Merge tag 'v3.10' into sched/core

Merge in a recent upstream commit:

  c2853c8df57f include/linux/math64.h: add div64_ul()

because:

  72a4cf20cb71 sched: Change cfs_rq load avg to unsigned long

relies on it.

[ We don't rebase sched/core for this, because the handful of
  followup commits after the broken commit are not behavioral
  changes so are unlikely to be needed during bisection. ]

Signed-off-by: Ingo Molnar <mingo@kernel.org>

22 files changed, 201 insertions, 89 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index ccfa383f1fd..f9281815595 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1649,6 +1649,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 	}
 	if (!snd_pcm_stream_linked(substream)) {
 		substream->group = group;
+		group = NULL;
 		spin_lock_init(&substream->group->lock);
 		INIT_LIST_HEAD(&substream->group->substreams);
 		list_add_tail(&substream->link_list, &substream->group->substreams);
@@ -1663,8 +1664,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
  _nolock:
 	snd_card_unref(substream1->pcm->card);
 	fput_light(file, fput_needed);
-	if (res < 0)
-		kfree(group);
+	kfree(group);
 	return res;
 }
 
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ae85bbd2e6f..4b1524a861f 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -788,6 +788,8 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable)
 		return;
 	if (codec->inv_eapd)
 		enable = !enable;
+	if (spec->keep_eapd_on && !enable)
+		return;
 	snd_hda_codec_update_cache(codec, pin, 0,
 				   AC_VERB_SET_EAPD_BTLENABLE,
 				   enable ? 0x02 : 0x00);
@@ -1938,17 +1940,7 @@ static int create_speaker_out_ctls(struct hda_codec *codec)
  * independent HP controls
  */
 
-/* update HP auto-mute state too */
-static void update_hp_automute_hook(struct hda_codec *codec)
-{
-	struct hda_gen_spec *spec = codec->spec;
-
-	if (spec->hp_automute_hook)
-		spec->hp_automute_hook(codec, NULL);
-	else
-		snd_hda_gen_hp_automute(codec, NULL);
-}
-
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack);
 static int indep_hp_info(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_info *uinfo)
 {
@@ -2009,7 +2001,7 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol,
 		else
 			*dacp = spec->alt_dac_nid;
 
-		update_hp_automute_hook(codec);
+		call_hp_automute(codec, NULL);
 		ret = 1;
 	}
  unlock:
@@ -2305,7 +2297,7 @@ static void update_hp_mic(struct hda_codec *codec, int adc_mux, bool force)
 		else
 			val = PIN_HP;
 		set_pin_target(codec, pin, val, true);
-		update_hp_automute_hook(codec);
+		call_hp_automute(codec, NULL);
 	}
 }
 
@@ -2714,7 +2706,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
 			val = snd_hda_get_default_vref(codec, nid);
 	}
 	snd_hda_set_pin_ctl_cache(codec, nid, val);
-	update_hp_automute_hook(codec);
+	call_hp_automute(codec, NULL);
 
 	return 1;
 }
@@ -3859,20 +3851,42 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 }
 EXPORT_SYMBOL_HDA(snd_hda_gen_mic_autoswitch);
 
-/* update jack retasking */
-static void update_automute_all(struct hda_codec *codec)
+/* call appropriate hooks */
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	if (spec->hp_automute_hook)
+		spec->hp_automute_hook(codec, jack);
+	else
+		snd_hda_gen_hp_automute(codec, jack);
+}
 
-	update_hp_automute_hook(codec);
+static void call_line_automute(struct hda_codec *codec,
+			       struct hda_jack_tbl *jack)
+{
+	struct hda_gen_spec *spec = codec->spec;
 	if (spec->line_automute_hook)
-		spec->line_automute_hook(codec, NULL);
+		spec->line_automute_hook(codec, jack);
 	else
-		snd_hda_gen_line_automute(codec, NULL);
+		snd_hda_gen_line_automute(codec, jack);
+}
+
+static void call_mic_autoswitch(struct hda_codec *codec,
+				struct hda_jack_tbl *jack)
+{
+	struct hda_gen_spec *spec = codec->spec;
 	if (spec->mic_autoswitch_hook)
-		spec->mic_autoswitch_hook(codec, NULL);
+		spec->mic_autoswitch_hook(codec, jack);
 	else
-		snd_hda_gen_mic_autoswitch(codec, NULL);
+		snd_hda_gen_mic_autoswitch(codec, jack);
+}
+
+/* update jack retasking */
+static void update_automute_all(struct hda_codec *codec)
+{
+	call_hp_automute(codec, NULL);
+	call_line_automute(codec, NULL);
+	call_mic_autoswitch(codec, NULL);
 }
 
 /*
@@ -4009,9 +4023,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
 		snd_printdd("hda-codec: Enable HP auto-muting on NID 0x%x\n",
 			    nid);
 		snd_hda_jack_detect_enable_callback(codec, nid, HDA_GEN_HP_EVENT,
-						    spec->hp_automute_hook ?
-						    spec->hp_automute_hook :
-						    snd_hda_gen_hp_automute);
+						    call_hp_automute);
 		spec->detect_hp = 1;
 	}
 
@@ -4024,9 +4036,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
 				snd_printdd("hda-codec: Enable Line-Out auto-muting on NID 0x%x\n", nid);
 				snd_hda_jack_detect_enable_callback(codec, nid,
 								    HDA_GEN_FRONT_EVENT,
-								    spec->line_automute_hook ?
-								    spec->line_automute_hook :
-								    snd_hda_gen_line_automute);
+								    call_line_automute);
 				spec->detect_lo = 1;
 			}
 		spec->automute_lo_possible = spec->detect_hp;
@@ -4068,9 +4078,7 @@ static bool auto_mic_check_imux(struct hda_codec *codec)
 		snd_hda_jack_detect_enable_callback(codec,
 						    spec->am_entry[i].pin,
 						    HDA_GEN_MIC_EVENT,
-						    spec->mic_autoswitch_hook ?
-						    spec->mic_autoswitch_hook :
-						    snd_hda_gen_mic_autoswitch);
+						    call_mic_autoswitch);
 	return true;
 }
 
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 54e66516037..76200314ee9 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -222,6 +222,7 @@ struct hda_gen_spec {
 	unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
 	unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
 	unsigned int own_eapd_ctl:1; /* set EAPD by own function */
+	unsigned int keep_eapd_on:1; /* don't turn off EAPD automatically */
 	unsigned int vmaster_mute_enum:1; /* add vmaster mute mode enum */
 	unsigned int indep_hp:1; /* independent HP supported */
 	unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index bd8d46cca2b..cccaf9c7a7b 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -58,6 +58,7 @@ enum {
 	CS420X_GPIO_23,
 	CS420X_MBP101,
 	CS420X_MBP81,
+	CS420X_MBA42,
 	CS420X_AUTO,
 	/* aliases */
 	CS420X_IMAC27_122 = CS420X_GPIO_23,
@@ -346,6 +347,7 @@ static const struct hda_model_fixup cs420x_models[] = {
 	{ .id = CS420X_APPLE, .name = "apple" },
 	{ .id = CS420X_MBP101, .name = "mbp101" },
 	{ .id = CS420X_MBP81, .name = "mbp81" },
+	{ .id = CS420X_MBA42, .name = "mba42" },
 	{}
 };
 
@@ -361,6 +363,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
 	SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
 	SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
+	SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
 	SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
 	{} /* terminator */
 };
@@ -414,6 +417,20 @@ static const struct hda_pintbl mbp101_pincfgs[] = {
 	{} /* terminator */
 };
 
+static const struct hda_pintbl mba42_pincfgs[] = {
+	{ 0x09, 0x012b4030 }, /* HP */
+	{ 0x0a, 0x400000f0 },
+	{ 0x0b, 0x90100120 }, /* speaker */
+	{ 0x0c, 0x400000f0 },
+	{ 0x0d, 0x90a00110 }, /* mic */
+	{ 0x0e, 0x400000f0 },
+	{ 0x0f, 0x400000f0 },
+	{ 0x10, 0x400000f0 },
+	{ 0x12, 0x400000f0 },
+	{ 0x15, 0x400000f0 },
+	{} /* terminator */
+};
+
 static void cs420x_fixup_gpio_13(struct hda_codec *codec,
 				 const struct hda_fixup *fix, int action)
 {
@@ -482,6 +499,12 @@ static const struct hda_fixup cs420x_fixups[] = {
 		.chained = true,
 		.chain_id = CS420X_GPIO_13,
 	},
+	[CS420X_MBA42] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = mba42_pincfgs,
+		.chained = true,
+		.chain_id = CS420X_GPIO_13,
+	},
 };
 
 static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 59d2e91a9ab..403010c9e82 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3483,6 +3483,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -3493,6 +3494,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
 	SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -3530,6 +3535,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -3593,6 +3599,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
 	{.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
+	{.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
+	{.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"},
 	{}
 };
 
@@ -4272,6 +4280,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
 	{.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"},
 	{.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"},
 	{.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
 	{}
 };
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e0dadcf2030..e5245544eb5 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -136,6 +136,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
 		spec->codec_type = VT1708S;
 	spec->no_pin_power_ctl = 1;
 	spec->gen.indep_hp = 1;
+	spec->gen.keep_eapd_on = 1;
 	spec->gen.pcm_playback_hook = via_playback_pcm_hook;
 	return spec;
 }
@@ -231,9 +232,14 @@ static void vt1708_update_hp_work(struct hda_codec *codec)
 
 static void set_widgets_power_state(struct hda_codec *codec)
 {
+#if 0 /* FIXME: the assumed connections don't match always with the
+       * actual routes by the generic parser, so better to disable
+       * the control for safety.
+       */
 	struct via_spec *spec = codec->spec;
 	if (spec->set_widgets_power_state)
 		spec->set_widgets_power_state(codec);
+#endif
 }
 
 static void update_power_state(struct hda_codec *codec, hda_nid_t nid,
@@ -478,7 +484,9 @@ static int via_suspend(struct hda_codec *codec)
 		/* Fix pop noise on headphones */
 		int i;
 		for (i = 0; i < spec->gen.autocfg.hp_outs; i++)
-			snd_hda_set_pin_ctl(codec, spec->gen.autocfg.hp_pins[i], 0);
+			snd_hda_codec_write(codec, spec->gen.autocfg.hp_pins[i],
+					    0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+					    0x00);
 	}
 
 	return 0;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index d59abe1682c..748e82d4d25 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card,
 	if (rc)
 		goto error_out;
 
-	if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) {
+	rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30));
+	if (rc < 0) {
 		dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA");
 		goto error_out_enabled;
 	}
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec0..987f728718c 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
 	{ CS42L52_BEEP_VOL, 0x00 },	/* r1D Beep Volume off Time */
 	{ CS42L52_BEEP_TONE_CTL, 0x00 },	/* r1E Beep Tone Cfg. */
 	{ CS42L52_TONE_CTL, 0x00 },	/* r1F Tone Ctl */
-	{ CS42L52_MASTERA_VOL, 0x88 },	/* r20 Master A Volume */
+	{ CS42L52_MASTERA_VOL, 0x00 },	/* r20 Master A Volume */
 	{ CS42L52_MASTERB_VOL, 0x00 },	/* r21 Master B Volume */
 	{ CS42L52_HPA_VOL, 0x00 },	/* r22 Headphone A Volume */
 	{ CS42L52_HPB_VOL, 0x00 },	/* r23 Headphone B Volume */
@@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
 
 static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
 
+static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
+
 static const unsigned int limiter_tlv[] = {
 	TLV_DB_RANGE_HEAD(2),
 	0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -225,7 +227,7 @@ static const char * const mic_bias_level_text[] = {
 };
 
 static const struct soc_enum mic_bias_level_enum =
-	SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+	SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
 			ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
 
 static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -260,7 +262,7 @@ static const char * const hp_gain_num_text[] = {
 };
 
 static const struct soc_enum hp_gain_enum =
-	SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+	SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5,
 		ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
 
 static const char * const beep_pitch_text[] = {
@@ -413,7 +415,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 	SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
 
 	SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
-			      CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+			      CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
 			      CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 	SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
 			    CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
-				6, 0x7f, 0x19, hl_tlv),
+				0, 0x7f, 0x19, mix_tlv),
 	SOC_DOUBLE_R("PCM Mixer Switch",
 		     CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
 
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c05907..4277012c471 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
 #define CS42L52_PB_CTL1_INV_PCMA		(1 << 2)
 #define CS42L52_PB_CTL1_MSTB_MUTE		(1 << 1)
 #define CS42L52_PB_CTL1_MSTA_MUTE		(1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK		0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK		0x03
 #define CS42L52_PB_CTL1_MUTE			3
 #define CS42L52_PB_CTL1_UNMUTE			0
 
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ce0d36412c9..8d14a76c724 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
 
 	ret = request_threaded_irq(max98090->irq, NULL,
-		max98090_interrupt, IRQF_TRIGGER_FALLING,
+		max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
 		"max98090_interrupt", codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "request_irq failed: %d\n",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 65d09d60b7c..1514bf845e4 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 
 			break;
 		}
-
-		if (found)
-			snd_soc_dapm_sync(widget->dapm);
 	}
 
-	ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
-
 	mutex_unlock(&widget->codec->mutex);
+
+	if (found)
+		snd_soc_dapm_sync(widget->dapm);
+
+	ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index e895d3939ee..100fdadda56 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
 ARIZONA_DSP_WIDGETS(DSP1, "DSP1"),
 
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
-		       ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux),
+		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+		       &wm5102_aec_loopback_mux),
 
 SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
 		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e0477..88ad7db52dd 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
 		 NULL, 0),
 
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
-		       ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux),
+		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+		       &wm5110_aec_loopback_mux),
 
 SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
 		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
@@ -976,6 +977,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
+	arizona_init_spk(codec);
+
 	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
 	priv->core.arizona->dapm = &codec->dapm;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1eb152cb109..29e95f93d48 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int drc = wm8994_get_drc(kcontrol->id.name);
 
+	if (drc < 0)
+		return drc;
 	ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
 
 	return 0;
@@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 
+	if (block < 0)
+		return block;
+
 	ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
 
 	return 0;
@@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_codec *codec = w->codec;
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	struct wm8994 *control = codec->control_data;
+	struct wm8994 *control = wm8994->wm8994;
 	int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
 	int i;
 	int dac;
@@ -3831,8 +3836,14 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 				ret);
 		} else if (!(ret & WM1811_JACKDET_LVL)) {
 			dev_dbg(codec->dev, "Ignoring removed jack\n");
-			return IRQ_HANDLED;
+			goto out;
 		}
+	} else if (!(reg & WM8958_MICD_STS)) {
+		snd_soc_jack_report(wm8994->micdet[0].jack, 0,
+				    SND_JACK_MECHANICAL | SND_JACK_HEADSET |
+				    wm8994->btn_mask);
+		wm8994->mic_detecting = true;
+		goto out;
 	}
 
 	if (wm8994->mic_detecting)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e..81490febac6 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 				       int word_length)
 {
 	u32 fmt;
-	u32 rotate = (word_length / 4) & 0x7;
+	u32 tx_rotate = (word_length / 4) & 0x7;
+	u32 rx_rotate = (32 - word_length) / 4;
 	u32 mask = (1ULL << word_length) - 1;
 
 	/*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
 				TXSSZ(fmt), TXSSZ(0x0F));
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
-				TXROT(rotate), TXROT(7));
+				TXROT(tx_rotate), TXROT(7));
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
-				RXROT(rotate), RXROT(7));
+				RXROT(rx_rotate), RXROT(7));
 		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
 				mask);
 	}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f2..06a8000aa07 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
 			goto err;
 	}
 
-	snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
-				SND_SOC_DAPM_STREAM_START);
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+					SND_SOC_DAPM_STREAM_START);
+	else
+		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+					SND_SOC_DAPM_STREAM_START);
 
 	/* cancel any delayed stream shutdown that is pending */
 	rtd->pop_wait = 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a80c883bb8b..c7051c457b7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -55,7 +55,8 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_clock_supply] = 1,
 	[snd_soc_dapm_micbias] = 2,
 	[snd_soc_dapm_dai_link] = 2,
-	[snd_soc_dapm_dai] = 3,
+	[snd_soc_dapm_dai_in] = 3,
+	[snd_soc_dapm_dai_out] = 3,
 	[snd_soc_dapm_aif_in] = 3,
 	[snd_soc_dapm_aif_out] = 3,
 	[snd_soc_dapm_mic] = 4,
@@ -92,7 +93,8 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_value_mux] = 9,
 	[snd_soc_dapm_aif_in] = 10,
 	[snd_soc_dapm_aif_out] = 10,
-	[snd_soc_dapm_dai] = 10,
+	[snd_soc_dapm_dai_in] = 10,
+	[snd_soc_dapm_dai_out] = 10,
 	[snd_soc_dapm_dai_link] = 11,
 	[snd_soc_dapm_clock_supply] = 12,
 	[snd_soc_dapm_regulator_supply] = 12,
@@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_in:
+	case snd_soc_dapm_dai_out:
 	case snd_soc_dapm_hp:
 	case snd_soc_dapm_mic:
 	case snd_soc_dapm_spk:
@@ -820,7 +823,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	switch (widget->id) {
 	case snd_soc_dapm_adc:
 	case snd_soc_dapm_aif_out:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_out:
 		if (widget->active) {
 			widget->outputs = snd_soc_dapm_suspend_check(widget);
 			return widget->outputs;
@@ -916,7 +919,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 	switch (widget->id) {
 	case snd_soc_dapm_dac:
 	case snd_soc_dapm_aif_in:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_in:
 		if (widget->active) {
 			widget->inputs = snd_soc_dapm_suspend_check(widget);
 			return widget->inputs;
@@ -1135,16 +1138,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
 	return out != 0 && in != 0;
 }
 
-static int dapm_dai_check_power(struct snd_soc_dapm_widget *w)
-{
-	DAPM_UPDATE_STAT(w, power_checks);
-
-	if (w->active)
-		return w->active;
-
-	return dapm_generic_check_power(w);
-}
-
 /* Check to see if an ADC has power */
 static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
 {
@@ -2318,7 +2311,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_in:
+	case snd_soc_dapm_dai_out:
 	case snd_soc_dapm_dai_link:
 		list_add(&path->list, &dapm->card->paths);
 		list_add(&path->list_sink, &wsink->sources);
@@ -3129,10 +3123,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		break;
 	case snd_soc_dapm_adc:
 	case snd_soc_dapm_aif_out:
+	case snd_soc_dapm_dai_out:
 		w->power_check = dapm_adc_check_power;
 		break;
 	case snd_soc_dapm_dac:
 	case snd_soc_dapm_aif_in:
+	case snd_soc_dapm_dai_in:
 		w->power_check = dapm_dac_check_power;
 		break;
 	case snd_soc_dapm_pga:
@@ -3152,9 +3148,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_clock_supply:
 		w->power_check = dapm_supply_check_power;
 		break;
-	case snd_soc_dapm_dai:
-		w->power_check = dapm_dai_check_power;
-		break;
 	default:
 		w->power_check = dapm_always_on_check_power;
 		break;
@@ -3375,7 +3368,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 	template.reg = SND_SOC_NOPM;
 
 	if (dai->driver->playback.stream_name) {
-		template.id = snd_soc_dapm_dai;
+		template.id = snd_soc_dapm_dai_in;
 		template.name = dai->driver->playback.stream_name;
 		template.sname = dai->driver->playback.stream_name;
 
@@ -3393,7 +3386,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 	}
 
 	if (dai->driver->capture.stream_name) {
-		template.id = snd_soc_dapm_dai;
+		template.id = snd_soc_dapm_dai_out;
 		template.name = dai->driver->capture.stream_name;
 		template.sname = dai->driver->capture.stream_name;
 
@@ -3423,8 +3416,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 
 	/* For each DAI widget... */
 	list_for_each_entry(dai_w, &card->widgets, list) {
-		if (dai_w->id != snd_soc_dapm_dai)
+		switch (dai_w->id) {
+		case snd_soc_dapm_dai_in:
+		case snd_soc_dapm_dai_out:
+			break;
+		default:
 			continue;
+		}
 
 		dai = dai_w->priv;
 
@@ -3433,8 +3431,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 			if (w->dapm != dai_w->dapm)
 				continue;
 
-			if (w->id == snd_soc_dapm_dai)
+			switch (w->id) {
+			case snd_soc_dapm_dai_in:
+			case snd_soc_dapm_dai_out:
 				continue;
+			default:
+				break;
+			}
 
 			if (!w->sname)
 				continue;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 73bb8eefa49..ccb6be4d658 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
 	/* Create any new FE <--> BE connections */
 	for (i = 0; i < list->num_widgets; i++) {
 
-		if (list->widgets[i]->id != snd_soc_dapm_dai)
+		switch (list->widgets[i]->id) {
+		case snd_soc_dapm_dai_in:
+		case snd_soc_dapm_dai_out:
+			break;
+		default:
 			continue;
+		}
 
 		/* is there a valid BE rtd for this widget */
 		be = dpcm_get_be(card, list->widgets[i], stream);
@@ -2011,9 +2016,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		if (cpu_dai->driver->capture.channels_min)
 			capture = 1;
 	} else {
-		if (codec_dai->driver->playback.channels_min)
+		if (codec_dai->driver->playback.channels_min &&
+		    cpu_dai->driver->playback.channels_min)
 			playback = 1;
-		if (codec_dai->driver->capture.channels_min)
+		if (codec_dai->driver->capture.channels_min &&
+		    cpu_dai->driver->capture.channels_min)
 			capture = 1;
 	}
 
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a1d9b0792a1..b9defcdeb7e 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = {
 	0x94, 0x01, 0x5c, 0x02  /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */
 };
 
-static const u8 known_fw_versions[][4] = {
-	{ 0x03, 0x01, 0x0b, 0x00 }
+static const u8 known_fw_versions[][2] = {
+	{ 0x03, 0x01 }
 };
 
 struct ihex_record {
@@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version)
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++)
-		if (!memcmp(version, known_fw_versions + i, 4))
+		if (!memcmp(version, known_fw_versions + i, 2))
 			return 0;
 
 	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 1a033177b83..64952e2d3ed 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -147,14 +147,32 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int
 		return -EINVAL;
 	}
 
+	alts = &iface->altsetting[0];
+	altsd = get_iface_desc(alts);
+
+	/*
+	 * Android with both accessory and audio interfaces enabled gets the
+	 * interface numbers wrong.
+	 */
+	if ((chip->usb_id == USB_ID(0x18d1, 0x2d04) ||
+	     chip->usb_id == USB_ID(0x18d1, 0x2d05)) &&
+	    interface == 0 &&
+	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+	    altsd->bInterfaceSubClass == USB_SUBCLASS_VENDOR_SPEC) {
+		interface = 2;
+		iface = usb_ifnum_to_if(dev, interface);
+		if (!iface)
+			return -EINVAL;
+		alts = &iface->altsetting[0];
+		altsd = get_iface_desc(alts);
+	}
+
 	if (usb_interface_claimed(iface)) {
 		snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n",
 						dev->devnum, ctrlif, interface);
 		return -EINVAL;
 	}
 
-	alts = &iface->altsetting[0];
-	altsd = get_iface_desc(alts);
 	if ((altsd->bInterfaceClass == USB_CLASS_AUDIO ||
 	     altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) &&
 	    altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) {
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ca4739c3f65..d5438083fd6 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -885,7 +885,9 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
 
 	case USB_ID(0x046d, 0x0808):
 	case USB_ID(0x046d, 0x0809):
+	case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */
 	case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
+	case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */
 	case USB_ID(0x046d, 0x0991):
 	/* Most audio usb devices lie about volume resolution.
 	 * Most Logitech webcams have res = 384.
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 7f1722f82c8..8b75bcf136f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -215,7 +215,13 @@
 	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
 },
 {
-	USB_DEVICE(0x046d, 0x0990),
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+		       USB_DEVICE_ID_MATCH_INT_CLASS |
+		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+	.idVendor = 0x046d,
+	.idProduct = 0x0990,
+	.bInterfaceClass = USB_CLASS_AUDIO,
+	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		.vendor_name = "Logitech, Inc.",
 		.product_name = "QuickCam Pro 9000",
@@ -1792,7 +1798,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		.ifnum = 0,
-		.type = QUIRK_MIDI_STANDARD_INTERFACE
+		.type = QUIRK_MIDI_FIXED_ENDPOINT,
+		.data = & (const struct snd_usb_midi_endpoint_info) {
+			.out_cables = 0x0007,
+			.in_cables  = 0x0007
+		}
 	}
 },
 {