Merge 3.8-rc5 into tty-next

This resolves a number of tty driver merge issues found in linux-next

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

26 files changed, 288 insertions, 155 deletions

diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 6fc0ae9..fff7753 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -18,6 +18,7 @@
 #include <linux/delay.h>
 #include <linux/module.h>
 #include <linux/io.h>
+#include <linux/gpio.h>
 
 #include <sound/ac97_codec.h>
 #include <sound/pxa2xx-lib.h>
@@ -148,6 +149,8 @@ static inline void pxa_ac97_warm_pxa27x(void)
 
 static inline void pxa_ac97_cold_pxa27x(void)
 {
+	unsigned int timeout;
+
 	GCR &=  GCR_COLD_RST;  /* clear everything but nCRST */
 	GCR &= ~GCR_COLD_RST;  /* then assert nCRST */
 
@@ -157,8 +160,10 @@ static inline void pxa_ac97_cold_pxa27x(void)
 	clk_enable(ac97conf_clk);
 	udelay(5);
 	clk_disable(ac97conf_clk);
-	GCR = GCR_COLD_RST;
-	udelay(50);
+	GCR = GCR_COLD_RST | GCR_WARM_RST;
+	timeout = 100;     /* wait for the codec-ready bit to be set */
+	while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
+		mdelay(1);
 }
 #endif
 
@@ -340,8 +345,21 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
 	}
 
 	if (cpu_is_pxa27x()) {
-		/* Use GPIO 113 as AC97 Reset on Bulverde */
+		/*
+		 * This gpio is needed for a work-around to a bug in the ac97
+		 * controller during warm reset.  The direction and level is set
+		 * here so that it is an output driven high when switching from
+		 * AC97_nRESET alt function to generic gpio.
+		 */
+		ret = gpio_request_one(reset_gpio, GPIOF_OUT_INIT_HIGH,
+				       "pxa27x ac97 reset");
+		if (ret < 0) {
+			pr_err("%s: gpio_request_one() failed: %d\n",
+			       __func__, ret);
+			goto err_conf;
+		}
 		pxa27x_assert_ac97reset(reset_gpio, 0);
+
 		ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
 		if (IS_ERR(ac97conf_clk)) {
 			ret = PTR_ERR(ac97conf_clk);
@@ -384,6 +402,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe);
 
 void pxa2xx_ac97_hw_remove(struct platform_device *dev)
 {
+	if (cpu_is_pxa27x())
+		gpio_free(reset_gpio);
 	GCR |= GCR_ACLINK_OFF;
 	free_irq(IRQ_AC97, NULL);
 	if (ac97conf_clk) {
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b8fb0a5..822df97 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3654,6 +3654,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 	hda_set_power_state(codec, AC_PWRST_D0);
 	restore_shutup_pins(codec);
 	hda_exec_init_verbs(codec);
+	snd_hda_jack_set_dirty_all(codec);
 	if (codec->patch_ops.resume)
 		codec->patch_ops.resume(codec);
 	else {
@@ -3665,10 +3666,8 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 
 	if (codec->jackpoll_interval)
 		hda_jackpoll_work(&codec->jackpoll_work.work);
-	else {
-		snd_hda_jack_set_dirty_all(codec);
+	else
 		snd_hda_jack_report_sync(codec);
-	}
 
 	codec->in_pm = 0;
 	snd_hda_power_down(codec); /* flag down before returning */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cca8727..0b6aeba 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -573,9 +573,12 @@ enum {
 #define AZX_DCAPS_PM_RUNTIME	(1 << 26)	/* runtime PM support */
 
 /* quirks for Intel PCH */
-#define AZX_DCAPS_INTEL_PCH \
+#define AZX_DCAPS_INTEL_PCH_NOPM \
 	(AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \
-	 AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME)
+	 AZX_DCAPS_COUNT_LPIB_DELAY)
+
+#define AZX_DCAPS_INTEL_PCH \
+	(AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME)
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci)
 static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* CPT */
 	{ PCI_DEVICE(0x8086, 0x1c20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
 	/* PBG */
 	{ PCI_DEVICE(0x8086, 0x1d20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
 	/* Panther Point */
 	{ PCI_DEVICE(0x8086, 0x1e20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
 	/* Lynx Point */
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 60890bf..009b77a 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int conexant_suspend(struct hda_codec *codec)
-{
-	snd_hda_shutup_pins(codec);
-	return 0;
-}
-#endif
-
 static const struct hda_codec_ops conexant_patch_ops = {
 	.build_controls = conexant_build_controls,
 	.build_pcms = conexant_build_pcms,
 	.init = conexant_init,
 	.free = conexant_free,
 	.set_power_state = conexant_set_power,
-#ifdef CONFIG_PM
-	.suspend = conexant_suspend,
-#endif
-	.reboot_notify = snd_hda_shutup_pins,
 };
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
@@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
 	.init = cx_auto_init,
 	.free = conexant_free,
 	.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
-	.suspend = conexant_suspend,
-#endif
-	.reboot_notify = snd_hda_shutup_pins,
 };
 
 /*
@@ -4652,6 +4636,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = {
 	  .patch = patch_conexant_auto },
 	{ .id = 0x14f15111, .name = "CX20753/4",
 	  .patch = patch_conexant_auto },
+	{ .id = 0x14f15113, .name = "CX20755",
+	  .patch = patch_conexant_auto },
+	{ .id = 0x14f15114, .name = "CX20756",
+	  .patch = patch_conexant_auto },
+	{ .id = 0x14f15115, .name = "CX20757",
+	  .patch = patch_conexant_auto },
 	{} /* terminator */
 };
 
@@ -4675,6 +4665,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150b9");
 MODULE_ALIAS("snd-hda-codec-id:14f1510f");
 MODULE_ALIAS("snd-hda-codec-id:14f15110");
 MODULE_ALIAS("snd-hda-codec-id:14f15111");
+MODULE_ALIAS("snd-hda-codec-id:14f15113");
+MODULE_ALIAS("snd-hda-codec-id:14f15114");
+MODULE_ALIAS("snd-hda-codec-id:14f15115");
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Conexant HD-audio codec");
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index b6c21ea..807a2aa 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1502,7 +1502,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol,
 	ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
 	substream = snd_pcm_chmap_substream(info, ctl_idx);
 	if (!substream || !substream->runtime)
-		return -EBADFD;
+		return 0; /* just for avoiding error from alsactl restore */
 	switch (substream->runtime->status->state) {
 	case SNDRV_PCM_STATE_OPEN:
 	case SNDRV_PCM_STATE_SETUP:
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 71ae23d..cf38861 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5817,6 +5817,9 @@ enum {
 	ALC269_TYPE_ALC269VB,
 	ALC269_TYPE_ALC269VC,
 	ALC269_TYPE_ALC269VD,
+	ALC269_TYPE_ALC280,
+	ALC269_TYPE_ALC282,
+	ALC269_TYPE_ALC284,
 };
 
 /*
@@ -5833,10 +5836,13 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	switch (spec->codec_variant) {
 	case ALC269_TYPE_ALC269VA:
 	case ALC269_TYPE_ALC269VC:
+	case ALC269_TYPE_ALC280:
+	case ALC269_TYPE_ALC284:
 		ssids = alc269va_ssids;
 		break;
 	case ALC269_TYPE_ALC269VB:
 	case ALC269_TYPE_ALC269VD:
+	case ALC269_TYPE_ALC282:
 		ssids = alc269_ssids;
 		break;
 	default:
@@ -6245,6 +6251,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x1977, "HP Pavilion 14", ALC269_FIXUP_MIC1_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
@@ -6259,6 +6266,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
@@ -6400,7 +6408,8 @@ static int patch_alc269(struct hda_codec *codec)
 
 	alc_auto_parse_customize_define(codec);
 
-	if (codec->vendor_id == 0x10ec0269) {
+	switch (codec->vendor_id) {
+	case 0x10ec0269:
 		spec->codec_variant = ALC269_TYPE_ALC269VA;
 		switch (alc_get_coef0(codec) & 0x00f0) {
 		case 0x0010:
@@ -6425,6 +6434,20 @@ static int patch_alc269(struct hda_codec *codec)
 			goto error;
 		spec->init_hook = alc269_fill_coef;
 		alc269_fill_coef(codec);
+		break;
+
+	case 0x10ec0280:
+	case 0x10ec0290:
+		spec->codec_variant = ALC269_TYPE_ALC280;
+		break;
+	case 0x10ec0282:
+	case 0x10ec0283:
+		spec->codec_variant = ALC269_TYPE_ALC282;
+		break;
+	case 0x10ec0284:
+	case 0x10ec0292:
+		spec->codec_variant = ALC269_TYPE_ALC284;
+		break;
 	}
 
 	/* automatic parse from the BIOS config */
@@ -7129,6 +7152,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
 	{ .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
 	{ .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
+	{ .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
 	{ .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
 	{ .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6e02e06..223c3d9 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 */
 /* status */
 #define HDSPM_AES32_wcLock	0x0200000
+#define HDSPM_AES32_wcSync	0x0100000
 #define HDSPM_AES32_wcFreq_bit  22
 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function
   HDSPM_bit2freq */
@@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm)
 	switch (hdspm->io_type) {
 	case AES32:
 		status = hdspm_read(hdspm, HDSPM_statusRegister);
-		if (status & HDSPM_wcSync)
-			return 2;
-		else if (status & HDSPM_wcLock)
-			return 1;
+		if (status & HDSPM_AES32_wcLock) {
+			if (status & HDSPM_AES32_wcSync)
+				return 2;
+			else
+				return 1;
+		}
 		return 0;
 		break;
 
@@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
 	unsigned int status;
 	unsigned int status2;
 	unsigned int timecode;
+	unsigned int wcLock, wcSync;
 	int pref_syncref;
 	char *autosync_ref;
 	int x;
@@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
 
 	snd_iprintf(buffer, "--- Status:\n");
 
+	wcLock = status & HDSPM_AES32_wcLock;
+	wcSync = wcLock && (status & HDSPM_AES32_wcSync);
+
 	snd_iprintf(buffer, "Word: %s  Frequency: %d\n",
-		    (status & HDSPM_AES32_wcLock) ? "Sync   " : "No Lock",
+		    (wcLock) ? (wcSync ? "Sync   " : "Lock   ") : "No Lock",
 		    HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF));
 
 	for (x = 0; x < 8; x++) {
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index adf397b..1d8bb59 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_DSP_A:
 		mode = 0;
 		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		mode = 1;
-		break;
 	case SND_SOC_DAIFMT_I2S:
 		mode = 2;
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		mode = 3;
-		break;
 	default:
 		arizona_aif_err(dai, "Unsupported DAI format %d\n",
 				fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
 				    ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
 		snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
-				    ARIZONA_AIF1_RATE_MASK, 8);
+				    ARIZONA_AIF1_RATE_MASK,
+				    8 << ARIZONA_AIF1_RATE_SHIFT);
 		break;
 	default:
 		arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 41dae1e..4deebeb 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -34,15 +34,15 @@
 
 #define ARIZONA_FLL_SRC_MCLK1      0
 #define ARIZONA_FLL_SRC_MCLK2      1
-#define ARIZONA_FLL_SRC_SLIMCLK    2
-#define ARIZONA_FLL_SRC_FLL1       3
-#define ARIZONA_FLL_SRC_FLL2       4
-#define ARIZONA_FLL_SRC_AIF1BCLK   5
-#define ARIZONA_FLL_SRC_AIF2BCLK   6
-#define ARIZONA_FLL_SRC_AIF3BCLK   7
-#define ARIZONA_FLL_SRC_AIF1LRCLK  8
-#define ARIZONA_FLL_SRC_AIF2LRCLK  9
-#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+#define ARIZONA_FLL_SRC_SLIMCLK    3
+#define ARIZONA_FLL_SRC_FLL1       4
+#define ARIZONA_FLL_SRC_FLL2       5
+#define ARIZONA_FLL_SRC_AIF1BCLK   8
+#define ARIZONA_FLL_SRC_AIF2BCLK   9
+#define ARIZONA_FLL_SRC_AIF3BCLK  10
+#define ARIZONA_FLL_SRC_AIF1LRCLK 12
+#define ARIZONA_FLL_SRC_AIF2LRCLK 13
+#define ARIZONA_FLL_SRC_AIF3LRCLK 14
 
 #define ARIZONA_MIXER_VOL_MASK             0x00FE
 #define ARIZONA_MIXER_VOL_SHIFT                 1
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 4f11279..ac8742a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec)
 	struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
 	int ret;
 	int gpio_nreset = -EINVAL;
-	int amutec_eq_bmutec = 0;
+	bool amutec_eq_bmutec = false;
 
 #ifdef CONFIG_OF
 	if (of_match_device(cs4271_dt_ids, codec->dev)) {
 		gpio_nreset = of_get_named_gpio(codec->dev->of_node,
 						"reset-gpio", 0);
 
-		if (!of_get_property(codec->dev->of_node,
+		if (of_get_property(codec->dev->of_node,
 				     "cirrus,amutec-eq-bmutec", NULL))
-			amutec_eq_bmutec = 1;
+			amutec_eq_bmutec = true;
 	}
 #endif
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 99bb1c6..9811a54 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = {
 
 static int cs42l52_get_clk(int mclk, int rate)
 {
-	int i, ret = 0;
+	int i, ret = -EINVAL;
 	u_int mclk1, mclk2 = 0;
 
 	for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
@@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate)
 			}
 		}
 	}
-	if (ret > ARRAY_SIZE(clk_map_table))
-		return -EINVAL;
 	return ret;
 }
 
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index d75257d..e19490c 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = {
 	{ 101, 0x00 },
 	{ 102, 0x00 },
 	{ 103, 0x01 },
-	{ 105, 0x01 },
-	{ 106, 0x00 },
-	{ 107, 0x01 },
+	{ 104, 0x01 },
+	{ 105, 0x00 },
+	{ 106, 0x01 },
 	{ 107, 0x00 },
 	{ 108, 0x00 },
 	{ 109, 0x00 },
@@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = {
 	{ 184, 0x00 },
 	{ 185, 0x00 },
 	{ 186, 0x00 },
-	{ 189, 0x00 },
+	{ 187, 0x00 },
 	{ 188, 0x00 },
-	{ 194, 0x00 },
-	{ 195, 0x00 },
-	{ 196, 0x00 },
-	{ 197, 0x00 },
-	{ 200, 0x00 },
-	{ 201, 0x00 },
-	{ 202, 0x00 },
-	{ 203, 0x00 },
-	{ 204, 0x00 },
-	{ 205, 0x00 },
-	{ 208, 0x00 },
+	{ 189, 0x00 },
+	{ 208, 0x06 },
 	{ 209, 0x00 },
-	{ 210, 0x00 },
-	{ 211, 0x00 },
-	{ 213, 0x00 },
-	{ 214, 0x00 },
-	{ 215, 0x00 },
-	{ 216, 0x00 },
-	{ 217, 0x00 },
-	{ 218, 0x00 },
-	{ 219, 0x00 },
+	{ 210, 0x08 },
+	{ 211, 0x54 },
+	{ 212, 0x14 },
+	{ 213, 0x0d },
+	{ 214, 0x0d },
+	{ 215, 0x14 },
+	{ 216, 0x60 },
 	{ 221, 0x00 },
 	{ 222, 0x00 },
+	{ 223, 0x00 },
 	{ 224, 0x00 },
-	{ 225, 0x00 },
-	{ 226, 0x00 },
-	{ 227, 0x00 },
-	{ 228, 0x00 },
-	{ 229, 0x00 },
-	{ 230, 0x13 },
-	{ 231, 0x00 },
-	{ 232, 0x80 },
-	{ 233, 0x0C },
-	{ 234, 0xDD },
-	{ 235, 0x00 },
-	{ 236, 0x04 },
-	{ 237, 0x00 },
-	{ 238, 0x00 },
-	{ 239, 0x00 },
-	{ 240, 0x00 },
-	{ 241, 0x00 },
-	{ 242, 0x00 },
-	{ 243, 0x00 },
-	{ 244, 0x00 },
-	{ 245, 0x00 },
 	{ 248, 0x00 },
 	{ 249, 0x00 },
-	{ 254, 0x00 },
+	{ 250, 0x00 },
 	{ 255, 0x00 },
 };
 
@@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
 };
 
 /* TLV Declarations */
-static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
-static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0);
+static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0);
 
 static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
 /* Sidetone supports mono only */
 SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 };
 
 static const struct snd_kcontrol_new lm49453_snd_controls[] = {
 	/* mic1 and mic2 supports mono only */
-	SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
-			0, digital_tlv),
-	SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
-			0, digital_tlv),
+	SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv),
+	SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv),
+
+	SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63,
+			0, adc_dac_tlv),
+	SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63,
+			0, adc_dac_tlv),
 
 	SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
-			  LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
-			  LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
 	SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
 	SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
@@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = {
 					  2, 1, 0),
 
 	SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
-			  LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
-			  LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-			  LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
-			  LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
 	SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-			0, 6, 0, digital_tlv),
+			0, 63, 0, adc_dac_tlv),
 
 	SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
 			0, 3, 0, port_tlv),
@@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	}
 
 	snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
-			    LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+			    LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5),
 			    (aif_val | mode | clk_phase));
 
 	snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index cb1675c..92bbfec 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 			5, 1, 0),
 
 	SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
-			0, 4, 0, mic_gain_tlv),
+			0, 3, 0, mic_gain_tlv),
 };
 
 /* mute the codec used by alsa core */
@@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 			SGTL5000_HP_ZCD_EN |
 			SGTL5000_ADC_ZCD_EN);
 
-	snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+	snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
 
 	/*
 	 * disable DAP
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index ab355c4..40c07be 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -74,9 +74,10 @@
 				SNDRV_PCM_FMTBIT_S32_LE)
 #define	S2PC_VALUE		0x98
 #define CLOCK_OUT		0x60
-#define LEFT_J_DATA_FORMAT	0x10
-#define I2S_DATA_FORMAT		0x12
-#define RIGHT_J_DATA_FORMAT	0x14
+#define DATA_FORMAT_MSK		0x0E
+#define LEFT_J_DATA_FORMAT	0x00
+#define I2S_DATA_FORMAT		0x02
+#define RIGHT_J_DATA_FORMAT	0x04
 #define CODEC_MUTE_VAL		0x80
 
 #define POWER_CNTLMSAK		0x40
@@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+	snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 1cbe88f..12bcae6 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 
 	ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
 	if (wm2000->speech_clarity)
-		ret &= ~WM2000_SPEECH_CLARITY;
-	else
 		ret |= WM2000_SPEECH_CLARITY;
+	else
+		ret &= ~WM2000_SPEECH_CLARITY;
 	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
 
 	wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index afcf31d..e6cefe1 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1566,15 +1566,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_DSP_A:
 		fmt_val = 0;
 		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		fmt_val = 1;
-		break;
 	case SND_SOC_DAIFMT_I2S:
 		fmt_val = 2;
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		fmt_val = 3;
-		break;
 	default:
 		dev_err(codec->dev, "Unsupported DAI format %d\n",
 			fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -1626,7 +1620,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 			    WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
 			    lrclk);
 	snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
-			    WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+			    WM2200_AIF1_FMT_MASK, fmt_val);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 5a5f369..54397a5 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_DSP_A:
 		mask = 0;
 		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		mask = 1;
-		break;
 	case SND_SOC_DAIFMT_I2S:
 		mask = 2;
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		mask = 3;
-		break;
 	default:
 		dev_err(codec->dev, "Unsupported DAI format %d\n",
 			fmt & SND_SOC_DAIFMT_FORMAT_MASK);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 688ade0..7a9048d 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,6 +36,9 @@
 struct wm5102_priv {
 	struct arizona_priv core;
 	struct arizona_fll fll[2];
+
+	unsigned int spk_ena:2;
+	unsigned int spk_ena_pending:1;
 };
 
 static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
 };
 
+static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol,
+			 int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+	struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+	if (arizona->rev < 1)
+		return 0;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (!wm5102->spk_ena) {
+			snd_soc_write(codec, 0x4f5, 0x25a);
+			wm5102->spk_ena_pending = true;
+		}
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		if (wm5102->spk_ena_pending) {
+			msleep(75);
+			snd_soc_write(codec, 0x4f5, 0xda);
+			wm5102->spk_ena_pending = false;
+			wm5102->spk_ena++;
+		}
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		wm5102->spk_ena--;
+		if (!wm5102->spk_ena)
+			snd_soc_write(codec, 0x4f5, 0x25a);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		if (!wm5102->spk_ena)
+			snd_soc_write(codec, 0x4f5, 0x0da);
+		break;
+	}
+
+	return 0;
+}
+
+
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
 		   ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
 		   ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ffc89fa..7b198c3 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 	const struct wm_adsp_region *mem;
 	const char *region_name;
 	char *file, *text;
+	void *buf;
 	unsigned int reg;
 	int regions = 0;
 	int ret, offset, type, sizes;
@@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 		}
 
 		if (reg) {
-			ret = regmap_raw_write(regmap, reg, region->data,
+			buf = kmemdup(region->data, le32_to_cpu(region->len),
+				      GFP_KERNEL);
+			if (!buf) {
+				adsp_err(dsp, "Out of memory\n");
+				return -ENOMEM;
+			}
+
+			ret = regmap_raw_write(regmap, reg, buf,
 					       le32_to_cpu(region->len));
+
+			kfree(buf);
+
 			if (ret != 0) {
 				adsp_err(dsp,
 					"%s.%d: Failed to write %d bytes at %d in %s: %d\n",
@@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 	const char *region_name;
 	int ret, pos, blocks, type, offset, reg;
 	char *file;
+	void *buf;
 
 	file = kzalloc(PAGE_SIZE, GFP_KERNEL);
 	if (file == NULL)
@@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 		}
 
 		if (reg) {
+			buf = kmemdup(blk->data, le32_to_cpu(blk->len),
+				      GFP_KERNEL);
+			if (!buf) {
+				adsp_err(dsp, "Out of memory\n");
+				return -ENOMEM;
+			}
+
 			ret = regmap_raw_write(regmap, reg, blk->data,
 					       le32_to_cpu(blk->len));
 			if (ret != 0) {
@@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 					"%s.%d: Failed to write to %x in %s\n",
 					file, blocks, reg, region_name);
 			}
+
+			kfree(buf);
 		}
 
 		pos += le32_to_cpu(blk->len) + sizeof(*blk);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 91d592f..2370063 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
 	ret = device_add(rtd->dev);
 	if (ret < 0) {
+		/* calling put_device() here to free the rtd->dev */
+		put_device(rtd->dev);
 		dev_err(card->dev,
 			"ASoC: failed to register runtime device: %d\n", ret);
 		return ret;
@@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
 		device_remove_file(rtd->dev, &dev_attr_codec_reg);
-		device_del(rtd->dev);
+		device_unregister(rtd->dev);
 		rtd->dev_registered = 0;
 	}
 
@@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
 	platform_max = mc->platform_max;
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 1;
+	uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
 	uinfo->value.integer.min = 0;
 	uinfo->value.integer.max = platform_max - min;
 
@@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	unsigned int reg = mc->reg;
+	unsigned int rreg = mc->rreg;
 	unsigned int shift = mc->shift;
 	int min = mc->min;
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
 	unsigned int val, val_mask;
+	int ret;
 
 	val = ((ucontrol->value.integer.value[0] + min) & mask);
 	if (invert)
@@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
 	val_mask = mask << shift;
 	val = val << shift;
 
-	return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+	ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
+	if (ret != 0)
+		return ret;
+
+	if (snd_soc_volsw_is_stereo(mc)) {
+		val = ((ucontrol->value.integer.value[1] + min) & mask);
+		if (invert)
+			val = max - val;
+		val_mask = mask << shift;
+		val = val << shift;
+
+		ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val);
+	}
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
 
@@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	unsigned int reg = mc->reg;
+	unsigned int rreg = mc->rreg;
 	unsigned int shift = mc->shift;
 	int min = mc->min;
 	int max = mc->max;
@@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
 	ucontrol->value.integer.value[0] =
 		ucontrol->value.integer.value[0] - min;
 
+	if (snd_soc_volsw_is_stereo(mc)) {
+		ucontrol->value.integer.value[1] =
+			(snd_soc_read(codec, rreg) >> shift) & mask;
+		if (invert)
+			ucontrol->value.integer.value[1] =
+				max - ucontrol->value.integer.value[1];
+		ucontrol->value.integer.value[1] =
+			ucontrol->value.integer.value[1] - min;
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d7711fc..cf191e6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
 		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
 			continue;
 
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index e71fe55..0e2ed3d 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -179,6 +179,15 @@ static struct usbmix_name_map audigy2nx_map[] = {
 	{ 0 } /* terminator */
 };
 
+static struct usbmix_selector_map c400_selectors[] = {
+	{
+		.id = 0x80,
+		.count = 2,
+		.names = (const char*[]) {"Internal", "SPDIF"}
+	},
+	{ 0 } /* terminator */
+};
+
 static struct usbmix_selector_map audigy2nx_selectors[] = {
 	{
 		.id = 14, /* Capture Source */
@@ -367,6 +376,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = hercules_usb51_map,
 	},
 	{
+		.id = USB_ID(0x0763, 0x2030),
+		.selector_map = c400_selectors,
+	},
+	{
 		.id = USB_ID(0x08bb, 0x2702),
 		.map = linex_map,
 		.ignore_ctl_error = 1,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 0422b13..15520de 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer)
  * are valid they presents mono controls as L and R channels of
  * stereo. So we provide a good mixer here.
  */
-struct std_mono_table ebox44_table[] = {
+static struct std_mono_table ebox44_table[] = {
 	{
 		.unitid = 4,
 		.control = 1,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index c659310..d82e378 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -511,6 +511,16 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs)
 	struct snd_usb_substream *sync_subs =
 		&subs->stream->substream[subs->direction ^ 1];
 
+	if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA ||
+	    !subs->stream)
+		return snd_usb_endpoint_set_params(subs->sync_endpoint,
+						   subs->pcm_format,
+						   subs->channels,
+						   subs->period_bytes,
+						   subs->cur_rate,
+						   subs->cur_audiofmt,
+						   NULL);
+
 	/* Try to find the best matching audioformat. */
 	list_for_each_entry(fp, &sync_subs->fmt_list, list) {
 		int score = match_endpoint_audioformats(fp, subs->cur_audiofmt,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 78e845e..64d25a7 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2289,7 +2289,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.rate_table = (unsigned int[]) {
 							44100, 48000, 88200, 96000
 					},
-					.clock = 0x81,
+					.clock = 0x80,
 				}
 			},
 			/* Capture */
@@ -2315,7 +2315,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.rate_table = (unsigned int[]) {
 						44100, 48000, 88200, 96000
 					},
-					.clock = 0x81,
+					.clock = 0x80,
 				}
 			},
 			/* MIDI */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index acc12f0..2c97185 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -387,11 +387,13 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev)
 		 * rules
 		 */
 		err = usb_driver_set_configuration(dev, 2);
-		if (err < 0) {
+		if (err < 0)
 			snd_printdd("error usb_driver_set_configuration: %d\n",
 				    err);
-			return -ENODEV;
-		}
+		/* Always return an error, so that we stop creating a device
+		   that will just be destroyed and recreated with a new
+		   configuration */
+		return -ENODEV;
 	} else
 		snd_printk(KERN_INFO "usb-audio: Fast Track Pro config OK\n");
 
@@ -859,6 +861,17 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep)
 	if ((le16_to_cpu(ep->chip->dev->descriptor.idVendor) == 0x23ba) &&
 	    ep->type == SND_USB_ENDPOINT_TYPE_SYNC)
 		ep->skip_packets = 4;
+
+	/*
+	 * M-Audio Fast Track C400 - when packets are not skipped, real world
+	 * latency varies by approx. +/- 50 frames (at 96KHz) each time the
+	 * stream is (re)started. When skipping packets 16 at endpoint start
+	 * up, the real world latency is stable within +/- 1 frame (also
+	 * across power cycles).
+	 */
+	if (ep->chip->usb_id == USB_ID(0x0763, 0x2030) &&
+	    ep->type == SND_USB_ENDPOINT_TYPE_DATA)
+		ep->skip_packets = 16;
 }
 
 void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,