Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This update became slightly bigger than usual for rc3, but most of the
  commits are small and trivial.  A large chunk is found for HD-audio
  ca0132 codec, which is mostly a clean up of the specific code, to make
  SPDIF working properly, and also in the new ASoC Arizona driver.

  One important fix is for usb-audio Oops fix since 3.5.  We still see
  some EHCI related bandwidth problem, but usb-audio should be more
  stabilized now.

  Other than that, a Kconfig fix is spread over files, and various
  HD-audio and ASoC fixes as usual, in addition to Julia's error path
  fixes."

* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (42 commits)
  ALSA: snd-als100: fix suspend/resume
  ALSA: hda - Fix leftover codec->power_transition
  ALSA: hda - don't create dysfunctional mixer controls for ca0132
  ALSA: sound/ppc/snd_ps3.c: fix error return code
  ALSA: sound/pci/rme9652/hdspm.c: fix error return code
  ALSA: sound/pci/sis7019.c: fix error return code
  ALSA: sound/pci/ctxfi/ctatc.c: fix error return code
  ALSA: sound/atmel/ac97c.c: fix error return code
  ALSA: sound/atmel/abdac.c: fix error return code
  ALSA: fix pcm.h kernel-doc warning and notation
  sound: oss/sb_audio: prevent divide by zero bug
  ASoC: wm9712: Fix inverted capture volume
  ASoC: wm9712: Fix microphone source selection
  ASoC: wm5102: Remove DRC2
  ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx
  ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream
  ALSA: lx6464es: Add a missing error check
  ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch
  ASoC: jack: Always notify full jack status
  ASoC: wm5110: Add missing input PGA routes
  ...

38 files changed, 249 insertions, 231 deletions

diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 0d7b25e8164..4e1fda75c1c 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 eb4ceb71123..277ebce23a4 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 bf47025bdf4..9052aff37f6 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/drivers/aloop.c b/sound/drivers/aloop.c
index 1128b35b2b0..5a34355e78e 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 f7d3bfc6bca..54bb6644a59 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 6ca59fc6dcb..ef171295f6d 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 2d67c78c9f4..f7cdaf51512 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 733b014ec7d..b2b3c014221 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 f75f5ffdfdf..a71d1c14a0f 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 8e40262d411..2f6e9c762d3 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/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 0bc2315b181..0849aac449f 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 88a9c20eb7a..f560051a949 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 c422d330ca5..7fbc1bcaf1a 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 c8aced182fd..60882c62f18 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 7e46258fc70..6894ec66258 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 d0d3540e39e..49750a96d64 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_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 94040ccf8e8..ea5775a1a7d 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 80d90cb4285..43077177691 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 d1ab4370673..5579b08bb35 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 b8ac8710f47..b12308b5ba2 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 512434efcc3..805ab6e9a78 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 f5ceb6f282d..210cafe0489 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 1aa52eff526..9b18b5243a5 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 318c5ba5360..dfb744381c4 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/wm5102.c b/sound/soc/codecs/wm5102.c
index 6537f16d383..e33d327396a 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 8033f706518..01ebbcc5c6a 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 aa9ce9dd7d8..ce672007379 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3733,21 +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 04ef03175c5..6c9eeca85b9 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -4038,6 +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 f16fb361a4e..c6d2076a796 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"},
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 95441bfc819..ce5e5cd254d 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 28dd76c7cb1..81d7728cf67 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 99a997f19bb..b6fa77678d9 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/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 34835e8a916..d33c48baaf7 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/samsung/pcm.c b/sound/soc/samsung/pcm.c
index b7b2a1f9142..89b064650f1 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 f81c5976b96..c501af6d8db 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;
 	}
@@ -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 7f8b3b7428b..0c172938b82 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/usb/endpoint.c b/sound/usb/endpoint.c
index 0f647d22cb4..c4118120268 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 a1298f37942..62ec808ed79 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)