REG(0x40400080)  /* Serial Audio Data Register (TX and RX FIFO access Register). */
+
+#define SACR0_RFTH(x)	((x) << 12)	/* Rx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_TFTH(x)	((x) << 8)	/* Tx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_STRF	(1 << 5)	/* FIFO Select for EFWR Special Function */
+#define SACR0_EFWR	(1 << 4)	/* Enable EFWR Function  */
+#define SACR0_RST	(1 << 3)	/* FIFO, i2s Register Reset */
+#define SACR0_BCKD	(1 << 2) 	/* Bit Clock Direction */
+#define SACR0_ENB	(1 << 0)	/* Enable I2S Link */
+#define SACR1_ENLBF	(1 << 5)	/* Enable Loopback */
+#define SACR1_DRPL	(1 << 4) 	/* Disable Replaying Function */
+#define SACR1_DREC	(1 << 3)	/* Disable Recording Function */
+#define SACR1_AMSL	(1 << 0)	/* Specify Alternate Mode */
+
+#define SASR0_I2SOFF	(1 << 7)	/* Controller Status */
+#define SASR0_ROR	(1 << 6)	/* Rx FIFO Overrun */
+#define SASR0_TUR	(1 << 5)	/* Tx FIFO Underrun */
+#define SASR0_RFS	(1 << 4)	/* Rx FIFO Service Request */
+#define SASR0_TFS	(1 << 3)	/* Tx FIFO Service Request */
+#define SASR0_BSY	(1 << 2)	/* I2S Busy */
+#define SASR0_RNE	(1 << 1)	/* Rx FIFO Not Empty */
+#define SASR0_TNF	(1 << 0) 	/* Tx FIFO Not Empty */
+
+#define SAICR_ROR	(1 << 6)	/* Clear Rx FIFO Overrun Interrupt */
+#define SAICR_TUR	(1 << 5)	/* Clear Tx FIFO Underrun Interrupt */
+
+#define SAIMR_ROR	(1 << 6)	/* Enable Rx FIFO Overrun Condition Interrupt */
+#define SAIMR_TUR	(1 << 5)	/* Enable Tx FIFO Underrun Condition Interrupt */
+#define SAIMR_RFS	(1 << 4)	/* Enable Rx FIFO Service Interrupt */
+#define SAIMR_TFS	(1 << 3)	/* Enable Tx FIFO Service Interrupt */
 
 struct pxa_i2s_port {
 	u32 sadiv;
@@ -54,7 +93,7 @@ static struct clk *clk_i2s;
 static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
 	.name			= "I2S PCM Stereo out",
 	.dev_addr		= __PREG(SADR),
-	.drcmr			= &DRCMRTXSADR,
+	.drcmr			= &DRCMR(3),
 	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
@@ -62,7 +101,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
 static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
 	.name			= "I2S PCM Stereo in",
 	.dev_addr		= __PREG(SADR),
-	.drcmr			= &DRCMRRXSADR,
+	.drcmr			= &DRCMR(2),
 	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
@@ -366,6 +405,6 @@ module_init(pxa2xx_i2s_init);
 module_exit(pxa2xx_i2s_exit);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
 MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 3d4738c06e7..d307b6757e9 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -19,16 +19,15 @@
 #include <linux/timer.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
+#include <linux/gpio.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/hardware/scoop.h>
 #include <mach/pxa-regs.h>
 #include <mach/hardware.h>
-#include <mach/akita.h>
 #include <mach/spitz.h>
 #include "../codecs/wm8750.h"
 #include "pxa2xx-pcm.h"
@@ -63,8 +62,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 		snd_soc_dapm_disable_pin(codec, "Mic Jack");
 		snd_soc_dapm_disable_pin(codec, "Line Jack");
 		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
-		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
 		break;
 	case SPITZ_MIC:
 		/* enable mic jack and bias, mute hp */
@@ -72,8 +71,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		snd_soc_dapm_disable_pin(codec, "Line Jack");
 		snd_soc_dapm_enable_pin(codec, "Mic Jack");
-		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
-		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
 		break;
 	case SPITZ_LINE:
 		/* enable line jack, disable mic bias and mute hp */
@@ -81,8 +80,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		snd_soc_dapm_disable_pin(codec, "Mic Jack");
 		snd_soc_dapm_enable_pin(codec, "Line Jack");
-		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
-		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
 		break;
 	case SPITZ_HEADSET:
 		/* enable and unmute headset jack enable mic bias, mute L hp */
@@ -90,8 +89,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 		snd_soc_dapm_enable_pin(codec, "Mic Jack");
 		snd_soc_dapm_disable_pin(codec, "Line Jack");
 		snd_soc_dapm_enable_pin(codec, "Headset Jack");
-		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
-		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
 		break;
 	case SPITZ_HP_OFF:
 
@@ -100,8 +99,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		snd_soc_dapm_disable_pin(codec, "Mic Jack");
 		snd_soc_dapm_disable_pin(codec, "Line Jack");
-		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
-		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
 		break;
 	}
 	snd_soc_dapm_sync(codec);
@@ -215,23 +214,14 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol,
 static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *k, int event)
 {
-	if (machine_is_borzoi() || machine_is_spitz()) {
-		if (SND_SOC_DAPM_EVENT_ON(event))
-			set_scoop_gpio(&spitzscoop2_device.dev,
-				SPITZ_SCP2_MIC_BIAS);
-		else
-			reset_scoop_gpio(&spitzscoop2_device.dev,
-				SPITZ_SCP2_MIC_BIAS);
-	}
+	if (machine_is_borzoi() || machine_is_spitz())
+		gpio_set_value(SPITZ_GPIO_MIC_BIAS,
+				SND_SOC_DAPM_EVENT_ON(event));
+
+	if (machine_is_akita())
+		gpio_set_value(AKITA_GPIO_MIC_BIAS,
+				SND_SOC_DAPM_EVENT_ON(event));
 
-	if (machine_is_akita()) {
-		if (SND_SOC_DAPM_EVENT_ON(event))
-			akita_set_ioexp(&akitaioexp_device.dev,
-				AKITA_IOEXP_MIC_BIAS);
-		else
-			akita_reset_ioexp(&akitaioexp_device.dev,
-				AKITA_IOEXP_MIC_BIAS);
-	}
 	return 0;
 }
 
@@ -291,13 +281,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* NC codec pins */
-	snd_soc_dapm_disable_pin(codec, "RINPUT1");
-	snd_soc_dapm_disable_pin(codec, "LINPUT2");
-	snd_soc_dapm_disable_pin(codec, "RINPUT2");
-	snd_soc_dapm_disable_pin(codec, "LINPUT3");
-	snd_soc_dapm_disable_pin(codec, "RINPUT3");
-	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "MONO1");
+	snd_soc_dapm_nc_pin(codec, "RINPUT1");
+	snd_soc_dapm_nc_pin(codec, "LINPUT2");
+	snd_soc_dapm_nc_pin(codec, "RINPUT2");
+	snd_soc_dapm_nc_pin(codec, "LINPUT3");
+	snd_soc_dapm_nc_pin(codec, "RINPUT3");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 2baaa750f12..afefe41b8c4 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "MONOOUT");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add tosa specific controls */
 	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 73a50e93a9a..87ddfefcc2f 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -511,21 +511,20 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	DBG("Entered %s\n", __func__);
 
 	/* set up NC codec pins */
-	snd_soc_dapm_disable_pin(codec, "LOUT2");
-	snd_soc_dapm_disable_pin(codec, "ROUT2");
-	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "OUT4");
-	snd_soc_dapm_disable_pin(codec, "LINE1");
-	snd_soc_dapm_disable_pin(codec, "LINE2");
-
-
-	/* set endpoints to default mode */
-	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+	snd_soc_dapm_nc_pin(codec, "LOUT2");
+	snd_soc_dapm_nc_pin(codec, "ROUT2");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "OUT4");
+	snd_soc_dapm_nc_pin(codec, "LINE1");
+	snd_soc_dapm_nc_pin(codec, "LINE2");
 
 	/* Add neo1973 specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
 				  ARRAY_SIZE(wm8753_dapm_widgets));
 
+	/* set endpoints to default mode */
+	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+
 	/* add neo1973 specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
 		err = snd_ctl_add(codec->card,
@@ -603,6 +602,8 @@ static int lm4857_i2c_probe(struct i2c_client *client,
 {
 	DBG("Entered %s\n", __func__);
 
+	i2c = client;
+
 	lm4857_write_regs();
 	return 0;
 }
@@ -611,6 +612,8 @@ static int lm4857_i2c_remove(struct i2c_client *client)
 {
 	DBG("Entered %s\n", __func__);
 
+	i2c = NULL;
+
 	return 0;
 }
 
@@ -650,7 +653,7 @@ static void lm4857_shutdown(struct i2c_client *dev)
 }
 
 static const struct i2c_device_id lm4857_i2c_id[] = {
-	{ "neo1973_lm4857", 0 }
+	{ "neo1973_lm4857", 0 },
 	{ }
 };
 
@@ -668,48 +671,6 @@ static struct i2c_driver lm4857_i2c_driver = {
 };
 
 static struct platform_device *neo1973_snd_device;
-static struct i2c_client *lm4857_client;
-
-static int __init neo1973_add_lm4857_device(struct platform_device *pdev,
-					    int i2c_bus,
-					    unsigned short i2c_address)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&lm4857_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add lm4857 driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = i2c_address;
-	strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
-	}
-
-	lm4857_client = client;
-	return 0;
-
-err_driver:
-	i2c_del_driver(&lm4857_i2c_driver);
-	return -ENODEV;
-}
 
 static int __init neo1973_init(void)
 {
@@ -736,8 +697,8 @@ static int __init neo1973_init(void)
 		return ret;
 	}
 
-	ret = neo1973_add_lm4857_device(neo1973_snd_device,
-					neo1973_wm8753_setup, 0x7C);
+	ret = i2c_add_driver(&lm4857_i2c_driver);
+
 	if (ret != 0)
 		platform_device_unregister(neo1973_snd_device);
 
@@ -748,7 +709,6 @@ static void __exit neo1973_exit(void)
 {
 	DBG("Entered %s\n", __func__);
 
-	i2c_unregister_device(lm4857_client);
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
 }
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ad381138fc2..16c7453f494 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -4,8 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *         with code, comments and ideas from :-
  *         Richard Purdie <richard@openedhand.com>
  *
@@ -96,8 +95,8 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 	codec->ac97->dev.parent = NULL;
 	codec->ac97->dev.release = soc_ac97_device_release;
 
-	snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
-		 codec->card->number, 0, codec->name);
+	dev_set_name(&codec->ac97->dev, "%d-%d:%s",
+		     codec->card->number, 0, codec->name);
 	err = device_register(&codec->ac97->dev);
 	if (err < 0) {
 		snd_printk(KERN_ERR "Can't register ac97 bus\n");
@@ -1463,7 +1462,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	int max = mc->max;
-	unsigned int shift = mc->min;
+	unsigned int shift = mc->shift;
 	unsigned int rshift = mc->rshift;
 
 	if (max == 1)
@@ -1886,7 +1885,7 @@ module_init(snd_soc_init);
 module_exit(snd_soc_exit);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
 MODULE_DESCRIPTION("ALSA SoC Core");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS("platform:soc-audio");
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 9ca9c08610f..7351db9606e 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2,8 +2,7 @@
  * soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -832,7 +831,7 @@ int snd_soc_dapm_sys_add(struct device *dev)
 		return ret;
 
 	asoc_debugfs = debugfs_create_dir("asoc", NULL);
-	if (!IS_ERR(asoc_debugfs))
+	if (!IS_ERR(asoc_debugfs) && asoc_debugfs)
 		debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
 				   &pop_time);
 	else
@@ -1484,6 +1483,26 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
 EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
 
 /**
+ * snd_soc_dapm_nc_pin - permanently disable pin.
+ * @codec: SoC codec
+ * @pin: pin name
+ *
+ * Marks the specified pin as being not connected, disabling it along
+ * any parent or child widgets.  At present this is identical to
+ * snd_soc_dapm_disable_pin() but in future it will be extended to do
+ * additional things such as disabling controls which only affect
+ * paths through the pin.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin)
+{
+	return snd_soc_dapm_set_pin(codec, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
+
+/**
  * snd_soc_dapm_get_pin_status - get audio pin status
  * @codec: audio codec
  * @pin: audio signal pin endpoint (or start point)
@@ -1521,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev)
 EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
 MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
 MODULE_LICENSE("GPL");
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 4ae07e236b3..10ba4218161 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -57,7 +57,7 @@ module_exit(cleanup_soundcore);
 /*
  *	OSS sound core handling. Breaks out sound functions to submodules
  *	
- *	Author:		Alan Cox <alan.cox@linux.org>
+ *	Author:		Alan Cox <alan@lxorguk.ukuu.org.uk>
  *
  *	Fixes:
  *
@@ -220,9 +220,8 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati
 	else
 		sprintf(s->name, "sound/%s%d", name, r / SOUND_STEP);
 
-	device_create_drvdata(sound_class, dev,
-			      MKDEV(SOUND_MAJOR, s->unit_minor),
-			      NULL, s->name+6);
+	device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor),
+		      NULL, s->name+6);
 	return r;
 
  fail:
@@ -458,7 +457,7 @@ EXPORT_SYMBOL(unregister_sound_mixer);
 
 void unregister_sound_midi(int unit)
 {
-	return sound_remove_unit(&chains[2], unit);
+	sound_remove_unit(&chains[2], unit);
 }
 
 EXPORT_SYMBOL(unregister_sound_midi);
@@ -475,7 +474,7 @@ EXPORT_SYMBOL(unregister_sound_midi);
 
 void unregister_sound_dsp(int unit)
 {
-	return sound_remove_unit(&chains[3], unit);
+	sound_remove_unit(&chains[3], unit);
 }
 
 
@@ -508,7 +507,7 @@ static struct sound_unit *__look_for_unit(int chain, int unit)
 	return NULL;
 }
 
-int soundcore_open(struct inode *inode, struct file *file)
+static int soundcore_open(struct inode *inode, struct file *file)
 {
 	int chain;
 	int unit = iminor(inode);
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index 49acee0c484..f87933e4881 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -1,6 +1,6 @@
 /*
  * Driver for AMD7930 sound chips found on Sparcs.
- * Copyright (C) 2002 David S. Miller <davem@redhat.com>
+ * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net>
  *
  * Based entirely upon drivers/sbus/audio/amd7930.c which is:
  * Copyright (C) 1996,1997 Thomas K. Dyas (tdyas@eden.rutgers.edu)
@@ -35,6 +35,8 @@
 #include <linux/init.h>
 #include <linux/interrupt.h>
 #include <linux/moduleparam.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -44,7 +46,6 @@
 
 #include <asm/io.h>
 #include <asm/irq.h>
-#include <asm/sbus.h>
 #include <asm/prom.h>
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
@@ -335,8 +336,8 @@ struct snd_amd7930 {
 	int			pgain;
 	int			mgain;
 
+	struct of_device	*op;
 	unsigned int		irq;
-	unsigned int		regs_size;
 	struct snd_amd7930	*next;
 };
 
@@ -905,13 +906,16 @@ static int __devinit snd_amd7930_mixer(struct snd_amd7930 *amd)
 
 static int snd_amd7930_free(struct snd_amd7930 *amd)
 {
+	struct of_device *op = amd->op;
+
 	amd7930_idle(amd);
 
 	if (amd->irq)
 		free_irq(amd->irq, amd);
 
 	if (amd->regs)
-		sbus_iounmap(amd->regs, amd->regs_size);
+		of_iounmap(&op->resource[0], amd->regs,
+			   resource_size(&op->resource[0]));
 
 	kfree(amd);
 
@@ -930,13 +934,12 @@ static struct snd_device_ops snd_amd7930_dev_ops = {
 };
 
 static int __devinit snd_amd7930_create(struct snd_card *card,
-					struct resource *rp,
-					unsigned int reg_size,
+					struct of_device *op,
 					int irq, int dev,
 					struct snd_amd7930 **ramd)
 {
-	unsigned long flags;
 	struct snd_amd7930 *amd;
+	unsigned long flags;
 	int err;
 
 	*ramd = NULL;
@@ -946,9 +949,10 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
 
 	spin_lock_init(&amd->lock);
 	amd->card = card;
-	amd->regs_size = reg_size;
+	amd->op = op;
 
-	amd->regs = sbus_ioremap(rp, 0, amd->regs_size, "amd7930");
+	amd->regs = of_ioremap(&op->resource[0], 0,
+			       resource_size(&op->resource[0]), "amd7930");
 	if (!amd->regs) {
 		snd_printk("amd7930-%d: Unable to map chip registers.\n", dev);
 		return -EIO;
@@ -997,12 +1001,15 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
 	return 0;
 }
 
-static int __devinit amd7930_attach_common(struct resource *rp, int irq)
+static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_device_id *match)
 {
+	struct resource *rp = &op->resource[0];
 	static int dev_num;
 	struct snd_card *card;
 	struct snd_amd7930 *amd;
-	int err;
+	int err, irq;
+
+	irq = op->irqs[0];
 
 	if (dev_num >= SNDRV_CARDS)
 		return -ENODEV;
@@ -1023,8 +1030,7 @@ static int __devinit amd7930_attach_common(struct resource *rp, int irq)
 		(unsigned long long)rp->start,
 		irq);
 
-	if ((err = snd_amd7930_create(card, rp,
-				      (rp->end - rp->start) + 1,
+	if ((err = snd_amd7930_create(card, op,
 				      irq, dev_num, &amd)) < 0)
 		goto out_err;
 
@@ -1049,43 +1055,7 @@ out_err:
 	return err;
 }
 
-static int __devinit amd7930_obio_attach(struct device_node *dp)
-{
-	const struct linux_prom_registers *regs;
-	const struct linux_prom_irqs *irqp;
-	struct resource res, *rp;
-	int len;
-
-	irqp = of_get_property(dp, "intr", &len);
-	if (!irqp) {
-		snd_printk("%s: Firmware node lacks IRQ property.\n",
-			   dp->full_name);
-		return -ENODEV;
-	}
-
-	regs = of_get_property(dp, "reg", &len);
-	if (!regs) {
-		snd_printk("%s: Firmware node lacks register property.\n",
-			   dp->full_name);
-		return -ENODEV;
-	}
-
-	rp = &res;
-	rp->start = regs->phys_addr;
-	rp->end = rp->start + regs->reg_size - 1;
-	rp->flags = IORESOURCE_IO | (regs->which_io & 0xff);
-
-	return amd7930_attach_common(rp, irqp->pri);
-}
-
-static int __devinit amd7930_sbus_probe(struct of_device *dev, const struct of_device_id *match)
-{
-	struct sbus_dev *sdev = to_sbus_device(&dev->dev);
-
-	return amd7930_attach_common(&sdev->resource[0], sdev->irqs[0]);
-}
-
-static struct of_device_id amd7930_match[] = {
+static const struct of_device_id amd7930_match[] = {
 	{
 		.name = "audio",
 	},
@@ -1100,20 +1070,7 @@ static struct of_platform_driver amd7930_sbus_driver = {
 
 static int __init amd7930_init(void)
 {
-	struct device_node *dp;
-
-	/* Try to find the sun4c "audio" node first. */
-	dp = of_find_node_by_path("/");
-	dp = dp->child;
-	while (dp) {
-		if (!strcmp(dp->name, "audio"))
-			amd7930_obio_attach(dp);
-
-		dp = dp->sibling;
-	}
-
-	/* Probe each SBUS for amd7930 chips. */
-	return of_register_driver(&amd7930_sbus_driver, &sbus_bus_type);
+	return of_register_driver(&amd7930_sbus_driver, &of_bus_type);
 }
 
 static void __exit amd7930_exit(void)
diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c
index 791d2fb821d..d44bf98e965 100644
--- a/sound/sparc/cs4231.c
+++ b/sound/sparc/cs4231.c
@@ -1,6 +1,6 @@
 /*
  * Driver for CS4231 sound chips found on Sparcs.
- * Copyright (C) 2002 David S. Miller <davem@redhat.com>
+ * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net>
  *
  * Based entirely upon drivers/sbus/audio/cs4231.c which is:
  * Copyright (C) 1996, 1997, 1998 Derrick J Brashear (shadow@andrew.cmu.edu)
@@ -17,7 +17,8 @@
 #include <linux/moduleparam.h>
 #include <linux/irq.h>
 #include <linux/io.h>
-
+#include <linux/of.h>
+#include <linux/of_device.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -29,13 +30,12 @@
 
 #ifdef CONFIG_SBUS
 #define SBUS_SUPPORT
-#include <asm/sbus.h>
 #endif
 
 #if defined(CONFIG_PCI) && defined(CONFIG_SPARC64)
 #define EBUS_SUPPORT
 #include <linux/pci.h>
-#include <asm/ebus.h>
+#include <asm/ebus_dma.h>
 #endif
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
@@ -70,8 +70,6 @@ struct cs4231_dma_control {
 	int		(*request)(struct cs4231_dma_control *dma_cont,
 				   dma_addr_t bus_addr, size_t len);
 	unsigned int	(*address)(struct cs4231_dma_control *dma_cont);
-	void		(*preallocate)(struct snd_cs4231 *chip,
-				       struct snd_pcm *pcm);
 #ifdef EBUS_SUPPORT
 	struct		ebus_dma_info	ebus_info;
 #endif
@@ -114,21 +112,12 @@ struct snd_cs4231 {
 	struct mutex		mce_mutex;	/* mutex for mce register */
 	struct mutex		open_mutex;	/* mutex for ALSA open/close */
 
-	union {
-#ifdef SBUS_SUPPORT
-		struct sbus_dev		*sdev;
-#endif
-#ifdef EBUS_SUPPORT
-		struct pci_dev		*pdev;
-#endif
-	} dev_u;
+	struct of_device	*op;
 	unsigned int		irq[2];
 	unsigned int		regs_size;
 	struct snd_cs4231	*next;
 };
 
-static struct snd_cs4231 *cs4231_list;
-
 /* Eventually we can use sound/isa/cs423x/cs4231_lib.c directly, but for
  * now....  -DaveM
  */
@@ -267,27 +256,19 @@ static unsigned char snd_cs4231_original_image[32] =
 
 static u8 __cs4231_readb(struct snd_cs4231 *cp, void __iomem *reg_addr)
 {
-#ifdef EBUS_SUPPORT
 	if (cp->flags & CS4231_FLAG_EBUS)
 		return readb(reg_addr);
 	else
-#endif
-#ifdef SBUS_SUPPORT
 		return sbus_readb(reg_addr);
-#endif
 }
 
 static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val,
 			    void __iomem *reg_addr)
 {
-#ifdef EBUS_SUPPORT
 	if (cp->flags & CS4231_FLAG_EBUS)
 		return writeb(val, reg_addr);
 	else
-#endif
-#ifdef SBUS_SUPPORT
 		return sbus_writeb(val, reg_addr);
-#endif
 }
 
 /*
@@ -1258,7 +1239,9 @@ static int __init snd_cs4231_pcm(struct snd_card *card)
 	pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
 	strcpy(pcm->name, "CS4231");
 
-	chip->p_dma.preallocate(chip, pcm);
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+					      &chip->op->dev,
+					      64 * 1024, 128 * 1024);
 
 	chip->pcm = pcm;
 
@@ -1627,8 +1610,7 @@ static int __init cs4231_attach_finish(struct snd_card *card)
 	if (err < 0)
 		goto out_err;
 
-	chip->next = cs4231_list;
-	cs4231_list = chip;
+	dev_set_drvdata(&chip->op->dev, chip);
 
 	dev++;
 	return 0;
@@ -1783,24 +1765,19 @@ static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont)
 	return sbus_readl(base->regs + base->dir + APCVA);
 }
 
-static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm)
-{
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_SBUS,
-					snd_dma_sbus_data(chip->dev_u.sdev),
-					64 * 1024, 128 * 1024);
-}
-
 /*
  * Init and exit routines
  */
 
 static int snd_cs4231_sbus_free(struct snd_cs4231 *chip)
 {
+	struct of_device *op = chip->op;
+
 	if (chip->irq[0])
 		free_irq(chip->irq[0], chip);
 
 	if (chip->port)
-		sbus_iounmap(chip->port, chip->regs_size);
+		of_iounmap(&op->resource[0], chip->port, chip->regs_size);
 
 	return 0;
 }
@@ -1817,7 +1794,7 @@ static struct snd_device_ops snd_cs4231_sbus_dev_ops = {
 };
 
 static int __init snd_cs4231_sbus_create(struct snd_card *card,
-					 struct sbus_dev *sdev,
+					 struct of_device *op,
 					 int dev)
 {
 	struct snd_cs4231 *chip = card->private_data;
@@ -1828,13 +1805,13 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card,
 	spin_lock_init(&chip->p_dma.sbus_info.lock);
 	mutex_init(&chip->mce_mutex);
 	mutex_init(&chip->open_mutex);
-	chip->dev_u.sdev = sdev;
-	chip->regs_size = sdev->reg_addrs[0].reg_size;
+	chip->op = op;
+	chip->regs_size = resource_size(&op->resource[0]);
 	memcpy(&chip->image, &snd_cs4231_original_image,
 	       sizeof(snd_cs4231_original_image));
 
-	chip->port = sbus_ioremap(&sdev->resource[0], 0,
-				  chip->regs_size, "cs4231");
+	chip->port = of_ioremap(&op->resource[0], 0,
+				chip->regs_size, "cs4231");
 	if (!chip->port) {
 		snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev);
 		return -EIO;
@@ -1849,22 +1826,20 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card,
 	chip->p_dma.enable = sbus_dma_enable;
 	chip->p_dma.request = sbus_dma_request;
 	chip->p_dma.address = sbus_dma_addr;
-	chip->p_dma.preallocate = sbus_dma_preallocate;
 
 	chip->c_dma.prepare = sbus_dma_prepare;
 	chip->c_dma.enable = sbus_dma_enable;
 	chip->c_dma.request = sbus_dma_request;
 	chip->c_dma.address = sbus_dma_addr;
-	chip->c_dma.preallocate = sbus_dma_preallocate;
 
-	if (request_irq(sdev->irqs[0], snd_cs4231_sbus_interrupt,
+	if (request_irq(op->irqs[0], snd_cs4231_sbus_interrupt,
 			IRQF_SHARED, "cs4231", chip)) {
 		snd_printdd("cs4231-%d: Unable to grab SBUS IRQ %d\n",
-			    dev, sdev->irqs[0]);
+			    dev, op->irqs[0]);
 		snd_cs4231_sbus_free(chip);
 		return -EBUSY;
 	}
-	chip->irq[0] = sdev->irqs[0];
+	chip->irq[0] = op->irqs[0];
 
 	if (snd_cs4231_probe(chip) < 0) {
 		snd_cs4231_sbus_free(chip);
@@ -1881,9 +1856,9 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card,
 	return 0;
 }
 
-static int __init cs4231_sbus_attach(struct sbus_dev *sdev)
+static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match)
 {
-	struct resource *rp = &sdev->resource[0];
+	struct resource *rp = &op->resource[0];
 	struct snd_card *card;
 	int err;
 
@@ -1895,9 +1870,9 @@ static int __init cs4231_sbus_attach(struct sbus_dev *sdev)
 		card->shortname,
 		rp->flags & 0xffL,
 		(unsigned long long)rp->start,
-		sdev->irqs[0]);
+		op->irqs[0]);
 
-	err = snd_cs4231_sbus_create(card, sdev, dev);
+	err = snd_cs4231_sbus_create(card, op, dev);
 	if (err < 0) {
 		snd_card_free(card);
 		return err;
@@ -1950,30 +1925,25 @@ static unsigned int _ebus_dma_addr(struct cs4231_dma_control *dma_cont)
 	return ebus_dma_addr(&dma_cont->ebus_info);
 }
 
-static void _ebus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm)
-{
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
-				      snd_dma_pci_data(chip->dev_u.pdev),
-				      64*1024, 128*1024);
-}
-
 /*
  * Init and exit routines
  */
 
 static int snd_cs4231_ebus_free(struct snd_cs4231 *chip)
 {
+	struct of_device *op = chip->op;
+
 	if (chip->c_dma.ebus_info.regs) {
 		ebus_dma_unregister(&chip->c_dma.ebus_info);
-		iounmap(chip->c_dma.ebus_info.regs);
+		of_iounmap(&op->resource[2], chip->c_dma.ebus_info.regs, 0x10);
 	}
 	if (chip->p_dma.ebus_info.regs) {
 		ebus_dma_unregister(&chip->p_dma.ebus_info);
-		iounmap(chip->p_dma.ebus_info.regs);
+		of_iounmap(&op->resource[1], chip->p_dma.ebus_info.regs, 0x10);
 	}
 
 	if (chip->port)
-		iounmap(chip->port);
+		of_iounmap(&op->resource[0], chip->port, 0x10);
 
 	return 0;
 }
@@ -1990,7 +1960,7 @@ static struct snd_device_ops snd_cs4231_ebus_dev_ops = {
 };
 
 static int __init snd_cs4231_ebus_create(struct snd_card *card,
-					 struct linux_ebus_device *edev,
+					 struct of_device *op,
 					 int dev)
 {
 	struct snd_cs4231 *chip = card->private_data;
@@ -2002,35 +1972,35 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card,
 	mutex_init(&chip->mce_mutex);
 	mutex_init(&chip->open_mutex);
 	chip->flags |= CS4231_FLAG_EBUS;
-	chip->dev_u.pdev = edev->bus->self;
+	chip->op = op;
 	memcpy(&chip->image, &snd_cs4231_original_image,
 	       sizeof(snd_cs4231_original_image));
 	strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)");
 	chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER;
 	chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback;
 	chip->c_dma.ebus_info.client_cookie = chip;
-	chip->c_dma.ebus_info.irq = edev->irqs[0];
+	chip->c_dma.ebus_info.irq = op->irqs[0];
 	strcpy(chip->p_dma.ebus_info.name, "cs4231(play)");
 	chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER;
 	chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback;
 	chip->p_dma.ebus_info.client_cookie = chip;
-	chip->p_dma.ebus_info.irq = edev->irqs[1];
+	chip->p_dma.ebus_info.irq = op->irqs[1];
 
 	chip->p_dma.prepare = _ebus_dma_prepare;
 	chip->p_dma.enable = _ebus_dma_enable;
 	chip->p_dma.request = _ebus_dma_request;
 	chip->p_dma.address = _ebus_dma_addr;
-	chip->p_dma.preallocate = _ebus_dma_preallocate;
 
 	chip->c_dma.prepare = _ebus_dma_prepare;
 	chip->c_dma.enable = _ebus_dma_enable;
 	chip->c_dma.request = _ebus_dma_request;
 	chip->c_dma.address = _ebus_dma_addr;
-	chip->c_dma.preallocate = _ebus_dma_preallocate;
 
-	chip->port = ioremap(edev->resource[0].start, 0x10);
-	chip->p_dma.ebus_info.regs = ioremap(edev->resource[1].start, 0x10);
-	chip->c_dma.ebus_info.regs = ioremap(edev->resource[2].start, 0x10);
+	chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231");
+	chip->p_dma.ebus_info.regs =
+		of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma");
+	chip->c_dma.ebus_info.regs =
+		of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma");
 	if (!chip->port || !chip->p_dma.ebus_info.regs ||
 	    !chip->c_dma.ebus_info.regs) {
 		snd_cs4231_ebus_free(chip);
@@ -2078,7 +2048,7 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card,
 	return 0;
 }
 
-static int __init cs4231_ebus_attach(struct linux_ebus_device *edev)
+static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match)
 {
 	struct snd_card *card;
 	int err;
@@ -2089,10 +2059,10 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev)
 
 	sprintf(card->longname, "%s at 0x%lx, irq %d",
 		card->shortname,
-		edev->resource[0].start,
-		edev->irqs[0]);
+		op->resource[0].start,
+		op->irqs[0]);
 
-	err = snd_cs4231_ebus_create(card, edev, dev);
+	err = snd_cs4231_ebus_create(card, op, dev);
 	if (err < 0) {
 		snd_card_free(card);
 		return err;
@@ -2102,68 +2072,57 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev)
 }
 #endif
 
-static int __init cs4231_init(void)
+static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match)
 {
-#ifdef SBUS_SUPPORT
-	struct sbus_bus *sbus;
-	struct sbus_dev *sdev;
-#endif
 #ifdef EBUS_SUPPORT
-	struct linux_ebus *ebus;
-	struct linux_ebus_device *edev;
+	if (!strcmp(op->node->parent->name, "ebus"))
+		return cs4231_ebus_probe(op, match);
 #endif
-	int found;
-
-	found = 0;
-
 #ifdef SBUS_SUPPORT
-	for_all_sbusdev(sdev, sbus) {
-		if (!strcmp(sdev->prom_name, "SUNW,CS4231")) {
-			if (cs4231_sbus_attach(sdev) == 0)
-				found++;
-		}
-	}
+	if (!strcmp(op->node->parent->name, "sbus") ||
+	    !strcmp(op->node->parent->name, "sbi"))
+		return cs4231_sbus_probe(op, match);
 #endif
-#ifdef EBUS_SUPPORT
-	for_each_ebus(ebus) {
-		for_each_ebusdev(edev, ebus) {
-			int match = 0;
-
-			if (!strcmp(edev->prom_node->name, "SUNW,CS4231")) {
-				match = 1;
-			} else if (!strcmp(edev->prom_node->name, "audio")) {
-				const char *compat;
-
-				compat = of_get_property(edev->prom_node,
-							 "compatible", NULL);
-				if (compat && !strcmp(compat, "SUNW,CS4231"))
-					match = 1;
-			}
+	return -ENODEV;
+}
 
-			if (match &&
-			    cs4231_ebus_attach(edev) == 0)
-				found++;
-		}
-	}
-#endif
+static int __devexit cs4231_remove(struct of_device *op)
+{
+	struct snd_cs4231 *chip = dev_get_drvdata(&op->dev);
 
+	snd_card_free(chip->card);
 
-	return (found > 0) ? 0 : -EIO;
+	return 0;
 }
 
-static void __exit cs4231_exit(void)
-{
-	struct snd_cs4231 *p = cs4231_list;
+static const struct of_device_id cs4231_match[] = {
+	{
+		.name = "SUNW,CS4231",
+	},
+	{
+		.name = "audio",
+		.compatible = "SUNW,CS4231",
+	},
+	{},
+};
 
-	while (p != NULL) {
-		struct snd_cs4231 *next = p->next;
+MODULE_DEVICE_TABLE(of, cs4231_match);
 
-		snd_card_free(p->card);
+static struct of_platform_driver cs4231_driver = {
+	.name		= "audio",
+	.match_table	= cs4231_match,
+	.probe		= cs4231_probe,
+	.remove		= __devexit_p(cs4231_remove),
+};
 
-		p = next;
-	}
+static int __init cs4231_init(void)
+{
+	return of_register_driver(&cs4231_driver, &of_bus_type);
+}
 
-	cs4231_list = NULL;
+static void __exit cs4231_exit(void)
+{
+	of_unregister_driver(&cs4231_driver);
 }
 
 module_init(cs4231_init);
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index c534a2a849f..23ed6f04a71 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -57,6 +57,7 @@
 #include <linux/delay.h>
 #include <linux/irq.h>
 #include <linux/io.h>
+#include <linux/dma-mapping.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -66,7 +67,7 @@
 #include <sound/initval.h>
 
 #include <linux/of.h>
-#include <asm/sbus.h>
+#include <linux/of_device.h>
 #include <asm/atomic.h>
 
 MODULE_AUTHOR("Rudolf Koenig, Brent Baccala and Martin Habets");
@@ -297,7 +298,7 @@ struct dbri_streaminfo {
 /* This structure holds the information for both chips (DBRI & CS4215) */
 struct snd_dbri {
 	int regs_size, irq;	/* Needed for unload */
-	struct sbus_dev *sdev;	/* SBUS device info */
+	struct of_device *op;	/* OF device info */
 	spinlock_t lock;
 
 	struct dbri_dma *dma;	/* Pointer to our DMA block */
@@ -2093,14 +2094,15 @@ static int snd_dbri_hw_params(struct snd_pcm_substream *substream,
 	 */
 	if (info->dvma_buffer == 0) {
 		if (DBRI_STREAMNO(substream) == DBRI_PLAY)
-			direction = SBUS_DMA_TODEVICE;
+			direction = DMA_TO_DEVICE;
 		else
-			direction = SBUS_DMA_FROMDEVICE;
+			direction = DMA_FROM_DEVICE;
 
-		info->dvma_buffer = sbus_map_single(dbri->sdev,
-					runtime->dma_area,
-					params_buffer_bytes(hw_params),
-					direction);
+		info->dvma_buffer =
+			dma_map_single(&dbri->op->dev,
+				       runtime->dma_area,
+				       params_buffer_bytes(hw_params),
+				       direction);
 	}
 
 	direction = params_buffer_bytes(hw_params);
@@ -2121,12 +2123,12 @@ static int snd_dbri_hw_free(struct snd_pcm_substream *substream)
 	 */
 	if (info->dvma_buffer) {
 		if (DBRI_STREAMNO(substream) == DBRI_PLAY)
-			direction = SBUS_DMA_TODEVICE;
+			direction = DMA_TO_DEVICE;
 		else
-			direction = SBUS_DMA_FROMDEVICE;
+			direction = DMA_FROM_DEVICE;
 
-		sbus_unmap_single(dbri->sdev, info->dvma_buffer,
-				  substream->runtime->buffer_size, direction);
+		dma_unmap_single(&dbri->op->dev, info->dvma_buffer,
+				 substream->runtime->buffer_size, direction);
 		info->dvma_buffer = 0;
 	}
 	if (info->pipe != -1) {
@@ -2519,31 +2521,34 @@ static void __devinit snd_dbri_proc(struct snd_card *card)
 static void snd_dbri_free(struct snd_dbri *dbri);
 
 static int __devinit snd_dbri_create(struct snd_card *card,
-				  struct sbus_dev *sdev,
-				  int irq, int dev)
+				     struct of_device *op,
+				     int irq, int dev)
 {
 	struct snd_dbri *dbri = card->private_data;
 	int err;
 
 	spin_lock_init(&dbri->lock);
-	dbri->sdev = sdev;
+	dbri->op = op;
 	dbri->irq = irq;
 
-	dbri->dma = sbus_alloc_consistent(sdev, sizeof(struct dbri_dma),
-					  &dbri->dma_dvma);
+	dbri->dma = dma_alloc_coherent(&op->dev,
+				       sizeof(struct dbri_dma),
+				       &dbri->dma_dvma, GFP_ATOMIC);
+	if (!dbri->dma)
+		return -ENOMEM;
 	memset((void *)dbri->dma, 0, sizeof(struct dbri_dma));
 
 	dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n",
 		dbri->dma, dbri->dma_dvma);
 
 	/* Map the registers into memory. */
-	dbri->regs_size = sdev->reg_addrs[0].reg_size;
-	dbri->regs = sbus_ioremap(&sdev->resource[0], 0,
-				  dbri->regs_size, "DBRI Registers");
+	dbri->regs_size = resource_size(&op->resource[0]);
+	dbri->regs = of_ioremap(&op->resource[0], 0,
+				dbri->regs_size, "DBRI Registers");
 	if (!dbri->regs) {
 		printk(KERN_ERR "DBRI: could not allocate registers\n");
-		sbus_free_consistent(sdev, sizeof(struct dbri_dma),
-				     (void *)dbri->dma, dbri->dma_dvma);
+		dma_free_coherent(&op->dev, sizeof(struct dbri_dma),
+				  (void *)dbri->dma, dbri->dma_dvma);
 		return -EIO;
 	}
 
@@ -2551,9 +2556,9 @@ static int __devinit snd_dbri_create(struct snd_card *card,
 			  "DBRI audio", dbri);
 	if (err) {
 		printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq);
-		sbus_iounmap(dbri->regs, dbri->regs_size);
-		sbus_free_consistent(sdev, sizeof(struct dbri_dma),
-				     (void *)dbri->dma, dbri->dma_dvma);
+		of_iounmap(&op->resource[0], dbri->regs, dbri->regs_size);
+		dma_free_coherent(&op->dev, sizeof(struct dbri_dma),
+				  (void *)dbri->dma, dbri->dma_dvma);
 		return err;
 	}
 
@@ -2577,27 +2582,23 @@ static void snd_dbri_free(struct snd_dbri *dbri)
 		free_irq(dbri->irq, dbri);
 
 	if (dbri->regs)
-		sbus_iounmap(dbri->regs, dbri->regs_size);
+		of_iounmap(&dbri->op->resource[0], dbri->regs, dbri->regs_size);
 
 	if (dbri->dma)
-		sbus_free_consistent(dbri->sdev, sizeof(struct dbri_dma),
-				     (void *)dbri->dma, dbri->dma_dvma);
+		dma_free_coherent(&dbri->op->dev,
+				  sizeof(struct dbri_dma),
+				  (void *)dbri->dma, dbri->dma_dvma);
 }
 
-static int __devinit dbri_probe(struct of_device *of_dev,
-				const struct of_device_id *match)
+static int __devinit dbri_probe(struct of_device *op, const struct of_device_id *match)
 {
-	struct sbus_dev *sdev = to_sbus_device(&of_dev->dev);
 	struct snd_dbri *dbri;
-	int irq;
 	struct resource *rp;
 	struct snd_card *card;
 	static int dev = 0;
+	int irq;
 	int err;
 
-	dprintk(D_GEN, "DBRI: Found %s in SBUS slot %d\n",
-		sdev->prom_name, sdev->slot);
-
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 	if (!enable[dev]) {
@@ -2605,7 +2606,7 @@ static int __devinit dbri_probe(struct of_device *of_dev,
 		return -ENOENT;
 	}
 
-	irq = sdev->irqs[0];
+	irq = op->irqs[0];
 	if (irq <= 0) {
 		printk(KERN_ERR "DBRI-%d: No IRQ.\n", dev);
 		return -ENODEV;
@@ -2618,12 +2619,12 @@ static int __devinit dbri_probe(struct of_device *of_dev,
 
 	strcpy(card->driver, "DBRI");
 	strcpy(card->shortname, "Sun DBRI");
-	rp = &sdev->resource[0];
+	rp = &op->resource[0];
 	sprintf(card->longname, "%s at 0x%02lx:0x%016Lx, irq %d",
 		card->shortname,
 		rp->flags & 0xffL, (unsigned long long)rp->start, irq);
 
-	err = snd_dbri_create(card, sdev, irq, dev);
+	err = snd_dbri_create(card, op, irq, dev);
 	if (err < 0) {
 		snd_card_free(card);
 		return err;
@@ -2640,7 +2641,7 @@ static int __devinit dbri_probe(struct of_device *of_dev,
 
 	/* /proc file handling */
 	snd_dbri_proc(card);
-	dev_set_drvdata(&of_dev->dev, card);
+	dev_set_drvdata(&op->dev, card);
 
 	err = snd_card_register(card);
 	if (err < 0)
@@ -2648,7 +2649,7 @@ static int __devinit dbri_probe(struct of_device *of_dev,
 
 	printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c)+CS4215(%d)\n",
 	       dev, dbri->regs,
-	       dbri->irq, sdev->prom_name[9], dbri->mm.version);
+	       dbri->irq, op->node->name[9], dbri->mm.version);
 	dev++;
 
 	return 0;
@@ -2659,19 +2660,19 @@ _err:
 	return err;
 }
 
-static int __devexit dbri_remove(struct of_device *dev)
+static int __devexit dbri_remove(struct of_device *op)
 {
-	struct snd_card *card = dev_get_drvdata(&dev->dev);
+	struct snd_card *card = dev_get_drvdata(&op->dev);
 
 	snd_dbri_free(card->private_data);
 	snd_card_free(card);
 
-	dev_set_drvdata(&dev->dev, NULL);
+	dev_set_drvdata(&op->dev, NULL);
 
 	return 0;
 }
 
-static struct of_device_id dbri_match[] = {
+static const struct of_device_id dbri_match[] = {
 	{
 		.name = "SUNW,DBRIe",
 	},
@@ -2693,7 +2694,7 @@ static struct of_platform_driver dbri_sbus_driver = {
 /* Probe for the dbri chip and then attach the driver. */
 static int __init dbri_init(void)
 {
-	return of_register_driver(&dbri_sbus_driver, &sbus_bus_type);
+	return of_register_driver(&dbri_sbus_driver, &of_bus_type);
 }
 
 static void __exit dbri_exit(void)
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 69689e79bf7..92115755d98 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -1480,6 +1480,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	/* Advanced modes of the Edirol UA-25EX.
+	 * For the standard mode, UA-25EX has ID 0582:00e7, which
+	 * offers only 16-bit PCM at 44.1 kHz and no MIDI.
+	 */
+	USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "EDIROL",
+		.product_name = "UA-25EX",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_EDIROL_UAXX
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_EDIROL_UAXX
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_EDIROL_UAXX
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Guillemot devices */
 {
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index b441fe2cd19..c2515b680f9 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -118,12 +118,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
 	void *vaddr;
 	struct us122l *us122l = area->vm_private_data;
 	struct usb_stream *s;
-	int vm_f = VM_FAULT_SIGBUS;
 
 	mutex_lock(&us122l->mutex);
 	s = us122l->sk.s;
 	if (!s)
-		goto out;
+		goto unlock;
 
 	offset = vmf->pgoff << PAGE_SHIFT;
 	if (offset < PAGE_ALIGN(s->read_size))
@@ -131,7 +130,7 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
 	else {
 		offset -= PAGE_ALIGN(s->read_size);
 		if (offset >= PAGE_ALIGN(s->write_size))
-			goto out;
+			goto unlock;
 
 		vaddr = us122l->sk.write_page + offset;
 	}
@@ -141,9 +140,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
 	mutex_unlock(&us122l->mutex);
 
 	vmf->page = page;
-	vm_f = 0;
-out:
-	return vm_f;
+
+	return 0;
+unlock:
+	mutex_unlock(&us122l->mutex);
+	return VM_FAULT_SIGBUS;
 }
 
 static void usb_stream_hwdep_vm_close(struct vm_area_struct *area)