Merge tag 'nfs-rdma-for-4.1-1' of git://git.linux-nfs.org/projects/anna/nfs-rdma

NFS: NFSoRDMA Client Changes

This patch series creates an operation vector for each of the different
memory registration modes.  This should make it easier to one day increase
credit limit, rsize, and wsize.

Signed-off-by: Anna Schumaker <Anna.Schumaker@Netapp.com>

37 files changed, 253 insertions, 127 deletions

diff --git a/sound/core/control.c b/sound/core/control.c
index 35324a8e83c8..eeb691d1911f 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 
 	if (info->count < 1)
 		return -EINVAL;
+	if (!*info->id.name)
+		return -EINVAL;
+	if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name))
+		return -EINVAL;
 	access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
 		(info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
 				 SNDRV_CTL_ELEM_ACCESS_INACTIVE|
diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index de7602bd69b5..27b044f84c81 100644
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -299,23 +299,23 @@
 #define RX_ISOCHRONOUS			0x008
 
 /*
+ * Index of first quadlet to be interpreted; read/write.  If > 0, that many
+ * quadlets at the beginning of each data block will be ignored, and all the
+ * audio and MIDI quadlets will follow.
+ */
+#define RX_SEQ_START			0x00c
+
+/*
  * The number of audio channels; read-only.  There will be one quadlet per
  * channel.
  */
-#define RX_NUMBER_AUDIO			0x00c
+#define RX_NUMBER_AUDIO			0x010
 
 /*
  * The number of MIDI ports, 0-8; read-only.  If > 0, there will be one
  * additional quadlet in each data block, following the audio quadlets.
  */
-#define RX_NUMBER_MIDI			0x010
-
-/*
- * Index of first quadlet to be interpreted; read/write.  If > 0, that many
- * quadlets at the beginning of each data block will be ignored, and all the
- * audio and MIDI quadlets will follow.
- */
-#define RX_SEQ_START			0x014
+#define RX_NUMBER_MIDI			0x014
 
 /*
  * Names of all audio channels; read-only.  Quadlets are byte-swapped.  Names
diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index ecfe20fd4de5..f5c1d1bced59 100644
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry,
 		} tx;
 		struct {
 			u32 iso;
+			u32 seq_start;
 			u32 number_audio;
 			u32 number_midi;
-			u32 seq_start;
 			char names[RX_NAMES_SIZE];
 			u32 ac3_caps;
 			u32 ac3_enable;
@@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry,
 			break;
 		snd_iprintf(buffer, "rx %u:\n", stream);
 		snd_iprintf(buffer, "  iso channel: %d\n", (int)buf.rx.iso);
+		snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
 		snd_iprintf(buffer, "  audio channels: %u\n",
 			    buf.rx.number_audio);
 		snd_iprintf(buffer, "  midi ports: %u\n", buf.rx.number_midi);
-		snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
 		if (quadlets >= 68) {
 			dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE);
 			snd_iprintf(buffer, "  names: %s\n", buf.rx.names);
diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c
index 5f17b77ee152..f0e4d502d604 100644
--- a/sound/firewire/iso-resources.c
+++ b/sound/firewire/iso-resources.c
@@ -26,7 +26,7 @@
 int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
 {
 	r->channels_mask = ~0uLL;
-	r->unit = fw_unit_get(unit);
+	r->unit = unit;
 	mutex_init(&r->mutex);
 	r->allocated = false;
 
@@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r)
 {
 	WARN_ON(r->allocated);
 	mutex_destroy(&r->mutex);
-	fw_unit_put(r->unit);
 }
 EXPORT_SYMBOL(fw_iso_resources_destroy);
 
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a2ce773bdc62..17c2637d842c 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 	}
 
-	if (!bus->no_response_fallback)
+	if (bus->no_response_fallback)
 		return -1;
 
 	if (!chip->polling_mode && chip->poll_count < 2) {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b680b4ec6331..8ec5289f8e05 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -687,12 +687,45 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
 	return val;
 }
 
+/* is this a stereo widget or a stereo-to-mono mix? */
+static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, int dir)
+{
+	unsigned int wcaps = get_wcaps(codec, nid);
+	hda_nid_t conn;
+
+	if (wcaps & AC_WCAP_STEREO)
+		return true;
+	if (dir != HDA_INPUT || get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
+		return false;
+	if (snd_hda_get_num_conns(codec, nid) != 1)
+		return false;
+	if (snd_hda_get_connections(codec, nid, &conn, 1) < 0)
+		return false;
+	return !!(get_wcaps(codec, conn) & AC_WCAP_STEREO);
+}
+
 /* initialize the amp value (only at the first time) */
 static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)
 {
 	unsigned int caps = query_amp_caps(codec, nid, dir);
 	int val = get_amp_val_to_activate(codec, nid, dir, caps, false);
-	snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+
+	if (is_stereo_amps(codec, nid, dir))
+		snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+	else
+		snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val);
+}
+
+/* update the amp, doing in stereo or mono depending on NID */
+static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx,
+		      unsigned int mask, unsigned int val)
+{
+	if (is_stereo_amps(codec, nid, dir))
+		return snd_hda_codec_amp_stereo(codec, nid, dir, idx,
+						mask, val);
+	else
+		return snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+						mask, val);
 }
 
 /* calculate amp value mask we can modify;
@@ -732,7 +765,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir,
 		return;
 
 	val &= mask;
-	snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val);
+	update_amp(codec, nid, dir, idx, mask, val);
 }
 
 static void activate_amp_out(struct hda_codec *codec, struct nid_path *path,
@@ -4424,13 +4457,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
 	has_amp = nid_has_mute(codec, mix, HDA_INPUT);
 	for (i = 0; i < nums; i++) {
 		if (has_amp)
-			snd_hda_codec_amp_stereo(codec, mix,
-						 HDA_INPUT, i,
-						 0xff, HDA_AMP_MUTE);
+			update_amp(codec, mix, HDA_INPUT, i,
+				   0xff, HDA_AMP_MUTE);
 		else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
-			snd_hda_codec_amp_stereo(codec, conn[i],
-						 HDA_OUTPUT, 0,
-						 0xff, HDA_AMP_MUTE);
+			update_amp(codec, conn[i], HDA_OUTPUT, 0,
+				   0xff, HDA_AMP_MUTE);
 	}
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4ca3d5d02436..a8a1e14272a1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1989,7 +1989,7 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
 	/* Sunrise Point */
 	{ PCI_DEVICE(0x8086, 0xa170),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
 	/* Sunrise Point-LP */
 	{ PCI_DEVICE(0x8086, 0x9d70),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ce5a6da83419..05e19f78b4cb 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -134,13 +134,38 @@ static void print_amp_caps(struct snd_info_buffer *buffer,
 		    (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT);
 }
 
+/* is this a stereo widget or a stereo-to-mono mix? */
+static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid,
+			   int dir, unsigned int wcaps, int indices)
+{
+	hda_nid_t conn;
+
+	if (wcaps & AC_WCAP_STEREO)
+		return true;
+	/* check for a stereo-to-mono mix; it must be:
+	 * only a single connection, only for input, and only a mixer widget
+	 */
+	if (indices != 1 || dir != HDA_INPUT ||
+	    get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
+		return false;
+
+	if (snd_hda_get_raw_connections(codec, nid, &conn, 1) < 0)
+		return false;
+	/* the connection source is a stereo? */
+	wcaps = snd_hda_param_read(codec, conn, AC_PAR_AUDIO_WIDGET_CAP);
+	return !!(wcaps & AC_WCAP_STEREO);
+}
+
 static void print_amp_vals(struct snd_info_buffer *buffer,
 			   struct hda_codec *codec, hda_nid_t nid,
-			   int dir, int stereo, int indices)
+			   int dir, unsigned int wcaps, int indices)
 {
 	unsigned int val;
+	bool stereo;
 	int i;
 
+	stereo = is_stereo_amps(codec, nid, dir, wcaps, indices);
+
 	dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
 	for (i = 0; i < indices; i++) {
 		snd_iprintf(buffer, " [");
@@ -757,12 +782,10 @@ static void print_codec_info(struct snd_info_entry *entry,
 			    (codec->single_adc_amp &&
 			     wid_type == AC_WID_AUD_IN))
 				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       1);
+					       wid_caps, 1);
 			else
 				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       conn_len);
+					       wid_caps, conn_len);
 		}
 		if (wid_caps & AC_WCAP_OUT_AMP) {
 			snd_iprintf(buffer, "  Amp-Out caps: ");
@@ -771,11 +794,10 @@ static void print_codec_info(struct snd_info_entry *entry,
 			if (wid_type == AC_WID_PIN &&
 			    codec->pin_amp_workaround)
 				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       conn_len);
+					       wid_caps, conn_len);
 			else
 				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-					       wid_caps & AC_WCAP_STEREO, 1);
+					       wid_caps, 1);
 		}
 
 		switch (wid_type) {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 1589c9bcce3e..dd2b3d92071f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -393,6 +393,7 @@ 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, 0x5600, "MacBookAir 5,2", CS420X_MBP81),
 	SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
 	SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
 	{} /* terminator */
@@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec)
 		return -ENOMEM;
 
 	spec->gen.automute_hook = cs_automute;
+	codec->single_adc_amp = 1;
 
 	snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
 			   cs420x_fixups);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fd3ed18670e9..da67ea8645a6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -223,6 +223,7 @@ enum {
 	CXT_PINCFG_LENOVO_TP410,
 	CXT_PINCFG_LEMOTE_A1004,
 	CXT_PINCFG_LEMOTE_A1205,
+	CXT_PINCFG_COMPAQ_CQ60,
 	CXT_FIXUP_STEREO_DMIC,
 	CXT_FIXUP_INC_MIC_BOOST,
 	CXT_FIXUP_HEADPHONE_MIC_PIN,
@@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = cxt_pincfg_lemote,
 	},
+	[CXT_PINCFG_COMPAQ_CQ60] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* 0x17 was falsely set up as a mic, it should 0x1d */
+			{ 0x17, 0x400001f0 },
+			{ 0x1d, 0x97a70120 },
+			{ }
+		}
+	},
 	[CXT_FIXUP_STEREO_DMIC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_stereo_dmic,
@@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = {
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
+	SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
 	{}
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 526398a4a442..74382137b9f5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on)
 {
 	/* We currently only handle front, HP */
 	static hda_nid_t pins[] = {
-		0x0f, 0x10, 0x14, 0x15, 0
+		0x0f, 0x10, 0x14, 0x15, 0x17, 0
 	};
 	hda_nid_t *p;
 	for (p = pins; *p; p++)
@@ -5036,6 +5036,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index b67480f1b1aa..4373ada95648 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -317,7 +317,7 @@ static int adav80x_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
-	unsigned int deemph = ucontrol->value.enumerated.item[0];
+	unsigned int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
@@ -333,7 +333,7 @@ static int adav80x_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = adav80x->deemph;
+	ucontrol->value.integer.value[0] = adav80x->deemph;
 	return 0;
 };
 
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 70861c7b1631..81b54a270bd8 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -76,7 +76,7 @@ static int ak4641_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.enumerated.item[0];
+	int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
@@ -92,7 +92,7 @@ static int ak4641_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = ak4641->deemph;
+	ucontrol->value.integer.value[0] = ak4641->deemph;
 	return 0;
 };
 
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 632e89f793a7..2a58b1dccd2f 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route ak4671_intercon[] = {
-	{"DAC Left", "NULL", "PMPLL"},
-	{"DAC Right", "NULL", "PMPLL"},
-	{"ADC Left", "NULL", "PMPLL"},
-	{"ADC Right", "NULL", "PMPLL"},
+	{"DAC Left", NULL, "PMPLL"},
+	{"DAC Right", NULL, "PMPLL"},
+	{"ADC Left", NULL, "PMPLL"},
+	{"ADC Right", NULL, "PMPLL"},
 
 	/* Outputs */
-	{"LOUT1", "NULL", "LOUT1 Mixer"},
-	{"ROUT1", "NULL", "ROUT1 Mixer"},
-	{"LOUT2", "NULL", "LOUT2 Mix Amp"},
-	{"ROUT2", "NULL", "ROUT2 Mix Amp"},
-	{"LOUT3", "NULL", "LOUT3 Mixer"},
-	{"ROUT3", "NULL", "ROUT3 Mixer"},
+	{"LOUT1", NULL, "LOUT1 Mixer"},
+	{"ROUT1", NULL, "ROUT1 Mixer"},
+	{"LOUT2", NULL, "LOUT2 Mix Amp"},
+	{"ROUT2", NULL, "ROUT2 Mix Amp"},
+	{"LOUT3", NULL, "LOUT3 Mixer"},
+	{"ROUT3", NULL, "ROUT3 Mixer"},
 
 	{"LOUT1 Mixer", "DACL", "DAC Left"},
 	{"ROUT1 Mixer", "DACR", "DAC Right"},
 	{"LOUT2 Mixer", "DACHL", "DAC Left"},
 	{"ROUT2 Mixer", "DACHR", "DAC Right"},
-	{"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"},
-	{"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"},
+	{"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"},
+	{"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"},
 	{"LOUT3 Mixer", "DACSL", "DAC Left"},
 	{"ROUT3 Mixer", "DACSR", "DAC Right"},
 
@@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = {
 	{"LIN2", NULL, "Mic Bias"},
 	{"RIN2", NULL, "Mic Bias"},
 
-	{"ADC Left", "NULL", "LIN MUX"},
-	{"ADC Right", "NULL", "RIN MUX"},
+	{"ADC Left", NULL, "LIN MUX"},
+	{"ADC Right", NULL, "RIN MUX"},
 
 	/* Analog Loops */
-	{"LIN1 Mixing Circuit", "NULL", "LIN1"},
-	{"RIN1 Mixing Circuit", "NULL", "RIN1"},
-	{"LIN2 Mixing Circuit", "NULL", "LIN2"},
-	{"RIN2 Mixing Circuit", "NULL", "RIN2"},
-	{"LIN3 Mixing Circuit", "NULL", "LIN3"},
-	{"RIN3 Mixing Circuit", "NULL", "RIN3"},
-	{"LIN4 Mixing Circuit", "NULL", "LIN4"},
-	{"RIN4 Mixing Circuit", "NULL", "RIN4"},
+	{"LIN1 Mixing Circuit", NULL, "LIN1"},
+	{"RIN1 Mixing Circuit", NULL, "RIN1"},
+	{"LIN2 Mixing Circuit", NULL, "LIN2"},
+	{"RIN2 Mixing Circuit", NULL, "RIN2"},
+	{"LIN3 Mixing Circuit", NULL, "LIN3"},
+	{"RIN3 Mixing Circuit", NULL, "RIN3"},
+	{"LIN4 Mixing Circuit", NULL, "LIN4"},
+	{"RIN4 Mixing Circuit", NULL, "RIN4"},
 
 	{"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"},
 	{"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"},
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 79a4efcb894c..7d3a6accaf9a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -286,7 +286,7 @@ static int cs4271_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = cs4271->deemph;
+	ucontrol->value.integer.value[0] = cs4271->deemph;
 	return 0;
 }
 
@@ -296,7 +296,7 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
 
-	cs4271->deemph = ucontrol->value.enumerated.item[0];
+	cs4271->deemph = ucontrol->value.integer.value[0];
 	return cs4271_set_deemph(codec);
 }
 
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index ffe96175a8a5..911c26c705fc 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -876,11 +876,11 @@ static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = {
 
 static const struct snd_soc_dapm_route da732x_dapm_routes[] = {
 	/* Inputs */
-	{"AUX1L PGA", "NULL", "AUX1L"},
-	{"AUX1R PGA", "NULL", "AUX1R"},
+	{"AUX1L PGA", NULL, "AUX1L"},
+	{"AUX1R PGA", NULL, "AUX1R"},
 	{"MIC1 PGA", NULL, "MIC1"},
-	{"MIC2 PGA", "NULL", "MIC2"},
-	{"MIC3 PGA", "NULL", "MIC3"},
+	{"MIC2 PGA", NULL, "MIC2"},
+	{"MIC3 PGA", NULL, "MIC3"},
 
 	/* Capture Path */
 	{"ADC1 Left MUX", "MIC1", "MIC1 PGA"},
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index f27325155ace..c5f35a07e8e4 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -120,7 +120,7 @@ static int es8328_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = es8328->deemph;
+	ucontrol->value.integer.value[0] = es8328->deemph;
 	return 0;
 }
 
@@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.enumerated.item[0];
+	int deemph = ucontrol->value.integer.value[0];
 	int ret;
 
 	if (deemph > 1)
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index a722a023c262..477e13d30971 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -118,7 +118,7 @@ static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = priv->deemph;
+	ucontrol->value.integer.value[0] = priv->deemph;
 
 	return 0;
 }
@@ -129,7 +129,7 @@ static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
 
-	priv->deemph = ucontrol->value.enumerated.item[0];
+	priv->deemph = ucontrol->value.integer.value[0];
 
 	return pcm1681_set_deemph(codec);
 }
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index f374840a5a7c..9b541e52da8c 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -1198,7 +1198,7 @@ static struct dmi_system_id dmi_dell_dino[] = {
 		.ident = "Dell Dino",
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
-			DMI_MATCH(DMI_BOARD_NAME, "0144P8")
+			DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343")
 		}
 	},
 	{ }
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index e182e6569bbd..3593a1496056 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1151,13 +1151,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 		/* Enable VDDC charge pump */
 		ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
 	} else if (vddio >= 3100 && vdda >= 3100) {
-		/*
-		 * if vddio and vddd > 3.1v,
-		 * charge pump should be clean before set ana_pwr
-		 */
-		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
-				SGTL5000_VDDC_CHRGPMP_POWERUP, 0);
-
+		ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP;
 		/* VDDC use VDDIO rail */
 		lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
 		lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 47b257e41809..82095d6cd070 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -538,8 +538,8 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = {
 	/* speaker map */
 	{ "IHFOUTL", NULL, "Speaker Rail"},
 	{ "IHFOUTR", NULL, "Speaker Rail"},
-	{ "IHFOUTL", "NULL", "Speaker Left Playback"},
-	{ "IHFOUTR", "NULL", "Speaker Right Playback"},
+	{ "IHFOUTL", NULL, "Speaker Left Playback"},
+	{ "IHFOUTR", NULL, "Speaker Right Playback"},
 	{ "Speaker Left Playback", NULL, "Speaker Left Filter"},
 	{ "Speaker Right Playback", NULL, "Speaker Right Filter"},
 	{ "Speaker Left Filter", NULL, "IHFDAC Left"},
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
index 249ef5c4c762..32942bed34b1 100644
--- a/sound/soc/codecs/tas5086.c
+++ b/sound/soc/codecs/tas5086.c
@@ -281,7 +281,7 @@ static int tas5086_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = priv->deemph;
+	ucontrol->value.integer.value[0] = priv->deemph;
 
 	return 0;
 }
@@ -292,7 +292,7 @@ static int tas5086_put_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
 
-	priv->deemph = ucontrol->value.enumerated.item[0];
+	priv->deemph = ucontrol->value.integer.value[0];
 
 	return tas5086_set_deemph(codec);
 }
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 8d9de49a5052..21d5402e343f 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -610,7 +610,7 @@ static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
 
-	ucontrol->value.enumerated.item[0] = wm2000->anc_active;
+	ucontrol->value.integer.value[0] = wm2000->anc_active;
 
 	return 0;
 }
@@ -620,7 +620,7 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
-	int anc_active = ucontrol->value.enumerated.item[0];
+	int anc_active = ucontrol->value.integer.value[0];
 	int ret;
 
 	if (anc_active > 1)
@@ -643,7 +643,7 @@ static int wm2000_speaker_get(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
 
-	ucontrol->value.enumerated.item[0] = wm2000->spk_ena;
+	ucontrol->value.integer.value[0] = wm2000->spk_ena;
 
 	return 0;
 }
@@ -653,7 +653,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
-	int val = ucontrol->value.enumerated.item[0];
+	int val = ucontrol->value.integer.value[0];
 	int ret;
 
 	if (val > 1)
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 098c143f44d6..c6d10533e2bd 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -125,7 +125,7 @@ static int wm8731_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = wm8731->deemph;
+	ucontrol->value.integer.value[0] = wm8731->deemph;
 
 	return 0;
 }
@@ -135,7 +135,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.enumerated.item[0];
+	int deemph = ucontrol->value.integer.value[0];
 	int ret = 0;
 
 	if (deemph > 1)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index dde462c082be..04b04f8e147c 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -442,7 +442,7 @@ static int wm8903_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = wm8903->deemph;
+	ucontrol->value.integer.value[0] = wm8903->deemph;
 
 	return 0;
 }
@@ -452,7 +452,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.enumerated.item[0];
+	int deemph = ucontrol->value.integer.value[0];
 	int ret = 0;
 
 	if (deemph > 1)
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index d3b3f57668cc..215e93c1ddf0 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -525,7 +525,7 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = wm8904->deemph;
+	ucontrol->value.integer.value[0] = wm8904->deemph;
 	return 0;
 }
 
@@ -534,7 +534,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.enumerated.item[0];
+	int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 1ab2d462afad..00bec915d652 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -393,7 +393,7 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = wm8955->deemph;
+	ucontrol->value.integer.value[0] = wm8955->deemph;
 	return 0;
 }
 
@@ -402,7 +402,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.enumerated.item[0];
+	int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index cf8fecf97f2c..3035d9856415 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -184,7 +184,7 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.enumerated.item[0] = wm8960->deemph;
+	ucontrol->value.integer.value[0] = wm8960->deemph;
 	return 0;
 }
 
@@ -193,7 +193,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.enumerated.item[0];
+	int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 9517571e820d..98c9525bd751 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -180,7 +180,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
 	struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
-	unsigned int val = ucontrol->value.enumerated.item[0];
+	unsigned int val = ucontrol->value.integer.value[0];
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int mixer, mask, shift, old;
@@ -193,7 +193,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
 
 	mutex_lock(&wm9712->lock);
 	old = wm9712->hp_mixer[mixer];
-	if (ucontrol->value.enumerated.item[0])
+	if (ucontrol->value.integer.value[0])
 		wm9712->hp_mixer[mixer] |= mask;
 	else
 		wm9712->hp_mixer[mixer] &= ~mask;
@@ -231,7 +231,7 @@ static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol,
 	mixer = mc->shift >> 8;
 	shift = mc->shift & 0xff;
 
-	ucontrol->value.enumerated.item[0] =
+	ucontrol->value.integer.value[0] =
 		(wm9712->hp_mixer[mixer] >> shift) & 1;
 
 	return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 68222917b396..79552953e1bd 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -255,7 +255,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
 	struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
-	unsigned int val = ucontrol->value.enumerated.item[0];
+	unsigned int val = ucontrol->value.integer.value[0];
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int mixer, mask, shift, old;
@@ -268,7 +268,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol,
 
 	mutex_lock(&wm9713->lock);
 	old = wm9713->hp_mixer[mixer];
-	if (ucontrol->value.enumerated.item[0])
+	if (ucontrol->value.integer.value[0])
 		wm9713->hp_mixer[mixer] |= mask;
 	else
 		wm9713->hp_mixer[mixer] &= ~mask;
@@ -306,7 +306,7 @@ static int wm9713_hp_mixer_get(struct snd_kcontrol *kcontrol,
 	mixer = mc->shift >> 8;
 	shift = mc->shift & 0xff;
 
-	ucontrol->value.enumerated.item[0] =
+	ucontrol->value.integer.value[0] =
 		(wm9713->hp_mixer[mixer] >> shift) & 1;
 
 	return 0;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 75870c0ea2c9..91eb3aef7f02 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1049,7 +1049,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 				enum spdif_txrate index, bool round)
 {
 	const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
-	bool is_sysclk = clk == spdif_priv->sysclk;
+	bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk);
 	u64 rate_ideal, rate_actual, sub;
 	u32 sysclk_dfmin, sysclk_dfmax;
 	u32 txclk_df, sysclk_df, arate;
@@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			spdif_priv->txclk_src[index], rate[index]);
 	dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",
 			spdif_priv->txclk_df[index], rate[index]);
-	if (spdif_priv->txclk[index] == spdif_priv->sysclk)
+	if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk))
 		dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",
 				spdif_priv->sysclk_df[index], rate[index]);
 	dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n",
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index b9fabbf69db6..6b0c8f717ec2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,7 +603,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	factor = (div2 + 1) * (7 * psr + 1) * 2;
 
 	for (i = 0; i < 255; i++) {
-		tmprate = freq * factor * (i + 2);
+		tmprate = freq * factor * (i + 1);
 
 		if (baudclk_is_used)
 			clkrate = clk_get_rate(ssi_private->baudclk);
@@ -1227,7 +1227,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 	ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0;
 	ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0;
 
-	ret = !of_property_read_u32_array(np, "dmas", dmas, 4);
+	ret = of_property_read_u32_array(np, "dmas", dmas, 4);
 	if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) {
 		ssi_private->use_dual_fifo = true;
 		/* When using dual fifo mode, we need to keep watermark
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index c42ffae5fe9f..402b728c0a06 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -207,9 +207,6 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
 		module = (void *)module + sizeof(*module) + module->mod_size;
 	}
 
-	/* allocate scratch mem regions */
-	sst_block_alloc_scratch(dsp);
-
 	return 0;
 }
 
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 394af5684c05..863a9ca34b8e 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -1732,6 +1732,7 @@ static void sst_hsw_drop_all(struct sst_hsw *hsw)
 int sst_hsw_dsp_load(struct sst_hsw *hsw)
 {
 	struct sst_dsp *dsp = hsw->dsp;
+	struct sst_fw *sst_fw, *t;
 	int ret;
 
 	dev_dbg(hsw->dev, "loading audio DSP....");
@@ -1748,12 +1749,17 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw)
 		return ret;
 	}
 
-	ret = sst_fw_reload(hsw->sst_fw);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: SST FW reload failed\n");
-		sst_dsp_dma_put_channel(dsp);
-		return -ENOMEM;
+	list_for_each_entry_safe_reverse(sst_fw, t, &dsp->fw_list, list) {
+		ret = sst_fw_reload(sst_fw);
+		if (ret < 0) {
+			dev_err(hsw->dev, "error: SST FW reload failed\n");
+			sst_dsp_dma_put_channel(dsp);
+			return -ENOMEM;
+		}
 	}
+	ret = sst_block_alloc_scratch(hsw->dsp);
+	if (ret < 0)
+		return -EINVAL;
 
 	sst_dsp_dma_put_channel(dsp);
 	return 0;
@@ -1809,12 +1815,17 @@ int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw)
 
 int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw)
 {
-	sst_fw_unload(hsw->sst_fw);
-	sst_block_free_scratch(hsw->dsp);
+	struct sst_fw *sst_fw, *t;
+	struct sst_dsp *dsp = hsw->dsp;
+
+	list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) {
+		sst_fw_unload(sst_fw);
+	}
+	sst_block_free_scratch(dsp);
 
 	hsw->boot_complete = false;
 
-	sst_dsp_sleep(hsw->dsp);
+	sst_dsp_sleep(dsp);
 
 	return 0;
 }
@@ -1943,6 +1954,11 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
 		goto fw_err;
 	}
 
+	/* allocate scratch mem regions */
+	ret = sst_block_alloc_scratch(hsw->dsp);
+	if (ret < 0)
+		goto boot_err;
+
 	/* wait for DSP boot completion */
 	sst_dsp_boot(hsw->dsp);
 	ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete,
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index def7d8260c4e..d19483081f9b 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -579,7 +579,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 	} else {
-		if (priv->extclk == priv->clk) {
+		if (clk_is_match(priv->extclk, priv->clk)) {
 			devm_clk_put(&pdev->dev, priv->extclk);
 			priv->extclk = ERR_PTR(-EINVAL);
 		} else {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 30579ca5bacb..e5c990889dcc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -347,6 +347,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
+	mutex_lock(&client_mutex);
+
 	list_for_each_entry(codec, &codec_list, list) {
 		len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
 			       codec->component.name);
@@ -358,6 +360,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf,
 		}
 	}
 
+	mutex_unlock(&client_mutex);
+
 	if (ret >= 0)
 		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 
@@ -382,6 +386,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
+	mutex_lock(&client_mutex);
+
 	list_for_each_entry(component, &component_list, list) {
 		list_for_each_entry(dai, &component->dai_list, list) {
 			len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
@@ -395,6 +401,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf,
 		}
 	}
 
+	mutex_unlock(&client_mutex);
+
 	ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 
 	kfree(buf);
@@ -418,6 +426,8 @@ static ssize_t platform_list_read_file(struct file *file,
 	if (!buf)
 		return -ENOMEM;
 
+	mutex_lock(&client_mutex);
+
 	list_for_each_entry(platform, &platform_list, list) {
 		len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
 			       platform->component.name);
@@ -429,6 +439,8 @@ static ssize_t platform_list_read_file(struct file *file,
 		}
 	}
 
+	mutex_unlock(&client_mutex);
+
 	ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 
 	kfree(buf);
@@ -836,6 +848,8 @@ static struct snd_soc_component *soc_find_component(
 {
 	struct snd_soc_component *component;
 
+	lockdep_assert_held(&client_mutex);
+
 	list_for_each_entry(component, &component_list, list) {
 		if (of_node) {
 			if (component->dev->of_node == of_node)
@@ -854,6 +868,8 @@ static struct snd_soc_dai *snd_soc_find_dai(
 	struct snd_soc_component *component;
 	struct snd_soc_dai *dai;
 
+	lockdep_assert_held(&client_mutex);
+
 	/* Find CPU DAI from registered DAIs*/
 	list_for_each_entry(component, &component_list, list) {
 		if (dlc->of_node && component->dev->of_node != dlc->of_node)
@@ -1508,6 +1524,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	struct snd_soc_codec *codec;
 	int ret, i, order;
 
+	mutex_lock(&client_mutex);
 	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
 
 	/* bind DAIs */
@@ -1662,6 +1679,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	card->instantiated = 1;
 	snd_soc_dapm_sync(&card->dapm);
 	mutex_unlock(&card->mutex);
+	mutex_unlock(&client_mutex);
 
 	return 0;
 
@@ -1680,6 +1698,7 @@ card_probe_error:
 
 base_error:
 	mutex_unlock(&card->mutex);
+	mutex_unlock(&client_mutex);
 
 	return ret;
 }
@@ -2713,13 +2732,6 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
 	list_del(&component->list);
 }
 
-static void snd_soc_component_del(struct snd_soc_component *component)
-{
-	mutex_lock(&client_mutex);
-	snd_soc_component_del_unlocked(component);
-	mutex_unlock(&client_mutex);
-}
-
 int snd_soc_register_component(struct device *dev,
 			       const struct snd_soc_component_driver *cmpnt_drv,
 			       struct snd_soc_dai_driver *dai_drv,
@@ -2767,14 +2779,17 @@ void snd_soc_unregister_component(struct device *dev)
 {
 	struct snd_soc_component *cmpnt;
 
+	mutex_lock(&client_mutex);
 	list_for_each_entry(cmpnt, &component_list, list) {
 		if (dev == cmpnt->dev && cmpnt->registered_as_component)
 			goto found;
 	}
+	mutex_unlock(&client_mutex);
 	return;
 
 found:
-	snd_soc_component_del(cmpnt);
+	snd_soc_component_del_unlocked(cmpnt);
+	mutex_unlock(&client_mutex);
 	snd_soc_component_cleanup(cmpnt);
 	kfree(cmpnt);
 }
@@ -2882,10 +2897,14 @@ struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev)
 {
 	struct snd_soc_platform *platform;
 
+	mutex_lock(&client_mutex);
 	list_for_each_entry(platform, &platform_list, list) {
-		if (dev == platform->dev)
+		if (dev == platform->dev) {
+			mutex_unlock(&client_mutex);
 			return platform;
+		}
 	}
+	mutex_unlock(&client_mutex);
 
 	return NULL;
 }
@@ -3090,15 +3109,15 @@ void snd_soc_unregister_codec(struct device *dev)
 {
 	struct snd_soc_codec *codec;
 
+	mutex_lock(&client_mutex);
 	list_for_each_entry(codec, &codec_list, list) {
 		if (dev == codec->dev)
 			goto found;
 	}
+	mutex_unlock(&client_mutex);
 	return;
 
 found:
-
-	mutex_lock(&client_mutex);
 	list_del(&codec->list);
 	snd_soc_component_del_unlocked(&codec->component);
 	mutex_unlock(&client_mutex);
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 67d476548dcf..07f984d5f516 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x0159),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "Roland", */
+		/* .product_name = "UA-22", */
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 /* this catches most recent vendor-specific Roland devices */
 {
 	.match_flags = USB_DEVICE_ID_MATCH_VENDOR |