Merge tag 'tegra-for-3.7-drivers-i2c' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra into next/drivers

From Stephen Warren:

ARM: tegra: i2c driver enhancements mostly related to clocking

This branch contains a number of fixes and cleanups to the Tegra I2C
driver related to clocks. These are based on the common clock conversion
in order to avoid duplicating the clock driver changes before and after
the conversion. Finally, a bug-fix related to I2C_M_NOSTART is included.

This branch is based on previous pull request tegra-for-3.7-common-clk.

* tag 'tegra-for-3.7-drivers-i2c' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra:
  i2c: tegra: dynamically control fast clk
  i2c: tegra: I2_M_NOSTART functionality not supported in Tegra20
  ARM: tegra: clock: remove unused clock entry for i2c
  ARM: tegra: clock: add connection name in i2c clock entry
  i2c: tegra: pass proper name for getting clock
  ARM: tegra: clock: add i2c fast clock entry in clock table
  ARM: Tegra: Add smp_twd clock for Tegra20
  ARM: tegra: cpu-tegra: explicitly manage re-parenting
  ARM: tegra: fix overflow in tegra20_pll_clk_round_rate()
  ARM: tegra: Fix data type for io address
  ARM: tegra: remove tegra_timer from tegra_list_clks
  ARM: tegra30: clocks: fix the wrong tegra_audio_sync_clk_ops name
  ARM: tegra: clocks: separate tegra_clk_32k_ops from Tegra20 and Tegra30
  ARM: tegra: Remove duplicate code
  ARM: tegra: Port tegra to generic clock framework
  ARM: tegra: Add clk_tegra structure and helper functions
  ARM: tegra: Rename tegra20 clock file
  ARM: tegra20: Separate out clk ops and clk data
  ARM: tegra30: Separate out clk ops and clk data
  ARM: tegra: fix U16 divider range check
  ...
  + sync to v3.6-rc4

Resolved remove/modify conflict in arch/arm/mach-sa1100/leds-hackkit.c
caused by the sync with v3.6-rc4.

Signed-off-by: Olof Johansson <olof@lixom.net>

58 files changed, 335 insertions, 260 deletions

diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 0d7b25e81643..4e1fda75c1c9 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 eb4ceb71123e..277ebce23a45 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 bf47025bdf45..9052aff37f64 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 4e7ec2b49873..d0f00356fc11 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 1128b35b2b05..5a34355e78e8 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 f7d3bfc6bca8..54bb6644a598 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 6ca59fc6dcb9..ef171295f6d4 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 2d67c78c9f4b..f7cdaf51512d 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 733b014ec7d1..b2b3c014221a 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 f75f5ffdfdfb..a71d1c14a0f6 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 8e40262d4117..2f6e9c762d3f 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 4f502a2bdc3c..0a436626182b 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 647218d69f68..4f7d2dfcef7b 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 0bc2315b181d..0849aac449f2 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 88a9c20eb7a2..f560051a949e 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 c422d330ca54..7fbc1bcaf1a9 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 c8aced182fd1..60882c62f180 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 7e46258fc700..6894ec66258c 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 d0d3540e39e7..49750a96d649 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 14361184ae1e..5e22a8f43d2e 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 69b928449789..8f23374fa642 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 344b221d2102..4f81dd44c837 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 94040ccf8e8f..ea5775a1a7db 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 80d90cb42853..430771776915 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 d1ab43706735..5579b08bb35b 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 b8ac8710f47f..b12308b5ba2a 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 512434efcc31..805ab6e9a78f 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 f5ceb6f282de..210cafe04890 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 1aa52eff526a..9b18b5243a56 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 318c5ba5360f..dfb744381c42 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 3c795921c5f6..23b40186f9b8 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 8c39dddd7d00..11b1b714b8b5 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 6276e352125f..8f726c063f42 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 8af6a5245b18..df2f99d1d428 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 982e437799a8..33c0f3d39c87 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 6537f16d383e..e33d327396ad 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 8033f7065189..01ebbcc5c6a4 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 eaf65863ec21..ce6720073798 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 bb62f4b3d563..6c9eeca85b95 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 099e6ec32125..c6d2076a796b 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 3eb19fb71d17..d0b8a3287a85 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 95441bfc8190..ce5e5cd254dd 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 28dd76c7cb1c..81d7728cf67f 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 99a997f19bb9..b6fa77678d97 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 aba71bfa33b1..b3030718c228 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 34835e8a9160..d33c48baaf71 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 1046083e90a0..acdd3ef14e08 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 5a649da9122a..f0feb06615f8 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 b7b2a1f91425..89b064650f14 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 f219b2f7ee68..c501af6d8dbe 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 7f8b3b7428bb..0c172938b82a 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 d684df294c0c..e463529b38bb 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 0c5bb33d258e..d4f14e492341 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 62ac0285bfaf..057e28ef770e 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 ee14d2dac2f5..5c472f335a64 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 7f71b4a0d4bc..2d9136da9865 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 0f647d22cb4a..c41181202688 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 a1298f379428..62ec808ed792 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)