Merge 3.6-rc3 into tty-next

This picks up all of the different fixes in Linus's tree that we also need here.

Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

58 files changed, 335 insertions, 260 deletions

diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 0d7b25e..4e1fda7 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
 	.prepare		= pxa2xx_ac97_pcm_prepare,
 };
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 
 static int pxa2xx_ac97_do_suspend(struct snd_card *card)
 {
@@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = {
 	.driver		= {
 		.name	= "pxa2xx-ac97",
 		.owner	= THIS_MODULE,
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 		.pm	= &pxa2xx_ac97_pm_ops,
 #endif
 	},
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index eb4ceb7..277ebce 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
 	dac->regs = ioremap(regs->start, resource_size(regs));
 	if (!dac->regs) {
 		dev_dbg(&pdev->dev, "could not remap register memory\n");
+		retval = -ENOMEM;
 		goto out_free_card;
 	}
 
@@ -534,7 +535,7 @@ out_put_pclk:
 	return retval;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int atmel_abdac_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index bf47025..9052aff 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream,
 	if (retval < 0)
 		return retval;
 	/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
-	if (cpu_is_at32ap7000()) {
-		if (retval < 0)
-			return retval;
-		/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
-		if (retval == 1)
-			if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
-				dw_dma_cyclic_free(chip->dma.rx_chan);
-	}
+	if (cpu_is_at32ap7000() && retval == 1)
+		if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
+			dw_dma_cyclic_free(chip->dma.rx_chan);
 
 	/* Set restrictions to params. */
 	mutex_lock(&opened_mutex);
@@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 
 	if (!chip->regs) {
 		dev_dbg(&pdev->dev, "could not remap register memory\n");
+		retval = -ENOMEM;
 		goto err_ioremap;
 	}
 
@@ -1134,7 +1130,7 @@ err_snd_card_new:
 	return retval;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int atmel_ac97c_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index 4e7ec2b..d0f0035 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -101,7 +101,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
 		if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device,
 						 chunk, &tmpb) < 0) {
 			if (!sgbuf->pages)
-				return NULL;
+				goto _failed;
 			if (!res_size)
 				goto _failed;
 			size = sgbuf->pages * PAGE_SIZE;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 1128b35..5a34355 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr)
 	return 0;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int loopback_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index f7d3bfc..54bb664 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr)
 	return 0;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int snd_dummy_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 6ca59fc..ef17129 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip)
 	pcspkr_stop_sound();
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int pcsp_suspend(struct device *dev)
 {
 	struct snd_pcsp *chip = dev_get_drvdata(dev);
@@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL);
 #define PCSP_PM_OPS	&pcsp_pm
 #else
 #define PCSP_PM_OPS	NULL
-#endif	/* CONFIG_PM */
+#endif	/* CONFIG_PM_SLEEP */
 
 static void pcsp_shutdown(struct platform_device *dev)
 {
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 2d67c78..f7cdaf5 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev,
 			irq[dev], dma8[dev], dma16[dev]);
 	}
 
-	if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
+	if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) {
 		snd_card_free(card);
 		return error;
 	}
diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c
index 733b014..b2b3c01 100644
--- a/sound/oss/sb_audio.c
+++ b/sound/oss/sb_audio.c
@@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed)
 	if (speed > 0)
 	{
 		int tmp;
-		int s = speed * devc->channels;
+		int s;
 
 		if (speed < 5000)
 			speed = 5000;
 		if (speed > 44100)
 			speed = 44100;
 
+		s = speed * devc->channels;
+
 		devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff;
 
 		tmp = 256 - devc->tconst;
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index f75f5ff..a71d1c1 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
 
 	if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
 		       codec_index != CS46XX_SECONDARY_CODEC_INDEX))
-		return -EINVAL;
+		return 0xffff;
 
 	chip->active_ctrl(chip, 1);
 
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 8e40262..2f6e9c7 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
 	atc_connect_resources(atc);
 
 	atc->timer = ct_timer_new(atc);
-	if (!atc->timer)
+	if (!atc->timer) {
+		err = -ENOMEM;
 		goto error1;
+	}
 
 	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops);
 	if (err < 0)
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 4f502a2..0a43662 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
 	for (page = blk->first_page; page <= blk->last_page; page++, idx++) {
 		unsigned long ofs = idx << PAGE_SHIFT;
 		dma_addr_t addr;
-		addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+		if (ofs >= runtime->dma_bytes)
+			addr = emu->silent_page.addr;
+		else
+			addr = snd_pcm_sgbuf_get_addr(substream, ofs);
 		if (! is_valid_page(emu, addr)) {
 			printk(KERN_ERR "emu: failure page = %d\n", idx);
 			mutex_unlock(&hdr->block_mutex);
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 647218d..4f7d2df 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	if (cfg->dig_outs)
 		snd_printd("   dig-out=0x%x/0x%x\n",
 			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
-	snd_printd("   inputs:");
+	snd_printd("   inputs:\n");
 	for (i = 0; i < cfg->num_inputs; i++) {
-		snd_printd(" %s=0x%x",
+		snd_printd("     %s=0x%x\n",
 			    hda_get_autocfg_input_label(codec, cfg, i),
 			    cfg->inputs[i].pin);
 	}
-	snd_printd("\n");
 	if (cfg->dig_in_pin)
 		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
 
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 0bc2315..0849aac 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -231,16 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
 
+static bool ctl_has_mute(struct snd_kcontrol *kcontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	return query_amp_caps(codec, get_amp_nid(kcontrol),
+			      get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE;
+}
+
 /* get/put callbacks for beep mute mixer switches */
 int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
-	if (beep) {
+	if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
 		ucontrol->value.integer.value[0] =
-			ucontrol->value.integer.value[1] =
-			beep->enabled;
+			ucontrol->value.integer.value[1] = beep->enabled;
 		return 0;
 	}
 	return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
@@ -252,9 +258,20 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
-	if (beep)
-		snd_hda_enable_beep_device(codec,
-					   *ucontrol->value.integer.value);
+	if (beep) {
+		u8 chs = get_amp_channels(kcontrol);
+		int enable = 0;
+		long *valp = ucontrol->value.integer.value;
+		if (chs & 1) {
+			enable |= *valp;
+			valp++;
+		}
+		if (chs & 2)
+			enable |= *valp;
+		snd_hda_enable_beep_device(codec, enable);
+	}
+	if (!ctl_has_mute(kcontrol))
+		return 0;
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88a9c20..f560051 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
 
+/* update the stream-id if changed */
+static void update_pcm_stream_id(struct hda_codec *codec,
+				 struct hda_cvt_setup *p, hda_nid_t nid,
+				 u32 stream_tag, int channel_id)
+{
+	unsigned int oldval, newval;
+
+	if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
+		oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
+		newval = (stream_tag << 4) | channel_id;
+		if (oldval != newval)
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_CHANNEL_STREAMID,
+					    newval);
+		p->stream_tag = stream_tag;
+		p->channel_id = channel_id;
+	}
+}
+
+/* update the format-id if changed */
+static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p,
+			      hda_nid_t nid, int format)
+{
+	unsigned int oldval;
+
+	if (p->format_id != format) {
+		oldval = snd_hda_codec_read(codec, nid, 0,
+					    AC_VERB_GET_STREAM_FORMAT, 0);
+		if (oldval != format) {
+			msleep(1);
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_STREAM_FORMAT,
+					    format);
+		}
+		p->format_id = format;
+	}
+}
+
 /**
  * snd_hda_codec_setup_stream - set up the codec for streaming
  * @codec: the CODEC to set up
@@ -1400,7 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 {
 	struct hda_codec *c;
 	struct hda_cvt_setup *p;
-	unsigned int oldval, newval;
 	int type;
 	int i;
 
@@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 	p = get_hda_cvt_setup(codec, nid);
 	if (!p)
 		return;
-	/* update the stream-id if changed */
-	if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
-		oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
-		newval = (stream_tag << 4) | channel_id;
-		if (oldval != newval)
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_CHANNEL_STREAMID,
-					    newval);
-		p->stream_tag = stream_tag;
-		p->channel_id = channel_id;
-	}
-	/* update the format-id if changed */
-	if (p->format_id != format) {
-		oldval = snd_hda_codec_read(codec, nid, 0,
-					    AC_VERB_GET_STREAM_FORMAT, 0);
-		if (oldval != format) {
-			msleep(1);
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_STREAM_FORMAT,
-					    format);
-		}
-		p->format_id = format;
-	}
+
+	if (codec->pcm_format_first)
+		update_pcm_format(codec, p, nid, format);
+	update_pcm_stream_id(codec, p, nid, stream_tag, channel_id);
+	if (!codec->pcm_format_first)
+		update_pcm_format(codec, p, nid, format);
+
 	p->active = 1;
 	p->dirty = 0;
 
@@ -3497,7 +3518,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg
 {
 	int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE);
 
-	if (sup < 0)
+	if (sup == -1)
 		return false;
 	if (sup & power_state)
 		return true;
@@ -4433,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 	 * then there is no need to go through power up here.
 	 */
 	if (codec->power_on) {
+		if (codec->power_transition < 0)
+			codec->power_transition = 0;
 		spin_unlock(&codec->power_lock);
 		return;
 	}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index c422d33..7fbc1bc 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -861,6 +861,7 @@ struct hda_codec {
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
 	unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
 	unsigned int no_jack_detect:1;	/* Machine has no jack-detection */
+	unsigned int pcm_format_first:1; /* PCM format must be set first */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	int power_transition;	/* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8aced1..60882c6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, CPT},"
 			 "{Intel, PPT},"
 			 "{Intel, LPT},"
+			 "{Intel, LPT_LP},"
 			 "{Intel, HPT},"
 			 "{Intel, PBG},"
 			 "{Intel, SCH},"
@@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
 	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Lynx Point-LP */
+	{ PCI_DEVICE(0x8086, 0x9c20),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Lynx Point-LP */
+	{ PCI_DEVICE(0x8086, 0x9c21),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* Haswell */
 	{ PCI_DEVICE(0x8086, 0x0c0c),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 7e46258..6894ec6 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
 	if (digi1 & AC_DIG1_EMPHASIS)
 		snd_iprintf(buffer, " Preemphasis");
 	if (digi1 & AC_DIG1_COPYRIGHT)
-		snd_iprintf(buffer, " Copyright");
+		snd_iprintf(buffer, " Non-Copyright");
 	if (digi1 & AC_DIG1_NONAUDIO)
 		snd_iprintf(buffer, " Non-Audio");
 	if (digi1 & AC_DIG1_PROFESSIONAL)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index d0d3540..49750a9 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_UNMUTE);
 	}
-	if (dac)
+	if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP))
 		snd_hda_codec_write(codec, dac, 0,
 				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
 }
@@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_IN_UNMUTE(0));
 	}
-	if (adc)
+	if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP))
 		snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
 				    AMP_IN_UNMUTE(0));
 }
@@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
 	int type = dir ? HDA_INPUT : HDA_OUTPUT;
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+	if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) {
+		snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid);
+		return 0;
+	}
 	sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
 	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
 }
@@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
 	int type = dir ? HDA_INPUT : HDA_OUTPUT;
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+	if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) {
+		snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid);
+		return 0;
+	}
 	sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
 	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
 }
@@ -464,50 +472,17 @@ exit:
 }
 
 /*
- * PCM stuffs
+ * PCM callbacks
  */
-static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid,
-				 u32 stream_tag,
-				 int channel_id, int format)
+static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				    struct hda_codec *codec,
+				    struct snd_pcm_substream *substream)
 {
-	unsigned int oldval, newval;
-
-	if (!nid)
-		return;
-
-	snd_printdd("ca0132_setup_stream: "
-		"NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
-		nid, stream_tag, channel_id, format);
-
-	/* update the format-id if changed */
-	oldval = snd_hda_codec_read(codec, nid, 0,
-				    AC_VERB_GET_STREAM_FORMAT,
-				    0);
-	if (oldval != format) {
-		msleep(20);
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_STREAM_FORMAT,
-				    format);
-	}
-
-	oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
-	newval = (stream_tag << 4) | channel_id;
-	if (oldval != newval) {
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_CHANNEL_STREAMID,
-				    newval);
-	}
-}
-
-static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
-{
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+	struct ca0132_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+					     hinfo);
 }
 
-/*
- * PCM callbacks
- */
 static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
 			unsigned int stream_tag,
@@ -515,10 +490,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+						stream_tag, format, substream);
 }
 
 static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
@@ -526,92 +499,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dacs[0]);
-
-	return 0;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 }
 
 /*
  * Digital out
  */
-static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			unsigned int stream_tag,
-			unsigned int format,
-			struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
 }
 
-static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			struct snd_pcm_substream *substream)
-{
-	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dig_out);
-
-	return 0;
-}
-
-/*
- * Analog capture
- */
-static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
 			unsigned int stream_tag,
 			unsigned int format,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->adcs[substream->number],
-			     stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+					     stream_tag, format, substream);
 }
 
-static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->adcs[substream->number]);
-
-	return 0;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
 }
 
-/*
- * Digital capture
- */
-static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			unsigned int stream_tag,
-			unsigned int format,
-			struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format);
-
-	return 0;
-}
-
-static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			struct snd_pcm_substream *substream)
-{
-	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dig_in);
-
-	return 0;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
 
 /*
@@ -621,6 +547,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.ops = {
+		.open = ca0132_playback_pcm_open,
 		.prepare = ca0132_playback_pcm_prepare,
 		.cleanup = ca0132_playback_pcm_cleanup
 	},
@@ -630,10 +557,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	.ops = {
-		.prepare = ca0132_capture_pcm_prepare,
-		.cleanup = ca0132_capture_pcm_cleanup
-	},
 };
 
 static struct hda_pcm_stream ca0132_pcm_digital_playback = {
@@ -641,6 +564,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.ops = {
+		.open = ca0132_dig_playback_pcm_open,
+		.close = ca0132_dig_playback_pcm_close,
 		.prepare = ca0132_dig_playback_pcm_prepare,
 		.cleanup = ca0132_dig_playback_pcm_cleanup
 	},
@@ -650,10 +575,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	.ops = {
-		.prepare = ca0132_dig_capture_pcm_prepare,
-		.cleanup = ca0132_dig_capture_pcm_cleanup
-	},
 };
 
 static int ca0132_build_pcms(struct hda_codec *codec)
@@ -928,18 +849,16 @@ static int ca0132_build_controls(struct hda_codec *codec)
 						    spec->dig_out);
 		if (err < 0)
 			return err;
-		err = add_out_volume(codec, spec->dig_out, "IEC958");
+		err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
 		if (err < 0)
 			return err;
+		/* spec->multiout.share_spdif = 1; */
 	}
 
 	if (spec->dig_in) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
 		if (err < 0)
 			return err;
-		err = add_in_volume(codec, spec->dig_in, "IEC958");
-		if (err < 0)
-			return err;
 	}
 	return 0;
 }
@@ -961,6 +880,9 @@ static void ca0132_config(struct hda_codec *codec)
 	struct ca0132_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 
+	codec->pcm_format_first = 1;
+	codec->no_sticky_stream = 1;
+
 	/* line-outs */
 	cfg->line_outs = 1;
 	cfg->line_out_pins[0] = 0x0b; /* front */
@@ -988,14 +910,24 @@ static void ca0132_config(struct hda_codec *codec)
 
 	/* Mic-in */
 	spec->input_pins[0] = 0x12;
-	spec->input_labels[0] = "Mic-In";
+	spec->input_labels[0] = "Mic";
 	spec->adcs[0] = 0x07;
 
 	/* Line-In */
 	spec->input_pins[1] = 0x11;
-	spec->input_labels[1] = "Line-In";
+	spec->input_labels[1] = "Line";
 	spec->adcs[1] = 0x08;
 	spec->num_inputs = 2;
+
+	/* SPDIF I/O */
+	spec->dig_out = 0x05;
+	spec->multiout.dig_out_nid = spec->dig_out;
+	cfg->dig_out_pins[0] = 0x0c;
+	cfg->dig_outs = 1;
+	cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+	spec->dig_in = 0x09;
+	cfg->dig_in_pin = 0x0e;
+	cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
 }
 
 static void ca0132_init_chip(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 1436118..5e22a8f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2967,12 +2967,10 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 };
 
 static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO),
 	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 Vostro 320", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
-	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
@@ -2988,14 +2986,10 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
-	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
-	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
-	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
-	SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 69b9284..8f23374 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -877,8 +877,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 	struct hdmi_eld *eld;
 	struct hdmi_spec_per_cvt *per_cvt = NULL;
 
-	hinfo->nid = 0; /* clear the leftover value */
-
 	/* Validate hinfo */
 	pin_idx = hinfo_to_pin_index(spec, hinfo);
 	if (snd_BUG_ON(pin_idx < 0))
@@ -1163,6 +1161,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
 }
 
+static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					     struct hda_codec *codec,
+					     struct snd_pcm_substream *substream)
+{
+	snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+	return 0;
+}
+
 static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 			  struct hda_codec *codec,
 			  struct snd_pcm_substream *substream)
@@ -1202,6 +1208,7 @@ static const struct hda_pcm_ops generic_ops = {
 	.open = hdmi_pcm_open,
 	.close = hdmi_pcm_close,
 	.prepare = generic_hdmi_playback_pcm_prepare,
+	.cleanup = generic_hdmi_playback_pcm_cleanup,
 };
 
 static int generic_hdmi_build_pcms(struct hda_codec *codec)
@@ -1220,7 +1227,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
 		pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
 		pstr->substreams = 1;
 		pstr->ops = generic_ops;
-		pstr->nid = 1; /* FIXME: just for avoiding a debug WARNING */
 		/* other pstr fields are set in open */
 	}
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 344b221..4f81dd4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6099,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = {
 	[ALC269_FIXUP_PCM_44K] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_pcm_44k,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_QUANTA_MUTE
 	},
 	[ALC269_FIXUP_STEREO_DMIC] = {
 		.type = ALC_FIXUP_FUNC,
@@ -6206,9 +6208,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
+	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 
 #if 0
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 94040cc..ea5775a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec)
 	unsigned int gpio;
 	int i;
 
-	snd_hda_sequence_write(codec, spec->init);
+	if (spec->init)
+		snd_hda_sequence_write(codec, spec->init);
 
 	/* power down adcs initially */
 	if (spec->powerdown_adcs)
@@ -5748,7 +5749,6 @@ again:
 		/* fallthru */
 	case 0x111d76b4: /* 6 Port without Analog Mixer */
 	case 0x111d76b5:
-		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92xx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
@@ -5773,7 +5773,6 @@ again:
 			spec->stream_delay = 40; /* 40 milliseconds */
 
 		/* disable VSW */
-		spec->init = stac92hd71bxx_core_init;
 		unmute_init++;
 		snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
 		snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
@@ -5788,7 +5787,6 @@ again:
 
 		/* fallthru */
 	default:
-		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92xx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
@@ -5796,6 +5794,9 @@ again:
 		break;
 	}
 
+	if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB)
+		spec->init = stac92hd71bxx_core_init;
+
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
 		snd_hda_sequence_write_cache(codec, unmute_init);
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 80d90cb..4307717 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	vt1708_stop_hp_work(spec);
+
+	if (spec->codec_type == VT1802) {
+		/* Fix pop noise on headphones */
+		int i;
+		for (i = 0; i < spec->autocfg.hp_outs; i++)
+			snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0);
+	}
+
 	return 0;
 }
 #endif
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d1ab437..5579b08 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip)
 	/* hardcoded device name & channel count */
 	err = snd_pcm_new(chip->card, (char *)card_name, 0,
 			  1, 1, &pcm);
+	if (err < 0)
+		return err;
 
 	pcm->private_data = chip;
 
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index b8ac871..b12308b 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
 		snd_printk(KERN_ERR "HDSPM: "
 				"unable to kmalloc Mixer memory of %d Bytes\n",
 				(int)sizeof(struct hdspm_mixer));
-		return err;
+		return -ENOMEM;
 	}
 
 	hdspm->port_names_in = NULL;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 512434e..805ab6e 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card,
 	if (rc)
 		goto error_out_cleanup;
 
-	if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
-			sis)) {
+	rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
+			 sis);
+	if (rc) {
 		dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq);
 		goto error_out_cleanup;
 	}
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index f5ceb6f..210cafe 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr)
 	return 0;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int snd_pmac_driver_suspend(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 1aa52ef..9b18b52 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
 				   GFP_KERNEL);
 	if (!the_card.null_buffer_start_vaddr) {
 		pr_info("%s: nullbuffer alloc failed\n", __func__);
+		ret = -ENOMEM;
 		goto clean_preallocate;
 	}
 	pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__,
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index 318c5ba5..dfb7443 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create);
 
 void sport_delete(struct sport_device *sport)
 {
+	if (sport->tx_desc)
+		dma_free_coherent(NULL, sport->tx_desc_size,
+				sport->tx_desc, 0);
+	if (sport->rx_desc)
+		dma_free_coherent(NULL, sport->rx_desc_size,
+				sport->rx_desc, 0);
 	sport_free_resource(sport);
+	kfree(sport);
 }
 EXPORT_SYMBOL(sport_delete);
 
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 3c79592..23b4018 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 
 	/* Setup AB8500 according to board-settings */
 	pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent);
+
+	/* Inform SoC Core that we have our own I/O arrangements. */
+	codec->control_data = (void *)true;
+
 	status = ab8500_audio_setup_mics(codec, &pdata->codec->amics);
 	if (status < 0) {
 		pr_err("%s: Failed to setup mics (%d)!\n", __func__, status);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 8c39ddd..11b1b71 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec)
 
 	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
 
+	codec->control_data = codec;	/* we don't use regmap! */
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0) {
 		printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 6276e35..8f726c0 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec)
 {
 	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
 
+	codec->control_data = priv->mc13xxx;
+
 	mc13xxx_lock(priv->mc13xxx);
 
 	/* these are the reset values */
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 8af6a52..df2f99d 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = {
 	{"Headphone Mux", "DAC", "DAC"},	/* dac --> hp_mux */
 	{"LO", NULL, "DAC"},			/* dac --> line_out */
 
+	{"LINE_IN", NULL, "VAG_POWER"},
 	{"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */
 	{"HP", NULL, "Headphone Mux"},		/* hp_mux --> hp */
 
@@ -1357,8 +1358,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 	if (ret)
 		goto err;
 
-	snd_soc_dapm_new_widgets(&codec->dapm);
-
 	return 0;
 
 err:
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 982e437..33c0f3d 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec)
 
 	printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
 
+	codec->control_data = codec;	/* we don't use regmap! */
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0)
 		goto codec_err;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 6537f16..e33d327 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
 
 ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
 
 SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
 		   ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
-SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
-		   ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
 
 ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
@@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
 
 ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
 
 ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
@@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
 		 NULL, 0),
 SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
 		 NULL, 0),
-SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
-		 NULL, 0),
-SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
-		 NULL, 0),
 
 SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
 		 NULL, 0),
@@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
 
 ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
 ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
-ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
-ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
 
 ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
 ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
@@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
 	{ name, "EQ4", "EQ4" }, \
 	{ name, "DRC1L", "DRC1L" }, \
 	{ name, "DRC1R", "DRC1R" }, \
-	{ name, "DRC2L", "DRC2L" }, \
-	{ name, "DRC2R", "DRC2R" }, \
 	{ name, "LHPF1", "LHPF1" }, \
 	{ name, "LHPF2", "LHPF2" }, \
 	{ name, "LHPF3", "LHPF3" }, \
@@ -639,6 +625,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	{ "AIF2 Capture", NULL, "SYSCLK" },
 	{ "AIF3 Capture", NULL, "SYSCLK" },
 
+	{ "IN1L PGA", NULL, "IN1L" },
+	{ "IN1R PGA", NULL, "IN1R" },
+
+	{ "IN2L PGA", NULL, "IN2L" },
+	{ "IN2R PGA", NULL, "IN2R" },
+
+	{ "IN3L PGA", NULL, "IN3L" },
+	{ "IN3R PGA", NULL, "IN3R" },
+
 	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
 	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
 	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
@@ -675,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 
 	ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
 	ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
-	ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
-	ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
 
 	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
 	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 8033f70..01ebbcc 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "AIF2 Capture", NULL, "SYSCLK" },
 	{ "AIF3 Capture", NULL, "SYSCLK" },
 
+	{ "IN1L PGA", NULL, "IN1L" },
+	{ "IN1R PGA", NULL, "IN1R" },
+
+	{ "IN2L PGA", NULL, "IN2L" },
+	{ "IN2R PGA", NULL, "IN2R" },
+
+	{ "IN3L PGA", NULL, "IN3L" },
+	{ "IN3R PGA", NULL, "IN3R" },
+
+	{ "IN4L PGA", NULL, "IN4L" },
+	{ "IN4R PGA", NULL, "IN4R" },
+
 	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
 	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
 	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index eaf6586..ce67200 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 		/* VMID 2*250k */
 		snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
 				    WM8962_VMID_SEL_MASK, 0x100);
+
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			msleep(100);
 		break;
 
 	case SND_SOC_BIAS_OFF:
@@ -3730,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev)
 
 	regcache_sync(wm8962->regmap);
 
-	regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP,
-			   WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA,
-			   WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA);
-
-	/* Bias enable at 2*50k for ramp */
-	regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
-			   WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA,
-			   WM8962_BIAS_ENA | 0x180);
-
-	msleep(5);
-
-	/* VMID back to 2x250k for standby */
-	regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
-			   WM8962_VMID_SEL_MASK, 0x100);
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index bb62f4b..6c9eeca 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	bclk_rate = params_rate(params) * 2;
+	bclk_rate = params_rate(params) * 4;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bclk_rate *= 16;
@@ -3253,10 +3253,13 @@ static void wm8994_mic_work(struct work_struct *work)
 	int ret;
 	int report;
 
+	pm_runtime_get_sync(dev);
+
 	ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, &reg);
 	if (ret < 0) {
 		dev_err(dev, "Failed to read microphone status: %d\n",
 			ret);
+		pm_runtime_put(dev);
 		return;
 	}
 
@@ -3299,6 +3302,8 @@ static void wm8994_mic_work(struct work_struct *work)
 
 	snd_soc_jack_report(priv->micdet[1].jack, report,
 			    SND_JACK_HEADSET | SND_JACK_BTN_0);
+
+	pm_runtime_put(dev);
 }
 
 static irqreturn_t wm8994_mic_irq(int irq, void *data)
@@ -3421,12 +3426,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 	int reg;
 	bool present;
 
+	pm_runtime_get_sync(codec->dev);
+
 	mutex_lock(&wm8994->accdet_lock);
 
 	reg = snd_soc_read(codec, WM1811_JACKDET_CTRL);
 	if (reg < 0) {
 		dev_err(codec->dev, "Failed to read jack status: %d\n", reg);
 		mutex_unlock(&wm8994->accdet_lock);
+		pm_runtime_put(codec->dev);
 		return IRQ_NONE;
 	}
 
@@ -3491,6 +3499,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 				    SND_JACK_MECHANICAL | SND_JACK_HEADSET |
 				    wm8994->btn_mask);
 
+	pm_runtime_put(codec->dev);
 	return IRQ_HANDLED;
 }
 
@@ -3602,6 +3611,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 	if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA))
 		return IRQ_HANDLED;
 
+	pm_runtime_get_sync(codec->dev);
+
 	/* We may occasionally read a detection without an impedence
 	 * range being provided - if that happens loop again.
 	 */
@@ -3612,6 +3623,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 			dev_err(codec->dev,
 				"Failed to read mic detect status: %d\n",
 				reg);
+			pm_runtime_put(codec->dev);
 			return IRQ_NONE;
 		}
 
@@ -3639,6 +3651,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 		dev_warn(codec->dev, "Accessory detection with no callback\n");
 
 out:
+	pm_runtime_put(codec->dev);
 	return IRQ_HANDLED;
 }
 
@@ -4025,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		break;
 	case WM8958:
 		if (wm8994->revision < 1) {
+			snd_soc_dapm_add_routes(dapm, wm8994_intercon,
+						ARRAY_SIZE(wm8994_intercon));
 			snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
 						ARRAY_SIZE(wm8994_revd_intercon));
 			snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 099e6ec..c6d2076 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
 
 SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
 SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
-SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
 SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
 
 SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv),
@@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]);
 
 /* Mic select */
 static const struct snd_kcontrol_new wm9712_mic_src_controls =
-SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
 
 /* diff select */
 static const struct snd_kcontrol_new wm9712_diff_sel_controls =
@@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
 	&wm9712_capture_selectl_controls),
 SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
 	&wm9712_capture_selectr_controls),
-SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0,
+	&wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
 	&wm9712_mic_src_controls),
 SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
 	&wm9712_diff_sel_controls),
@@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
 SND_SOC_DAPM_OUTPUT("MONOOUT"),
 SND_SOC_DAPM_OUTPUT("HPOUTL"),
@@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = {
 	{"Mic PGA", NULL, "MIC1"},
 	{"Mic PGA", NULL, "MIC2"},
 
+	/* microphones */
+	{"Differential Mic", NULL, "MIC1"},
+	{"Differential Mic", NULL, "MIC2"},
+	{"Left Mic Select Source", "Mic 1", "MIC1"},
+	{"Left Mic Select Source", "Mic 2", "MIC2"},
+	{"Left Mic Select Source", "Stereo", "MIC1"},
+	{"Left Mic Select Source", "Differential", "Differential Mic"},
+	{"Right Mic Select Source", "Mic 1", "MIC1"},
+	{"Right Mic Select Source", "Mic 2", "MIC2"},
+	{"Right Mic Select Source", "Stereo", "MIC2"},
+	{"Right Mic Select Source", "Differential", "Differential Mic"},
+
 	/* left capture selector */
 	{"Left Capture Select", "Mic", "MIC1"},
 	{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
@@ -619,6 +634,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec)
 {
 	int ret = 0;
 
+	codec->control_data = codec;	/* we don't use regmap! */
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0) {
 		printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 3eb19fb..d0b8a32 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec)
 	if (wm9713 == NULL)
 		return -ENOMEM;
 	snd_soc_codec_set_drvdata(codec, wm9713);
+	codec->control_data = wm9713;	/* we don't use regmap! */
 
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 95441bf..ce5e5cd 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
 static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
 {
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (dev->txnumevt)	/* enable FIFO */
+		if (dev->txnumevt) {	/* enable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+								FIFO_ENABLE);
 			mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
 								FIFO_ENABLE);
+		}
 		mcasp_start_tx(dev);
 	} else {
-		if (dev->rxnumevt)	/* enable FIFO */
+		if (dev->rxnumevt) {	/* enable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+								FIFO_ENABLE);
 			mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
 								FIFO_ENABLE);
+		}
 		mcasp_start_rx(dev);
 	}
 }
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 28dd76c..81d7728 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
 static struct snd_soc_dai_driver imx_ssi_dai = {
 	.probe = imx_ssi_dai_probe,
 	.playback = {
-		.channels_min = 1,
+		/* The SSI does not support monaural audio. */
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
index 99a997f..b6fa776 100644
--- a/sound/soc/mxs/Kconfig
+++ b/sound/soc/mxs/Kconfig
@@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC
 if SND_MXS_SOC
 
 config SND_SOC_MXS_SGTL5000
-	tristate "SoC Audio support for i.MX boards with sgtl5000"
+	tristate "SoC Audio support for MXS boards with sgtl5000"
 	depends on I2C
 	select SND_SOC_SGTL5000
 	help
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index aba71bf..b303071 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *cpu_dai)
 {
 	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+	struct mxs_saif *master_saif;
 	u32 scr, stat;
 	int ret;
 
+	master_saif = mxs_saif_get_master(saif);
+	if (!master_saif)
+		return -EINVAL;
+
 	/* mclk should already be set */
 	if (!saif->mclk && saif->mclk_in_use) {
 		dev_err(cpu_dai->dev, "set mclk first\n");
@@ -420,6 +425,25 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
+	/* prepare clk in hw_param, enable in trigger */
+	clk_prepare(saif->clk);
+	if (saif != master_saif) {
+		/*
+		* Set an initial clock rate for the saif internal logic to work
+		* properly. This is important when working in EXTMASTER mode
+		* that uses the other saif's BITCLK&LRCLK but it still needs a
+		* basic clock which should be fast enough for the internal
+		* logic.
+		*/
+		clk_enable(saif->clk);
+		ret = clk_set_rate(saif->clk, 24000000);
+		clk_disable(saif->clk);
+		if (ret)
+			return ret;
+
+		clk_prepare(master_saif->clk);
+	}
+
 	scr = __raw_readl(saif->base + SAIF_CTRL);
 
 	scr &= ~BM_SAIF_CTRL_WORD_LENGTH;
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 34835e8..d33c48b 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux)
 {
 	const char *signal, *src;
 
-	if (mcbsp->pdata->mux_signal)
+	if (!mcbsp->pdata->mux_signal)
 		return -EINVAL;
 
 	switch (mux) {
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 1046083..acdd3ef 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver);
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-mcbsp");
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 5a649da..f0feb06 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver);
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-pcm-audio");
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index b7b2a1f..89b0646 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -20,7 +20,7 @@
 #include <sound/pcm_params.h>
 
 #include <plat/audio.h>
-#include <plat/dma.h>
+#include <mach/dma.h>
 
 #include "dma.h"
 #include "pcm.h"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f219b2f..c501af6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 	}
 
 	if (!rtd->cpu_dai) {
-		dev_dbg(card->dev, "CPU DAI %s not registered\n",
+		dev_err(card->dev, "CPU DAI %s not registered\n",
 			dai_link->cpu_dai_name);
 		return -EPROBE_DEFER;
 	}
@@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 		}
 
 		if (!rtd->codec_dai) {
-			dev_dbg(card->dev, "CODEC DAI %s not registered\n",
+			dev_err(card->dev, "CODEC DAI %s not registered\n",
 				dai_link->codec_dai_name);
 			return -EPROBE_DEFER;
 		}
 	}
 
 	if (!rtd->codec) {
-		dev_dbg(card->dev, "CODEC %s not registered\n",
+		dev_err(card->dev, "CODEC %s not registered\n",
 			dai_link->codec_name);
 		return -EPROBE_DEFER;
 	}
@@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 		rtd->platform = platform;
 	}
 	if (!rtd->platform) {
-		dev_dbg(card->dev, "platform %s not registered\n",
+		dev_err(card->dev, "platform %s not registered\n",
 			dai_link->platform_name);
 		return -EPROBE_DEFER;
 	}
@@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card,
 	}
 
 	/* If the driver didn't set I/O up try regmap */
-	if (!codec->control_data)
+	if (!codec->write && dev_get_regmap(codec->dev, NULL))
 		snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
 
 	if (driver->controls)
@@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num)
 			return 0;
 	}
 
+	dev_err(card->dev, "%s not registered\n", aux_dev->codec_name);
+
 	return -EPROBE_DEFER;
 }
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7f8b3b7..0c17293 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	}
 
 	/* Report before the DAPM sync to help users updating micbias status */
-	blocking_notifier_call_chain(&jack->notifier, status, jack);
+	blocking_notifier_call_chain(&jack->notifier, jack->status, jack);
 
 	snd_soc_dapm_sync(dapm);
 
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index d684df2..e463529 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -177,7 +177,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 	}
 
 	alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
-	if (alc5632->gpio_hp_det == -ENODEV)
+	if (alc5632->gpio_hp_det == -EPROBE_DEFER)
 		return -EPROBE_DEFER;
 
 	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0c5bb33..d4f14e4 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 	} else if (np) {
 		pdata->gpio_spkr_en = of_get_named_gpio(np,
 						"nvidia,spkr-en-gpios", 0);
-		if (pdata->gpio_spkr_en == -ENODEV)
+		if (pdata->gpio_spkr_en == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_hp_mute = of_get_named_gpio(np,
 						"nvidia,hp-mute-gpios", 0);
-		if (pdata->gpio_hp_mute == -ENODEV)
+		if (pdata->gpio_hp_mute == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_hp_det = of_get_named_gpio(np,
 						"nvidia,hp-det-gpios", 0);
-		if (pdata->gpio_hp_det == -ENODEV)
+		if (pdata->gpio_hp_det == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_int_mic_en = of_get_named_gpio(np,
 						"nvidia,int-mic-en-gpios", 0);
-		if (pdata->gpio_int_mic_en == -ENODEV)
+		if (pdata->gpio_int_mic_en == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
 						"nvidia,ext-mic-en-gpios", 0);
-		if (pdata->gpio_ext_mic_en == -ENODEV)
+		if (pdata->gpio_ext_mic_en == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 	}
 
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index 62ac028..057e28e 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -21,7 +21,7 @@
 #include <linux/mfd/dbx500-prcmu.h>
 
 #include <mach/hardware.h>
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
 
 #include <sound/soc.h>
 #include <sound/soc-dai.h>
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index ee14d2d..5c472f3 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -19,7 +19,7 @@
 #include <linux/slab.h>
 
 #include <mach/hardware.h>
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
 
 #include <sound/soc.h>
 
diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h
index 7f71b4a..2d9136d 100644
--- a/sound/soc/ux500/ux500_msp_i2s.h
+++ b/sound/soc/ux500/ux500_msp_i2s.h
@@ -17,7 +17,7 @@
 
 #include <linux/platform_device.h>
 
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
 
 #define MSP_INPUT_FREQ_APB 48000000
 
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 0f647d2..c411812 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -821,10 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
 	if (++ep->use_count != 1)
 		return 0;
 
-	/* just to be sure */
-	deactivate_urbs(ep, 0, 1);
-	wait_clear_urbs(ep);
-
 	ep->active_mask = 0;
 	ep->unlink_mask = 0;
 	ep->phase = 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a1298f3..62ec808 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -544,6 +544,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 	subs->last_frame_number = 0;
 	runtime->delay = 0;
 
+	/* clear the pending deactivation on the target EPs */
+	deactivate_endpoints(subs);
+
 	/* for playback, submit the URBs now; otherwise, the first hwptr_done
 	 * updates for all URBs would happen at the same time when starting */
 	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)