Merge tag 'v4.0-rc7' into asoc-core

Linux 4.0-rc7

64 files changed, 477 insertions, 258 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/core/pcm_native.c b/sound/core/pcm_native.c
index b03a638b420c..279e24f61305 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1552,6 +1552,8 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state)
 			if (! snd_pcm_playback_empty(substream)) {
 				snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING);
 				snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING);
+			} else {
+				runtime->status->state = SNDRV_PCM_STATE_SETUP;
 			}
 			break;
 		case SNDRV_PCM_STATE_RUNNING:
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index f62780ed64ad..7821b07415a7 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -105,6 +105,8 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum,
 		int pitchbend = chan->midi_pitchbend;
 		int segment;
 
+		if (pitchbend < -0x2000)
+			pitchbend = -0x2000;
 		if (pitchbend > 0x1FFF)
 			pitchbend = 0x1FFF;
 
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 0d580186ef1a..5cc356db5351 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -33,7 +33,7 @@
  */
 #define MAX_MIDI_RX_BLOCKS	8
 
-#define TRANSFER_DELAY_TICKS	0x2e00 /* 479.17 µs */
+#define TRANSFER_DELAY_TICKS	0x2e00 /* 479.17 microseconds */
 
 /* isochronous header parameters */
 #define ISO_DATA_LENGTH_SHIFT	16
@@ -78,7 +78,7 @@ static void pcm_period_tasklet(unsigned long data);
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 		      enum amdtp_stream_direction dir, enum cip_flags flags)
 {
-	s->unit = fw_unit_get(unit);
+	s->unit = unit;
 	s->direction = dir;
 	s->flags = flags;
 	s->context = ERR_PTR(-1);
@@ -102,7 +102,6 @@ void amdtp_stream_destroy(struct amdtp_stream *s)
 {
 	WARN_ON(amdtp_stream_running(s));
 	mutex_destroy(&s->mutex);
-	fw_unit_put(s->unit);
 }
 EXPORT_SYMBOL(amdtp_stream_destroy);
 
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index fc19c99654aa..611b7dae7ee5 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -116,11 +116,22 @@ end:
 	return err;
 }
 
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
 static void
 bebob_card_free(struct snd_card *card)
 {
 	struct snd_bebob *bebob = card->private_data;
 
+	snd_bebob_stream_destroy_duplex(bebob);
+	fw_unit_put(bebob->unit);
+
+	kfree(bebob->maudio_special_quirk);
+
 	if (bebob->card_index >= 0) {
 		mutex_lock(&devices_mutex);
 		clear_bit(bebob->card_index, devices_used);
@@ -205,7 +216,7 @@ bebob_probe(struct fw_unit *unit,
 	card->private_free = bebob_card_free;
 
 	bebob->card = card;
-	bebob->unit = unit;
+	bebob->unit = fw_unit_get(unit);
 	bebob->spec = spec;
 	mutex_init(&bebob->mutex);
 	spin_lock_init(&bebob->lock);
@@ -306,10 +317,11 @@ static void bebob_remove(struct fw_unit *unit)
 	if (bebob == NULL)
 		return;
 
-	kfree(bebob->maudio_special_quirk);
+	/* Awake bus-reset waiters. */
+	if (!completion_done(&bebob->bus_reset))
+		complete_all(&bebob->bus_reset);
 
-	snd_bebob_stream_destroy_duplex(bebob);
-	snd_card_disconnect(bebob->card);
+	/* No need to wait for releasing card object in this context. */
 	snd_card_free_when_closed(bebob->card);
 }
 
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 0ebcabfdc7ce..98e4fc8121a1 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -410,8 +410,6 @@ break_both_connections(struct snd_bebob *bebob)
 static void
 destroy_both_connections(struct snd_bebob *bebob)
 {
-	break_both_connections(bebob);
-
 	cmp_connection_destroy(&bebob->in_conn);
 	cmp_connection_destroy(&bebob->out_conn);
 }
@@ -712,22 +710,16 @@ void snd_bebob_stream_update_duplex(struct snd_bebob *bebob)
 	mutex_unlock(&bebob->mutex);
 }
 
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
 void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob)
 {
-	mutex_lock(&bebob->mutex);
-
-	amdtp_stream_pcm_abort(&bebob->rx_stream);
-	amdtp_stream_pcm_abort(&bebob->tx_stream);
-
-	amdtp_stream_stop(&bebob->rx_stream);
-	amdtp_stream_stop(&bebob->tx_stream);
-
 	amdtp_stream_destroy(&bebob->rx_stream);
 	amdtp_stream_destroy(&bebob->tx_stream);
 
 	destroy_both_connections(bebob);
-
-	mutex_unlock(&bebob->mutex);
 }
 
 /*
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index fa9cf761b610..07dbd01d7a6b 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -311,14 +311,21 @@ end:
 	return err;
 }
 
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
 static void destroy_stream(struct snd_dice *dice, struct amdtp_stream *stream)
 {
-	amdtp_stream_destroy(stream);
+	struct fw_iso_resources *resources;
 
 	if (stream == &dice->tx_stream)
-		fw_iso_resources_destroy(&dice->tx_resources);
+		resources = &dice->tx_resources;
 	else
-		fw_iso_resources_destroy(&dice->rx_resources);
+		resources = &dice->rx_resources;
+
+	amdtp_stream_destroy(stream);
+	fw_iso_resources_destroy(resources);
 }
 
 int snd_dice_stream_init_duplex(struct snd_dice *dice)
@@ -332,6 +339,8 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice)
 		goto end;
 
 	err = init_stream(dice, &dice->rx_stream);
+	if (err < 0)
+		destroy_stream(dice, &dice->tx_stream);
 end:
 	return err;
 }
@@ -340,10 +349,7 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice)
 {
 	snd_dice_transaction_clear_enable(dice);
 
-	stop_stream(dice, &dice->tx_stream);
 	destroy_stream(dice, &dice->tx_stream);
-
-	stop_stream(dice, &dice->rx_stream);
 	destroy_stream(dice, &dice->rx_stream);
 
 	dice->substreams_counter = 0;
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 90d8f40ff727..70a111d7f428 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -226,11 +226,20 @@ static void dice_card_strings(struct snd_dice *dice)
 	strcpy(card->mixername, "DICE");
 }
 
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
 static void dice_card_free(struct snd_card *card)
 {
 	struct snd_dice *dice = card->private_data;
 
+	snd_dice_stream_destroy_duplex(dice);
 	snd_dice_transaction_destroy(dice);
+	fw_unit_put(dice->unit);
+
 	mutex_destroy(&dice->mutex);
 }
 
@@ -251,7 +260,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
 
 	dice = card->private_data;
 	dice->card = card;
-	dice->unit = unit;
+	dice->unit = fw_unit_get(unit);
 	card->private_free = dice_card_free;
 
 	spin_lock_init(&dice->lock);
@@ -305,10 +314,7 @@ static void dice_remove(struct fw_unit *unit)
 {
 	struct snd_dice *dice = dev_get_drvdata(&unit->device);
 
-	snd_card_disconnect(dice->card);
-
-	snd_dice_stream_destroy_duplex(dice);
-
+	/* No need to wait for releasing card object in this context. */
 	snd_card_free_when_closed(dice->card);
 }
 
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 3e2ed8e82cbc..2682e7e3e5c9 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -173,11 +173,23 @@ end:
 	return err;
 }
 
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
 static void
 efw_card_free(struct snd_card *card)
 {
 	struct snd_efw *efw = card->private_data;
 
+	snd_efw_stream_destroy_duplex(efw);
+	snd_efw_transaction_remove_instance(efw);
+	fw_unit_put(efw->unit);
+
+	kfree(efw->resp_buf);
+
 	if (efw->card_index >= 0) {
 		mutex_lock(&devices_mutex);
 		clear_bit(efw->card_index, devices_used);
@@ -185,7 +197,6 @@ efw_card_free(struct snd_card *card)
 	}
 
 	mutex_destroy(&efw->mutex);
-	kfree(efw->resp_buf);
 }
 
 static int
@@ -218,7 +229,7 @@ efw_probe(struct fw_unit *unit,
 	card->private_free = efw_card_free;
 
 	efw->card = card;
-	efw->unit = unit;
+	efw->unit = fw_unit_get(unit);
 	mutex_init(&efw->mutex);
 	spin_lock_init(&efw->lock);
 	init_waitqueue_head(&efw->hwdep_wait);
@@ -289,10 +300,7 @@ static void efw_remove(struct fw_unit *unit)
 {
 	struct snd_efw *efw = dev_get_drvdata(&unit->device);
 
-	snd_efw_stream_destroy_duplex(efw);
-	snd_efw_transaction_remove_instance(efw);
-
-	snd_card_disconnect(efw->card);
+	/* No need to wait for releasing card object in this context. */
 	snd_card_free_when_closed(efw->card);
 }
 
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 4f440e163667..c55db1bddc80 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -100,17 +100,22 @@ end:
 	return err;
 }
 
+/*
+ * This function should be called before starting the stream or after stopping
+ * the streams.
+ */
 static void
 destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream)
 {
-	stop_stream(efw, stream);
-
-	amdtp_stream_destroy(stream);
+	struct cmp_connection *conn;
 
 	if (stream == &efw->tx_stream)
-		cmp_connection_destroy(&efw->out_conn);
+		conn = &efw->out_conn;
 	else
-		cmp_connection_destroy(&efw->in_conn);
+		conn = &efw->in_conn;
+
+	amdtp_stream_destroy(stream);
+	cmp_connection_destroy(&efw->out_conn);
 }
 
 static int
@@ -319,12 +324,8 @@ void snd_efw_stream_update_duplex(struct snd_efw *efw)
 
 void snd_efw_stream_destroy_duplex(struct snd_efw *efw)
 {
-	mutex_lock(&efw->mutex);
-
 	destroy_stream(efw, &efw->rx_stream);
 	destroy_stream(efw, &efw->tx_stream);
-
-	mutex_unlock(&efw->mutex);
 }
 
 void snd_efw_stream_lock_changed(struct snd_efw *efw)
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/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index bda845afb470..e6757cd85724 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -171,9 +171,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
 	}
 
 	/* Wait first packet */
-	err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT);
-	if (err < 0)
+	if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) {
 		stop_stream(oxfw, stream);
+		err = -ETIMEDOUT;
+	}
 end:
 	return err;
 }
@@ -337,6 +338,10 @@ void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw,
 	stop_stream(oxfw, stream);
 }
 
+/*
+ * This function should be called before starting the stream or after stopping
+ * the streams.
+ */
 void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw,
 				     struct amdtp_stream *stream)
 {
@@ -347,8 +352,6 @@ void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw,
 	else
 		conn = &oxfw->in_conn;
 
-	stop_stream(oxfw, stream);
-
 	amdtp_stream_destroy(stream);
 	cmp_connection_destroy(conn);
 }
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 60e5cad0531a..8c6ce019f437 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -104,11 +104,23 @@ end:
 	return err;
 }
 
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
 static void oxfw_card_free(struct snd_card *card)
 {
 	struct snd_oxfw *oxfw = card->private_data;
 	unsigned int i;
 
+	snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
+	if (oxfw->has_output)
+		snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
+
+	fw_unit_put(oxfw->unit);
+
 	for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
 		kfree(oxfw->tx_stream_formats[i]);
 		kfree(oxfw->rx_stream_formats[i]);
@@ -136,7 +148,7 @@ static int oxfw_probe(struct fw_unit *unit,
 	oxfw = card->private_data;
 	oxfw->card = card;
 	mutex_init(&oxfw->mutex);
-	oxfw->unit = unit;
+	oxfw->unit = fw_unit_get(unit);
 	oxfw->device_info = (const struct device_info *)id->driver_data;
 	spin_lock_init(&oxfw->lock);
 	init_waitqueue_head(&oxfw->hwdep_wait);
@@ -212,12 +224,7 @@ static void oxfw_remove(struct fw_unit *unit)
 {
 	struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device);
 
-	snd_card_disconnect(oxfw->card);
-
-	snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
-	if (oxfw->has_output)
-		snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
-
+	/* No need to wait for releasing card object in this context. */
 	snd_card_free_when_closed(oxfw->card);
 }
 
diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
index 17e49a071af4..b408540798c1 100644
--- a/sound/isa/msnd/msnd_pinnacle_mixer.c
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card)
 	spin_lock_init(&chip->mixer_lock);
 	strcpy(card->mixername, "MSND Pinnacle Mixer");
 
-	for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++)
+	for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) {
 		err = snd_ctl_add(card,
 				  snd_ctl_new1(snd_msnd_controls + idx, chip));
 		if (err < 0)
 			return err;
+	}
 
 	return 0;
 }
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index dfcb5e929f9f..17c2637d842c 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -961,7 +961,6 @@ static int azx_alloc_cmd_io(struct azx *chip)
 		dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n");
 	return err;
 }
-EXPORT_SYMBOL_GPL(azx_alloc_cmd_io);
 
 static void azx_init_cmd_io(struct azx *chip)
 {
@@ -1026,7 +1025,6 @@ static void azx_init_cmd_io(struct azx *chip)
 	azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN);
 	spin_unlock_irq(&chip->reg_lock);
 }
-EXPORT_SYMBOL_GPL(azx_init_cmd_io);
 
 static void azx_free_cmd_io(struct azx *chip)
 {
@@ -1036,7 +1034,6 @@ static void azx_free_cmd_io(struct azx *chip)
 	azx_writeb(chip, CORBCTL, 0);
 	spin_unlock_irq(&chip->reg_lock);
 }
-EXPORT_SYMBOL_GPL(azx_free_cmd_io);
 
 static unsigned int azx_command_addr(u32 cmd)
 {
@@ -1167,7 +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) {
@@ -1316,7 +1313,6 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
 	else
 		return azx_corb_send_cmd(bus, val);
 }
-EXPORT_SYMBOL_GPL(azx_send_cmd);
 
 /* get a response */
 static unsigned int azx_get_response(struct hda_bus *bus,
@@ -1330,7 +1326,6 @@ static unsigned int azx_get_response(struct hda_bus *bus,
 	else
 		return azx_rirb_get_response(bus, addr);
 }
-EXPORT_SYMBOL_GPL(azx_get_response);
 
 #ifdef CONFIG_SND_HDA_DSP_LOADER
 /*
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 36d2f20db7a4..a8a1e14272a1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1966,7 +1966,7 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
 	/* Panther Point */
 	{ PCI_DEVICE(0x8086, 0x1e20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
 	/* Lynx Point */
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
@@ -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 b2b24a8b3dac..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),
@@ -5209,6 +5210,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x17, 0x40000000},
 		{0x1d, 0x40700001},
 		{0x21, 0x02211040}),
+	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+		ALC255_STANDARD_PINS,
+		{0x12, 0x90a60170},
+		{0x14, 0x90170140},
+		{0x17, 0x40000000},
+		{0x1d, 0x40700001},
+		{0x21, 0x02211050}),
 	SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
 		{0x12, 0x90a60130},
 		{0x13, 0x40000000},
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6d36c5b78805..87eff3173ce9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -79,6 +79,7 @@ enum {
 	STAC_ALIENWARE_M17X,
 	STAC_92HD89XX_HP_FRONT_JACK,
 	STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK,
+	STAC_92HD73XX_ASUS_MOBO,
 	STAC_92HD73XX_MODELS
 };
 
@@ -1911,7 +1912,18 @@ static const struct hda_fixup stac92hd73xx_fixups[] = {
 	[STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs,
-	}
+	},
+	[STAC_92HD73XX_ASUS_MOBO] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* enable 5.1 and SPDIF out */
+			{ 0x0c, 0x01014411 },
+			{ 0x0d, 0x01014410 },
+			{ 0x0e, 0x01014412 },
+			{ 0x22, 0x014b1180 },
+			{ }
+		}
+	},
 };
 
 static const struct hda_model_fixup stac92hd73xx_models[] = {
@@ -1923,6 +1935,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = {
 	{ .id = STAC_DELL_M6_BOTH, .name = "dell-m6" },
 	{ .id = STAC_DELL_EQ, .name = "dell-eq" },
 	{ .id = STAC_ALIENWARE_M17X, .name = "alienware" },
+	{ .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" },
 	{}
 };
 
@@ -1975,6 +1988,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = {
 				"HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17,
 				"unknown HP", STAC_92HD89XX_HP_FRONT_JACK),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_ASUSTEK, 0x83f8, "ASUS AT4NM10",
+		      STAC_92HD73XX_ASUS_MOBO),
 	{} /* terminator */
 };
 
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f5ad214663f9..8de836165cf2 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -46,8 +46,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/mach-types.h>
-
 #include "../codecs/wm8731.h"
 #include "atmel-pcm.h"
 #include "atmel_ssc_dai.h"
@@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
 	int ret;
 
 	if (!np) {
-		if (!(machine_is_at91sam9g20ek() ||
-			machine_is_at91sam9g20ek_2mmc()))
-			return -ENODEV;
+		return -ENODEV;
 	}
 
 	ret = atmel_ssc_set_audio(0);
@@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 
 	/* Parse device node info */
-	if (np) {
-		ret = snd_soc_of_parse_card_name(card, "atmel,model");
-		if (ret)
-			goto err;
-
-		ret = snd_soc_of_parse_audio_routing(card,
-			"atmel,audio-routing");
-		if (ret)
-			goto err;
-
-		/* Parse codec info */
-		at91sam9g20ek_dai.codec_name = NULL;
-		codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
-		if (!codec_np) {
-			dev_err(&pdev->dev, "codec info missing\n");
-			return -EINVAL;
-		}
-		at91sam9g20ek_dai.codec_of_node = codec_np;
-
-		/* Parse dai and platform info */
-		at91sam9g20ek_dai.cpu_dai_name = NULL;
-		at91sam9g20ek_dai.platform_name = NULL;
-		cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
-		if (!cpu_np) {
-			dev_err(&pdev->dev, "dai and pcm info missing\n");
-			return -EINVAL;
-		}
-		at91sam9g20ek_dai.cpu_of_node = cpu_np;
-		at91sam9g20ek_dai.platform_of_node = cpu_np;
-
-		of_node_put(codec_np);
-		of_node_put(cpu_np);
+	ret = snd_soc_of_parse_card_name(card, "atmel,model");
+	if (ret)
+		goto err;
+
+	ret = snd_soc_of_parse_audio_routing(card,
+		"atmel,audio-routing");
+	if (ret)
+		goto err;
+
+	/* Parse codec info */
+	at91sam9g20ek_dai.codec_name = NULL;
+	codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "codec info missing\n");
+		return -EINVAL;
+	}
+	at91sam9g20ek_dai.codec_of_node = codec_np;
+
+	/* Parse dai and platform info */
+	at91sam9g20ek_dai.cpu_dai_name = NULL;
+	at91sam9g20ek_dai.platform_name = NULL;
+	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "dai and pcm info missing\n");
+		return -EINVAL;
 	}
+	at91sam9g20ek_dai.cpu_of_node = cpu_np;
+	at91sam9g20ek_dai.platform_of_node = cpu_np;
+
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 7b7fbcd49e5e..c7cd60f009e9 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97
 
 config SND_EP93XX_SOC_SNAPPERCL15
         tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
-        depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
+        depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C
         select SND_EP93XX_SOC_I2S
         select SND_SOC_TLV320AIC23_I2C
         help
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 064e6c18e109..ea9f0e31f9d4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX98090 if I2C
 	select SND_SOC_MAX98095 if I2C
-	select SND_SOC_MAX98357A
+	select SND_SOC_MAX98357A if GPIOLIB
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9768 if I2C
 	select SND_SOC_MAX9877 if I2C
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/max98357a.c b/sound/soc/codecs/max98357a.c
index 1806333ea29e..e9e6efbc21dd 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -12,9 +12,19 @@
  * max98357a.c -- MAX98357A ALSA SoC Codec driver
  */
 
-#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/err.h>
 #include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/kernel.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
 #include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
 
 #define DRV_NAME "max98357a"
 
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/rt5670.c b/sound/soc/codecs/rt5670.c
index e1a4a45c57e2..fd102613d20d 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg)
 	case RT5670_ADC_EQ_CTRL1:
 	case RT5670_EQ_CTRL1:
 	case RT5670_ALC_CTRL_1:
-	case RT5670_IRQ_CTRL1:
 	case RT5670_IRQ_CTRL2:
 	case RT5670_INT_IRQ_ST:
 	case RT5670_IL_CMD:
@@ -2703,6 +2702,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 
 	regmap_write(rt5670->regmap, RT5670_RESET, 0);
 
+	regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val);
+	if (val >= 4)
+		regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980);
+	else
+		regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00);
+
 	ret = regmap_register_patch(rt5670->regmap, init_list,
 				    ARRAY_SIZE(init_list));
 	if (ret != 0)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5d0bb8748dd1..fb9c20eace3f 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "IB45 Bypass Mux", "Bypass", "IB45 Mux" },
 	{ "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" },
 
-	{ "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" },
-	{ "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" },
+	{ "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+	{ "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
 	{ "IB6 Mux", "SLB DAC 6", "SLB DAC6" },
 	{ "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" },
 	{ "IB6 Mux", "IF4 DAC L", "IF4 DAC L" },
@@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" },
 	{ "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" },
 
-	{ "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" },
-	{ "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" },
+	{ "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+	{ "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
 	{ "IB7 Mux", "SLB DAC 7", "SLB DAC7" },
 	{ "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" },
 	{ "IB7 Mux", "IF4 DAC R", "IF4 DAC R" },
@@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "DAC1 FS", NULL, "DAC1 MIXL" },
 	{ "DAC1 FS", NULL, "DAC1 MIXR" },
 
-	{ "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" },
-	{ "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" },
+	{ "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" },
+	{ "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" },
 	{ "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" },
 	{ "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" },
 	{ "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" },
 	{ "DAC2 L Mux", "OB 2", "OutBound2" },
 
-	{ "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" },
-	{ "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" },
+	{ "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" },
+	{ "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" },
 	{ "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" },
 	{ "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" },
 	{ "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" },
@@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "DAC2 R Mux", "Haptic Generator", "Haptic Generator" },
 	{ "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" },
 
-	{ "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" },
-	{ "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" },
+	{ "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" },
+	{ "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" },
 	{ "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" },
 	{ "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" },
 	{ "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" },
 	{ "DAC3 L Mux", "OB 4", "OutBound4" },
 
-	{ "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" },
-	{ "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" },
+	{ "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" },
+	{ "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" },
 	{ "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" },
 	{ "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" },
 	{ "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" },
 	{ "DAC3 R Mux", "OB 5", "OutBound5" },
 
-	{ "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" },
-	{ "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" },
+	{ "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+	{ "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
 	{ "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" },
 	{ "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" },
 	{ "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" },
 	{ "DAC4 L Mux", "OB 6", "OutBound6" },
 
-	{ "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" },
-	{ "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" },
+	{ "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+	{ "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
 	{ "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" },
 	{ "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" },
 	{ "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" },
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/sta32x.c b/sound/soc/codecs/sta32x.c
index 3a1343fa109b..007a0e3bc273 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = {
 };
 
 static const struct regmap_range sta32x_write_regs_range[] = {
-	regmap_reg_range(STA32X_CONFA,  STA32X_AUTO2),
-	regmap_reg_range(STA32X_C1CFG,  STA32X_FDRC2),
+	regmap_reg_range(STA32X_CONFA,  STA32X_FDRC2),
 };
 
 static const struct regmap_range sta32x_read_regs_range[] = {
-	regmap_reg_range(STA32X_CONFA,  STA32X_AUTO2),
-	regmap_reg_range(STA32X_C1CFG,  STA32X_FDRC2),
+	regmap_reg_range(STA32X_CONFA,  STA32X_FDRC2),
 };
 
 static const struct regmap_range sta32x_volatile_regs_range[] = {
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 2595611e8a6d..6b0c8f717ec2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,17 +603,20 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	factor = (div2 + 1) * (7 * psr + 1) * 2;
 
 	for (i = 0; i < 255; i++) {
-		/* The bclk rate must be smaller than 1/5 sysclk rate */
-		if (factor * (i + 1) < 5)
-			continue;
-
-		tmprate = freq * factor * (i + 2);
+		tmprate = freq * factor * (i + 1);
 
 		if (baudclk_is_used)
 			clkrate = clk_get_rate(ssi_private->baudclk);
 		else
 			clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
 
+		/*
+		 * Hardware limitation: The bclk rate must be
+		 * never greater than 1/5 IPG clock rate
+		 */
+		if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+			continue;
+
 		clkrate /= factor;
 		afreq = clkrate / (i + 1);
 
@@ -1224,7 +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/generic/simple-card.c b/sound/soc/generic/simple-card.c
index f7c6734bd5da..fb550b5869d2 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
 			    strlen(dai_link->cpu_dai_name)   +
 			    strlen(dai_link->codec_dai_name) + 2,
 			    GFP_KERNEL);
+	if (!name) {
+		ret = -ENOMEM;
+		goto dai_link_of_err;
+	}
+
 	sprintf(name, "%s-%s", dai_link->cpu_dai_name,
 				dai_link->codec_dai_name);
 	dai_link->name = dai_link->stream_name = name;
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index dfebfdd5eb2a..daecc58f28af 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -150,7 +150,7 @@ enum sst_cmd_type {
 
 enum sst_task {
 	SST_TASK_SBA = 1,
-	SST_TASK_MMX,
+	SST_TASK_MMX = 3,
 };
 
 enum sst_type {
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/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 8a8d56a146e7..11c578651c1c 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx,
 
 	spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
 
-	shim_regs->imrx = sst_shim_read64(shim, SST_IMRX),
+	shim_regs->imrx = sst_shim_read64(shim, SST_IMRX);
+	shim_regs->csr = sst_shim_read64(shim, SST_CSR);
+
 
 	spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
 }
@@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx,
 	 */
 	spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
 	sst_shim_write64(shim, SST_IMRX, shim_regs->imrx),
+	sst_shim_write64(shim, SST_CSR, shim_regs->csr),
 	spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
 }
 
@@ -379,6 +382,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx)
 	 * initially active. So change the state to active before
 	 * enabling the pm
 	 */
+
+	if (!acpi_disabled)
+		pm_runtime_set_active(ctx->dev);
+
 	pm_runtime_enable(ctx->dev);
 
 	if (acpi_disabled)
@@ -409,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev)
 	synchronize_irq(ctx->irq_num);
 	flush_workqueue(ctx->post_msg_wq);
 
+	ctx->ops->reset(ctx);
 	/* save the shim registers because PMC doesn't save state */
 	sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64);
 
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/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index ccfb41c22e53..f7eb42aa3f38 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 		return ret;
 
 	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+	if (!card)
+		return -ENOMEM;
+
 	card->name = devm_kasprintf(dev, GFP_KERNEL,
 				    "HDMI %s", dev_name(ad->dssdev));
 	card->owner = THIS_MODULE;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index c7eb9dd67f60..fd99d89de6a8 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
 	case OMAP_MCBSP_SYSCLK_CLKX_EXT:
 		regs->srgr2	|= CLKSM;
+		regs->pcr0	|= SCLKME;
+		/*
+		 * If McBSP is master but yet the CLKX/CLKR pin drives the SRG,
+		 * disable output on those pins. This enables to inject the
+		 * reference clock through CLKX/CLKR. For this to work
+		 * set_dai_sysclk() _needs_ to be called after set_dai_fmt().
+		 */
+		regs->pcr0	&= ~CLKXM;
+		break;
 	case OMAP_MCBSP_SYSCLK_CLKR_EXT:
 		regs->pcr0	|= SCLKME;
+		/* Disable ouput on CLKR pin in master mode */
+		regs->pcr0	&= ~CLKRM;
 		break;
 	default:
 		err = -ENODEV;
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index f4b05bc23e4b..1343ecbf0bd5 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64));
+	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 3cebf6ca03df..0632a36852c8 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM
 
 config SND_SOC_SPEYSIDE
 	tristate "Audio support for Wolfson Speyside"
-	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM8996
 	select SND_SOC_WM9081
@@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY
 
 config SND_SOC_BELLS
 	tristate "Audio support for Wolfson Bells"
-	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA
+	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM5102
 	select SND_SOC_WM5110
@@ -206,7 +206,7 @@ config SND_SOC_LOWLAND
 
 config SND_SOC_LITTLEMILL
 	tristate "Audio support for Wolfson Littlemill"
-	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C
 	select SND_SAMSUNG_I2S
 	select MFD_WM8994
 	select SND_SOC_WM8994
@@ -223,7 +223,7 @@ config SND_SOC_SNOW
 
 config SND_SOC_ODROIDX2
 	tristate "Audio support for Odroid-X2 and Odroid-U3"
-	depends on SND_SOC_SAMSUNG
+	depends on SND_SOC_SAMSUNG && I2C
 	select SND_SOC_MAX98090
 	select SND_SAMSUNG_I2S
 	help
@@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2
 
 config SND_SOC_ARNDALE_RT5631_ALC5631
         tristate "Audio support for RT5631(ALC5631) on Arndale Board"
-        depends on SND_SOC_SAMSUNG
+        depends on SND_SOC_SAMSUNG && I2C
         select SND_SAMSUNG_I2S
         select SND_SOC_RT5631
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 1b53605f7154..110577c52317 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1252,6 +1252,8 @@ static int rsnd_probe(struct platform_device *pdev)
 			goto exit_snd_probe;
 	}
 
+	dev_set_drvdata(dev, priv);
+
 	/*
 	 *	asoc register
 	 */
@@ -1268,8 +1270,6 @@ static int rsnd_probe(struct platform_device *pdev)
 		goto exit_snd_soc;
 	}
 
-	dev_set_drvdata(dev, priv);
-
 	pm_runtime_enable(dev);
 
 	dev_info(dev, "probed\n");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4443581d69f2..71585d0562fa 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)
@@ -1507,6 +1523,7 @@ 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 */
@@ -1661,6 +1678,7 @@ 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);
 
 	soc_init_card_debugfs(card);
 
@@ -1681,6 +1699,7 @@ card_probe_error:
 
 base_error:
 	mutex_unlock(&card->mutex);
+	mutex_unlock(&client_mutex);
 
 	return ret;
 }
@@ -2712,13 +2731,6 @@ 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,
@@ -2766,14 +2778,17 @@ 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);
 }
@@ -2881,10 +2896,14 @@ 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;
 }
@@ -3089,15 +3108,15 @@ 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/line6/playback.c b/sound/usb/line6/playback.c
index 05dee690f487..97ed593f6010 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[],
 		for (; p < buf_end; ++p) {
 			short pv = le16_to_cpu(*p);
 			int val = (pv * volume[chn & 1]) >> 8;
-			pv = clamp(val, 0x7fff, -0x8000);
+			pv = clamp(val, -0x8000, 0x7fff);
 			*p = cpu_to_le16(pv);
 			++chn;
 		}
@@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[],
 
 			val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16);
 			val = (val * volume[chn & 1]) >> 8;
-			val = clamp(val, 0x7fffff, -0x800000);
+			val = clamp(val, -0x800000, 0x7fffff);
 			p[0] = val;
 			p[1] = val >> 8;
 			p[2] = val >> 16;
@@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal,
 			short pov = le16_to_cpu(*po);
 			short piv = le16_to_cpu(*pi);
 			int val = pov + ((piv * volume) >> 8);
-			pov = clamp(val, 0x7fff, -0x8000);
+			pov = clamp(val, -0x8000, 0x7fff);
 			*po = cpu_to_le16(pov);
 		}
 	}
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 |