Merge branch 'master' into for-next

Conflicts:
	MAINTAINERS
	arch/arm/mach-omap2/pm24xx.c
	drivers/scsi/bfa/bfa_fcpim.c

Needed to update to apply fixes for which the old branch was too
outdated.

97 files changed, 708 insertions, 438 deletions

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index f2f41c854221..6e2409181895 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -420,9 +420,9 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
 		return PTR_ERR(pclk);
 	}
 	sample_clk = clk_get(&pdev->dev, "sample_clk");
-	if (IS_ERR(pclk)) {
+	if (IS_ERR(sample_clk)) {
 		dev_dbg(&pdev->dev, "no sample clock\n");
-		retval = PTR_ERR(pclk);
+		retval = PTR_ERR(sample_clk);
 		goto out_put_pclk;
 	}
 	clk_enable(pclk);
diff --git a/sound/core/info.c b/sound/core/info.c
index b70564ed8b37..7077f601da5a 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -23,7 +23,6 @@
 #include <linux/time.h>
 #include <linux/mm.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/string.h>
 #include <sound/core.h>
 #include <sound/minors.h>
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 5c8c7dff8ede..b753ec661fcf 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1510,16 +1510,19 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use
 static int snd_pcm_oss_reset(struct snd_pcm_oss_file *pcm_oss_file)
 {
 	struct snd_pcm_substream *substream;
+	struct snd_pcm_runtime *runtime;
+	int i;
 
-	substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
-	if (substream != NULL) {
-		snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
-		substream->runtime->oss.prepare = 1;
-	}
-	substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
-	if (substream != NULL) {
+	for (i = 0; i < 2; i++) { 
+		substream = pcm_oss_file->streams[i];
+		if (!substream)
+			continue;
+		runtime = substream->runtime;
 		snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
-		substream->runtime->oss.prepare = 1;
+		runtime->oss.prepare = 1;
+		runtime->oss.buffer_used = 0;
+		runtime->oss.prev_hw_ptr_period = 0;
+		runtime->oss.period_ptr = 0;
 	}
 	return 0;
 }
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a1707cca9c66..b75db8e9cc0f 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -223,7 +223,7 @@ static void xrun_log(struct snd_pcm_substream *substream,
 	entry->jiffies = jiffies;
 	entry->pos = pos;
 	entry->period_size = runtime->period_size;
-	entry->buffer_size = runtime->buffer_size;;
+	entry->buffer_size = runtime->buffer_size;
 	entry->old_hw_ptr = runtime->status->hw_ptr;
 	entry->hw_ptr_base = runtime->hw_ptr_base;
 	log->idx = (log->idx + 1) % XRUN_LOG_CNT;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 8e022fc70f55..c0ebb5162e95 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -22,7 +22,6 @@
 #include <linux/mm.h>
 #include <linux/file.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/time.h>
 #include <linux/pm_qos_params.h>
 #include <linux/uio.h>
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 62a093efb453..66691fe437e6 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -21,7 +21,6 @@
 
 #include <linux/init.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/time.h>
 #include <linux/device.h>
 #include <linux/moduleparam.h>
diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c
index 727bdb9ba2dc..d8cf3e58dc76 100644
--- a/sound/oss/dev_table.c
+++ b/sound/oss/dev_table.c
@@ -71,7 +71,7 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
 	if (sound_nblocks >= MAX_MEM_BLOCKS)
 		sound_nblocks = MAX_MEM_BLOCKS - 1;
 
-	op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations)));
+	op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct audio_operations)));
 	sound_nblocks++;
 	if (sound_nblocks >= MAX_MEM_BLOCKS)
 		sound_nblocks = MAX_MEM_BLOCKS - 1;
@@ -81,7 +81,6 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
 		sound_unload_audiodev(num);
 		return -(ENOMEM);
 	}
-	memset((char *) op, 0, sizeof(struct audio_operations));
 	init_waitqueue_head(&op->in_sleeper);
 	init_waitqueue_head(&op->out_sleeper);	
 	init_waitqueue_head(&op->poll_sleeper);
@@ -128,7 +127,7 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
 	/* FIXME: This leaks a mixer_operations struct every time its called
 	   until you unload sound! */
 	   
-	op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations)));
+	op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct mixer_operations)));
 	sound_nblocks++;
 	if (sound_nblocks >= MAX_MEM_BLOCKS)
 		sound_nblocks = MAX_MEM_BLOCKS - 1;
@@ -137,7 +136,6 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
 		printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
 		return -ENOMEM;
 	}
-	memset((char *) op, 0, sizeof(struct mixer_operations));
 	memcpy((char *) op, (char *) driver, driver_size);
 
 	strlcpy(op->name, name, sizeof(op->name));
diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
index 782b3b84dac6..ceedb1eff203 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -178,7 +178,7 @@ int MIDIbuf_open(int dev, struct file *file)
 		return err;
 
 	parms[dev].prech_timeout = MAX_SCHEDULE_TIMEOUT;
-	midi_in_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+	midi_in_buf[dev] = vmalloc(sizeof(struct midi_buf));
 
 	if (midi_in_buf[dev] == NULL)
 	{
@@ -188,7 +188,7 @@ int MIDIbuf_open(int dev, struct file *file)
 	}
 	midi_in_buf[dev]->len = midi_in_buf[dev]->head = midi_in_buf[dev]->tail = 0;
 
-	midi_out_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+	midi_out_buf[dev] = vmalloc(sizeof(struct midi_buf));
 
 	if (midi_out_buf[dev] == NULL)
 	{
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index e19dd5dcc2de..9b800ce5100e 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -859,7 +859,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
 			return 0;
 
 		case SNDCTL_COPR_LOAD:
-			buf = (copr_buffer *) vmalloc(sizeof(copr_buffer));
+			buf = vmalloc(sizeof(copr_buffer));
 			if (buf == NULL)
 				return -ENOSPC;
 			if (copy_from_user(buf, arg, sizeof(copr_buffer))) {
@@ -871,7 +871,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
 			return err;
 		
 		case SNDCTL_COPR_SENDMSG:
-			mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+			mbuf = vmalloc(sizeof(copr_msg));
 			if (mbuf == NULL)
 				return -ENOSPC;
 			if (copy_from_user(mbuf, arg, sizeof(copr_msg))) {
@@ -895,7 +895,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
 
 		case SNDCTL_COPR_RCVMSG:
 			err = 0;
-			mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+			mbuf = vmalloc(sizeof(copr_msg));
 			if (mbuf == NULL)
 				return -ENOSPC;
 			data = (unsigned short *)mbuf->data;
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index e85789e53816..5ea1098ac427 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -1646,13 +1646,13 @@ void sequencer_init(void)
 {
 	if (sequencer_ok)
 		return;
-	queue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * EV_SZ);
+	queue = vmalloc(SEQ_MAX_QUEUE * EV_SZ);
 	if (queue == NULL)
 	{
 		printk(KERN_ERR "sequencer: Can't allocate memory for sequencer output queue\n");
 		return;
 	}
-	iqueue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * IEV_SZ);
+	iqueue = vmalloc(SEQ_MAX_QUEUE * IEV_SZ);
 	if (iqueue == NULL)
 	{
 		printk(KERN_ERR "sequencer: Can't allocate memory for sequencer input queue\n");
diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c
index f7e374ec4414..1b9bf9395cfe 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -625,6 +625,8 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao,
 			control_cache_size, (struct hpi_control_cache_info *)
 			&phw->control_cache[0]
 			);
+		if (!phw->p_cache)
+			pao->has_control_cache = 0;
 	} else
 		pao->has_control_cache = 0;
 
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 22c5fc625533..2672f6591ceb 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -644,6 +644,8 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 				interface->control_cache.size_in_bytes,
 				(struct hpi_control_cache_info *)
 				p_control_cache_virtual);
+			if (!phw->p_cache)
+				err = HPI_ERROR_MEMORY_ALLOC;
 		}
 		if (!err) {
 			err = hpios_locked_mem_get_phys_addr(&phw->
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index dda4f1c6f658..d67f4d3db911 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -571,14 +571,20 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32
 {
 	struct hpi_control_cache *p_cache =
 		kmalloc(sizeof(*p_cache), GFP_KERNEL);
+	if (!p_cache)
+		return NULL;
+	p_cache->p_info =
+		kmalloc(sizeof(*p_cache->p_info) * number_of_controls,
+			GFP_KERNEL);
+	if (!p_cache->p_info) {
+		kfree(p_cache);
+		return NULL;
+	}
 	p_cache->cache_size_in_bytes = size_in_bytes;
 	p_cache->control_count = number_of_controls;
 	p_cache->p_cache =
 		(struct hpi_control_cache_single *)pDSP_control_buffer;
 	p_cache->init = 0;
-	p_cache->p_info =
-		kmalloc(sizeof(*p_cache->p_info) * p_cache->control_count,
-		GFP_KERNEL);
 	return p_cache;
 }
 
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 62895a719fcb..22dbd91811a4 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -435,7 +435,7 @@ void __devexit asihpi_adapter_remove(struct pci_dev *pci_dev)
 	struct hpi_message hm;
 	struct hpi_response hr;
 	struct hpi_adapter *pa;
-	pa = (struct hpi_adapter *)pci_get_drvdata(pci_dev);
+	pa = pci_get_drvdata(pci_dev);
 
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM,
 		HPI_SUBSYS_DELETE_ADAPTER);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 4679ed83a43b..2f3cacbd5528 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1129,10 +1129,11 @@ snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip,
 
 		count_areas = size/2;
 		addr_area2 = addr+count_areas;
-		count_areas--; /* max. index */
 		snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n",
 				addr, count_areas, addr_area2, count_areas);
 
+		count_areas--; /* max. index */
+
 		/* build combined I/O buffer length word */
 		lengths = (count_areas << 16) | (count_areas);
 		spin_lock_irqsave(&chip->reg_lock, flags);
@@ -1740,11 +1741,15 @@ static const struct snd_pcm_hardware snd_azf3328_hardware =
 	.rate_max =		AZF_FREQ_66200,
 	.channels_min =		1,
 	.channels_max =		2,
-	.buffer_bytes_max =	65536,
-	.period_bytes_min =	64,
-	.period_bytes_max =	65536,
-	.periods_min =		1,
-	.periods_max =		1024,
+	.buffer_bytes_max =	(64*1024),
+	.period_bytes_min =	1024,
+	.period_bytes_max =	(32*1024),
+	/* We simply have two DMA areas (instead of a list of descriptors
+	   such as other cards); I believe that this is a fixed hardware
+	   attribute and there isn't much driver magic to be done to expand it.
+	   Thus indicate that we have at least and at most 2 periods. */
+	.periods_min =		2,
+	.periods_max =		2,
 	/* FIXME: maybe that card actually has a FIFO?
 	 * Hmm, it seems newer revisions do have one, but we still don't know
 	 * its size... */
@@ -1980,8 +1985,13 @@ snd_azf3328_timer_stop(struct snd_timer *timer)
 	chip = snd_timer_chip(timer);
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	/* disable timer countdown and interrupt */
-	/* FIXME: should we write TIMER_IRQ_ACK here? */
-	snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0);
+	/* Hmm, should we write TIMER_IRQ_ACK here?
+	   YES indeed, otherwise a rogue timer operation - which prompts
+	   ALSA(?) to call repeated stop() in vain, but NOT start() -
+	   will never end (value 0x03 is kept shown in control byte).
+	   Simply manually poking 0x04 _once_ immediately successfully stops
+	   the hardware/ALSA interrupt activity. */
+	snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x04);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 	snd_azf3328_dbgcallleave();
 	return 0;
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index 3e5ca8fb519f..e377287192aa 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -225,39 +225,25 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip)
 {
 	struct dsp_spos_instance * ins = kzalloc(sizeof(struct dsp_spos_instance), GFP_KERNEL);
 
-	if (ins == NULL) 
+	if (ins == NULL)
 		return NULL;
 
 	/* better to use vmalloc for this big table */
-	ins->symbol_table.nsymbols = 0;
 	ins->symbol_table.symbols = vmalloc(sizeof(struct dsp_symbol_entry) *
 					    DSP_MAX_SYMBOLS);
-	ins->symbol_table.highest_frag_index = 0;
-
-	if (ins->symbol_table.symbols == NULL) {
+	ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL);
+	ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL);
+	if (!ins->symbol_table.symbols || !ins->code.data || !ins->modules) {
 		cs46xx_dsp_spos_destroy(chip);
 		goto error;
 	}
-
+	ins->symbol_table.nsymbols = 0;
+	ins->symbol_table.highest_frag_index = 0;
 	ins->code.offset = 0;
 	ins->code.size = 0;
-	ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL);
-
-	if (ins->code.data == NULL) {
-		cs46xx_dsp_spos_destroy(chip);
-		goto error;
-	}
-
 	ins->nscb = 0;
 	ins->ntask = 0;
-
 	ins->nmodules = 0;
-	ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL);
-
-	if (ins->modules == NULL) {
-		cs46xx_dsp_spos_destroy(chip);
-		goto error;
-	}
 
 	/* default SPDIF input sample rate
 	   to 48000 khz */
@@ -271,8 +257,8 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip)
 
 	/* set left and right validity bits and
 	   default channel status */
-	ins->spdif_csuv_default = 
-		ins->spdif_csuv_stream =  
+	ins->spdif_csuv_default =
+		ins->spdif_csuv_stream =
 	 /* byte 0 */  ((unsigned int)_wrap_all_bits(  (SNDRV_PCM_DEFAULT_CON_SPDIF        & 0xff)) << 24) |
 	 /* byte 1 */  ((unsigned int)_wrap_all_bits( ((SNDRV_PCM_DEFAULT_CON_SPDIF >> 8) & 0xff)) << 16) |
 	 /* byte 3 */   (unsigned int)_wrap_all_bits(  (SNDRV_PCM_DEFAULT_CON_SPDIF >> 24) & 0xff) |
@@ -281,6 +267,9 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip)
 	return ins;
 
 error:
+	kfree(ins->modules);
+	kfree(ins->code.data);
+	vfree(ins->symbol_table.symbols);
 	kfree(ins);
 	return NULL;
 }
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 85ab43e89212..457d21189b0d 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -129,8 +129,6 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
 
 	apcm->substream = substream;
 	apcm->interrupt = ct_atc_pcm_interrupt;
-	runtime->private_data = apcm;
-	runtime->private_free = ct_atc_pcm_free_substream;
 	if (IEC958 == substream->pcm->device) {
 		runtime->hw = ct_spdif_passthru_playback_hw;
 		atc->spdif_out_passthru(atc, 1);
@@ -155,8 +153,12 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
 	}
 
 	apcm->timer = ct_timer_instance_new(atc->timer, apcm);
-	if (!apcm->timer)
+	if (!apcm->timer) {
+		kfree(apcm);
 		return -ENOMEM;
+	}
+	runtime->private_data = apcm;
+	runtime->private_free = ct_atc_pcm_free_substream;
 
 	return 0;
 }
@@ -278,8 +280,6 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
 	apcm->started = 0;
 	apcm->substream = substream;
 	apcm->interrupt = ct_atc_pcm_interrupt;
-	runtime->private_data = apcm;
-	runtime->private_free = ct_atc_pcm_free_substream;
 	runtime->hw = ct_pcm_capture_hw;
 	runtime->hw.rate_max = atc->rsr * atc->msr;
 
@@ -298,8 +298,12 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
 	}
 
 	apcm->timer = ct_timer_instance_new(atc->timer, apcm);
-	if (!apcm->timer)
+	if (!apcm->timer) {
+		kfree(apcm);
 		return -ENOMEM;
+	}
+	runtime->private_data = apcm;
+	runtime->private_free = ct_atc_pcm_free_substream;
 
 	return 0;
 }
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index cb0c23a6b473..4a663471dadc 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -189,6 +189,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a,
 	a->channels = GRAB_BITS(buf, 0, 0, 3);
 	a->channels++;
 
+	a->sample_bits = 0;
+	a->max_bitrate = 0;
+
 	a->format = GRAB_BITS(buf, 0, 3, 4);
 	switch (a->format) {
 	case AUDIO_CODING_TYPE_REF_STREAM_HEADER:
@@ -198,7 +201,6 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a,
 
 	case AUDIO_CODING_TYPE_LPCM:
 		val = GRAB_BITS(buf, 2, 0, 3);
-		a->sample_bits = 0;
 		for (i = 0; i < 3; i++)
 			if (val & (1 << i))
 				a->sample_bits |= cea_sample_sizes[i + 1];
@@ -598,24 +600,19 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
 {
 	int i;
 
-	pcm->rates = 0;
-	pcm->formats = 0;
-	pcm->maxbps = 0;
-	pcm->channels_min = -1;
-	pcm->channels_max = 0;
+	/* assume basic audio support (the basic audio flag is not in ELD;
+	 * however, all audio capable sinks are required to support basic
+	 * audio) */
+	pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
+	pcm->formats = SNDRV_PCM_FMTBIT_S16_LE;
+	pcm->maxbps = 16;
+	pcm->channels_max = 2;
 	for (i = 0; i < eld->sad_count; i++) {
 		struct cea_sad *a = &eld->sad[i];
 		pcm->rates |= a->rates;
-		if (a->channels < pcm->channels_min)
-			pcm->channels_min = a->channels;
 		if (a->channels > pcm->channels_max)
 			pcm->channels_max = a->channels;
 		if (a->format == AUDIO_CODING_TYPE_LPCM) {
-			if (a->sample_bits & AC_SUPPCM_BITS_16) {
-				pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE;
-				if (pcm->maxbps < 16)
-					pcm->maxbps = 16;
-			}
 			if (a->sample_bits & AC_SUPPCM_BITS_20) {
 				pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
 				if (pcm->maxbps < 20)
@@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
 	/* restrict the parameters by the values the codec provides */
 	pcm->rates &= codec_pars->rates;
 	pcm->formats &= codec_pars->formats;
-	pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min);
 	pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max);
 	pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 21aa9b0e28f6..b030c8eba21f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2296,6 +2296,7 @@ static int azx_dev_free(struct snd_device *device)
  */
 static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 460fb2ef7e39..18af38ebf757 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1166,6 +1166,7 @@ static const char *cs420x_models[CS420X_MODELS] = {
 
 static struct snd_pci_quirk cs420x_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53),
+	SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55),
 	SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
 	SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55),
 	SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6361f752b5f3..76bd58a0e2b6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2116,8 +2116,8 @@ static void cxt5066_update_speaker(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 	unsigned int pinctl;
 
-	snd_printdd("CXT5066: update speaker, hp_present=%d\n",
-		spec->hp_present);
+	snd_printdd("CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n",
+		    spec->hp_present, spec->cur_eapd);
 
 	/* Port A (HP) */
 	pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0;
@@ -2125,11 +2125,20 @@ static void cxt5066_update_speaker(struct hda_codec *codec)
 			pinctl);
 
 	/* Port D (HP/LO) */
-	pinctl = ((spec->hp_present & 2) && spec->cur_eapd)
-		? spec->port_d_mode : 0;
-	/* Mute if Port A is connected on Thinkpad */
-	if (spec->thinkpad && (spec->hp_present & 1))
-		pinctl = 0;
+	if (spec->dell_automute) {
+		/* DELL AIO Port Rule: PortA>  PortD>  IntSpk */
+		pinctl = (!(spec->hp_present & 1) && spec->cur_eapd)
+			? PIN_OUT : 0;
+	} else if (spec->thinkpad) {
+		if (spec->cur_eapd)
+			pinctl = spec->port_d_mode;
+		/* Mute dock line-out if Port A (laptop HP) is present */
+		if (spec->hp_present&  1)
+			pinctl = 0;
+	} else {
+		pinctl = ((spec->hp_present & 2) && spec->cur_eapd)
+			? spec->port_d_mode : 0;
+	}
 	snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 			pinctl);
 
@@ -2137,14 +2146,6 @@ static void cxt5066_update_speaker(struct hda_codec *codec)
 	pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
 	snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 			pinctl);
-
-	if (spec->dell_automute) {
-		/* DELL AIO Port Rule: PortA > PortD > IntSpk */
-		pinctl = (!(spec->hp_present & 1) && spec->cur_eapd)
-			? PIN_OUT : 0;
-		snd_hda_codec_write(codec, 0x1c, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
-	}
 }
 
 /* turn on/off EAPD (+ mute HP) as a master switch */
@@ -3095,11 +3096,11 @@ static const char *cxt5066_models[CXT5066_MODELS] = {
 static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
-	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
-		      CXT5066_DELL_LAPTOP),
+	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
+	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
 	SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3108,8 +3109,10 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
  	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index d3e49aa5b9ec..31df7747990d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -834,7 +834,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 			return -ENODEV;
 	} else {
 		/* fallback to the codec default */
-		hinfo->channels_min = codec_pars->channels_min;
 		hinfo->channels_max = codec_pars->channels_max;
 		hinfo->rates = codec_pars->rates;
 		hinfo->formats = codec_pars->formats;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5f00589cb791..427da45d7906 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1614,6 +1614,7 @@ do_sku:
 		spec->init_amp = ALC_INIT_GPIO3;
 		break;
 	case 5:
+	default:
 		spec->init_amp = ALC_INIT_DEFAULT;
 		break;
 	}
@@ -2014,6 +2015,36 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
 };
 
 /*
+ *ALC888 Acer Aspire 7730G model
+ */
+
+static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+/* Bias voltage on for external mic port */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Front Mic: set to PIN_IN (empty by default) */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/*Enable internal subwoofer */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{0x17, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+/*
  * ALC889 Acer Aspire 8930G model
  */
 
@@ -2200,6 +2231,16 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x17;
 }
 
+static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->autocfg.speaker_pins[2] = 0x17;
+}
+
 static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4554,6 +4595,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
 	SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
 	SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
+	SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG),
 	SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
 	SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW),
 	SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
@@ -9524,13 +9566,6 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-	{ } /* end */
-};
-
 static void alc888_6st_dell_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -9831,7 +9866,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
-	SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
 	SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
 	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
@@ -10328,7 +10362,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.const_channel_count = 6,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.setup = alc888_acer_aspire_6530g_setup,
+		.setup = alc888_acer_aspire_7730g_setup,
 		.init_hook = alc_automute_amp,
 	},
 	[ALC883_MEDION] = {
@@ -10796,7 +10830,8 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i, err;
+	int i, err, type;
+	int type_idx = 0;
 	hda_nid_t nid;
 
 	for (i = 0; i < cfg->num_inputs; i++) {
@@ -10805,9 +10840,15 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
 		nid = cfg->inputs[i].pin;
 		if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) {
 			char label[32];
+			type = cfg->inputs[i].type;
+			if (i > 0 && type == cfg->inputs[i - 1].type)
+				type_idx++;
+			else
+				type_idx = 0;
 			snprintf(label, sizeof(label), "%s Boost",
 				 hda_get_autocfg_input_label(codec, cfg, i));
-			err = add_control(spec, ALC_CTL_WIDGET_VOL, label, 0,
+			err = add_control(spec, ALC_CTL_WIDGET_VOL, label,
+					  type_idx,
 				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
 			if (err < 0)
 				return err;
@@ -14623,7 +14664,10 @@ static int alc275_setup_dual_adc(struct hda_codec *codec)
 /* different alc269-variants */
 enum {
 	ALC269_TYPE_NORMAL,
+	ALC269_TYPE_ALC258,
 	ALC269_TYPE_ALC259,
+	ALC269_TYPE_ALC269VB,
+	ALC269_TYPE_ALC270,
 	ALC269_TYPE_ALC271X,
 };
 
@@ -14763,6 +14807,8 @@ static int alc269_resume(struct hda_codec *codec)
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC269_FIXUP_DELL_M101Z,
+	ALC269_FIXUP_LENOVO_EDGE14,
+	ALC269_FIXUP_ASUS_G73JW,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -14780,11 +14826,22 @@ static const struct alc_fixup alc269_fixups[] = {
 			{}
 		}
 	},
+	[ALC269_FIXUP_LENOVO_EDGE14] = {
+		.sku = ALC_FIXUP_SKU_IGNORE,
+	},
+	[ALC269_FIXUP_ASUS_G73JW] = {
+		.pins = (const struct alc_pincfg[]) {
+			{ 0x17, 0x99130111 }, /* subwoofer */
+			{ }
+		}
+	},
 };
 
 static struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_LENOVO_EDGE14),
+	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	{}
 };
 
@@ -15023,7 +15080,7 @@ static int alc269_fill_coef(struct hda_codec *codec)
 static int patch_alc269(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int board_config;
+	int board_config, coef;
 	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -15034,14 +15091,23 @@ static int patch_alc269(struct hda_codec *codec)
 
 	alc_auto_parse_customize_define(codec);
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){
+	coef = alc_read_coef_idx(codec, 0);
+	if ((coef & 0x00f0) == 0x0010) {
 		if (codec->bus->pci->subsystem_vendor == 0x1025 &&
 		    spec->cdefine.platform_type == 1) {
 			alc_codec_rename(codec, "ALC271X");
 			spec->codec_variant = ALC269_TYPE_ALC271X;
-		} else {
+		} else if ((coef & 0xf000) == 0x1000) {
+			spec->codec_variant = ALC269_TYPE_ALC270;
+		} else if ((coef & 0xf000) == 0x2000) {
 			alc_codec_rename(codec, "ALC259");
 			spec->codec_variant = ALC269_TYPE_ALC259;
+		} else if ((coef & 0xf000) == 0x3000) {
+			alc_codec_rename(codec, "ALC258");
+			spec->codec_variant = ALC269_TYPE_ALC258;
+		} else {
+			alc_codec_rename(codec, "ALC269VB");
+			spec->codec_variant = ALC269_TYPE_ALC269VB;
 		}
 	} else
 		alc_fix_pll_init(codec, 0x20, 0x04, 15);
@@ -15104,7 +15170,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->stream_digital_capture = &alc269_pcm_digital_capture;
 
 	if (!spec->adc_nids) { /* wasn't filled automatically? use default */
-		if (spec->codec_variant != ALC269_TYPE_NORMAL) {
+		if (spec->codec_variant == ALC269_TYPE_NORMAL) {
 			spec->adc_nids = alc269_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
 			spec->capsrc_nids = alc269_capsrc_nids;
@@ -16898,7 +16964,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+	return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x22, 0);
 }
 
 
@@ -18952,6 +19018,8 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
 		return 0x02;
 	else if (nid >= 0x0c && nid <= 0x0e)
 		return nid - 0x0c + 0x02;
+	else if (nid == 0x26) /* ALC887-VD has this DAC too */
+		return 0x25;
 	else
 		return 0;
 }
@@ -18960,7 +19028,7 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
 static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
 				   hda_nid_t dac)
 {
-	hda_nid_t mix[4];
+	hda_nid_t mix[5];
 	int i, num;
 
 	num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
@@ -19298,6 +19366,7 @@ static const struct alc_fixup alc662_fixups[] = {
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
+	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	{}
@@ -19419,7 +19488,10 @@ static int patch_alc888(struct hda_codec *codec)
 {
 	if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){
 		kfree(codec->chip_name);
-		codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
+		if (codec->vendor_id == 0x10ec0887)
+			codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL);
+		else
+			codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
 		if (!codec->chip_name) {
 			alc_free(codec);
 			return -ENOMEM;
@@ -19909,7 +19981,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
-	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
+	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
 	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 93fa59cc60ef..efa4225f5fd6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -389,6 +389,11 @@ static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
 	0x11, 0x20, 0
 };
 
+#define STAC92HD87B_NUM_DMICS	 1
+static hda_nid_t stac92hd87b_dmic_nids[STAC92HD87B_NUM_DMICS + 1] = {
+	0x11, 0
+};
+
 #define STAC92HD83XXX_NUM_CAPS	2
 static unsigned long stac92hd83xxx_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
@@ -1622,6 +1627,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1,
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
+		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	{} /* terminator */
 };
 
@@ -3486,10 +3493,8 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 				return err;
 		}
 
-		if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) {
+		if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1)
 			snd_hda_add_imux_item(imux, label, index, NULL);
-			spec->num_analog_muxes++;
-		}
 	}
 
 	return 0;
@@ -5452,12 +5457,17 @@ again:
 				stac92hd83xxx_brd_tbl[spec->board_config]);
 
 	switch (codec->vendor_id) {
+	case 0x111d76d1:
+	case 0x111d76d9:
+		spec->dmic_nids = stac92hd87b_dmic_nids;
+		spec->num_dmics = stac92xx_connected_ports(codec,
+				stac92hd87b_dmic_nids,
+				STAC92HD87B_NUM_DMICS);
+		/* Fall through */
 	case 0x111d7666:
 	case 0x111d7667:
 	case 0x111d7668:
 	case 0x111d7669:
-	case 0x111d76d1:
-	case 0x111d76d9:
 		spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids);
 		spec->pin_nids = stac92hd88xxx_pin_nids;
 		spec->mono_nid = 0;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 400f9ebd243e..629a5494347a 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1866,6 +1866,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 	},
 	{
 		.subvendor = 0x1028,
+		.subdevice = 0x0182,
+		.name = "Dell Latitude D610",	/* STAC9750/51 */
+		.type = AC97_TUNE_HP_ONLY
+	},
+	{
+		.subvendor = 0x1028,
 		.subdevice = 0x0186,
 		.name = "Dell Latitude D810", /* cf. Malone #41015 */
 		.type = AC97_TUNE_HP_MUTE_LED
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index ef9af3f4ace2..1bd7a540fd49 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -425,7 +425,7 @@ exit:
 static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream)
 {
 	struct snd_pcm_substream *substream = lx_stream->stream;
-	const int is_capture = lx_stream->is_capture;
+	const unsigned int is_capture = lx_stream->is_capture;
 
 	int err;
 
@@ -473,7 +473,7 @@ static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream)
 
 static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream)
 {
-	const int is_capture = lx_stream->is_capture;
+	const unsigned int is_capture = lx_stream->is_capture;
 	int err;
 
 	snd_printd(LXP "stopping: stopping stream\n");
diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h
index 51afc048961d..aea621eafbb5 100644
--- a/sound/pci/lx6464es/lx6464es.h
+++ b/sound/pci/lx6464es/lx6464es.h
@@ -60,7 +60,7 @@ struct lx_stream {
 	snd_pcm_uframes_t          frame_pos;
 	enum lx_stream_status      status; /* free, open, running, draining
 					    * pause */
-	int                        is_capture:1;
+	unsigned int               is_capture:1;
 };
 
 
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index 3086b751da4a..617f98b0cbae 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -1152,7 +1152,7 @@ static int lx_interrupt_request_new_buffer(struct lx6464es *chip,
 					   struct lx_stream *lx_stream)
 {
 	struct snd_pcm_substream *substream = lx_stream->stream;
-	int is_capture = lx_stream->is_capture;
+	const unsigned int is_capture = lx_stream->is_capture;
 	int err;
 	unsigned long flags;
 
diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h
index a46f5083db99..812e288ef2e7 100644
--- a/sound/pci/mixart/mixart_hwdep.h
+++ b/sound/pci/mixart/mixart_hwdep.h
@@ -25,11 +25,21 @@
 
 #include <sound/hwdep.h>
 
+#ifndef readl_be
 #define readl_be(x) be32_to_cpu(__raw_readl(x))
+#endif
+
+#ifndef writel_be
 #define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr)
+#endif
 
+#ifndef readl_le
 #define readl_le(x) le32_to_cpu(__raw_readl(x))
+#endif
+
+#ifndef writel_le
 #define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr)
+#endif
 
 #define MIXART_MEM(mgr,x)	((mgr)->mem[0].virt + (x))
 #define MIXART_REG(mgr,x)	((mgr)->mem[1].virt + (x))
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index 85081172403f..b47cfd45b3b9 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -1228,10 +1228,8 @@ int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
 					       chip->rsrc[i].start + 1,
 					       rnames[i]) == NULL) {
 				printk(KERN_ERR "snd: can't request rsrc "
-				       " %d (%s: 0x%016llx:%016llx)\n",
-				       i, rnames[i],
-				       (unsigned long long)chip->rsrc[i].start,
-				       (unsigned long long)chip->rsrc[i].end);
+				       " %d (%s: %pR)\n",
+				       i, rnames[i], &chip->rsrc[i]);
 				err = -ENODEV;
 				goto __error;
 			}
@@ -1256,10 +1254,8 @@ int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
 					       chip->rsrc[i].start + 1,
 					       rnames[i]) == NULL) {
 				printk(KERN_ERR "snd: can't request rsrc "
-				       " %d (%s: 0x%016llx:%016llx)\n",
-				       i, rnames[i],
-				       (unsigned long long)chip->rsrc[i].start,
-				       (unsigned long long)chip->rsrc[i].end);
+				       " %d (%s: %pR)\n",
+				       i, rnames[i], &chip->rsrc[i]);
 				err = -ENODEV;
 				goto __error;
 			}
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index e720d5e6f04c..bee3c94f58b0 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -16,7 +16,8 @@ config SND_ATMEL_SOC_SSC
 
 config SND_AT91_SOC_SAM9G20_WM8731
 	tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
-	depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+	depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC && \
+                   AT91_PROGRAMMABLE_CLOCKS
 	select SND_ATMEL_SOC_SSC
 	select SND_SOC_WM8731
 	help
@@ -25,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731
 
 config SND_AT32_SOC_PLAYPAQ
         tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
         select SND_ATMEL_SOC_SSC
         select SND_SOC_WM8510
         help
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 293569dfd0ed..e521ada80542 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -222,9 +222,9 @@ static int __init at91sam9g20ek_init(void)
 	}
 
 	pllb = clk_get(NULL, "pllb");
-	if (IS_ERR(mclk)) {
+	if (IS_ERR(pllb)) {
 		printk(KERN_ERR "ASoC: Failed to get PLLB\n");
-		ret = PTR_ERR(mclk);
+		ret = PTR_ERR(pllb);
 		goto err_mclk;
 	}
 	ret = clk_set_parent(mclk, pllb);
@@ -240,6 +240,7 @@ static int __init at91sam9g20ek_init(void)
 	if (!at91sam9g20ek_snd_device) {
 		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
 		ret = -ENOMEM;
+		goto err_mclk;
 	}
 
 	platform_set_drvdata(at91sam9g20ek_snd_device,
@@ -248,11 +249,13 @@ static int __init at91sam9g20ek_init(void)
 	ret = platform_device_add(at91sam9g20ek_snd_device);
 	if (ret) {
 		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
-		platform_device_put(at91sam9g20ek_snd_device);
+		goto err_device_add;
 	}
 
 	return ret;
 
+err_device_add:
+	platform_device_put(at91sam9g20ek_snd_device);
 err_mclk:
 	clk_put(mclk);
 	mclk = NULL;
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index e3d283561c19..86e0f8586dc3 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -167,7 +167,6 @@ static int __init afeb9260_soc_init(void)
 
 	return 0;
 err1:
-	platform_device_del(afeb9260_snd_device);
 	platform_device_put(afeb9260_snd_device);
 	return err;
 }
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 94a9d06b9027..3b5690d28b8b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -25,8 +25,9 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS42L51 if I2C
 	select SND_SOC_CS4270 if I2C
+	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
-	select SND_SOC_JZ4740 if SOC_JZ4740
+	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 43316a70be05..5182e76e7eb3 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -28,6 +28,11 @@
 #include <sound/max98088.h>
 #include "max98088.h"
 
+enum max98088_type {
+       MAX98088,
+       MAX98089,
+};
+
 struct max98088_cdata {
        unsigned int rate;
        unsigned int fmt;
@@ -36,6 +41,7 @@ struct max98088_cdata {
 
 struct max98088_priv {
        u8 reg_cache[M98088_REG_CNT];
+       enum max98088_type devtype;
        void *control_data;
        struct max98088_pdata *pdata;
        unsigned int sysclk;
@@ -2013,7 +2019,10 @@ err_access:
 
 static int max98088_remove(struct snd_soc_codec *codec)
 {
+       struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
+
        max98088_set_bias_level(codec, SND_SOC_BIAS_OFF);
+       kfree(max98088->eq_texts);
 
        return 0;
 }
@@ -2040,6 +2049,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
        if (max98088 == NULL)
                return -ENOMEM;
 
+       max98088->devtype = id->driver_data;
+
        i2c_set_clientdata(i2c, max98088);
        max98088->control_data = i2c;
        max98088->pdata = i2c->dev.platform_data;
@@ -2059,7 +2070,8 @@ static int __devexit max98088_i2c_remove(struct i2c_client *client)
 }
 
 static const struct i2c_device_id max98088_i2c_id[] = {
-       { "max98088", 0 },
+       { "max98088", MAX98088 },
+       { "max98089", MAX98089 },
        { }
 };
 MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 00d67cc8e206..061f9e5a497b 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -383,6 +383,7 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
 	.reg_cache_size = sizeof(stac9766_reg),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_step = 2,
+	.reg_cache_default = stac9766_reg,
 };
 
 static __devinit int stac9766_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index fc687790188b..77b8f9ae29be 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1176,7 +1176,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_gpio);
 int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
 {
 	u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
-	u8 val, bit = gpio ? 2: 1;
+	u8 val = 0, bit = gpio ? 2 : 1;
 
 	aic3x_read(codec, reg, &val);
 	return (val >> bit) & 1;
@@ -1204,7 +1204,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
 
 int aic3x_headset_detected(struct snd_soc_codec *codec)
 {
-	u8 val;
+	u8 val = 0;
 	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
 	return (val >> 4) & 1;
 }
@@ -1212,7 +1212,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 
 int aic3x_button_pressed(struct snd_soc_codec *codec)
 {
-	u8 val;
+	u8 val = 0;
 	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
 	return (val >> 5) & 1;
 }
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index d251ff54a2d3..c5ab8c805771 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -58,7 +58,7 @@
 	(1000000000 / ((rate * 1000) / samples))
 
 #define US_TO_SAMPLES(rate, us) \
-	(rate / (1000000 / us))
+	(rate / (1000000 / (us < 1000000 ? us : 1000000)))
 
 #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
 	((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate)))
@@ -200,7 +200,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 		      u8 *value)
 {
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-	int val;
+	int val, ret = 0;
 
 	*value = reg & 0xff;
 
@@ -210,6 +210,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 		if (val < 0) {
 			dev_err(codec->dev, "Read failed (%d)\n", val);
 			value[0] = dac33_read_reg_cache(codec, reg);
+			ret = val;
 		} else {
 			value[0] = val;
 			dac33_write_reg_cache(codec, reg, val);
@@ -218,7 +219,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 		value[0] = dac33_read_reg_cache(codec, reg);
 	}
 
-	return 0;
+	return ret;
 }
 
 static int dac33_write(struct snd_soc_codec *codec, unsigned int reg,
@@ -329,13 +330,18 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
 		    dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
 }
 
-static inline void dac33_read_id(struct snd_soc_codec *codec)
+static inline int dac33_read_id(struct snd_soc_codec *codec)
 {
+	int i, ret = 0;
 	u8 reg;
 
-	dac33_read(codec, DAC33_DEVICE_ID_MSB, &reg);
-	dac33_read(codec, DAC33_DEVICE_ID_LSB, &reg);
-	dac33_read(codec, DAC33_DEVICE_REV_ID, &reg);
+	for (i = 0; i < 3; i++) {
+		ret = dac33_read(codec, DAC33_DEVICE_ID_MSB + i, &reg);
+		if (ret < 0)
+			break;
+	}
+
+	return ret;
 }
 
 static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
@@ -1076,6 +1082,9 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 		/* Number of samples under i2c latency */
 		dac33->alarm_threshold = US_TO_SAMPLES(rate,
 						dac33->mode1_latency);
+		nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
+				dac33->alarm_threshold;
+
 		if (dac33->auto_fifo_config) {
 			if (period_size <= dac33->alarm_threshold)
 				/*
@@ -1086,6 +1095,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 				       ((dac33->alarm_threshold / period_size) +
 				       (dac33->alarm_threshold % period_size ?
 				       1 : 0));
+			else if (period_size > nsample_limit)
+				dac33->nsample = nsample_limit;
 			else
 				dac33->nsample = period_size;
 		} else {
@@ -1097,8 +1108,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 			 */
 			dac33->nsample_max = substream->runtime->buffer_size -
 						period_size;
-			nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
-					dac33->alarm_threshold;
+
 			if (dac33->nsample_max > nsample_limit)
 				dac33->nsample_max = nsample_limit;
 
@@ -1414,9 +1424,15 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
 		dev_err(codec->dev, "Failed to power up codec: %d\n", ret);
 		goto err_power;
 	}
-	dac33_read_id(codec);
+	ret = dac33_read_id(codec);
 	dac33_hard_power(codec, 0);
 
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to read chip ID: %d\n", ret);
+		ret = -ENODEV;
+		goto err_power;
+	}
+
 	/* Check if the IRQ number is valid and request it */
 	if (dac33->irq >= 0) {
 		ret = request_irq(dac33->irq, dac33_interrupt_handler,
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 329acc1a2074..d2c243095673 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -78,8 +78,10 @@ static int tpa6130a2_i2c_write(int reg, u8 value)
 
 	if (data->power_state) {
 		val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value);
-		if (val < 0)
+		if (val < 0) {
 			dev_err(&tpa6130a2_client->dev, "Write failed\n");
+			return val;
+		}
 	}
 
 	/* Either powered on or off, we save the context */
@@ -119,13 +121,13 @@ static int tpa6130a2_power(int power)
 {
 	struct	tpa6130a2_data *data;
 	u8	val;
-	int	ret;
+	int	ret = 0;
 
 	BUG_ON(tpa6130a2_client == NULL);
 	data = i2c_get_clientdata(tpa6130a2_client);
 
 	mutex_lock(&data->mutex);
-	if (power) {
+	if (power && !data->power_state) {
 		/* Power on */
 		if (data->power_gpio >= 0)
 			gpio_set_value(data->power_gpio, 1);
@@ -153,7 +155,7 @@ static int tpa6130a2_power(int power)
 		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
 		val &= ~TPA6130A2_SWS;
 		tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
-	} else {
+	} else if (!power && data->power_state) {
 		/* set SWS */
 		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
 		val |= TPA6130A2_SWS;
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 7540a509a6f5..464f0cfa4c7a 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -597,6 +597,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
 	.resume =       uda134x_soc_resume,
 	.reg_cache_size = sizeof(uda134x_reg),
 	.reg_word_size = sizeof(u8),
+	.reg_cache_default = uda134x_reg,
 	.reg_cache_step = 1,
 	.read = uda134x_read_reg_cache,
 	.write = uda134x_write,
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f4f1fba38eb9..7611add7f8c3 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -831,7 +831,7 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	}
 
 	/* MCLK direction */
-	if (dir == WM8350_MCLK_DIR_OUT)
+	if (dir == SND_SOC_CLOCK_OUT)
 		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
 				WM8350_MCLK_DIR);
 	else
@@ -1586,6 +1586,13 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
 			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
 
+	/* Make sure AIF tristating is disabled by default */
+	wm8350_clear_bits(wm8350, WM8350_AI_FORMATING, WM8350_AIF_TRI);
+
+	/* Make sure we've got a sane companding setup too */
+	wm8350_clear_bits(wm8350, WM8350_ADC_DAC_COMP,
+			  WM8350_DAC_COMP | WM8350_LOOPBACK);
+
 	/* Make sure jack detect is disabled to start off with */
 	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 712ef7c76f90..9a433a5396cb 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -146,7 +146,6 @@ static int wm8523_startup(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	return 0;
 	snd_pcm_hw_constraint_list(substream->runtime, 0,
 				   SNDRV_PCM_HW_PARAM_RATE,
 				   &wm8523->rate_constraint);
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index a2e0ed59b376..8725d4e75431 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -161,7 +161,7 @@
 static const u16 wm8580_reg[] = {
 	0x0121, 0x017e, 0x007d, 0x0014, /*R3*/
 	0x0121, 0x017e, 0x007d, 0x0194, /*R7*/
-	0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/
+	0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/
 	0x0182, 0x0082, 0x000a, 0x0024, /*R15*/
 	0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/
 	0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/
@@ -491,16 +491,16 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
 		paifa |= 0x8;
 		break;
 	case SNDRV_PCM_FORMAT_S20_3LE:
-		paifa |= 0x10;
+		paifa |= 0x0;
 		paifb |= WM8580_AIF_LENGTH_20;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		paifa |= 0x10;
+		paifa |= 0x0;
 		paifb |= WM8580_AIF_LENGTH_24;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
-		paifa |= 0x10;
-		paifb |= WM8580_AIF_LENGTH_24;
+		paifa |= 0x0;
+		paifb |= WM8580_AIF_LENGTH_32;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 631385802eb4..e725c09a3e79 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -526,7 +526,7 @@ static int wm8731_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0);
 
 	/* Disable bypass path by default */
-	snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0);
+	snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0);
 
 	snd_soc_add_controls(codec, wm8731_snd_controls,
 			     ARRAY_SIZE(wm8731_snd_controls));
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 04182c464e35..0132a27140ae 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -34,7 +34,6 @@
 /* codec private data */
 struct wm8776_priv {
 	enum snd_soc_control_type control_type;
-	u16 reg_cache[WM8776_CACHEREGNUM];
 	int sysclk[2];
 };
 
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index b4f11724a63f..aca4b1ea10bb 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg)
 {
 	switch (reg) {
 	case WM8900_REG_ID:
-	case WM8900_REG_POWER1:
 		return 1;
 	default:
 		return 0;
@@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec)
 		return -ENODEV;
 	}
 
-	/* Read back from the chip */
-	reg = snd_soc_read(codec, WM8900_REG_POWER1);
-	reg = (reg >> 12) & 0xf;
-	dev_info(codec->dev, "WM8900 revision %d\n", reg);
-
 	wm8900_reset(codec);
 
 	/* Turn the chip on */
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 33be84e506ea..9001cc48ba13 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -818,7 +818,8 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
-	return wm8904->deemph;
+	ucontrol->value.enumerated.item[0] = wm8904->deemph;
+	return 0;
 }
 
 static int wm8904_put_deemph(struct snd_kcontrol *kcontrol,
@@ -2498,6 +2499,8 @@ static int wm8904_remove(struct snd_soc_codec *codec)
 
 	wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
+	kfree(wm8904->retune_mobile_texts);
+	kfree(wm8904->drc_texts);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index f89ad6c9a80b..9cbab8e1de01 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -380,7 +380,8 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 
-	return wm8955->deemph;
+	ucontrol->value.enumerated.item[0] = wm8955->deemph;
+	return 0;
 }
 
 static int wm8955_put_deemph(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 8d5efb333c33..21986c42272f 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -138,7 +138,8 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 
-	return wm8960->deemph;
+	ucontrol->value.enumerated.item[0] = wm8960->deemph;
+	return 0;
 }
 
 static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 4f326f604104..8340485c9851 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -711,7 +711,7 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream,
 	if (fs <= 24000)
 		reg |= WM8961_DACSLOPE;
 	else
-		reg &= WM8961_DACSLOPE;
+		reg &= ~WM8961_DACSLOPE;
 	snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg);
 
 	return 0;
@@ -736,7 +736,7 @@ static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 		freq /= 2;
 	} else {
 		dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq);
-		reg &= WM8961_MCLKDIV;
+		reg &= ~WM8961_MCLKDIV;
 	}
 
 	snd_soc_write(codec, WM8961_CLOCKING1, reg);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 894d0cd3aa9b..1304ca91a11c 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3339,7 +3339,7 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 	int mask;
 	int active;
 
-	mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2);
+	mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK);
 
 	active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2);
 	active &= ~mask;
@@ -3500,8 +3500,11 @@ static ssize_t wm8962_beep_set(struct device *dev,
 {
 	struct wm8962_priv *wm8962 = dev_get_drvdata(dev);
 	long int time;
+	int ret;
 
-	strict_strtol(buf, 10, &time);
+	ret = strict_strtol(buf, 10, &time);
+	if (ret != 0)
+		return ret;
 
 	input_event(wm8962->beep, EV_SND, SND_TONE, time);
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0db59c3aa5d4..4d3e6f1ac584 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3903,6 +3903,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		return -ENOMEM;
 	snd_soc_codec_set_drvdata(codec, wm8994);
 
+	codec->reg_cache = &wm8994->reg_cache;
+
 	wm8994->pdata = dev_get_platdata(codec->dev->parent);
 	wm8994->codec = codec;
 
@@ -4059,6 +4061,8 @@ static int  wm8994_codec_remove(struct snd_soc_codec *codec)
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
+	kfree(wm8994->retune_mobile_texts);
+	kfree(wm8994->drc_texts);
 	kfree(wm8994);
 
 	return 0;
@@ -4071,6 +4075,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = {
 	.resume =	wm8994_resume,
 	.read = wm8994_read,
 	.write = wm8994_write,
+	.readable_register = wm8994_readable,
+	.volatile_register = wm8994_volatile,
 	.set_bias_level = wm8994_set_bias_level,
 };
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 2cb81538cd91..0e24092722c3 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -123,7 +123,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 			reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
 			break;
 		default:
-			WARN(1, "Unknown DCS readback method");
+			WARN(1, "Unknown DCS readback method\n");
 			break;
 		}
 
@@ -293,7 +293,7 @@ SOC_DOUBLE_R("Speaker Switch",
 SOC_DOUBLE_R("Speaker ZC Switch",
 	     WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
 	     7, 1, 0),
-SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 0, 3, 7, 0,
+SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 3, 0, 7, 0,
 	       spkboost_tlv),
 SOC_ENUM("Speaker Reference", speaker_ref),
 SOC_ENUM("Speaker Mode", speaker_mode),
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 2b07b17a6b2d..bc9e6b0b3f6f 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -157,12 +157,23 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 }
 
 /* davinci-evm digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link evm_dai = {
+static struct snd_soc_dai_link dm6446_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-mcasp.0",
+	.cpu_dai_name = "davinci-mcbsp",
 	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.0-001a",
+	.codec_name = "tlv320aic3x-codec.1-001b",
+	.platform_name = "davinci-pcm-audio",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link dm355_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai_name = "davinci-mcbsp.1",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.codec_name = "tlv320aic3x-codec.1-001b",
 	.platform_name = "davinci-pcm-audio",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
@@ -172,10 +183,10 @@ static struct snd_soc_dai_link dm365_evm_dai = {
 #ifdef CONFIG_SND_DM365_AIC3X_CODEC
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-i2s",
+	.cpu_dai_name = "davinci-mcbsp",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.init = evm_aic3x_init,
-	.codec_name = "tlv320aic3x-codec.0-001a",
+	.codec_name = "tlv320aic3x-codec.1-0018",
 	.ops = &evm_ops,
 #elif defined(CONFIG_SND_DM365_VOICE_CODEC)
 	.name = "Voice Codec - CQ93VC",
@@ -219,10 +230,17 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
 	.ops = &evm_ops,
 };
 
-/* davinci dm6446, dm355 evm audio machine driver */
-static struct snd_soc_card snd_soc_card_evm = {
-	.name = "DaVinci EVM",
-	.dai_link = &evm_dai,
+/* davinci dm6446 evm audio machine driver */
+static struct snd_soc_card dm6446_snd_soc_card_evm = {
+	.name = "DaVinci DM6446 EVM",
+	.dai_link = &dm6446_evm_dai,
+	.num_links = 1,
+};
+
+/* davinci dm355 evm audio machine driver */
+static struct snd_soc_card dm355_snd_soc_card_evm = {
+	.name = "DaVinci DM355 EVM",
+	.dai_link = &dm355_evm_dai,
 	.num_links = 1,
 };
 
@@ -261,10 +279,10 @@ static int __init evm_init(void)
 	int ret;
 
 	if (machine_is_davinci_evm()) {
-		evm_snd_dev_data = &snd_soc_card_evm;
+		evm_snd_dev_data = &dm6446_snd_soc_card_evm;
 		index = 0;
 	} else if (machine_is_davinci_dm355_evm()) {
-		evm_snd_dev_data = &snd_soc_card_evm;
+		evm_snd_dev_data = &dm355_snd_soc_card_evm;
 		index = 1;
 	} else if (machine_is_davinci_dm365_evm()) {
 		evm_snd_dev_data = &dm365_snd_soc_card_evm;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index d46b545d41f4..9e0e565e6ed9 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -426,9 +426,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 	snd_pcm_format_t fmt;
 	unsigned element_cnt = 1;
 
-	dai->capture_dma_data = dev->dma_params;
-	dai->playback_dma_data = dev->dma_params;
-
 	/* general line settings */
 	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -601,6 +598,15 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+	return 0;
+}
+
 static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
@@ -612,6 +618,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
 static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+	.startup	= davinci_i2s_startup,
 	.shutdown	= davinci_i2s_shutdown,
 	.prepare	= davinci_i2s_prepare,
 	.trigger	= davinci_i2s_trigger,
@@ -749,7 +756,7 @@ static struct platform_driver davinci_mcbsp_driver = {
 	.probe		= davinci_i2s_probe,
 	.remove		= davinci_i2s_remove,
 	.driver		= {
-		.name	= "davinci-i2s",
+		.name	= "davinci-mcbsp",
 		.owner	= THIS_MODULE,
 	},
 };
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 86918ee12419..fb55d2c5d704 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -715,9 +715,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	int word_length;
 	u8 fifo_level;
 
-	cpu_dai->capture_dma_data = dev->dma_params;
-	cpu_dai->playback_dma_data = dev->dma_params;
-
 	davinci_hw_common_param(dev, substream->stream);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		fifo_level = dev->txnumevt;
@@ -799,7 +796,17 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+	return 0;
+}
+
 static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+	.startup	= davinci_mcasp_startup,
 	.trigger	= davinci_mcasp_trigger,
 	.hw_params	= davinci_mcasp_hw_params,
 	.set_fmt	= davinci_mcasp_set_dai_fmt,
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 009b6521a1bf..6c6666a1f942 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -84,7 +84,7 @@ static struct snd_soc_ops sffsdr_ops = {
 static struct snd_soc_dai_link sffsdr_dai = {
 	.name = "PCM3008", /* Codec name */
 	.stream_name = "PCM3008 HiFi",
-	.cpu_dai_name = "davinci-asp.0",
+	.cpu_dai_name = "davinci-mcbsp",
 	.codec_dai_name = "pcm3008-hifi",
 	.codec_name = "pcm3008-codec",
 	.platform_name = "davinci-pcm-audio",
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index ea232f6a2c21..9d2afccc3a2d 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -97,9 +97,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream,
 			&davinci_vcif_dev->dma_params[substream->stream];
 	u32 w;
 
-	dai->capture_dma_data = davinci_vcif_dev->dma_params;
-	dai->playback_dma_data = davinci_vcif_dev->dma_params;
-
 	/* Restart the codec before setup */
 	davinci_vcif_stop(substream);
 	davinci_vcif_start(substream);
@@ -174,9 +171,19 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static int davinci_vcif_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+	return 0;
+}
+
 #define DAVINCI_VCIF_RATES	SNDRV_PCM_RATE_8000_48000
 
 static struct snd_soc_dai_ops davinci_vcif_dai_ops = {
+	.startup	= davinci_vcif_startup,
 	.trigger	= davinci_vcif_trigger,
 	.hw_params	= davinci_vcif_hw_params,
 };
@@ -240,7 +247,10 @@ fail:
 
 static int davinci_vcif_remove(struct platform_device *pdev)
 {
+	struct davinci_vcif_dev *davinci_vcif_dev = dev_get_drvdata(&pdev->dev);
+
 	snd_soc_unregister_dai(&pdev->dev);
+	kfree(davinci_vcif_dev);
 
 	return 0;
 }
diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c
index 4b0d19913728..286817946c56 100644
--- a/sound/soc/ep93xx/simone.c
+++ b/sound/soc/ep93xx/simone.c
@@ -54,24 +54,26 @@ static int __init simone_init(void)
 
 	ret = platform_device_add(simone_snd_ac97_device);
 	if (ret)
-		goto fail;
+		goto fail1;
 
 	simone_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!simone_snd_device) {
 		ret = -ENOMEM;
-		goto fail;
+		goto fail2;
 	}
 
 	platform_set_drvdata(simone_snd_device, &snd_soc_simone);
 	ret = platform_device_add(simone_snd_device);
-	if (ret) {
-		platform_device_put(simone_snd_device);
-		goto fail;
-	}
+	if (ret)
+		goto fail3;
 
-	return ret;
+	return 0;
 
-fail:
+fail3:
+	platform_device_put(simone_snd_device);
+fail2:
+	platform_device_del(simone_snd_ac97_device);
+fail1:
 	platform_device_put(simone_snd_ac97_device);
 	return ret;
 }
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 53251e6b5bd5..108b5d8bd0e9 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -76,6 +76,7 @@ static __init int efika_fabric_init(void)
 	rc = platform_device_add(pdev);
 	if (rc) {
 		pr_err("efika_fabric_init: platform_device_add() failed\n");
+		platform_device_put(pdev);
 		return -ENODEV;
 	}
 	return 0;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index dce6b551cd78..f92dca07cd35 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -9,7 +9,6 @@
 #include <linux/module.h>
 #include <linux/of_device.h>
 #include <linux/slab.h>
-#include <linux/of_device.h>
 #include <linux/of_platform.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 74ffed41340f..9018fa5bf0db 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -160,7 +160,7 @@ static int __devinit psc_i2s_of_probe(struct platform_device *op,
 	rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
 	if (rc != 0) {
 		pr_err("Failed to register DAI\n");
-		return 0;
+		return rc;
 	}
 
 	psc_dma = dev_get_drvdata(&op->dev);
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 0d7dcf1e4863..7d7847a1e66b 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -498,6 +498,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "platform device add failed\n");
 		goto error;
 	}
+	dev_set_drvdata(&pdev->dev, sound_device);
 
 	of_node_put(codec_np);
 
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 63b9eaa1ebc2..026b756961e0 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -498,6 +498,7 @@ static int p1022_ds_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "platform device add failed\n");
 		goto error;
 	}
+	dev_set_drvdata(&pdev->dev, sound_device);
 
 	of_node_put(codec_np);
 
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 25f27ec1dd6e..ba4d85e317ed 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -76,6 +76,7 @@ static __init int pcm030_fabric_init(void)
 	rc = platform_device_add(pdev);
 	if (rc) {
 		pr_err("pcm030_fabric_init: platform_device_add() failed\n");
+		platform_device_put(pdev);
 		return -ENODEV;
 	}
 	return 0;
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index b59675257ce5..dd4fffdbd177 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -34,8 +34,8 @@ static int eukrea_tlv320_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;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret;
 
 	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
@@ -79,10 +79,10 @@ static struct snd_soc_ops eukrea_tlv320_snd_ops = {
 static struct snd_soc_dai_link eukrea_tlv320_dai = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
-	.codec_dai	= "tlv320aic23-hifi",
+	.codec_dai_name	= "tlv320aic23-hifi",
 	.platform_name	= "imx-pcm-audio.0",
 	.codec_name	= "tlv320aic23-codec.0-001a",
-	.cpu_dai = "imx-ssi.0",
+	.cpu_dai_name	= "imx-ssi.0",
 	.ops		= &eukrea_tlv320_snd_ops,
 };
 
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index fd493ee1428e..671ef8dd524c 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -20,6 +20,7 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
+#include <linux/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/initval.h>
@@ -27,165 +28,146 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <mach/dma-mx1-mx2.h>
+#include <mach/dma.h>
 
 #include "imx-ssi.h"
 
 struct imx_pcm_runtime_data {
-	int sg_count;
-	struct scatterlist *sg_list;
-	int period;
+	int period_bytes;
 	int periods;
-	unsigned long dma_addr;
 	int dma;
-	struct snd_pcm_substream *substream;
 	unsigned long offset;
 	unsigned long size;
-	unsigned long period_cnt;
 	void *buf;
 	int period_time;
+	struct dma_async_tx_descriptor *desc;
+	struct dma_chan *dma_chan;
+	struct imx_dma_data dma_data;
 };
 
-/* Called by the DMA framework when a period has elapsed */
-static void imx_ssi_dma_progression(int channel, void *data,
-					struct scatterlist *sg)
+static void audio_dma_irq(void *data)
 {
-	struct snd_pcm_substream *substream = data;
+	struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 
-	if (!sg)
-		return;
-
-	runtime = iprtd->substream->runtime;
+	iprtd->offset += iprtd->period_bytes;
+	iprtd->offset %= iprtd->period_bytes * iprtd->periods;
 
-	iprtd->offset = sg->dma_address - runtime->dma_addr;
-
-	snd_pcm_period_elapsed(iprtd->substream);
+	snd_pcm_period_elapsed(substream);
 }
 
-static void imx_ssi_dma_callback(int channel, void *data)
+static bool filter(struct dma_chan *chan, void *param)
 {
-	pr_err("%s shouldn't be called\n", __func__);
-}
+	struct imx_pcm_runtime_data *iprtd = param;
 
-static void snd_imx_dma_err_callback(int channel, void *data, int err)
-{
-	struct snd_pcm_substream *substream = data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct imx_pcm_dma_params *dma_params = 
-		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
-	int ret;
+	if (!imx_dma_is_general_purpose(chan))
+		return false;
 
-	pr_err("DMA timeout on channel %d -%s%s%s%s\n",
-		 channel,
-		 err & IMX_DMA_ERR_BURST ?    " burst" : "",
-		 err & IMX_DMA_ERR_REQUEST ?  " request" : "",
-		 err & IMX_DMA_ERR_TRANSFER ? " transfer" : "",
-		 err & IMX_DMA_ERR_BUFFER ?   " buffer" : "");
+        chan->private = &iprtd->dma_data;
 
-	imx_dma_disable(iprtd->dma);
-	ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
-			IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
-			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-			DMA_MODE_WRITE : DMA_MODE_READ);
-	if (!ret)
-		imx_dma_enable(iprtd->dma);
+        return true;
 }
 
-static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
+static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct imx_pcm_dma_params *dma_params;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+	struct dma_slave_config slave_config;
+	dma_cap_mask_t mask;
+	enum dma_slave_buswidth buswidth;
 	int ret;
 
 	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
-	iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
-	if (iprtd->dma < 0) {
-		pr_err("Failed to claim the audio DMA\n");
-		return -ENODEV;
-	}
+	iprtd->dma_data.peripheral_type = IMX_DMATYPE_SSI;
+	iprtd->dma_data.priority = DMA_PRIO_HIGH;
+	iprtd->dma_data.dma_request = dma_params->dma;
 
-	ret = imx_dma_setup_handlers(iprtd->dma,
-				imx_ssi_dma_callback,
-				snd_imx_dma_err_callback, substream);
-	if (ret)
-		goto out;
+	/* Try to grab a DMA channel */
+	dma_cap_zero(mask);
+	dma_cap_set(DMA_SLAVE, mask);
+	iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+	if (!iprtd->dma_chan)
+		return -EINVAL;
 
-	ret = imx_dma_setup_progression_handler(iprtd->dma,
-			imx_ssi_dma_progression);
-	if (ret) {
-		pr_err("Failed to setup the DMA handler\n");
-		goto out;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+	case SNDRV_PCM_FORMAT_S24_LE:
+		buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
+		break;
+	default:
+		return 0;
 	}
 
-	ret = imx_dma_config_channel(iprtd->dma,
-			IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
-			IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
-			dma_params->dma, 1);
-	if (ret < 0) {
-		pr_err("Cannot configure DMA channel: %d\n", ret);
-		goto out;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		slave_config.direction = DMA_TO_DEVICE;
+		slave_config.dst_addr = dma_params->dma_addr;
+		slave_config.dst_addr_width = buswidth;
+		slave_config.dst_maxburst = dma_params->burstsize;
+	} else {
+		slave_config.direction = DMA_FROM_DEVICE;
+		slave_config.src_addr = dma_params->dma_addr;
+		slave_config.src_addr_width = buswidth;
+		slave_config.src_maxburst = dma_params->burstsize;
 	}
 
-	imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2);
+	ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config);
+	if (ret)
+		return ret;
 
 	return 0;
-out:
-	imx_dma_free(iprtd->dma);
-	return ret;
 }
 
 static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
-	int i;
 	unsigned long dma_addr;
+	struct dma_chan *chan;
+	struct imx_pcm_dma_params *dma_params;
+	int ret;
 
-	imx_ssi_dma_alloc(substream);
+	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	ret = imx_ssi_dma_alloc(substream, params);
+	if (ret)
+		return ret;
+	chan = iprtd->dma_chan;
 
 	iprtd->size = params_buffer_bytes(params);
 	iprtd->periods = params_periods(params);
-	iprtd->period = params_period_bytes(params);
+	iprtd->period_bytes = params_period_bytes(params);
 	iprtd->offset = 0;
 	iprtd->period_time = HZ / (params_rate(params) /
 			params_period_size(params));
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 
-	if (iprtd->sg_count != iprtd->periods) {
-		kfree(iprtd->sg_list);
-
-		iprtd->sg_list = kcalloc(iprtd->periods + 1,
-				sizeof(struct scatterlist), GFP_KERNEL);
-		if (!iprtd->sg_list)
-			return -ENOMEM;
-		iprtd->sg_count = iprtd->periods + 1;
-	}
-
-	sg_init_table(iprtd->sg_list, iprtd->sg_count);
 	dma_addr = runtime->dma_addr;
 
-	for (i = 0; i < iprtd->periods; i++) {
-		iprtd->sg_list[i].page_link = 0;
-		iprtd->sg_list[i].offset = 0;
-		iprtd->sg_list[i].dma_address = dma_addr;
-		iprtd->sg_list[i].length = iprtd->period;
-		dma_addr += iprtd->period;
+	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
+
+	iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+			iprtd->period_bytes * iprtd->periods,
+			iprtd->period_bytes,
+			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+			DMA_TO_DEVICE : DMA_FROM_DEVICE);
+	if (!iprtd->desc) {
+		dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+		return -EINVAL;
 	}
 
-	/* close the loop */
-	iprtd->sg_list[iprtd->sg_count - 1].offset = 0;
-	iprtd->sg_list[iprtd->sg_count - 1].length = 0;
-	iprtd->sg_list[iprtd->sg_count - 1].page_link =
-			((unsigned long) iprtd->sg_list | 0x01) & ~0x02;
+	iprtd->desc->callback = audio_dma_irq;
+	iprtd->desc->callback_param = substream;
+
 	return 0;
 }
 
@@ -194,41 +176,21 @@ static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 
-	if (iprtd->dma >= 0) {
-		imx_dma_free(iprtd->dma);
-		iprtd->dma = -EINVAL;
+	if (iprtd->dma_chan) {
+		dma_release_channel(iprtd->dma_chan);
+		iprtd->dma_chan = NULL;
 	}
 
-	kfree(iprtd->sg_list);
-	iprtd->sg_list = NULL;
-
 	return 0;
 }
 
 static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct imx_pcm_dma_params *dma_params;
-	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
-	int err;
 
 	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
-	iprtd->substream = substream;
-	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
-	iprtd->period_cnt = 0;
-
-	pr_debug("%s: buf: %p period: %d periods: %d\n",
-			__func__, iprtd->buf, iprtd->period, iprtd->periods);
-
-	err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
-			IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
-			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-			DMA_MODE_WRITE : DMA_MODE_READ);
-	if (err)
-		return err;
-
 	return 0;
 }
 
@@ -241,14 +203,14 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		imx_dma_enable(iprtd->dma);
+		dmaengine_submit(iprtd->desc);
 
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		imx_dma_disable(iprtd->dma);
+		dmaengine_terminate_all(iprtd->dma_chan);
 
 		break;
 	default:
@@ -263,6 +225,9 @@ static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 
+	pr_debug("%s: %ld %ld\n", __func__, iprtd->offset,
+			bytes_to_frames(substream->runtime, iprtd->offset));
+
 	return bytes_to_frames(substream->runtime, iprtd->offset);
 }
 
@@ -279,7 +244,7 @@ static struct snd_pcm_hardware snd_imx_hardware = {
 	.channels_max = 2,
 	.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
 	.period_bytes_min = 128,
-	.period_bytes_max = 16 * 1024,
+	.period_bytes_max = 65535, /* Limited by SDMA engine */
 	.periods_min = 2,
 	.periods_max = 255,
 	.fifo_size = 0,
@@ -304,11 +269,23 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
 	}
 
 	snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
+
+	return 0;
+}
+
+static int snd_imx_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+	kfree(iprtd);
+
 	return 0;
 }
 
 static struct snd_pcm_ops imx_pcm_ops = {
 	.open		= snd_imx_open,
+	.close		= snd_imx_close,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= snd_imx_pcm_hw_params,
 	.hw_free	= snd_imx_pcm_hw_free,
@@ -340,7 +317,6 @@ static struct platform_driver imx_pcm_driver = {
 			.name = "imx-pcm-audio",
 			.owner = THIS_MODULE,
 	},
-
 	.probe = imx_soc_platform_probe,
 	.remove = __devexit_p(imx_soc_platform_remove),
 };
@@ -356,4 +332,3 @@ static void __exit snd_imx_pcm_exit(void)
 	platform_driver_unregister(&imx_pcm_driver);
 }
 module_exit(snd_imx_pcm_exit);
-
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index d4bd345b0a8d..390b6ffc2658 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -439,7 +439,22 @@ void imx_pcm_free(struct snd_pcm *pcm)
 }
 EXPORT_SYMBOL_GPL(imx_pcm_free);
 
+static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
+{
+	struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
+	uint32_t val;
+
+	snd_soc_dai_set_drvdata(dai, ssi);
+
+	val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
+		SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
+	writel(val, ssi->base + SSI_SFCSR);
+
+	return 0;
+}
+
 static struct snd_soc_dai_driver imx_ssi_dai = {
+	.probe = imx_ssi_dai_probe,
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
@@ -455,20 +470,6 @@ static struct snd_soc_dai_driver imx_ssi_dai = {
 	.ops = &imx_ssi_pcm_dai_ops,
 };
 
-static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
-{
-	struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
-	uint32_t val;
-
-	snd_soc_dai_set_drvdata(dai, ssi);
-
-	val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
-		SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
-	writel(val, ssi->base + SSI_SFCSR);
-
-	return 0;
-}
-
 static struct snd_soc_dai_driver imx_ac97_dai = {
 	.probe = imx_ssi_dai_probe,
 	.ac97_control = 1,
@@ -677,9 +678,25 @@ static int imx_ssi_probe(struct platform_device *pdev)
 		goto failed_register;
 	}
 
-	ssi->soc_platform_pdev = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
-	if (!ssi->soc_platform_pdev)
+	ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
+	if (!ssi->soc_platform_pdev_fiq) {
+		ret = -ENOMEM;
+		goto failed_pdev_fiq_alloc;
+	}
+
+	platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi);
+	ret = platform_device_add(ssi->soc_platform_pdev_fiq);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to add platform device\n");
+		goto failed_pdev_fiq_add;
+	}
+
+	ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id);
+	if (!ssi->soc_platform_pdev) {
+		ret = -ENOMEM;
 		goto failed_pdev_alloc;
+	}
+
 	platform_set_drvdata(ssi->soc_platform_pdev, ssi);
 	ret = platform_device_add(ssi->soc_platform_pdev);
 	if (ret) {
@@ -692,6 +709,10 @@ static int imx_ssi_probe(struct platform_device *pdev)
 failed_pdev_add:
 	platform_device_put(ssi->soc_platform_pdev);
 failed_pdev_alloc:
+	platform_device_del(ssi->soc_platform_pdev_fiq);
+failed_pdev_fiq_add:
+	platform_device_put(ssi->soc_platform_pdev_fiq);
+failed_pdev_fiq_alloc:
 	snd_soc_unregister_dai(&pdev->dev);
 failed_register:
 failed_ac97:
@@ -712,8 +733,8 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev)
 	struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	struct imx_ssi *ssi = platform_get_drvdata(pdev);
 
-	platform_device_del(ssi->soc_platform_pdev);
-	platform_device_put(ssi->soc_platform_pdev);
+	platform_device_unregister(ssi->soc_platform_pdev);
+	platform_device_unregister(ssi->soc_platform_pdev_fiq);
 
 	snd_soc_unregister_dai(&pdev->dev);
 
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index 53b780d9b2b0..a4406a134892 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -185,6 +185,9 @@
 
 #define DRV_NAME "imx-ssi"
 
+#include <linux/dmaengine.h>
+#include <mach/dma.h>
+
 struct imx_pcm_dma_params {
 	int dma;
 	unsigned long dma_addr;
@@ -212,6 +215,7 @@ struct imx_ssi {
 	int enabled;
 
 	struct platform_device *soc_platform_pdev;
+	struct platform_device *soc_platform_pdev_fiq;
 };
 
 struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev,
diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c
index 6a65dd705519..9eabc28667e6 100644
--- a/sound/soc/imx/phycore-ac97.c
+++ b/sound/soc/imx/phycore-ac97.c
@@ -20,9 +20,6 @@
 #include <sound/soc-dapm.h>
 #include <asm/mach-types.h>
 
-#include "../codecs/wm9712.h"
-#include "imx-ssi.h"
-
 static struct snd_soc_card imx_phycore;
 
 static struct snd_soc_ops imx_phycore_hifi_ops = {
@@ -41,11 +38,12 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = {
 };
 
 static struct snd_soc_card imx_phycore = {
-	.name		= "PhyCORE-audio",
+	.name		= "PhyCORE-ac97-audio",
 	.dai_link	= imx_phycore_dai_ac97,
 	.num_links	= ARRAY_SIZE(imx_phycore_dai_ac97),
 };
 
+static struct platform_device *imx_phycore_snd_ac97_device;
 static struct platform_device *imx_phycore_snd_device;
 
 static int __init imx_phycore_init(void)
@@ -56,29 +54,42 @@ static int __init imx_phycore_init(void)
 		/* return happy. We might run on a totally different machine */
 		return 0;
 
-	imx_phycore_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!imx_phycore_snd_device)
+	imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+	if (!imx_phycore_snd_ac97_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(imx_phycore_snd_device, &imx_phycore);
-	ret = platform_device_add(imx_phycore_snd_device);
+	platform_set_drvdata(imx_phycore_snd_ac97_device, &imx_phycore);
+	ret = platform_device_add(imx_phycore_snd_ac97_device);
+	if (ret)
+		goto fail1;
 
 	imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1);
-	if (!imx_phycore_snd_device)
-		return -ENOMEM;
+	if (!imx_phycore_snd_device) {
+		ret = -ENOMEM;
+		goto fail2;
+	}
 	ret = platform_device_add(imx_phycore_snd_device);
 
 	if (ret) {
 		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
-		platform_device_put(imx_phycore_snd_device);
+		goto fail3;
 	}
 
+	return 0;
+
+fail3:
+	platform_device_put(imx_phycore_snd_device);
+fail2:
+	platform_device_del(imx_phycore_snd_ac97_device);
+fail1:
+	platform_device_put(imx_phycore_snd_ac97_device);
 	return ret;
 }
 
 static void __exit imx_phycore_exit(void)
 {
 	platform_device_unregister(imx_phycore_snd_device);
+	platform_device_unregister(imx_phycore_snd_ac97_device);
 }
 
 late_initcall(imx_phycore_init);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 293dc748797c..dac6732da969 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -49,7 +49,7 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97,
 	mutex_lock(&ac97_mutex);
 
 	val = nuc900_checkready();
-	if (!!val) {
+	if (val) {
 		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
 		goto out;
 	}
@@ -102,7 +102,7 @@ static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	mutex_lock(&ac97_mutex);
 
 	tmp = nuc900_checkready();
-	if (!!tmp)
+	if (tmp)
 		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
 
 	/* clear the R_WB bit and write register index */
@@ -149,7 +149,7 @@ static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97)
 	udelay(100);
 
 	val = nuc900_checkready();
-	if (!!val)
+	if (val)
 		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
 
 	mutex_unlock(&ac97_mutex);
@@ -263,8 +263,7 @@ static int nuc900_ac97_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static int nuc900_ac97_probe(struct platform_device *pdev,
-					struct snd_soc_dai *dai)
+static int nuc900_ac97_probe(struct snd_soc_dai *dai)
 {
 	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
 	unsigned long val;
@@ -284,12 +283,12 @@ static int nuc900_ac97_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void nuc900_ac97_remove(struct platform_device *pdev,
-						struct snd_soc_dai *dai)
+static int nuc900_ac97_remove(struct snd_soc_dai *dai)
 {
 	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
 
 	clk_disable(nuc900_audio->clk);
+	return 0;
 }
 
 static struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
@@ -313,7 +312,7 @@ static struct snd_soc_dai_driver nuc900_ac97_dai = {
 		.channels_max	= 2,
 	},
 	.ops = &nuc900_ac97_dai_ops,
-}
+};
 
 static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
 {
@@ -384,7 +383,6 @@ out0:
 
 static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
 {
-
 	snd_soc_unregister_dai(&pdev->dev);
 
 	clk_put(nuc900_ac97_data->clk);
@@ -392,6 +390,7 @@ static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
 	release_mem_region(nuc900_ac97_data->res->start,
 				resource_size(nuc900_ac97_data->res));
 
+	kfree(nuc900_ac97_data);
 	nuc900_ac97_data = NULL;
 
 	return 0;
diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h
index aeed8ead2b2b..59f7e8ed1a68 100644
--- a/sound/soc/nuc900/nuc900-audio.h
+++ b/sound/soc/nuc900/nuc900-audio.h
@@ -110,4 +110,6 @@ struct nuc900_audio {
 
 };
 
+extern struct nuc900_audio *nuc900_ac97_data;
+
 #endif /*end _NUC900_AUDIO_H */
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 195d1ac94771..8263f56dc665 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -50,12 +50,12 @@ static int nuc900_dma_hw_params(struct snd_pcm_substream *substream,
 	unsigned long flags;
 	int ret = 0;
 
-	spin_lock_irqsave(&nuc900_audio->lock, flags);
-
 	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
 	if (ret < 0)
 		return ret;
 
+	spin_lock_irqsave(&nuc900_audio->lock, flags);
+
 	nuc900_audio->substream = substream;
 	nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr;
 	nuc900_audio->buffersize[substream->stream] =
@@ -169,6 +169,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct nuc900_audio *nuc900_audio = runtime->private_data;
 	unsigned long flags, val;
+	int ret = 0;
 
 	spin_lock_irqsave(&nuc900_audio->lock, flags);
 
@@ -197,10 +198,10 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
 		AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
 		break;
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 	spin_unlock_irqrestore(&nuc900_audio->lock, flags);
-	return 0;
+	return ret;
 }
 
 static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -332,7 +333,7 @@ static struct snd_soc_platform_driver nuc900_soc_platform = {
 	.ops		= &nuc900_dma_ops,
 	.pcm_new	= nuc900_dma_new,
 	.pcm_free	= nuc900_dma_free_dma_buffers,
-}
+};
 
 static int __devinit nuc900_soc_platform_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index d542ea2ff6be..a088db6d5091 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -12,8 +12,8 @@ config SND_OMAP_SOC_MCPDM
 config SND_OMAP_SOC_N810
 	tristate "SoC Audio support for Nokia N810"
 	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
+	depends on OMAP_MUX
 	select SND_OMAP_SOC_MCBSP
-	select OMAP_MUX
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d211c9fa5a91..7e84f24b9a88 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -644,15 +644,23 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
 
 	case OMAP_MCBSP_CLKR_SRC_CLKR:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR);
 		break;
 	case OMAP_MCBSP_CLKR_SRC_CLKX:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX);
 		break;
 	case OMAP_MCBSP_FSR_SRC_FSR:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR);
 		break;
 	case OMAP_MCBSP_FSR_SRC_FSX:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX);
 		break;
 	default:
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index dbd9d96b5f92..4ee33ce2cb98 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -306,6 +306,7 @@ static int __init omap3pandora_soc_init(void)
 		pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n",
 			dev_name(&omap3pandora_snd_device->dev),
 			PTR_ERR(omap3pandora_dac_reg));
+		ret = PTR_ERR(omap3pandora_dac_reg);
 		goto fail3;
 	}
 
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index f0e662556428..65ae00e976ef 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -177,7 +177,8 @@ static int __init osk_soc_init(void)
 	tlv320aic23_mclk = clk_get(dev, "mclk");
 	if (IS_ERR(tlv320aic23_mclk)) {
 		printk(KERN_ERR "Could not get mclk clock\n");
-		return -ENODEV;
+		err = PTR_ERR(tlv320aic23_mclk);
+		goto err2;
 	}
 
 	/*
@@ -188,7 +189,7 @@ static int __init osk_soc_init(void)
 		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
 			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
 			err = -ECANCELED;
-			goto err1;
+			goto err3;
 		}
 	}
 
@@ -196,9 +197,12 @@ static int __init osk_soc_init(void)
 	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
 
 	return 0;
-err1:
+
+err3:
 	clk_put(tlv320aic23_mclk);
+err2:
 	platform_device_del(osk_snd_device);
+err1:
 	platform_device_put(osk_snd_device);
 
 	return err;
@@ -207,6 +211,7 @@ err1:
 
 static void __exit osk_soc_exit(void)
 {
+	clk_put(tlv320aic23_mclk);
 	platform_device_unregister(osk_snd_device);
 }
 
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 37f191bbfdd9..580f48571303 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,6 +1,7 @@
 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
 	depends on ARCH_PXA
+	select SND_ARM
 	select SND_PXA2XX_LIB
 	help
 	  Say Y or M if you want to add support for codecs attached to
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 97e9423615c9..f451acd4935b 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -100,8 +100,13 @@ static int corgi_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	corgi_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index b8207ced4072..5ef0526924b9 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -72,9 +72,13 @@ static int magician_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	magician_ext_control(codec);
 
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index af84ee9c5e11..84edd0385a21 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -77,8 +77,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	poodle_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index f470f360f4dd..0b30d7de24ec 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -108,8 +108,13 @@ static int spitz_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	spitz_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index a3bfb2e8b70f..7b983f935454 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -79,10 +79,15 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
+
+	mutex_lock(&codec->mutex);
 
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 8a6b53ccd203..d85bf8a0abb2 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -2,6 +2,7 @@ config SND_S3C24XX_SOC
 	tristate "SoC Audio for the Samsung S3CXXXX chips"
 	depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210
 	select S3C64XX_DMA if ARCH_S3C64XX
+	select S3C2410_DMA if ARCH_S3C2410
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the S3C24XX AC97 or I2S interfaces. You will also need to
diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c
index ffd5cf2fb0a9..468cc11fdf47 100644
--- a/sound/soc/s3c24xx/rx1950_uda1380.c
+++ b/sound/soc/s3c24xx/rx1950_uda1380.c
@@ -50,7 +50,6 @@ static unsigned int rates[] = {
 	16000,
 	44100,
 	48000,
-	88200,
 };
 
 static struct snd_pcm_hw_constraint_list hw_rates = {
@@ -130,7 +129,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 };
 
 static struct platform_device *s3c24xx_snd_device;
-static struct clk *xtal;
 
 static int rx1950_startup(struct snd_pcm_substream *substream)
 {
@@ -179,10 +177,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
 	case 44100:
 	case 88200:
 		clk_source = S3C24XX_CLKSRC_MPLL;
-		fs_mode = S3C2410_IISMOD_256FS;
-		div = clk_get_rate(xtal) / (256 * rate);
-		if (clk_get_rate(xtal) % (256 * rate) > (128 * rate))
-			div++;
+		fs_mode = S3C2410_IISMOD_384FS;
+		div = 1;
 		break;
 	default:
 		printk(KERN_ERR "%s: rate %d is not supported\n",
@@ -210,7 +206,7 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
 
 	/* set MCLK division for sample rate */
 	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
-		S3C2410_IISMOD_384FS);
+		fs_mode);
 	if (ret < 0)
 		return ret;
 
@@ -295,17 +291,8 @@ static int __init rx1950_init(void)
 		goto err_plat_add;
 	}
 
-	xtal = clk_get(&s3c24xx_snd_device->dev, "xtal");
-
-	if (IS_ERR(xtal)) {
-		ret = PTR_ERR(xtal);
-		platform_device_unregister(s3c24xx_snd_device);
-		goto err_clk;
-	}
-
 	return 0;
 
-err_clk:
 err_plat_add:
 err_plat_alloc:
 err_gpio_conf:
@@ -320,7 +307,6 @@ static void __exit rx1950_exit(void)
 	platform_device_unregister(s3c24xx_snd_device);
 	snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
 		hp_jack_gpios);
-	clk_put(xtal);
 	gpio_free(S3C2410_GPA(1));
 }
 
diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c
index 11c88b13dd2e..761e02b4a6e0 100644
--- a/sound/soc/s3c24xx/smdk_spdif.c
+++ b/sound/soc/s3c24xx/smdk_spdif.c
@@ -38,7 +38,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev)
 	}
 
 	mout_epll = clk_get(NULL, "mout_epll");
-	if (IS_ERR(fout_epll)) {
+	if (IS_ERR(mout_epll)) {
 		printk(KERN_WARNING "%s: Cannot find mout_epll.\n",
 				__func__);
 		ret = -EINVAL;
@@ -54,7 +54,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev)
 	}
 
 	sclk_spdif = clk_get(NULL, "sclk_spdif");
-	if (IS_ERR(fout_epll)) {
+	if (IS_ERR(sclk_spdif)) {
 		printk(KERN_WARNING "%s: Cannot find sclk_spdif.\n",
 				__func__);
 		ret = -EINVAL;
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 8778faa174a6..3052f64b2403 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -434,7 +434,7 @@ static struct snd_soc_dai_driver s6000_i2s_dai = {
 		.rate_max = 1562500,
 	},
 	.ops = &s6000_i2s_dai_ops,
-}
+};
 
 static int __devinit s6000_i2s_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 271fd222bf19..ab3ccaec72d2 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -473,7 +473,7 @@ static int s6000_pcm_new(struct snd_card *card,
 	}
 
 	res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED,
-			  s6000_soc_platform.name, pcm);
+			  "s6000-audio", pcm);
 	if (res) {
 		printk(KERN_ERR "s6000-pcm couldn't get IRQ\n");
 		return res;
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 96c05e137538..c1244c5bc730 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -167,7 +167,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 
 	snd_soc_dapm_sync(codec);
 
-	snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec));
+	snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec));
 
 	return 0;
 }
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 507e709f2807..4c2404b1b862 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -132,6 +132,8 @@ struct fsi_priv {
 	struct fsi_stream playback;
 	struct fsi_stream capture;
 
+	long rate;
+
 	u32 mst_ctrl;
 };
 
@@ -854,10 +856,17 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	int is_play = fsi_is_play(substream);
+	struct fsi_master *master = fsi_get_master(fsi);
+	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 
 	fsi_irq_disable(fsi, is_play);
 	fsi_clk_ctrl(fsi, 0);
 
+	set_rate = master->info->set_rate;
+	if (set_rate && fsi->rate)
+		set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
+	fsi->rate = 0;
+
 	pm_runtime_put_sync(dai->dev);
 }
 
@@ -891,20 +900,20 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	struct fsi_master *master = fsi_get_master(fsi);
-	int (*set_rate)(int is_porta, int rate) = master->info->set_rate;
+	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 	int fsi_ver = master->core->ver;
-	int is_play = fsi_is_play(substream);
+	long rate = params_rate(params);
 	int ret;
 
-	/* if slave mode, set_rate is not needed */
-	if (!fsi_is_master_mode(fsi, is_play))
+	set_rate = master->info->set_rate;
+	if (!set_rate)
 		return 0;
 
-	/* it is error if no set_rate */
-	if (!set_rate)
-		return -EIO;
+	ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
+	if (ret < 0) /* error */
+		return ret;
 
-	ret = set_rate(fsi_is_port_a(fsi), params_rate(params));
+	fsi->rate = rate;
 	if (ret > 0) {
 		u32 data = 0;
 
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 40bbdf1591dc..05192d97b377 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -387,7 +387,7 @@ static int __devinit sh4_soc_dai_probe(struct platform_device *pdev)
 
 static int __devexit sh4_soc_dai_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_dai(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai));
+	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai));
 	return 0;
 }
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1c8f3f507f54..85b7d548f167 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev,
 {
 	struct snd_soc_pcm_runtime *rtd =
 			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	int ret;
 
-	strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	if (ret)
+		return ret;
 
 	return count;
 }
@@ -1616,12 +1619,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	/* register any AC97 codecs */
 	for (i = 0; i < card->num_rtd; i++) {
-			ret = soc_register_ac97_dai_link(&card->rtd[i]);
-			if (ret < 0) {
-				printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name);
-				goto probe_dai_err;
-			}
+		ret = soc_register_ac97_dai_link(&card->rtd[i]);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name);
+			while (--i >= 0)
+				soc_unregister_ac97_dai_link(&card->rtd[i]);
+			goto probe_dai_err;
 		}
+	}
 #endif
 
 	card->instantiated = 1;
@@ -3040,8 +3045,10 @@ int snd_soc_register_dais(struct device *dev,
 	for (i = 0; i < count; i++) {
 
 		dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL);
-		if (dai == NULL)
-			return -ENOMEM;
+		if (dai == NULL) {
+			ret = -ENOMEM;
+			goto err;
+		}
 
 		/* create DAI component name */
 		dai->name = fmt_multiple_name(dev, &dai_drv[i]);
@@ -3067,7 +3074,9 @@ int snd_soc_register_dais(struct device *dev,
 		pr_debug("Registered DAI '%s'\n", dai->name);
 	}
 
+	mutex_lock(&client_mutex);
 	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
 	return 0;
 
 err:
@@ -3260,9 +3269,6 @@ int snd_soc_register_codec(struct device *dev,
 	return 0;
 
 error:
-	for (i--; i >= 0; i--)
-		snd_soc_unregister_dai(dev);
-
 	if (codec->reg_cache)
 		kfree(codec->reg_cache);
 	kfree(codec->name);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7d85c6496afa..c721502833bc 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -683,12 +683,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
 			    struct snd_soc_dapm_widget *b,
 			    int sort[])
 {
-	if (a->codec != b->codec)
-		return (unsigned long)a - (unsigned long)b;
 	if (sort[a->id] != sort[b->id])
 		return sort[a->id] - sort[b->id];
 	if (a->reg != b->reg)
 		return a->reg - b->reg;
+	if (a->codec != b->codec)
+		return (unsigned long)a->codec - (unsigned long)b->codec;
 
 	return 0;
 }
@@ -944,6 +944,9 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 		case SND_SOC_DAPM_STREAM_RESUME:
 			sys_power = 1;
 			break;
+		case SND_SOC_DAPM_STREAM_STOP:
+			sys_power = !!codec->active;
+			break;
 		case SND_SOC_DAPM_STREAM_SUSPEND:
 			sys_power = 0;
 			break;
diff --git a/sound/sound_core.c b/sound/sound_core.c
index c03bbaefdbc3..5580aced8730 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -104,7 +104,6 @@ module_exit(cleanup_soundcore);
 
 #include <linux/init.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/types.h>
 #include <linux/kernel.h>
 #include <linux/sound.h>
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 1bc56b2b94e2..337a00241a1f 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -155,7 +155,7 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip)
 	if (max_tries < 1)
 		max_tries = 1;
 
-	/* ssc_div must be a power of 2. */
+	/* ssc_div must be even. */
 	ssc_div = (ssc_div + 1) & ~1UL;
 
 	if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) {
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 7dae05d8783e..782f741cd00a 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -60,7 +60,7 @@ static const struct rc_config {
 	{ USB_ID(0x041e, 0x3000), 0, 1, 2, 1,  18, 0x0013 }, /* Extigy       */
 	{ USB_ID(0x041e, 0x3020), 2, 1, 6, 6,  18, 0x0013 }, /* Audigy 2 NX  */
 	{ USB_ID(0x041e, 0x3040), 2, 2, 6, 6,  2,  0x6e91 }, /* Live! 24-bit */
-	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi */
+	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 */
 	{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6,  2,  0x6e91 }, /* Toshiba SB0500 */
 };
 
@@ -183,7 +183,13 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	if (value > 1)
 		return -EINVAL;
 	changed = value != mixer->audigy2nx_leds[index];
-	err = snd_usb_ctl_msg(mixer->chip->dev,
+	if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042))
+		err = snd_usb_ctl_msg(mixer->chip->dev,
+			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
+			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+			      !value, 0, NULL, 0, 100);
+	else
+		err = snd_usb_ctl_msg(mixer->chip->dev,
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
 			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
 			      value, index + 2, NULL, 0, 100);
@@ -225,8 +231,12 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer)
 	int i, err;
 
 	for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) {
+		/* USB X-Fi S51 doesn't have a CMSS LED */
+		if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0)
+			continue;
 		if (i > 1 && /* Live24ext has 2 LEDs only */
 			(mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+			 mixer->chip->usb_id == USB_ID(0x041e, 0x3042) ||
 			 mixer->chip->usb_id == USB_ID(0x041e, 0x3048)))
 			break; 
 		err = snd_ctl_add(mixer->chip->card,
@@ -365,6 +375,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 
 	if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) ||
 	    mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+	    mixer->chip->usb_id == USB_ID(0x041e, 0x3042) ||
 	    mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) {
 		if ((err = snd_audigy2nx_controls_create(mixer)) < 0)
 			return err;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index cff3a3c465d7..4132522ac90f 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -676,8 +676,10 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime,
 	if (!needs_knot)
 		return 0;
 
-	subs->rate_list.count = count;
 	subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL);
+	if (!subs->rate_list.list)
+		return -ENOMEM;
+	subs->rate_list.count = count;
 	subs->rate_list.mask = 0;
 	count = 0;
 	list_for_each_entry(fp, &subs->fmt_list, list) {