Merge commit 'v2.6.39' into for-2.6.40/core

Since for-2.6.40/core was forked off the 2.6.39 devel tree, we've
had churn in the core area that makes it difficult to handle
patches for eg cfq or blk-throttle. Instead of requiring that they
be based in older versions with bugs that have been fixed later
in the rc cycle, merge in 2.6.39 final.

Also fixes up conflicts in the below files.

Conflicts:
	drivers/block/paride/pcd.c
	drivers/cdrom/viocd.c
	drivers/ide/ide-cd.c

Signed-off-by: Jens Axboe <jaxboe@fusionio.com>

22 files changed, 160 insertions, 82 deletions

diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index 58804c7..fd2188c 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -170,7 +170,7 @@ static void tas_set_volume(struct tas *tas)
 	/* analysing the volume and mixer tables shows
 	 * that they are similar enough when we shift
 	 * the mixer table down by 4 bits. The error
-	 * is minuscule, in just one item the error
+	 * is miniscule, in just one item the error
 	 * is 1, at a value of 0x07f17b (mixer table
 	 * value is 0x07f17a) */
 	tmp = tas_gaintable[left];
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 33f0ba5..62e9591 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -44,10 +44,10 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = {
 	.channels_min = 1,
 	.channels_max = 2,
 	.buffer_bytes_max = 0x10000,
-	.period_bytes_min = 0x1,
+	.period_bytes_min = 0x20,
 	.period_bytes_max = 0x1000,
 	.periods_min = 2,
-	.periods_max = 32,
+	.periods_max = 1024,
 };
 
 #ifndef CHIP_AU8820
@@ -140,6 +140,9 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
 					SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0)
 		return err;
 
+	snd_pcm_hw_constraint_step(runtime, 0,
+					SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64);
+
 	if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
 #ifndef CHIP_AU8820
 		if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) {
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 430f41d..759ade1 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -937,6 +937,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
 static void restore_shutup_pins(struct hda_codec *codec)
 {
@@ -953,6 +954,7 @@ static void restore_shutup_pins(struct hda_codec *codec)
 	}
 	codec->pins_shutup = 0;
 }
+#endif
 
 static void init_hda_cache(struct hda_cache_rec *cache,
 			   unsigned int record_size);
@@ -1329,6 +1331,7 @@ static void purify_inactive_streams(struct hda_codec *codec)
 	}
 }
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* clean up all streams; called from suspend */
 static void hda_cleanup_all_streams(struct hda_codec *codec)
 {
@@ -1340,6 +1343,7 @@ static void hda_cleanup_all_streams(struct hda_codec *codec)
 			really_cleanup_stream(codec, p);
 	}
 }
+#endif
 
 /*
  * amp access functions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 52928d9..c82979a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1704,11 +1704,11 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
 				   codec->chip_name, fix->type);
 			break;
 		}
-		if (!fix[id].chained)
+		if (!fix->chained)
 			break;
 		if (++depth > 10)
 			break;
-		id = fix[id].chain_id;
+		id = fix->chain_id;
 	}
 }
 
@@ -5645,6 +5645,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
 static struct snd_pci_quirk beep_white_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
 	SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
+	SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
 	SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
 	{}
 };
@@ -9863,6 +9864,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
 
 	SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
@@ -10699,7 +10701,6 @@ enum {
 	PINFIX_LENOVO_Y530,
 	PINFIX_PB_M5210,
 	PINFIX_ACER_ASPIRE_7736,
-	PINFIX_GIGABYTE_880GM,
 };
 
 static const struct alc_fixup alc882_fixups[] = {
@@ -10731,13 +10732,6 @@ static const struct alc_fixup alc882_fixups[] = {
 		.type = ALC_FIXUP_SKU,
 		.v.sku = ALC_FIXUP_SKU_IGNORE,
 	},
-	[PINFIX_GIGABYTE_880GM] = {
-		.type = ALC_FIXUP_PINS,
-		.v.pins = (const struct alc_pincfg[]) {
-			{ 0x14, 0x1114410 }, /* set as speaker */
-			{ }
-		}
-	},
 };
 
 static struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -10745,7 +10739,6 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
-	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM),
 	{}
 };
 
@@ -14868,6 +14861,23 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
 	alc_write_coef_idx(codec, 0x1e, coef | 0x80);
 }
 
+static void alc271_fixup_dmic(struct hda_codec *codec,
+			      const struct alc_fixup *fix, int action)
+{
+	static struct hda_verb verbs[] = {
+		{0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
+		{0x20, AC_VERB_SET_PROC_COEF, 0x4000},
+		{}
+	};
+	unsigned int cfg;
+
+	if (strcmp(codec->chip_name, "ALC271X"))
+		return;
+	cfg = snd_hda_codec_get_pincfg(codec, 0x12);
+	if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
+		snd_hda_sequence_write(codec, verbs);
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -14876,6 +14886,7 @@ enum {
 	ALC269_FIXUP_ASUS_G73JW,
 	ALC269_FIXUP_LENOVO_EAPD,
 	ALC275_FIXUP_SONY_HWEQ,
+	ALC271_FIXUP_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -14929,7 +14940,11 @@ static const struct alc_fixup alc269_fixups[] = {
 		.v.func = alc269_fixup_hweq,
 		.chained = true,
 		.chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
-	}
+	},
+	[ALC271_FIXUP_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc271_fixup_dmic,
+	},
 };
 
 static struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -14938,6 +14953,7 @@ static 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_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
@@ -18782,6 +18798,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 		      ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
 	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
+	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+		      ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
@@ -19455,7 +19473,7 @@ enum {
 	ALC662_FIXUP_IDEAPAD,
 	ALC272_FIXUP_MARIO,
 	ALC662_FIXUP_CZC_P10T,
-	ALC662_FIXUP_GIGABYTE,
+	ALC662_FIXUP_SKU_IGNORE,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -19484,20 +19502,17 @@ static const struct alc_fixup alc662_fixups[] = {
 			{}
 		}
 	},
-	[ALC662_FIXUP_GIGABYTE] = {
-		.type = ALC_FIXUP_PINS,
-		.v.pins = (const struct alc_pincfg[]) {
-			{ 0x14, 0x1114410 }, /* set as speaker */
-			{ }
-		}
+	[ALC662_FIXUP_SKU_IGNORE] = {
+		.type = ALC_FIXUP_SKU,
+		.v.sku = ALC_FIXUP_SKU_IGNORE,
 	},
 };
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
-	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 1371b57c..0997031 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1292,14 +1292,18 @@ static void notify_aa_path_ctls(struct hda_codec *codec)
 {
 	int i;
 	struct snd_ctl_elem_id id;
-	const char *labels[] = {"Mic", "Front Mic", "Line"};
+	const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"};
+	struct snd_kcontrol *ctl;
 
 	memset(&id, 0, sizeof(id));
 	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 	for (i = 0; i < ARRAY_SIZE(labels); i++) {
 		sprintf(id.name, "%s Playback Volume", labels[i]);
-		snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
-			       &id);
+		ctl = snd_hda_find_mixer_ctl(codec, id.name);
+		if (ctl)
+			snd_ctl_notify(codec->bus->card,
+					SNDRV_CTL_EVENT_MASK_VALUE,
+					&ctl->id);
 	}
 }
 
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index f7cd346f..f5ccdbf 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
 	snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
 		ARRAY_SIZE(jz4740_codec_dapm_routes));
 
-	snd_soc_dapm_new_widgets(codec);
-
 	jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index a54d2a5..4d9fb27 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -927,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = {
 		.owner		= THIS_MODULE,
 	},
 	.probe		= sn95031_device_probe,
-	.remove		= sn95031_device_remove,
+	.remove		= __devexit_p(sn95031_device_remove),
 };
 
 static int __init sn95031_init(void)
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 2727bef..b04d280 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -139,7 +139,7 @@ SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
 SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
 
 SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
-SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 7, 1, 0),
+SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
 SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
 
 SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1),
@@ -602,7 +602,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
 	.read = ssm2602_read_reg_cache,
 	.write = ssm2602_write,
 	.set_bias_level = ssm2602_set_bias_level,
-	.reg_cache_size = sizeof(ssm2602_reg),
+	.reg_cache_size = ARRAY_SIZE(ssm2602_reg),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = ssm2602_reg,
 };
@@ -614,7 +614,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
  *    low  = 0x1a
  *    high = 0x1b
  */
-static int ssm2602_i2c_probe(struct i2c_client *i2c,
+static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
 	struct ssm2602_priv *ssm2602;
@@ -635,7 +635,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c,
 	return ret;
 }
 
-static int ssm2602_i2c_remove(struct i2c_client *client)
+static int __devexit ssm2602_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
 	kfree(i2c_get_clientdata(client));
@@ -655,7 +655,7 @@ static struct i2c_driver ssm2602_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe = ssm2602_i2c_probe,
-	.remove = ssm2602_i2c_remove,
+	.remove = __devexit_p(ssm2602_i2c_remove),
 	.id_table = ssm2602_i2c_id,
 };
 #endif
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 48ffd40..a7b8f30 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -601,9 +601,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
 	.reg_cache_step = 1,
 	.read = uda134x_read_reg_cache,
 	.write = uda134x_write,
-#ifdef POWER_OFF_ON_STANDBY
 	.set_bias_level = uda134x_set_bias_level,
-#endif
 };
 
 static int __devinit uda134x_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ae1cadf..824d1c8 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re
 	case WM8903_REVISION_NUMBER:
 	case WM8903_INTERRUPT_STATUS_1:
 	case WM8903_WRITE_SEQUENCER_4:
-	case WM8903_POWER_MANAGEMENT_3:
-	case WM8903_POWER_MANAGEMENT_2:
 	case WM8903_DC_SERVO_READBACK_1:
 	case WM8903_DC_SERVO_READBACK_2:
 	case WM8903_DC_SERVO_READBACK_3:
@@ -694,7 +692,7 @@ SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
 SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
 
 SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
-		 WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv),
+		 WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
 SOC_ENUM("ADC Companding Mode", adc_companding),
 SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
 
@@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
 SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
 		   right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
 
-SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
-		   4, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
+		   1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
 		   0, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0,
 		   NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0,
 		   NULL, 0),
 
 SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0),
 
 SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
 		   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
 		   NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
 		   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
 		   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
 		   NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
 		   NULL, 0),
 
 SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
@@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Left Speaker PGA", NULL, "Left Speaker Mixer" },
 	{ "Right Speaker PGA", NULL, "Right Speaker Mixer" },
 
-	{ "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
-	{ "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
-	{ "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
-	{ "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
+	{ "HPL_ENA", NULL, "Left Headphone Output PGA" },
+	{ "HPR_ENA", NULL, "Right Headphone Output PGA" },
+	{ "HPL_ENA_DLY", NULL, "HPL_ENA" },
+	{ "HPR_ENA_DLY", NULL, "HPR_ENA" },
+	{ "LINEOUTL_ENA", NULL, "Left Line Output PGA" },
+	{ "LINEOUTR_ENA", NULL, "Right Line Output PGA" },
+	{ "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" },
+	{ "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" },
 
 	{ "HPL_DCS", NULL, "DCS Master" },
 	{ "HPR_DCS", NULL, "DCS Master" },
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3290333..84e1bd1 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch volume updates (right only; we always do left then right). */
+	snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME,
+			    WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME,
 			    WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
+	snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME,
+			    WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME,
 			    WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
+	snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME,
+			    WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
 	snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME,
 			    WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
+	snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME,
+			    WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME,
 			    WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
+	snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME,
+			    WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME,
 			    WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
+	snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME,
+			    WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
 	snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME,
 			    WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
+	snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME,
+			    WM8994_DAC1_VU, WM8994_DAC1_VU);
 	snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME,
 			    WM8994_DAC1_VU, WM8994_DAC1_VU);
+	snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME,
+			    WM8994_DAC2_VU, WM8994_DAC2_VU);
 	snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME,
 			    WM8994_DAC2_VU, WM8994_DAC2_VU);
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 7b6b3c1..4005e9a 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 
 	{ "SPKL", "Input Switch", "MIXINL" },
 	{ "SPKL", "IN1LP Switch", "IN1LP" },
-	{ "SPKL", "Output Switch", "Left Output Mixer" },
+	{ "SPKL", "Output Switch", "Left Output PGA" },
 	{ "SPKL", NULL, "TOCLK" },
 
 	{ "SPKR", "Input Switch", "MIXINR" },
 	{ "SPKR", "IN1RP Switch", "IN1RP" },
-	{ "SPKR", "Output Switch", "Right Output Mixer" },
+	{ "SPKR", "Output Switch", "Right Output PGA" },
 	{ "SPKR", NULL, "TOCLK" },
 
 	{ "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
@@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "SPKOUTRP", NULL, "SPKR Driver" },
 	{ "SPKOUTRN", NULL, "SPKR Driver" },
 
-	{ "Left Headphone Mux", "Mixer", "Left Output Mixer" },
-	{ "Right Headphone Mux", "Mixer", "Right Output Mixer" },
+	{ "Left Headphone Mux", "Mixer", "Left Output PGA" },
+	{ "Right Headphone Mux", "Mixer", "Right Output PGA" },
 
 	{ "Headphone PGA", NULL, "Left Headphone Mux" },
 	{ "Headphone PGA", NULL, "Right Headphone Mux" },
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index a5af834..4ddc6d3 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -434,17 +434,21 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
 		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26));
+		mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+				ACLKX | AHCLKX | AFSX);
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
 		/* codec is clock master and frame slave */
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
 		mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
 		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26));
+		mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+				ACLKX | ACLKR);
+		mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+				AFSX | AFSR);
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		/* codec is clock and frame master */
@@ -454,7 +458,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
 		mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-		mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26));
+		mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+				ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
 		break;
 
 	default:
@@ -644,7 +649,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
 		mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
 		mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
 
-		if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+		if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
 			mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
 					FSXMOD(dev->tdm_slots), FSXMOD(0x1FF));
 		else
@@ -660,7 +665,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
 				AHCLKRE);
 		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
 
-		if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+		if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
 			mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG,
 					FSRMOD(dev->tdm_slots), FSRMOD(0x1FF));
 		else
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 419bf4f..cd22a54 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -133,7 +133,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream,
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	uint32_t conf;
 
-	if (!dai->active)
+	if (dai->active)
 		return;
 
 	conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index b2e9198..6b1f9d3 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = {
 static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
 					int state)
 {
-	spin_lock(&stream->status_lock);
+	unsigned long flags;
+	spin_lock_irqsave(&stream->status_lock, flags);
 	stream->stream_status = state;
-	spin_unlock(&stream->status_lock);
+	spin_unlock_irqrestore(&stream->status_lock, flags);
 }
 
 static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
 {
 	int state;
+	unsigned long flags;
 
-	spin_lock(&stream->status_lock);
+	spin_lock_irqsave(&stream->status_lock, flags);
 	state = stream->stream_status;
-	spin_unlock(&stream->status_lock);
+	spin_unlock_irqrestore(&stream->status_lock, flags);
 	return state;
 }
 
@@ -374,6 +376,11 @@ static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
 static struct snd_pcm_ops sst_platform_ops = {
 	.open = sst_platform_open,
 	.close = sst_platform_close,
@@ -382,6 +389,7 @@ static struct snd_pcm_ops sst_platform_ops = {
 	.trigger = sst_platform_pcm_trigger,
 	.pointer = sst_platform_pcm_pointer,
 	.hw_params = sst_platform_pcm_hw_params,
+	.hw_free = sst_platform_pcm_hw_free,
 };
 
 static void sst_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index f6b3a3c..0e80dae 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -236,18 +236,18 @@ static struct snd_soc_dai_link goni_dai[] = {
 	.name = "WM8994",
 	.stream_name = "WM8994 HiFi",
 	.cpu_dai_name = "samsung-i2s.0",
-	.codec_dai_name = "wm8994-hifi",
+	.codec_dai_name = "wm8994-aif1",
 	.platform_name = "samsung-audio",
-	.codec_name = "wm8994-codec.0-0x1a",
+	.codec_name = "wm8994-codec.0-001a",
 	.init = goni_wm8994_init,
 	.ops = &goni_hifi_ops,
 }, {
 	.name = "WM8994 Voice",
 	.stream_name = "Voice",
 	.cpu_dai_name = "goni-voice-dai",
-	.codec_dai_name = "wm8994-voice",
+	.codec_dai_name = "wm8994-aif2",
 	.platform_name = "samsung-audio",
-	.codec_name = "wm8994-codec.0-0x1a",
+	.codec_name = "wm8994-codec.0-001a",
 	.ops = &goni_voice_ops,
 },
 };
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 38aac7d..9c7e8b4 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai,
 	ctl = readl(regs + S3C_PCM_CTL);
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_NB_NF:
-		/* Nothing to do, NB_NF by default */
+	case SND_SOC_DAIFMT_IB_NF:
+		/* Nothing to do, IB_NF by default */
 		break;
 	default:
 		dev_err(pcm->dev, "Unsupported clock inversion!\n");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 0c9997e..23c0e83 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev)
 	master->fsib.master	= master;
 
 	pm_runtime_enable(&pdev->dev);
-	pm_runtime_resume(&pdev->dev);
 	dev_set_drvdata(&pdev->dev, master);
 
+	pm_runtime_get_sync(&pdev->dev);
 	fsi_soft_all_reset(master);
+	pm_runtime_put_sync(&pdev->dev);
 
 	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
 			  id_entry->name, master);
@@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev)
 		goto exit_free_irq;
 	}
 
-	return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+	ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
+				    ARRAY_SIZE(fsi_soc_dai));
+	if (ret < 0) {
+		dev_err(&pdev->dev, "cannot snd dai register\n");
+		goto exit_snd_soc;
+	}
+
+	return ret;
 
+exit_snd_soc:
+	snd_soc_unregister_platform(&pdev->dev);
 exit_free_irq:
 	free_irq(irq, master);
 exit_iounmap:
@@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev)
 
 	master = dev_get_drvdata(&pdev->dev);
 
-	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
-	snd_soc_unregister_platform(&pdev->dev);
-
+	free_irq(master->irq, master);
 	pm_runtime_disable(&pdev->dev);
 
-	free_irq(master->irq, master);
+	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
+	snd_soc_unregister_platform(&pdev->dev);
 
 	iounmap(master->base);
 	kfree(master);
@@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit);
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_ALIAS("platform:fsi-pcm-audio");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b76b74d..dd55d10 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -629,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			runtime->hw.rates |= codec_dai_drv->capture.rates;
 	}
 
+	ret = -EINVAL;
 	snd_pcm_limit_hw_rates(runtime);
 	if (!runtime->hw.rates) {
 		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
@@ -640,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			codec_dai->name, cpu_dai->name);
 		goto config_err;
 	}
-	if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
+	if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+	    runtime->hw.channels_min > runtime->hw.channels_max) {
 		printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
 				codec_dai->name, cpu_dai->name);
 		goto config_err;
@@ -2060,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = {
 	.resume = snd_soc_resume,
 	.poweroff = snd_soc_poweroff,
 };
+EXPORT_SYMBOL_GPL(snd_soc_pm_ops);
 
 /* ASoC platform driver */
 static struct platform_driver soc_driver = {
@@ -3288,6 +3291,8 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	if (!card->name || !card->dev)
 		return -EINVAL;
 
+	dev_set_drvdata(card->dev, card);
+
 	snd_soc_initialize_card_lists(card);
 
 	soc_init_card_debugfs(card);
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index 8585957..556a571 100644
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = {
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
 	},
 	.probe = tegra_snd_harmony_probe,
 	.remove = __devexit_p(tegra_snd_harmony_remove),
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5b792d2..f079b5e 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -176,9 +176,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 			if (!rate)
 				continue;
 			/* C-Media CM6501 mislabels its 96 kHz altsetting */
+			/* Terratec Aureon 7.1 USB C-Media 6206, too */
 			if (rate == 48000 && nr_rates == 1 &&
 			    (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
-			     chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
+			     chip->usb_id == USB_ID(0x0d8c, 0x0102) ||
+			     chip->usb_id == USB_ID(0x0ccd, 0x00b1)) &&
 			    fp->altsetting == 5 && fp->maxpacksize == 392)
 				rate = 96000;
 			/* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ec07e62..1b94ec3 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -533,6 +533,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 
 	case USB_ID(0x0d8c, 0x0102):
 		/* C-Media CM6206 / CM106-Like Sound Device */
+	case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */
 		return snd_usb_cm6206_boot_quirk(dev);
 
 	case USB_ID(0x133e, 0x0815):