Merge commit 'v3.0.8' into android-3.0

29 files changed, 185 insertions, 160 deletions

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index f134130..3388442 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1758,6 +1758,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 	snd_pcm_uframes_t avail = 0;
 	long wait_time, tout;
 
+	init_waitqueue_entry(&wait, current);
+	set_current_state(TASK_INTERRUPTIBLE);
+	add_wait_queue(&runtime->tsleep, &wait);
+
 	if (runtime->no_period_wakeup)
 		wait_time = MAX_SCHEDULE_TIMEOUT;
 	else {
@@ -1768,16 +1772,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 		}
 		wait_time = msecs_to_jiffies(wait_time * 1000);
 	}
-	init_waitqueue_entry(&wait, current);
-	add_wait_queue(&runtime->tsleep, &wait);
+
 	for (;;) {
 		if (signal_pending(current)) {
 			err = -ERESTARTSYS;
 			break;
 		}
+
+		/*
+		 * We need to check if space became available already
+		 * (and thus the wakeup happened already) first to close
+		 * the race of space already having become available.
+		 * This check must happen after been added to the waitqueue
+		 * and having current state be INTERRUPTIBLE.
+		 */
+		if (is_playback)
+			avail = snd_pcm_playback_avail(runtime);
+		else
+			avail = snd_pcm_capture_avail(runtime);
+		if (avail >= runtime->twake)
+			break;
 		snd_pcm_stream_unlock_irq(substream);
-		tout = schedule_timeout_interruptible(wait_time);
+
+		tout = schedule_timeout(wait_time);
+
 		snd_pcm_stream_lock_irq(substream);
+		set_current_state(TASK_INTERRUPTIBLE);
 		switch (runtime->status->state) {
 		case SNDRV_PCM_STATE_SUSPENDED:
 			err = -ESTRPIPE;
@@ -1803,14 +1823,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 			err = -EIO;
 			break;
 		}
-		if (is_playback)
-			avail = snd_pcm_playback_avail(runtime);
-		else
-			avail = snd_pcm_capture_avail(runtime);
-		if (avail >= runtime->twake)
-			break;
 	}
  _endloop:
+	set_current_state(TASK_RUNNING);
 	remove_wait_queue(&runtime->tsleep, &wait);
 	*availp = avail;
 	return err;
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7c1cbf0..950eed0 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -531,6 +531,8 @@ int snd_timer_stop(struct snd_timer_instance *timeri)
 	if (err < 0)
 		return err;
 	timer = timeri->timer;
+	if (!timer)
+		return -EINVAL;
 	spin_lock_irqsave(&timer->lock, flags);
 	timeri->cticks = timeri->ticks;
 	timeri->pticks = 0;
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1..a872d0a 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
 	0x103c0944, /* HP nc6220 */
 	0x103c0934, /* HP nc8220 */
 	0x103c006d, /* HP nx9105 */
+	0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
 	0x17340088, /* FSC Scenic-W */
 	0 /* end */
 };
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index a7ec703..ecce948 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
 module_param_array(tea575x_tuner, int, NULL, 0444);
 MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
 
+#define TUNER_DISABLED		(1<<3)
 #define TUNER_ONLY		(1<<4)
 #define TUNER_TYPE_MASK		(~TUNER_ONLY & 0xFFFF)
 
@@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip)
 
       __end_hw:
 #ifdef CONFIG_SND_FM801_TEA575X_BOOL
-	snd_tea575x_exit(&chip->tea);
+	if (!(chip->tea575x_tuner & TUNER_DISABLED))
+		snd_tea575x_exit(&chip->tea);
 #endif
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1236,7 +1238,6 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
 		if (snd_tea575x_init(&chip->tea)) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
 			return -ENODEV;
 		}
 	} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1251,11 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		}
 		if (tea575x_tuner == 4) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
-			return -ENODEV;
+			chip->tea575x_tuner = TUNER_DISABLED;
 		}
 	}
-	strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
+	if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
+		strlcpy(chip->tea.card,
+			snd_fm801_tea575x_gpios[(tea575x_tuner &
+						 TUNER_TYPE_MASK) - 1].name,
+			sizeof(chip->tea.card));
+	}
 #endif
 
 	*rchip = chip;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 486f6de..981b6fd 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2352,6 +2352,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1028, 0x02c6, "Dell Inspiron 1010", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 26a1521..fb6fbe4 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -508,7 +508,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
 		      int index, unsigned int pval, int dir,
 		      struct snd_kcontrol **kctlp)
 {
-	char tmp[32];
+	char tmp[44];
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
 	knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7bbc5f2..cf1fa36 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3097,6 +3097,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	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 Thinkpad", CXT5066_THINKPAD),
+	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),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 524ff26..4c7cd6b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -397,7 +397,7 @@ struct alc_spec {
 	unsigned int auto_mic:1;
 	unsigned int automute:1;	/* HP automute enabled */
 	unsigned int detect_line:1;	/* Line-out detection enabled */
-	unsigned int automute_lines:1;	/* automute line-out as well */
+	unsigned int automute_lines:1;	/* automute line-out as well; NOP when automute_hp_lo isn't set */
 	unsigned int automute_hp_lo:1;	/* both HP and LO available */
 
 	/* other flags */
@@ -1161,7 +1161,7 @@ static void update_speakers(struct hda_codec *codec)
 	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
 	    spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
 		return;
-	if (!spec->automute_lines || !spec->automute)
+	if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
 		on = 0;
 	else
 		on = spec->jack_present;
@@ -1494,7 +1494,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 	if (!spec->automute)
 		val = 0;
-	else if (!spec->automute_lines)
+	else if (!spec->automute_hp_lo || !spec->automute_lines)
 		val = 1;
 	else
 		val = 2;
@@ -1515,7 +1515,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
 		spec->automute = 0;
 		break;
 	case 1:
-		if (spec->automute && !spec->automute_lines)
+		if (spec->automute &&
+		    (!spec->automute_hp_lo || !spec->automute_lines))
 			return 0;
 		spec->automute = 1;
 		spec->automute_lines = 0;
@@ -1858,7 +1859,9 @@ do_sku:
 	 * 15   : 1 --> enable the function "Mute internal speaker
 	 *	        when the external headphone out jack is plugged"
 	 */
-	if (!spec->autocfg.hp_pins[0]) {
+	if (!spec->autocfg.hp_pins[0] &&
+	    !(spec->autocfg.line_out_pins[0] &&
+	      spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
 		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7f81cc2..5c42f3e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5470,6 +5470,7 @@ again:
 	switch (codec->vendor_id) {
 	case 0x111d76d1:
 	case 0x111d76d9:
+	case 0x111d76df:
 	case 0x111d76e5:
 	case 0x111d7666:
 	case 0x111d7667:
@@ -6399,6 +6400,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+	{ .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c0..2f0f836 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
 		.cpu_dai_name = "bfin-tdm.0",
 		.codec_dai_name ="ad193x-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad193x.5",
+		.codec_name = "spi0.5",
 		.ops = &bf5xx_ad193x_ops,
 	},
 	{
@@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
 		.cpu_dai_name = "bfin-tdm.1",
 		.codec_dai_name ="ad193x-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad193x.5",
+		.codec_name = "spi0.5",
 		.ops = &bf5xx_ad193x_ops,
 	},
 };
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5..f1a8be5 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -307,7 +307,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
 
 	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
-	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+		| (word_len << AD193X_DAC_WORD_LEN_SHFT);
 	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
 
 	reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b54..cccc2e8 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
 #define AD193X_DAC_LEFT_HIGH    (1 << 3)
 #define AD193X_DAC_BCLK_INV     (1 << 7)
 #define AD193X_DAC_CTRL2        0x804
-#define AD193X_DAC_WORD_LEN_MASK	0xC
+#define AD193X_DAC_WORD_LEN_SHFT        3
+#define AD193X_DAC_WORD_LEN_MASK        0x18
 #define AD193X_DAC_MASTER_MUTE  1
 #define AD193X_DAC_CHNL_MUTE    0x805
 #define AD193X_DACL1_MUTE       0
@@ -63,7 +64,7 @@
 #define AD193X_ADC_CTRL1        0x80f
 #define AD193X_ADC_SERFMT_MASK		0x60
 #define AD193X_ADC_SERFMT_STEREO	(0 << 5)
-#define AD193X_ADC_SERFMT_TDM		(1 << 2)
+#define AD193X_ADC_SERFMT_TDM		(1 << 5)
 #define AD193X_ADC_SERFMT_AUX		(2 << 5)
 #define AD193X_ADC_WORD_LEN_MASK	0x3
 #define AD193X_ADC_CTRL2        0x810
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index ff29380..c49317c 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -33,73 +33,31 @@
 #define SGTL5000_DAP_REG_OFFSET	0x0100
 #define SGTL5000_MAX_REG_OFFSET	0x013A
 
-/* default value of sgtl5000 registers except DAP */
-static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] =  {
-	0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */
-	0x0000, /* 0x0002, CHIP_DIG_POWER. */
-	0x0008, /* 0x0004, CHIP_CKL_CTRL */
-	0x0010, /* 0x0006, CHIP_I2S_CTRL */
-	0x0000, /* 0x0008, reserved */
-	0x0008, /* 0x000A, CHIP_SSS_CTRL */
-	0x0000, /* 0x000C, reserved */
-	0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */
-	0x3c3c, /* 0x0010, CHIP_DAC_VOL */
-	0x0000, /* 0x0012, reserved */
-	0x015f, /* 0x0014, CHIP_PAD_STRENGTH */
-	0x0000, /* 0x0016, reserved */
-	0x0000, /* 0x0018, reserved */
-	0x0000, /* 0x001A, reserved */
-	0x0000, /* 0x001E, reserved */
-	0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */
-	0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */
-	0x0111, /* 0x0024, CHIP_ANN_CTRL */
-	0x0000, /* 0x0026, CHIP_LINREG_CTRL */
-	0x0000, /* 0x0028, CHIP_REF_CTRL */
-	0x0000, /* 0x002A, CHIP_MIC_CTRL */
-	0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */
-	0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */
-	0x7060, /* 0x0030, CHIP_ANA_POWER */
-	0x5000, /* 0x0032, CHIP_PLL_CTRL */
-	0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */
-	0x0000, /* 0x0036, CHIP_ANA_STATUS */
-	0x0000, /* 0x0038, reserved */
-	0x0000, /* 0x003A, CHIP_ANA_TEST2 */
-	0x0000, /* 0x003C, CHIP_SHORT_CTRL */
-	0x0000, /* reserved */
-};
-
-/* default value of dap registers */
-static const u16 sgtl5000_dap_regs[] = {
-	0x0000, /* 0x0100, DAP_CONTROL */
-	0x0000, /* 0x0102, DAP_PEQ */
-	0x0040, /* 0x0104, DAP_BASS_ENHANCE */
-	0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */
-	0x0000, /* 0x0108, DAP_AUDIO_EQ */
-	0x0040, /* 0x010A, DAP_SGTL_SURROUND */
-	0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */
-	0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */
-	0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */
-	0x0000, /* 0x0112, reserved */
-	0x0000, /* 0x0114, reserved */
-	0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */
-	0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */
-	0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */
-	0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */
-	0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */
-	0x8000, /* 0x0120, DAP_MAIN_CHAN */
-	0x0000, /* 0x0122, DAP_MIX_CHAN */
-	0x0510, /* 0x0124, DAP_AVC_CTRL */
-	0x1473, /* 0x0126, DAP_AVC_THRESHOLD */
-	0x0028, /* 0x0128, DAP_AVC_ATTACK */
-	0x0050, /* 0x012A, DAP_AVC_DECAY */
-	0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */
-	0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */
-	0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */
-	0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */
-	0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */
-	0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */
-	0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */
-	0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */
+/* default value of sgtl5000 registers */
+static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] =  {
+	[SGTL5000_CHIP_CLK_CTRL] = 0x0008,
+	[SGTL5000_CHIP_I2S_CTRL] = 0x0010,
+	[SGTL5000_CHIP_SSS_CTRL] = 0x0008,
+	[SGTL5000_CHIP_DAC_VOL] = 0x3c3c,
+	[SGTL5000_CHIP_PAD_STRENGTH] = 0x015f,
+	[SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818,
+	[SGTL5000_CHIP_ANA_CTRL] = 0x0111,
+	[SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404,
+	[SGTL5000_CHIP_ANA_POWER] = 0x7060,
+	[SGTL5000_CHIP_PLL_CTRL] = 0x5000,
+	[SGTL5000_DAP_BASS_ENHANCE] = 0x0040,
+	[SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f,
+	[SGTL5000_DAP_SURROUND] = 0x0040,
+	[SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f,
+	[SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f,
+	[SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f,
+	[SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f,
+	[SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f,
+	[SGTL5000_DAP_MAIN_CHAN] = 0x8000,
+	[SGTL5000_DAP_AVC_CTRL] = 0x0510,
+	[SGTL5000_DAP_AVC_THRESHOLD] = 0x1473,
+	[SGTL5000_DAP_AVC_ATTACK] = 0x0028,
+	[SGTL5000_DAP_AVC_DECAY] = 0x0050,
 };
 
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */
@@ -1022,12 +980,10 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state)
 static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
 {
 	u16 *cache = codec->reg_cache;
-	int i;
-	int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1;
+	u16 reg;
 
 	/* restore regular registers */
-	for (i = 0; i < regular_regs; i++) {
-		int reg = i << 1;
+	for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) {
 
 		/* this regs depends on the others */
 		if (reg == SGTL5000_CHIP_ANA_POWER ||
@@ -1037,35 +993,31 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
 			reg == SGTL5000_CHIP_CLK_CTRL)
 			continue;
 
-		snd_soc_write(codec, reg, cache[i]);
+		snd_soc_write(codec, reg, cache[reg]);
 	}
 
 	/* restore dap registers */
-	for (i = SGTL5000_DAP_REG_OFFSET >> 1;
-			i < SGTL5000_MAX_REG_OFFSET >> 1; i++) {
-		int reg = i << 1;
-
-		snd_soc_write(codec, reg, cache[i]);
-	}
+	for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2)
+		snd_soc_write(codec, reg, cache[reg]);
 
 	/*
 	 * restore power and other regs according
 	 * to set_power() and set_clock()
 	 */
 	snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
-			cache[SGTL5000_CHIP_LINREG_CTRL >> 1]);
+			cache[SGTL5000_CHIP_LINREG_CTRL]);
 
 	snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER,
-			cache[SGTL5000_CHIP_ANA_POWER >> 1]);
+			cache[SGTL5000_CHIP_ANA_POWER]);
 
 	snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL,
-			cache[SGTL5000_CHIP_CLK_CTRL >> 1]);
+			cache[SGTL5000_CHIP_CLK_CTRL]);
 
 	snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL,
-			cache[SGTL5000_CHIP_REF_CTRL >> 1]);
+			cache[SGTL5000_CHIP_REF_CTRL]);
 
 	snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
-			cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]);
+			cache[SGTL5000_CHIP_LINE_OUT_CTRL]);
 	return 0;
 }
 
@@ -1460,16 +1412,6 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client,
 	if (!sgtl5000)
 		return -ENOMEM;
 
-	/*
-	 * copy DAP default values to default value array.
-	 * sgtl5000 register space has a big hole, merge it
-	 * at init phase makes life easy.
-	 * FIXME: should we drop 'const' of sgtl5000_regs?
-	 */
-	memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)),
-			sgtl5000_dap_regs,
-			SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET);
-
 	i2c_set_clientdata(client, sgtl5000);
 
 	ret = snd_soc_register_codec(&client->dev,
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 84f4ad5..9801cd7 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -431,7 +431,8 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	u16 reg = snd_soc_read(codec, SSM2602_PWR) & 0xff7f;
+	u16 reg = snd_soc_read(codec, SSM2602_PWR);
+	reg &= ~(PWR_POWER_OFF | PWR_OSC_PDN);
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index ffa2ffe..aa091a0 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1454,8 +1454,8 @@ static int wm8753_probe(struct snd_soc_codec *codec)
 	/* set the update bits */
 	snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100);
-	snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100);
-	snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100);
+	snd_soc_update_bits(codec, WM8753_LADC, 0x0100, 0x0100);
+	snd_soc_update_bits(codec, WM8753_RADC, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_LOUT1V, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_ROUT1V, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_LOUT2V, 0x0100, 0x0100);
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index fff695c..cbaf8b7 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -368,7 +368,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
 	.pcm_free	= &psc_dma_free,
 };
 
-static int mpc5200_hpcd_probe(struct of_device *op)
+static int mpc5200_hpcd_probe(struct platform_device *op)
 {
 	phys_addr_t fifo;
 	struct psc_dma *psc_dma;
@@ -486,7 +486,7 @@ out_unmap:
 	return ret;
 }
 
-static int mpc5200_hpcd_remove(struct of_device *op)
+static int mpc5200_hpcd_remove(struct platform_device *op)
 {
 	struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
 
@@ -518,7 +518,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match);
 static struct platform_driver mpc5200_hpcd_of_driver = {
 	.probe		= mpc5200_hpcd_probe,
 	.remove		= mpc5200_hpcd_remove,
-	.dev = {
+	.driver = {
 		.owner		= THIS_MODULE,
 		.name		= "mpc5200-pcm-audio",
 		.of_match_table    = mpc5200_hpcd_match,
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 07b7723..4b82290 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int err = 0;
 
+	if (mcbsp_data->active)
+		if (freq == mcbsp_data->in_freq)
+			return 0;
+		else
+			return -EBUSY;
+
 	/* The McBSP signal muxing functions are only available on McBSP1 */
 	if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR ||
 	    clk_id == OMAP_MCBSP_CLKR_SRC_CLKX ||
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index b644575..2b8350b 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -196,20 +196,20 @@ static int zylonite_probe(struct snd_soc_card *card)
 	if (clk_pout) {
 		pout = clk_get(NULL, "CLK_POUT");
 		if (IS_ERR(pout)) {
-			dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n",
+			dev_err(card->dev, "Unable to obtain CLK_POUT: %ld\n",
 				PTR_ERR(pout));
 			return PTR_ERR(pout);
 		}
 
 		ret = clk_enable(pout);
 		if (ret != 0) {
-			dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+			dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
 				ret);
 			clk_put(pout);
 			return ret;
 		}
 
-		dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n",
+		dev_dbg(card->dev, "MCLK enabled at %luHz\n",
 			clk_get_rate(pout));
 	}
 
@@ -241,7 +241,7 @@ static int zylonite_resume_pre(struct snd_soc_card *card)
 	if (clk_pout) {
 		ret = clk_enable(pout);
 		if (ret != 0)
-			dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+			dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
 				ret);
 	}
 
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 3b53ad5..14eb6ea 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = {
 	.cpu_dai_name	= "s3c2412-i2s",
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name	= "samsung-audio",
-	.codec_name	= "wm8750-codec.0-0x1a",
+	.codec_name	= "wm8750-codec.0-001a",
 	.init		= jive_wm8750_init,
 	.ops		= &jive_ops,
 };
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98..fa31d9c 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 
 	snd_soc_dapm_sync(dapm);
 
-	snd_jack_report(jack->jack, status);
+	snd_jack_report(jack->jack, jack->status);
 
 out:
 	mutex_unlock(&codec->mutex);
@@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 					      IRQF_TRIGGER_FALLING,
 					      gpios[i].name,
 					      &gpios[i]);
-		if (ret)
+		if (ret < 0)
 			goto err;
 
 		if (gpios[i].wake) {
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 3c271f9..6201710 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -309,9 +309,14 @@ static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 
 static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 {
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+
+	substream = pcm->streams[stream].substream;
+	if (!substream)
+		return;
 
+	buf = &substream->dma_buffer;
 	if (!buf->area)
 		return;
 
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0d6738a..7766478 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -56,6 +56,7 @@
 #define GPIO_HP_MUTE    BIT(1)
 #define GPIO_INT_MIC_EN BIT(2)
 #define GPIO_EXT_MIC_EN BIT(3)
+#define GPIO_HP_DET     BIT(4)
 
 struct tegra_wm8903 {
 	struct tegra_asoc_utils_data util_data;
@@ -304,6 +305,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 		snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack,
 					1,
 					&tegra_wm8903_hp_jack_gpio);
+		machine->gpio_requested |= GPIO_HP_DET;
 	}
 
 	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
@@ -429,10 +431,10 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = machine->pdata;
 
-	snd_soc_unregister_card(card);
-
-	tegra_asoc_utils_fini(&machine->util_data);
-
+	if (machine->gpio_requested & GPIO_HP_DET)
+		snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
+					1,
+					&tegra_wm8903_hp_jack_gpio);
 	if (machine->gpio_requested & GPIO_EXT_MIC_EN)
 		gpio_free(pdata->gpio_ext_mic_en);
 	if (machine->gpio_requested & GPIO_INT_MIC_EN)
@@ -441,6 +443,11 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 		gpio_free(pdata->gpio_hp_mute);
 	if (machine->gpio_requested & GPIO_SPKR_EN)
 		gpio_free(pdata->gpio_spkr_en);
+	machine->gpio_requested = 0;
+
+	snd_soc_unregister_card(card);
+
+	tegra_asoc_utils_fini(&machine->util_data);
 
 	kfree(machine);
 
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index d0d493c..2cf87f5 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev)
 
 	for (i = 0; i < N_URBS; i++) {
 		usb_kill_urb(dev->data_urbs_in[i]);
-		usb_kill_urb(dev->data_urbs_out[i]);
+
+		if (test_bit(i, &dev->outurb_active_mask))
+			usb_kill_urb(dev->data_urbs_out[i]);
 	}
+
+	dev->outurb_active_mask = 0;
 }
 
 static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
@@ -612,8 +616,9 @@ static void read_completed(struct urb *urb)
 {
 	struct snd_usb_caiaq_cb_info *info = urb->context;
 	struct snd_usb_caiaqdev *dev;
-	struct urb *out;
-	int frame, len, send_it = 0, outframe = 0;
+	struct urb *out = NULL;
+	int i, frame, len, send_it = 0, outframe = 0;
+	size_t offset = 0;
 
 	if (urb->status || !info)
 		return;
@@ -623,7 +628,17 @@ static void read_completed(struct urb *urb)
 	if (!dev->streaming)
 		return;
 
-	out = dev->data_urbs_out[info->index];
+	/* find an unused output urb that is unused */
+	for (i = 0; i < N_URBS; i++)
+		if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) {
+			out = dev->data_urbs_out[i];
+			break;
+		}
+
+	if (!out) {
+		log("Unable to find an output urb to use\n");
+		goto requeue;
+	}
 
 	/* read the recently received packet and send back one which has
 	 * the same layout */
@@ -634,7 +649,8 @@ static void read_completed(struct urb *urb)
 		len = urb->iso_frame_desc[outframe].actual_length;
 		out->iso_frame_desc[outframe].length = len;
 		out->iso_frame_desc[outframe].actual_length = 0;
-		out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+		out->iso_frame_desc[outframe].offset = offset;
+		offset += len;
 
 		if (len > 0) {
 			spin_lock(&dev->spinlock);
@@ -650,11 +666,15 @@ static void read_completed(struct urb *urb)
 	}
 
 	if (send_it) {
-		out->number_of_packets = FRAMES_PER_URB;
+		out->number_of_packets = outframe;
 		out->transfer_flags = URB_ISO_ASAP;
 		usb_submit_urb(out, GFP_ATOMIC);
+	} else {
+		struct snd_usb_caiaq_cb_info *oinfo = out->context;
+		clear_bit(oinfo->index, &dev->outurb_active_mask);
 	}
 
+requeue:
 	/* re-submit inbound urb */
 	for (frame = 0; frame < FRAMES_PER_URB; frame++) {
 		urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
@@ -676,6 +696,8 @@ static void write_completed(struct urb *urb)
 		dev->output_running = 1;
 		wake_up(&dev->prepare_wait_queue);
 	}
+
+	clear_bit(info->index, &dev->outurb_active_mask);
 }
 
 static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
@@ -827,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	if (!dev->data_cb_info)
 		return -ENOMEM;
 
+	dev->outurb_active_mask = 0;
+	BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8));
+
 	for (i = 0; i < N_URBS; i++) {
 		dev->data_cb_info[i].dev = dev;
 		dev->data_cb_info[i].index = i;
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index b2b3101..3f9c633 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -96,6 +96,7 @@ struct snd_usb_caiaqdev {
 	int input_panic, output_panic, warned;
 	char *audio_in_buf, *audio_out_buf;
 	unsigned int samplerates, bpp;
+	unsigned long outurb_active_mask;
 
 	struct snd_pcm_substream *sub_playback[MAX_STREAMS];
 	struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 4432ef7..a213813 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] =  { KEY_C, KEY_B, KEY_A };
 static unsigned short keycode_rk2[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
 					 KEY_5, KEY_6, KEY_7 };
 static unsigned short keycode_rk3[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
-					 KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
+					 KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
 
 static unsigned short keycode_kore[] = {
 	KEY_FN_F1,      /* "menu"               */
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 220c616..57a8e2d 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -529,8 +529,11 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
 	return chip;
 
  __error:
-	if (chip && !chip->num_interfaces)
-		snd_card_free(chip->card);
+	if (chip) {
+		if (!chip->num_interfaces)
+			snd_card_free(chip->card);
+		chip->probing = 0;
+	}
 	mutex_unlock(&register_mutex);
  __err_val:
 	return NULL;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index b0ef9f5..05842c8 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -352,7 +352,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 			continue;
 		}
 		if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
-		    ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
+		    ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) {
 			snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
 				   dev->devnum, iface_no, altno);
 			continue;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c22fa76..cdd19d7 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p,
 	if (p && p->dB) {
 		cval->dBmin = p->dB->min;
 		cval->dBmax = p->dB->max;
+		cval->initialized = 1;
 	}
 }
 
@@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 				" Switch" : " Volume");
 		if (control == UAC_FU_VOLUME) {
 			check_mapped_dB(map, cval);
-			if (cval->dBmin < cval->dBmax) {
+			if (cval->dBmin < cval->dBmax || !cval->initialized) {
 				kctl->tlv.c = mixer_vol_tlv;
 				kctl->vd[0].access |= 
 					SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1191,6 +1192,11 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 
 	if (state->mixer->protocol == UAC_VERSION_1) {
 		csize = hdr->bControlSize;
+		if (!csize) {
+			snd_printdd(KERN_ERR "usbaudio: unit %u: "
+				    "invalid bControlSize == 0\n", unitid);
+			return -EINVAL;
+		}
 		channels = (hdr->bLength - 7) / csize - 1;
 		bmaControls = hdr->bmaControls;
 	} else {
@@ -1934,15 +1940,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 	struct mixer_build state;
 	int err;
 	const struct usbmix_ctl_map *map;
-	struct usb_host_interface *hostif;
 	void *p;
 
-	hostif = mixer->chip->ctrl_intf;
 	memset(&state, 0, sizeof(state));
 	state.chip = mixer->chip;
 	state.mixer = mixer;
-	state.buffer = hostif->extra;
-	state.buflen = hostif->extralen;
+	state.buffer = mixer->hostif->extra;
+	state.buflen = mixer->hostif->extralen;
 
 	/* check the mapping table */
 	for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1955,7 +1959,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 	}
 
 	p = NULL;
-	while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
+	while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
+					    p, UAC_OUTPUT_TERMINAL)) != NULL) {
 		if (mixer->protocol == UAC_VERSION_1) {
 			struct uac1_output_terminal_descriptor *desc = p;
 
@@ -2162,17 +2167,15 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
 /* create the handler for the optional status interrupt endpoint */
 static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
 {
-	struct usb_host_interface *hostif;
 	struct usb_endpoint_descriptor *ep;
 	void *transfer_buffer;
 	int buffer_length;
 	unsigned int epnum;
 
-	hostif = mixer->chip->ctrl_intf;
 	/* we need one interrupt input endpoint */
-	if (get_iface_desc(hostif)->bNumEndpoints < 1)
+	if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
 		return 0;
-	ep = get_endpoint(hostif, 0);
+	ep = get_endpoint(mixer->hostif, 0);
 	if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
 		return 0;
 
@@ -2202,7 +2205,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 	};
 	struct usb_mixer_interface *mixer;
 	struct snd_info_entry *entry;
-	struct usb_host_interface *host_iface;
 	int err;
 
 	strcpy(chip->card->mixername, "USB Mixer");
@@ -2219,8 +2221,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 		return -ENOMEM;
 	}
 
-	host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
-	switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+	mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+	switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
 	case UAC_VERSION_1:
 	default:
 		mixer->protocol = UAC_VERSION_1;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ae1a14d..81b2d8a 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -3,6 +3,7 @@
 
 struct usb_mixer_interface {
 	struct snd_usb_audio *chip;
+	struct usb_host_interface *hostif;
 	struct list_head list;
 	unsigned int ignore_ctl_error;
 	struct urb *urb;