Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (41 commits)
  ALSA: hda - Identify more variants for ALC269
  ALSA: hda - Fix wrong ALC269 variant check
  ALSA: hda - Enable jack sense for Thinkpad Edge 11
  ALSA: Revert "ALSA: hda - Fix switching between dmic and mic using the same mux on IDT/STAC"
  ALSA: hda - Fixed ALC887-VD initial error
  ALSA: atmel - Fix the return value in error path
  ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J
  ALSA: snd-atmel-abdac: test wrong variable
  ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer
  ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup
  ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of pci_get_drvdata
  ALSA: sound/core/pcm_lib.c: Remove unnecessary semicolons
  ALSA: sound/ppc: Use printf extension %pR for struct resource
  ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and Headphone controls
  ASoC: uda134x - set reg_cache_default to uda134x_reg
  ASoC: Add support for MAX98089 CODEC
  ASoC: davinci: fixes for multi-component
  ASoC: Fix register cache setup WM8994 for multi-component
  ASoC: Fix dapm_seq_compare() for multi-component
  ASoC: RX1950: Fix hw_params function
  ...

44 files changed, 367 insertions, 248 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/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/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/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/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/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6361f752b5f3..846d1ead47fd 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3100,6 +3100,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	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),
@@ -3110,6 +3111,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
 	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_realtek.c b/sound/pci/hda/patch_realtek.c
index 5f00589cb791..0ac6aed0c889 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14623,7 +14623,10 @@ 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,
 };
 
@@ -15023,7 +15026,7 @@ 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 +15037,23 @@ 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 +15116,7 @@ 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;
@@ -19298,6 +19310,7 @@ 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 +19432,10 @@ 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 +19925,7 @@ 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..5c710807dfe5 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),
@@ -3486,10 +3491,8 @@ 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 +5455,17 @@ 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/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/codecs/max98088.c b/sound/soc/codecs/max98088.c
index bc22ee93a75d..470cb93b1d1f 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;
@@ -2040,6 +2046,8 @@ 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 +2067,8 @@ 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/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/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/wm8962.c b/sound/soc/codecs/wm8962.c
index 894d0cd3aa9b..e8092745a207 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -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..830dfdd66c5f 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;
 
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..fb4cc1edf339 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,
 };
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/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..d2d98c75ee8a 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,
@@ -677,7 +678,17 @@ 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);
+	ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
+	if (!ssi->soc_platform_pdev_fiq)
+		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)
 		goto failed_pdev_alloc;
 	platform_set_drvdata(ssi->soc_platform_pdev, ssi);
@@ -692,6 +703,9 @@ static int imx_ssi_probe(struct platform_device *pdev)
 failed_pdev_add:
 	platform_device_put(ssi->soc_platform_pdev);
 failed_pdev_alloc:
+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:
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..39f23734781a 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 = {
@@ -41,7 +38,7 @@ 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),
 };
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/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 73d0edd8ded9..7b983f935454 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -81,8 +81,13 @@ static int tosa_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 */
 	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/soc-core.c b/sound/soc/soc-core.c
index 614a8b30d87b..441285ade024 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3043,8 +3043,10 @@ 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]);
@@ -3263,9 +3265,6 @@ 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..75ed6491222d 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;
 }
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) {