Merge tag 'v4.1.11' into toradex_vf_4.1-next

This is the 4.1.11 stable release

30 files changed, 193 insertions, 110 deletions

diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 885683a3b0bd..e0406211716b 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -9,6 +9,14 @@ menuconfig SND_ARM
 	  Drivers that are implemented on ASoC can be found in
 	  "ALSA for SoC audio support" section.
 
+config SND_PXA2XX_LIB
+	tristate
+	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
+	select SND_DMAENGINE_PCM
+
+config SND_PXA2XX_LIB_AC97
+	bool
+
 if SND_ARM
 
 config SND_ARMAACI
@@ -21,13 +29,6 @@ config SND_PXA2XX_PCM
 	tristate
 	select SND_PCM
 
-config SND_PXA2XX_LIB
-	tristate
-	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
-
-config SND_PXA2XX_LIB_AC97
-	bool
-
 config SND_PXA2XX_AC97
 	tristate "AC97 driver for the Intel PXA2xx chip"
 	depends on ARCH_PXA
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index e061355f535f..bf20593d3085 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -730,8 +730,9 @@ static void handle_in_packet(struct amdtp_stream *s,
 	    s->data_block_counter != UINT_MAX)
 		data_block_counter = s->data_block_counter;
 
-	if (((s->flags & CIP_SKIP_DBC_ZERO_CHECK) && data_block_counter == 0) ||
-	    (s->data_block_counter == UINT_MAX)) {
+	if (((s->flags & CIP_SKIP_DBC_ZERO_CHECK) &&
+	     data_block_counter == s->tx_first_dbc) ||
+	    s->data_block_counter == UINT_MAX) {
 		lost = false;
 	} else if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
 		lost = data_block_counter != s->data_block_counter;
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index 8a03a91e728b..25c905537658 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -153,6 +153,8 @@ struct amdtp_stream {
 
 	/* quirk: fixed interval of dbc between previos/current packets. */
 	unsigned int tx_dbc_interval;
+	/* quirk: indicate the value of dbc field in a first packet. */
+	unsigned int tx_first_dbc;
 
 	bool callbacked;
 	wait_queue_head_t callback_wait;
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 2682e7e3e5c9..c94a432f7cc6 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -248,8 +248,16 @@ efw_probe(struct fw_unit *unit,
 	err = get_hardware_info(efw);
 	if (err < 0)
 		goto error;
+	/* AudioFire8 (since 2009) and AudioFirePre8 */
 	if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9)
 		efw->is_af9 = true;
+	/* These models uses the same firmware. */
+	if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2 ||
+	    entry->model_id == MODEL_ECHO_AUDIOFIRE_4 ||
+	    entry->model_id == MODEL_ECHO_AUDIOFIRE_9 ||
+	    entry->model_id == MODEL_GIBSON_RIP ||
+	    entry->model_id == MODEL_GIBSON_GOLDTOP)
+		efw->is_fireworks3 = true;
 
 	snd_efw_proc_init(efw);
 
diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 4f0201a95222..084d414b228c 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -71,6 +71,7 @@ struct snd_efw {
 
 	/* for quirks */
 	bool is_af9;
+	bool is_fireworks3;
 	u32 firmware_version;
 
 	unsigned int midi_in_ports;
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index c55db1bddc80..7e353f1f7bff 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -172,6 +172,15 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw)
 	efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT;
 	/* Fireworks reset dbc at bus reset. */
 	efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK;
+	/*
+	 * But Recent firmwares starts packets with non-zero dbc.
+	 * Driver version 5.7.6 installs firmware version 5.7.3.
+	 */
+	if (efw->is_fireworks3 &&
+	    (efw->firmware_version == 0x5070000 ||
+	     efw->firmware_version == 0x5070300 ||
+	     efw->firmware_version == 0x5080000))
+		efw->tx_stream.tx_first_dbc = 0x02;
 	/* AudioFire9 always reports wrong dbs. */
 	if (efw->is_af9)
 		efw->tx_stream.flags |= CIP_WRONG_DBS;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5645481af3d9..36e8f1236637 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3259,7 +3259,7 @@ static int add_std_chmaps(struct hda_codec *codec)
 			struct snd_pcm_chmap *chmap;
 			const struct snd_pcm_chmap_elem *elem;
 
-			if (!pcm || pcm->own_chmap ||
+			if (!pcm || !pcm->pcm || pcm->own_chmap ||
 			    !hinfo->substreams)
 				continue;
 			elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ac0db1679f09..5bc7f2e2715c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -671,7 +671,8 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid,
 		}
 		for (i = 0; i < path->depth; i++) {
 			if (path->path[i] == nid) {
-				if (dir == HDA_OUTPUT || path->idx[i] == idx)
+				if (dir == HDA_OUTPUT || idx == -1 ||
+				    path->idx[i] == idx)
 					return true;
 				break;
 			}
@@ -682,7 +683,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid,
 
 /* check whether the NID is referred by any active paths */
 #define is_active_nid_for_any(codec, nid) \
-	is_active_nid(codec, nid, HDA_OUTPUT, 0)
+	is_active_nid(codec, nid, HDA_OUTPUT, -1)
 
 /* get the default amp value for the target state */
 static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
@@ -883,8 +884,7 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path,
 	struct hda_gen_spec *spec = codec->spec;
 	int i;
 
-	if (!enable)
-		path->active = false;
+	path->active = enable;
 
 	/* make sure the widget is powered up */
 	if (enable && (spec->power_down_unused || codec->power_save_node))
@@ -902,9 +902,6 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path,
 		if (has_amp_out(codec, path, i))
 			activate_amp_out(codec, path, i, enable);
 	}
-
-	if (enable)
-		path->active = true;
 }
 EXPORT_SYMBOL_GPL(snd_hda_activate_path);
 
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 50e9dd675579..b791529bf31c 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -634,6 +634,7 @@ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11),
 	SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6),
 	SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6),
+	SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11),
 	{} /* terminator */
 };
 
@@ -1001,9 +1002,7 @@ static void cs4210_spdif_automute(struct hda_codec *codec,
 
 	spec->spdif_present = spdif_present;
 	/* SPDIF TX on/off */
-	if (spdif_present)
-		snd_hda_set_pin_ctl(codec, spdif_pin,
-				    spdif_present ? PIN_OUT : 0);
+	snd_hda_set_pin_ctl(codec, spdif_pin, spdif_present ? PIN_OUT : 0);
 
 	cs_automute(codec);
 }
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 78b719b5b34d..06cc9d57ba3d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -200,12 +200,33 @@ static int cx_auto_init(struct hda_codec *codec)
 	return 0;
 }
 
-#define cx_auto_free	snd_hda_gen_free
+static void cx_auto_reboot_notify(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	if (codec->core.vendor_id != 0x14f150f2)
+		return;
+
+	/* Turn the CX20722 codec into D3 to avoid spurious noises
+	   from the internal speaker during (and after) reboot */
+	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
+
+	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+}
+
+static void cx_auto_free(struct hda_codec *codec)
+{
+	cx_auto_reboot_notify(codec);
+	snd_hda_gen_free(codec);
+}
 
 static const struct hda_codec_ops cx_auto_patch_ops = {
 	.build_controls = cx_auto_build_controls,
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = cx_auto_init,
+	.reboot_notify = cx_auto_reboot_notify,
 	.free = cx_auto_free,
 	.unsol_event = snd_hda_jack_unsol_event,
 #ifdef CONFIG_PM
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 590bcfb0e82f..57bb5a559f8e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1134,7 +1134,7 @@ static const struct hda_fixup alc880_fixups[] = {
 		/* override all pins as BIOS on old Amilo is broken */
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
-			{ 0x14, 0x0121411f }, /* HP */
+			{ 0x14, 0x0121401f }, /* HP */
 			{ 0x15, 0x99030120 }, /* speaker */
 			{ 0x16, 0x99030130 }, /* bass speaker */
 			{ 0x17, 0x411111f0 }, /* N/A */
@@ -1154,7 +1154,7 @@ static const struct hda_fixup alc880_fixups[] = {
 		/* almost compatible with FUJITSU, but no bass and SPDIF */
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
-			{ 0x14, 0x0121411f }, /* HP */
+			{ 0x14, 0x0121401f }, /* HP */
 			{ 0x15, 0x99030120 }, /* speaker */
 			{ 0x16, 0x411111f0 }, /* N/A */
 			{ 0x17, 0x411111f0 }, /* N/A */
@@ -1363,7 +1363,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810),
 	SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM),
 	SND_PCI_QUIRK(0x1631, 0xe011, "PB 13201056", ALC880_FIXUP_6ST_AUTOMUTE),
-	SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FIXUP_F1734),
+	SND_PCI_QUIRK(0x1734, 0x107c, "FSC Amilo M1437", ALC880_FIXUP_FUJITSU),
 	SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU),
 	SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_FIXUP_F1734),
 	SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU),
@@ -4182,6 +4182,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
 	}
 }
 
+/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */
+static void alc_fixup_tpt440_dock(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	static const struct hda_pintbl pincfgs[] = {
+		{ 0x16, 0x21211010 }, /* dock headphone */
+		{ 0x19, 0x21a11010 }, /* dock mic */
+		{ }
+	};
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
+		codec->power_save_node = 0; /* avoid click noises */
+		snd_hda_apply_pincfgs(codec, pincfgs);
+	}
+}
+
 static void alc_shutup_dell_xps13(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4507,7 +4525,6 @@ enum {
 	ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
 	ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC292_FIXUP_TPT440_DOCK,
-	ALC292_FIXUP_TPT440_DOCK2,
 	ALC283_FIXUP_BXBT2807_MIC,
 	ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
 	ALC282_FIXUP_ASPIRE_V5_PINS,
@@ -4972,17 +4989,7 @@ static const struct hda_fixup alc269_fixups[] = {
 	},
 	[ALC292_FIXUP_TPT440_DOCK] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
-		.chained = true,
-		.chain_id = ALC292_FIXUP_TPT440_DOCK2
-	},
-	[ALC292_FIXUP_TPT440_DOCK2] = {
-		.type = HDA_FIXUP_PINS,
-		.v.pins = (const struct hda_pintbl[]) {
-			{ 0x16, 0x21211010 }, /* dock headphone */
-			{ 0x19, 0x21a11010 }, /* dock mic */
-			{ }
-		},
+		.v.func = alc_fixup_tpt440_dock,
 		.chained = true,
 		.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
 	},
@@ -5118,6 +5125,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+	SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+	SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+	SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+	SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
 	SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5221,6 +5233,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
@@ -6452,6 +6465,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05fe, "Dell XPS 15", ALC668_FIXUP_DELL_XPS13),
 	SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_XPS13),
+	SND_PCI_QUIRK(0x1028, 0x060d, "Dell M3800", ALC668_FIXUP_DELL_XPS13),
 	SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0696, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 25f0f45e6640..b1bc66783974 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4522,7 +4522,11 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 		return err;
 
 	spec = codec->spec;
-	codec->power_save_node = 1;
+	/* enable power_save_node only for new 92HD89xx chips, as it causes
+	 * click noises on old 92HD73xx chips.
+	 */
+	if ((codec->core.vendor_id & 0xfffffff0) != 0x111d7670)
+		codec->power_save_node = 1;
 	spec->linear_tone_beep = 0;
 	spec->gen.mixer_nid = 0x1d;
 	spec->have_spdif_mux = 1;
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index c75995f2779c..b914a08258ea 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -129,6 +129,8 @@ static struct snd_soc_dai_link db1300_i2s_dai = {
 	.cpu_dai_name	= "au1xpsc_i2s.2",
 	.platform_name	= "au1xpsc-pcm.2",
 	.codec_name	= "wm8731.0-001b",
+	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &db1200_i2s_wm8731_ops,
 };
 
@@ -146,6 +148,8 @@ static struct snd_soc_dai_link db1550_i2s_dai = {
 	.cpu_dai_name	= "au1xpsc_i2s.3",
 	.platform_name	= "au1xpsc-pcm.3",
 	.codec_name	= "wm8731.0-001b",
+	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &db1200_i2s_wm8731_ops,
 };
 
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 4373ada95648..3a91a00fb973 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -864,7 +864,6 @@ const struct regmap_config adav80x_regmap_config = {
 	.val_bits = 8,
 	.pad_bits = 1,
 	.reg_bits = 7,
-	.read_flag_mask = 0x01,
 
 	.max_register = ADAV80X_PLL_OUTE,
 
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index eff4b4d512b7..ee91edcf3cb0 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1610,17 +1610,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id)
 }
 EXPORT_SYMBOL_GPL(arizona_init_dai);
 
-static irqreturn_t arizona_fll_clock_ok(int irq, void *data)
-{
-	struct arizona_fll *fll = data;
-
-	arizona_fll_dbg(fll, "clock OK\n");
-
-	complete(&fll->ok);
-
-	return IRQ_HANDLED;
-}
-
 static struct {
 	unsigned int min;
 	unsigned int max;
@@ -1902,17 +1891,18 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll)
 static int arizona_enable_fll(struct arizona_fll *fll)
 {
 	struct arizona *arizona = fll->arizona;
-	unsigned long time_left;
 	bool use_sync = false;
 	int already_enabled = arizona_is_enabled_fll(fll);
 	struct arizona_fll_cfg cfg;
+	int i;
+	unsigned int val;
 
 	if (already_enabled < 0)
 		return already_enabled;
 
 	if (already_enabled) {
 		/* Facilitate smooth refclk across the transition */
-		regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7,
+		regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9,
 					 ARIZONA_FLL1_GAIN_MASK, 0);
 		regmap_update_bits_async(fll->arizona->regmap, fll->base + 1,
 					 ARIZONA_FLL1_FREERUN,
@@ -1964,9 +1954,6 @@ static int arizona_enable_fll(struct arizona_fll *fll)
 	if (!already_enabled)
 		pm_runtime_get(arizona->dev);
 
-	/* Clear any pending completions */
-	try_wait_for_completion(&fll->ok);
-
 	regmap_update_bits_async(arizona->regmap, fll->base + 1,
 				 ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
 	if (use_sync)
@@ -1978,10 +1965,24 @@ static int arizona_enable_fll(struct arizona_fll *fll)
 		regmap_update_bits_async(arizona->regmap, fll->base + 1,
 					 ARIZONA_FLL1_FREERUN, 0);
 
-	time_left = wait_for_completion_timeout(&fll->ok,
-					  msecs_to_jiffies(250));
-	if (time_left == 0)
+	arizona_fll_dbg(fll, "Waiting for FLL lock...\n");
+	val = 0;
+	for (i = 0; i < 15; i++) {
+		if (i < 5)
+			usleep_range(200, 400);
+		else
+			msleep(20);
+
+		regmap_read(arizona->regmap,
+			    ARIZONA_INTERRUPT_RAW_STATUS_5,
+			    &val);
+		if (val & (ARIZONA_FLL1_CLOCK_OK_STS << (fll->id - 1)))
+			break;
+	}
+	if (i == 15)
 		arizona_fll_warn(fll, "Timed out waiting for lock\n");
+	else
+		arizona_fll_dbg(fll, "FLL locked (%d polls)\n", i);
 
 	return 0;
 }
@@ -2066,11 +2067,8 @@ EXPORT_SYMBOL_GPL(arizona_set_fll);
 int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
 		     int ok_irq, struct arizona_fll *fll)
 {
-	int ret;
 	unsigned int val;
 
-	init_completion(&fll->ok);
-
 	fll->id = id;
 	fll->base = base;
 	fll->arizona = arizona;
@@ -2092,13 +2090,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
 	snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name),
 		 "FLL%d clock OK", id);
 
-	ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name,
-				  arizona_fll_clock_ok, fll);
-	if (ret != 0) {
-		dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n",
-			id, ret);
-	}
-
 	regmap_update_bits(arizona->regmap, fll->base + 1,
 			   ARIZONA_FLL1_FREERUN, 0);
 
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 11ff899b0272..14e8485b5585 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -233,7 +233,6 @@ struct arizona_fll {
 	int id;
 	unsigned int base;
 	unsigned int vco_mult;
-	struct completion ok;
 
 	unsigned int fout;
 	int sync_src;
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 477e13d30971..e7ba557979cb 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -102,7 +102,7 @@ static int pcm1681_set_deemph(struct snd_soc_codec *codec)
 
 	if (val != -1) {
 		regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL,
-					PCM1681_DEEMPH_RATE_MASK, val);
+				   PCM1681_DEEMPH_RATE_MASK, val << 3);
 		enable = 1;
 	} else
 		enable = 0;
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 178e55d4d481..06317f7d945f 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -985,6 +985,35 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rt5640_lout_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		hp_amp_power_on(codec);
+		snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
+			RT5640_PWR_LM, RT5640_PWR_LM);
+		snd_soc_update_bits(codec, RT5640_OUTPUT,
+			RT5640_L_MUTE | RT5640_R_MUTE, 0);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(codec, RT5640_OUTPUT,
+			RT5640_L_MUTE | RT5640_R_MUTE,
+			RT5640_L_MUTE | RT5640_R_MUTE);
+		snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
+			RT5640_PWR_LM, 0);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
 			   struct snd_kcontrol *kcontrol, int event)
 {
@@ -1180,13 +1209,16 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 		0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)),
 	SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0,
 		0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)),
-	SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0,
+	SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0,
 		rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)),
 	SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM,
 		0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
 		rt5640_hp_event,
 		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0,
+		rt5640_lout_event,
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_L_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1,
@@ -1501,8 +1533,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HP R Playback", "Switch", "HP Amp"},
 	{"HPOL", NULL, "HP L Playback"},
 	{"HPOR", NULL, "HP R Playback"},
-	{"LOUTL", NULL, "LOUT MIX"},
-	{"LOUTR", NULL, "LOUT MIX"},
+
+	{"LOUT amp", NULL, "LOUT MIX"},
+	{"LOUTL", NULL, "LOUT amp"},
+	{"LOUTR", NULL, "LOUT amp"},
 };
 
 static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = {
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 3593a1496056..3a29c0ac5d8a 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1339,8 +1339,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 			sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
 
 	snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
-			SGTL5000_BIAS_R_MASK,
-			sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
+			SGTL5000_BIAS_VOLT_MASK,
+			sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT);
 	/*
 	 * disable DAP
 	 * TODO:
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index a984485108cd..f7549cc7ea85 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -315,7 +315,13 @@ static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	if (invert_fclk)
 		ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC;
 
-	return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1);
+	return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1,
+			SSM4567_SAI_CTRL_1_BCLK |
+			SSM4567_SAI_CTRL_1_FSYNC |
+			SSM4567_SAI_CTRL_1_LJ |
+			SSM4567_SAI_CTRL_1_TDM |
+			SSM4567_SAI_CTRL_1_PDM,
+			ctrl1);
 }
 
 static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable)
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index a3e97b46b64e..0d28e3b356f6 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -131,10 +131,10 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		for (i = 0; i < 4; i++)
-			i2s_write_reg(dev->i2s_base, TOR(i), 0);
+			i2s_read_reg(dev->i2s_base, TOR(i));
 	} else {
 		for (i = 0; i < 4; i++)
-			i2s_write_reg(dev->i2s_base, ROR(i), 0);
+			i2s_read_reg(dev->i2s_base, ROR(i));
 	}
 }
 
diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c
index 7b50a9d17ec1..edc186908358 100644
--- a/sound/soc/intel/atom/sst/sst_drv_interface.c
+++ b/sound/soc/intel/atom/sst/sst_drv_interface.c
@@ -42,6 +42,11 @@
 #define MIN_FRAGMENT_SIZE (50 * 1024)
 #define MAX_FRAGMENT_SIZE (1024 * 1024)
 #define SST_GET_BYTES_PER_SAMPLE(pcm_wd_sz)  (((pcm_wd_sz + 15) >> 4) << 1)
+#ifdef CONFIG_PM
+#define GET_USAGE_COUNT(dev) (atomic_read(&dev->power.usage_count))
+#else
+#define GET_USAGE_COUNT(dev) 1
+#endif
 
 int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id)
 {
@@ -141,15 +146,9 @@ static int sst_power_control(struct device *dev, bool state)
 	int ret = 0;
 	int usage_count = 0;
 
-#ifdef CONFIG_PM
-	usage_count = atomic_read(&dev->power.usage_count);
-#else
-	usage_count = 1;
-#endif
-
 	if (state == true) {
 		ret = pm_runtime_get_sync(dev);
-
+		usage_count = GET_USAGE_COUNT(dev);
 		dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count);
 		if (ret < 0) {
 			dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret);
@@ -164,6 +163,7 @@ static int sst_power_control(struct device *dev, bool state)
 			}
 		}
 	} else {
+		usage_count = GET_USAGE_COUNT(dev);
 		dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count);
 		return sst_pm_runtime_put(ctx);
 	}
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 39cea80846c3..f2bf8661dd21 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,7 +1,6 @@
 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
 	depends on ARCH_PXA
-	select SND_ARM
 	select SND_PXA2XX_LIB
 	help
 	  Say Y or M if you want to add support for codecs attached to
@@ -25,7 +24,6 @@ config SND_PXA2XX_AC97
 config SND_PXA2XX_SOC_AC97
 	tristate
 	select AC97_BUS
-	select SND_ARM
 	select SND_PXA2XX_LIB_AC97
 	select SND_SOC_AC97_BUS
 
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1f6054650991..9e4b04e0fbd1 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_cold_reset,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c
index 8bf2e2c4bafb..9e371eb3e4fa 100644
--- a/sound/soc/samsung/arndale_rt5631.c
+++ b/sound/soc/samsung/arndale_rt5631.c
@@ -116,15 +116,6 @@ static int arndale_audio_probe(struct platform_device *pdev)
 	return ret;
 }
 
-static int arndale_audio_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-
-	return 0;
-}
-
 static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = {
 	{ .compatible = "samsung,arndale-rt5631", },
 	{ .compatible = "samsung,arndale-alc5631", },
@@ -139,7 +130,6 @@ static struct platform_driver arndale_audio_driver = {
 		.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
 	},
 	.probe = arndale_audio_probe,
-	.remove = arndale_audio_remove,
 };
 
 module_platform_driver(arndale_audio_driver);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 158204d08924..b6c12dccb259 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1811,6 +1811,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 					   size_t count, loff_t *ppos)
 {
 	struct snd_soc_dapm_widget *w = file->private_data;
+	struct snd_soc_card *card = w->dapm->card;
 	char *buf;
 	int in, out;
 	ssize_t ret;
@@ -1820,6 +1821,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 	if (!buf)
 		return -ENOMEM;
 
+	mutex_lock(&card->dapm_mutex);
+
 	/* Supply widgets are not handled by is_connected_{input,output}_ep() */
 	if (w->is_supply) {
 		in = 0;
@@ -1866,6 +1869,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 					p->sink->name);
 	}
 
+	mutex_unlock(&card->dapm_mutex);
+
 	ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 
 	kfree(buf);
@@ -2140,11 +2145,15 @@ static ssize_t dapm_widget_show(struct device *dev,
 	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 	int i, count = 0;
 
+	mutex_lock(&rtd->card->dapm_mutex);
+
 	for (i = 0; i < rtd->num_codecs; i++) {
 		struct snd_soc_codec *codec = rtd->codec_dais[i]->codec;
 		count += dapm_widget_show_codec(codec, buf + count);
 	}
 
+	mutex_unlock(&rtd->card->dapm_mutex);
+
 	return count;
 }
 
@@ -3100,16 +3109,10 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	}
 
 	prefix = soc_dapm_prefix(dapm);
-	if (prefix) {
+	if (prefix)
 		w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name);
-		if (widget->sname)
-			w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix,
-					     widget->sname);
-	} else {
+	else
 		w->name = kasprintf(GFP_KERNEL, "%s", widget->name);
-		if (widget->sname)
-			w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname);
-	}
 	if (w->name == NULL) {
 		kfree(w);
 		return NULL;
@@ -3557,7 +3560,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 				break;
 			}
 
-			if (!w->sname || !strstr(w->sname, dai_w->name))
+			if (!w->sname || !strstr(w->sname, dai_w->sname))
 				continue;
 
 			if (dai_w->id == snd_soc_dapm_dai_in) {
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
index 82e350e9501c..ac75816ada7c 100644
--- a/sound/synth/emux/emux_oss.c
+++ b/sound/synth/emux/emux_oss.c
@@ -69,7 +69,8 @@ snd_emux_init_seq_oss(struct snd_emux *emu)
 	struct snd_seq_oss_reg *arg;
 	struct snd_seq_device *dev;
 
-	if (snd_seq_device_new(emu->card, 0, SNDRV_SEQ_DEV_ID_OSS,
+	/* using device#1 here for avoiding conflicts with OPL3 */
+	if (snd_seq_device_new(emu->card, 1, SNDRV_SEQ_DEV_ID_OSS,
 			       sizeof(struct snd_seq_oss_reg), &dev) < 0)
 		return;
 
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 1fab9778807a..0450593980fd 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -638,7 +638,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip)
 	int err = -ENODEV;
 
 	down_read(&chip->shutdown_rwsem);
-	if (chip->probing && chip->in_pm)
+	if (chip->probing || chip->in_pm)
 		err = 0;
 	else if (!chip->shutdown)
 		err = usb_autopm_get_interface(chip->pm_intf);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 8b7e391dd0b8..cd8ed2e393a2 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2522,7 +2522,7 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list)
 		for (c = 0; c < MAX_CHANNELS; c++) {
 			if (!(cval->cmask & (1 << c)))
 				continue;
-			if (cval->cached & (1 << c)) {
+			if (cval->cached & (1 << (c + 1))) {
 				err = snd_usb_set_cur_mix_value(cval, c + 1, idx,
 							cval->cache_val[idx]);
 				if (err < 0)
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 754e689596a2..00ebc0ca008e 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1268,6 +1268,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;
 
+	case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */
 	case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */
 	case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */
 		if (fp->altsetting == 3)