19 files changed, 167 insertions, 171 deletions

diff --git a/sound/core/sound.c b/sound/core/sound.c
index 185cec01ee25..5fc93d00572a 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -186,7 +186,7 @@ static const struct file_operations snd_fops =
 };
 
 #ifdef CONFIG_SND_DYNAMIC_MINORS
-static int snd_find_free_minor(int type)
+static int snd_find_free_minor(int type, struct snd_card *card, int dev)
 {
 	int minor;
 
@@ -209,7 +209,7 @@ static int snd_find_free_minor(int type)
 	return -EBUSY;
 }
 #else
-static int snd_kernel_minor(int type, struct snd_card *card, int dev)
+static int snd_find_free_minor(int type, struct snd_card *card, int dev)
 {
 	int minor;
 
@@ -237,6 +237,8 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev)
 	}
 	if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OS_MINORS))
 		return -EINVAL;
+	if (snd_minors[minor])
+		return -EBUSY;
 	return minor;
 }
 #endif
@@ -276,13 +278,7 @@ int snd_register_device(int type, struct snd_card *card, int dev,
 	preg->private_data = private_data;
 	preg->card_ptr = card;
 	mutex_lock(&sound_mutex);
-#ifdef CONFIG_SND_DYNAMIC_MINORS
-	minor = snd_find_free_minor(type);
-#else
-	minor = snd_kernel_minor(type, card, dev);
-	if (minor >= 0 && snd_minors[minor])
-		minor = -EBUSY;
-#endif
+	minor = snd_find_free_minor(type, card, dev);
 	if (minor < 0) {
 		err = minor;
 		goto error;
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/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index bda845afb470..29ccb3637164 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -337,6 +337,10 @@ 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 +351,6 @@ 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/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 227990bc02e3..375e94f4cf52 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -329,8 +329,8 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev)
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	hda->regs = devm_ioremap_resource(dev, res);
-	if (IS_ERR(chip->remap_addr))
-		return PTR_ERR(chip->remap_addr);
+	if (IS_ERR(hda->regs))
+		return PTR_ERR(hda->regs);
 
 	chip->remap_addr = hda->regs + HDA_BAR0;
 	chip->addr = res->start + HDA_BAR0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ddb93083a2af..b2b24a8b3dac 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4937,6 +4937,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY),
 	/* ALC282 */
+	SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
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/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 2c363fdca9fd..cb666c73712d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6043,23 +6043,30 @@ hdspm_hw_constraints_aes32_sample_rates = {
 	.mask = 0
 };
 
-static int snd_hdspm_playback_open(struct snd_pcm_substream *substream)
+static int snd_hdspm_open(struct snd_pcm_substream *substream)
 {
 	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
 	spin_lock_irq(&hdspm->lock);
-
 	snd_pcm_set_sync(substream);
+	runtime->hw = (playback) ? snd_hdspm_playback_subinfo :
+		snd_hdspm_capture_subinfo;
 
+	if (playback) {
+		if (hdspm->capture_substream == NULL)
+			hdspm_stop_audio(hdspm);
 
-	runtime->hw = snd_hdspm_playback_subinfo;
-
-	if (hdspm->capture_substream == NULL)
-		hdspm_stop_audio(hdspm);
+		hdspm->playback_pid = current->pid;
+		hdspm->playback_substream = substream;
+	} else {
+		if (hdspm->playback_substream == NULL)
+			hdspm_stop_audio(hdspm);
 
-	hdspm->playback_pid = current->pid;
-	hdspm->playback_substream = substream;
+		hdspm->capture_pid = current->pid;
+		hdspm->capture_substream = substream;
+	}
 
 	spin_unlock_irq(&hdspm->lock);
 
@@ -6082,6 +6089,9 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream)
 		snd_pcm_hw_constraint_minmax(runtime,
 					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
 					     64, 8192);
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_PERIODS,
+					     2, 2);
 		break;
 	}
 
@@ -6091,105 +6101,42 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream)
 				&hdspm_hw_constraints_aes32_sample_rates);
 	} else {
 		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				snd_hdspm_hw_rule_rate_out_channels, hdspm,
+				(playback ?
+				 snd_hdspm_hw_rule_rate_out_channels :
+				 snd_hdspm_hw_rule_rate_in_channels), hdspm,
 				SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 	}
 
 	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_out_channels, hdspm,
+			(playback ? snd_hdspm_hw_rule_out_channels :
+			 snd_hdspm_hw_rule_in_channels), hdspm,
 			SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 
 	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_out_channels_rate, hdspm,
+			(playback ? snd_hdspm_hw_rule_out_channels_rate :
+			 snd_hdspm_hw_rule_in_channels_rate), hdspm,
 			SNDRV_PCM_HW_PARAM_RATE, -1);
 
 	return 0;
 }
 
-static int snd_hdspm_playback_release(struct snd_pcm_substream *substream)
+static int snd_hdspm_release(struct snd_pcm_substream *substream)
 {
 	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+	bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
 	spin_lock_irq(&hdspm->lock);
 
-	hdspm->playback_pid = -1;
-	hdspm->playback_substream = NULL;
-
-	spin_unlock_irq(&hdspm->lock);
-
-	return 0;
-}
-
-
-static int snd_hdspm_capture_open(struct snd_pcm_substream *substream)
-{
-	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	spin_lock_irq(&hdspm->lock);
-	snd_pcm_set_sync(substream);
-	runtime->hw = snd_hdspm_capture_subinfo;
-
-	if (hdspm->playback_substream == NULL)
-		hdspm_stop_audio(hdspm);
-
-	hdspm->capture_pid = current->pid;
-	hdspm->capture_substream = substream;
-
-	spin_unlock_irq(&hdspm->lock);
-
-	snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
-	snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
-
-	switch (hdspm->io_type) {
-	case AIO:
-	case RayDAT:
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-					     32, 4096);
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
-					     16384, 16384);
-		break;
-
-	default:
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-					     64, 8192);
-		break;
-	}
-
-	if (AES32 == hdspm->io_type) {
-		runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
-		snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				&hdspm_hw_constraints_aes32_sample_rates);
+	if (playback) {
+		hdspm->playback_pid = -1;
+		hdspm->playback_substream = NULL;
 	} else {
-		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				snd_hdspm_hw_rule_rate_in_channels, hdspm,
-				SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+		hdspm->capture_pid = -1;
+		hdspm->capture_substream = NULL;
 	}
 
-	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_in_channels, hdspm,
-			SNDRV_PCM_HW_PARAM_CHANNELS, -1);
-
-	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_in_channels_rate, hdspm,
-			SNDRV_PCM_HW_PARAM_RATE, -1);
-
-	return 0;
-}
-
-static int snd_hdspm_capture_release(struct snd_pcm_substream *substream)
-{
-	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
-
-	spin_lock_irq(&hdspm->lock);
-
-	hdspm->capture_pid = -1;
-	hdspm->capture_substream = NULL;
-
 	spin_unlock_irq(&hdspm->lock);
+
 	return 0;
 }
 
@@ -6407,21 +6354,9 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
 	return 0;
 }
 
-static struct snd_pcm_ops snd_hdspm_playback_ops = {
-	.open = snd_hdspm_playback_open,
-	.close = snd_hdspm_playback_release,
-	.ioctl = snd_hdspm_ioctl,
-	.hw_params = snd_hdspm_hw_params,
-	.hw_free = snd_hdspm_hw_free,
-	.prepare = snd_hdspm_prepare,
-	.trigger = snd_hdspm_trigger,
-	.pointer = snd_hdspm_hw_pointer,
-	.page = snd_pcm_sgbuf_ops_page,
-};
-
-static struct snd_pcm_ops snd_hdspm_capture_ops = {
-	.open = snd_hdspm_capture_open,
-	.close = snd_hdspm_capture_release,
+static struct snd_pcm_ops snd_hdspm_ops = {
+	.open = snd_hdspm_open,
+	.close = snd_hdspm_release,
 	.ioctl = snd_hdspm_ioctl,
 	.hw_params = snd_hdspm_hw_params,
 	.hw_free = snd_hdspm_hw_free,
@@ -6515,9 +6450,9 @@ static int snd_hdspm_create_pcm(struct snd_card *card,
 	strcpy(pcm->name, hdspm->card_name);
 
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
-			&snd_hdspm_playback_ops);
+			&snd_hdspm_ops);
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
-			&snd_hdspm_capture_ops);
+			&snd_hdspm_ops);
 
 	pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
 
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index d6fa9d5514e1..7e21e8f85885 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -91,7 +91,8 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = {
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME |
-				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+				  SNDRV_PCM_INFO_DRAIN_TRIGGER,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
 				  SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= PAGE_SIZE,
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 4864392bfcba..c9917ca5de1a 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -151,7 +151,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
 			hw.info |= SNDRV_PCM_INFO_BATCH;
 
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			addr_widths = dma_caps.dstn_addr_widths;
+			addr_widths = dma_caps.dst_addr_widths;
 		else
 			addr_widths = dma_caps.src_addr_widths;
 	}
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 03fed6611d9e..2ed260b10f6d 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -303,6 +303,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
 		return err;
 	}
 
+	/* Don't check the sample rate for devices which we know don't
+	 * support reading */
+	if (snd_usb_get_sample_rate_quirk(chip))
+		return 0;
+
 	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
 				   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a7398412310b..753a47de8459 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1111,6 +1111,11 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	}
 }
 
+bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
+{
+	/* MS Lifecam HD-5000 doesn't support reading the sample rate. */
+	return chip->usb_id == USB_ID(0x045E, 0x076D);
+}
 
 /* Marantz/Denon USB DACs need a vendor cmd to switch
  * between PCM and native DSD mode
@@ -1122,6 +1127,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs,
 	int err;
 
 	switch (subs->stream->chip->usb_id) {
+	case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
 	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
 	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
 
@@ -1201,6 +1207,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) {
 
 		switch (le16_to_cpu(dev->descriptor.idProduct)) {
+		case 0x1003: /* Denon DA300-USB */
 		case 0x3005: /* Marantz HD-DAC1 */
 		case 0x3006: /* Marantz SA-14S1 */
 			mdelay(20);
@@ -1262,6 +1269,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 
 	/* Denon/Marantz devices with USB DAC functionality */
 	switch (chip->usb_id) {
+	case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */
 	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
 	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
 		if (fp->altsetting == 2)
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index 1b862386577d..2cd71ed1201f 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -21,6 +21,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 			      struct audioformat *fmt);
 
+bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip);
+
 int snd_usb_is_big_endian_format(struct snd_usb_audio *chip,
 				 struct audioformat *fp);