Merge branch 'for-linus' into for-next

Sync with the latest 3.19-rc state for applying other ALSA sequencer
core fixes.

22 files changed, 158 insertions, 90 deletions

diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index ec667f158f19..5d905d90d504 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -82,36 +82,6 @@ struct snd_seq_dummy_port {
 static int my_client = -1;
 
 /*
- * unuse callback - send ALL_SOUNDS_OFF and RESET_CONTROLLERS events
- * to subscribers.
- * Note: this callback is called only after all subscribers are removed.
- */
-static int
-dummy_unuse(void *private_data, struct snd_seq_port_subscribe *info)
-{
-	struct snd_seq_dummy_port *p;
-	int i;
-	struct snd_seq_event ev;
-
-	p = private_data;
-	memset(&ev, 0, sizeof(ev));
-	if (p->duplex)
-		ev.source.port = p->connect;
-	else
-		ev.source.port = p->port;
-	ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
-	ev.type = SNDRV_SEQ_EVENT_CONTROLLER;
-	for (i = 0; i < 16; i++) {
-		ev.data.control.channel = i;
-		ev.data.control.param = MIDI_CTL_ALL_SOUNDS_OFF;
-		snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
-		ev.data.control.param = MIDI_CTL_RESET_CONTROLLERS;
-		snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
-	}
-	return 0;
-}
-
-/*
  * event input callback - just redirect events to subscribers
  */
 static int
@@ -175,7 +145,6 @@ create_port(int idx, int type)
 		| SNDRV_SEQ_PORT_TYPE_PORT;
 	memset(&pcb, 0, sizeof(pcb));
 	pcb.owner = THIS_MODULE;
-	pcb.unuse = dummy_unuse;
 	pcb.event_input = dummy_input;
 	pcb.private_free = dummy_free;
 	pcb.private_data = rec;
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 3badc70124ab..0d580186ef1a 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -21,7 +21,19 @@
 #define CYCLES_PER_SECOND	8000
 #define TICKS_PER_SECOND	(TICKS_PER_CYCLE * CYCLES_PER_SECOND)
 
-#define TRANSFER_DELAY_TICKS	0x2e00 /* 479.17 µs */
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND	3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS	8
+
+#define TRANSFER_DELAY_TICKS	0x2e00 /* 479.17 µs */
 
 /* isochronous header parameters */
 #define ISO_DATA_LENGTH_SHIFT	16
@@ -78,8 +90,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->callbacked = false;
 	s->sync_slave = NULL;
 
-	s->rx_blocks_for_midi = UINT_MAX;
-
 	return 0;
 }
 EXPORT_SYMBOL(amdtp_stream_init);
@@ -222,6 +232,14 @@ sfc_found:
 	for (i = 0; i < pcm_channels; i++)
 		s->pcm_positions[i] = i;
 	s->midi_position = s->pcm_channels;
+
+	/*
+	 * We do not know the actual MIDI FIFO size of most devices.  Just
+	 * assume two bytes, i.e., one byte can be received over the bus while
+	 * the previous one is transmitted over MIDI.
+	 * (The value here is adjusted for midi_ratelimit_per_packet().)
+	 */
+	s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
 }
 EXPORT_SYMBOL(amdtp_stream_set_parameters);
 
@@ -463,6 +481,36 @@ static void amdtp_fill_pcm_silence(struct amdtp_stream *s,
 	}
 }
 
+/*
+ * To avoid sending MIDI bytes at too high a rate, assume that the receiving
+ * device has a FIFO, and track how much it is filled.  This values increases
+ * by one whenever we send one byte in a packet, but the FIFO empties at
+ * a constant rate independent of our packet rate.  One packet has syt_interval
+ * samples, so the number of bytes that empty out of the FIFO, per packet(!),
+ * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate.  To avoid storing
+ * fractional values, the values in midi_fifo_used[] are measured in bytes
+ * multiplied by the sample rate.
+ */
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+	int used;
+
+	used = s->midi_fifo_used[port];
+	if (used == 0) /* common shortcut */
+		return true;
+
+	used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+	used = max(used, 0);
+	s->midi_fifo_used[port] = used;
+
+	return used < s->midi_fifo_limit;
+}
+
+static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
+{
+	s->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
+}
+
 static void amdtp_fill_midi(struct amdtp_stream *s,
 			    __be32 *buffer, unsigned int frames)
 {
@@ -470,16 +518,21 @@ static void amdtp_fill_midi(struct amdtp_stream *s,
 	u8 *b;
 
 	for (f = 0; f < frames; f++) {
-		buffer[s->midi_position] = 0;
 		b = (u8 *)&buffer[s->midi_position];
 
 		port = (s->data_block_counter + f) % 8;
-		if ((f >= s->rx_blocks_for_midi) ||
-		    (s->midi[port] == NULL) ||
-		    (snd_rawmidi_transmit(s->midi[port], b + 1, 1) <= 0))
-			b[0] = 0x80;
-		else
+		if (f < MAX_MIDI_RX_BLOCKS &&
+		    midi_ratelimit_per_packet(s, port) &&
+		    s->midi[port] != NULL &&
+		    snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) {
+			midi_rate_use_one_byte(s, port);
 			b[0] = 0x81;
+		} else {
+			b[0] = 0x80;
+			b[1] = 0;
+		}
+		b[2] = 0;
+		b[3] = 0;
 
 		buffer += s->data_block_quadlets;
 	}
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index e6e8926275b0..8a03a91e728b 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -148,13 +148,12 @@ struct amdtp_stream {
 	bool double_pcm_frames;
 
 	struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
+	int midi_fifo_limit;
+	int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
 
 	/* quirk: fixed interval of dbc between previos/current packets. */
 	unsigned int tx_dbc_interval;
 
-	/* quirk: the first count of data blocks in an rx packet for MIDI */
-	unsigned int rx_blocks_for_midi;
-
 	bool callbacked;
 	wait_queue_head_t callback_wait;
 	struct amdtp_stream *sync_slave;
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 1aab0a32870c..0ebcabfdc7ce 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -484,13 +484,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
 		amdtp_stream_destroy(&bebob->rx_stream);
 		destroy_both_connections(bebob);
 	}
-	/*
-	 * The firmware for these devices ignore MIDI messages in more than
-	 * first 8 data blocks of an received AMDTP packet.
-	 */
-	if (bebob->spec == &maudio_fw410_spec ||
-	    bebob->spec == &maudio_special_spec)
-		bebob->rx_stream.rx_blocks_for_midi = 8;
 end:
 	return err;
 }
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index b985fc5ebdc6..4f440e163667 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -179,11 +179,6 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw)
 		destroy_stream(efw, &efw->tx_stream);
 		goto end;
 	}
-	/*
-	 * Fireworks ignores MIDI messages in more than first 8 data
-	 * blocks of an received AMDTP packet.
-	 */
-	efw->rx_stream.rx_blocks_for_midi = 8;
 
 	/* set IEC61883 compliant mode (actually not fully compliant...) */
 	err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883);
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
index 7752860f7230..4c23381727a1 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev)
 	if (ret)
 		goto err_clk_disable;
 
+	return 0;
+
 err_clk_disable:
 	clk_disable_unprepare(i2s->clk);
 	return ret;
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index e5f2fb884bf3..30c673cdc12e 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -188,8 +188,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0);
 static const char * const pcm512x_dsp_program_texts[] = {
 	"FIR interpolation with de-emphasis",
 	"Low latency IIR with de-emphasis",
-	"Fixed process flow",
 	"High attenuation with de-emphasis",
+	"Fixed process flow",
 	"Ringing-less low latency FIR",
 };
 
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 2cd4fe463102..1d1c7f8a9af2 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -861,10 +861,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream,
 		RT286_I2S_CTRL1, 0x0018, d_len_code << 3);
 	dev_dbg(codec->dev, "format val = 0x%x\n", val);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
-	else
-		snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
+	snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
+	snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 890022171359..d27630accf03 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2174,10 +2174,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w,
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
 	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
+	case SND_SOC_DAPM_PRE_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2);
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0);
 		break;
+
 	default:
 		return 0;
 	}
@@ -2192,10 +2196,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w,
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
 	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
+	case SND_SOC_DAPM_PRE_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2);
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0);
 		break;
+
 	default:
 		return 0;
 	}
@@ -2303,9 +2311,11 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
 
 static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
-		0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU),
+		0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU |
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT,
-		0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU),
+		0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU |
+		SND_SOC_DAPM_POST_PMU),
 
 	/* ASRC */
 	SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5677_ASRC_1, 0, 0, NULL, 0),
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
index 1d1205702d23..9f2dced046de 100644
--- a/sound/soc/codecs/ts3a227e.c
+++ b/sound/soc/codecs/ts3a227e.c
@@ -254,6 +254,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
 	struct ts3a227e *ts3a227e;
 	struct device *dev = &i2c->dev;
 	int ret;
+	unsigned int acc_reg;
 
 	ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL);
 	if (ts3a227e == NULL)
@@ -283,6 +284,11 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
 			   INTB_DISABLE | ADC_COMPLETE_INT_DISABLE,
 			   ADC_COMPLETE_INT_DISABLE);
 
+	/* Read jack status because chip might not trigger interrupt at boot. */
+	regmap_read(ts3a227e->regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg);
+	ts3a227e_new_jack_state(ts3a227e, acc_reg);
+	ts3a227e_jack_report(ts3a227e);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 4d2d2b1380d5..75b87c5c0f04 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
 	{ "Right Capture PGA", NULL, "Right Capture Mux" },
 	{ "Right Capture PGA", NULL, "Right Capture Inverting Mux" },
 
-	{ "AIFOUTL", "Left",  "ADCL" },
-	{ "AIFOUTL", "Right", "ADCR" },
-	{ "AIFOUTR", "Left",  "ADCL" },
-	{ "AIFOUTR", "Right", "ADCR" },
+	{ "AIFOUTL Mux", "Left", "ADCL" },
+	{ "AIFOUTL Mux", "Right", "ADCR" },
+	{ "AIFOUTR Mux", "Left", "ADCL" },
+	{ "AIFOUTR Mux", "Right", "ADCR" },
+
+	{ "AIFOUTL", NULL, "AIFOUTL Mux" },
+	{ "AIFOUTR", NULL, "AIFOUTR Mux" },
 
 	{ "ADCL", NULL, "CLK_DSP" },
 	{ "ADCL", NULL, "Left Capture PGA" },
@@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
 };
 
 static const struct snd_soc_dapm_route dac_intercon[] = {
-	{ "DACL", "Right", "AIFINR" },
-	{ "DACL", "Left",  "AIFINL" },
+	{ "DACL Mux", "Left", "AIFINL" },
+	{ "DACL Mux", "Right", "AIFINR" },
+
+	{ "DACR Mux", "Left", "AIFINL" },
+	{ "DACR Mux", "Right", "AIFINR" },
+
+	{ "DACL", NULL, "DACL Mux" },
 	{ "DACL", NULL, "CLK_DSP" },
 
-	{ "DACR", "Right", "AIFINR" },
-	{ "DACR", "Left",  "AIFINL" },
+	{ "DACR", NULL, "DACR Mux" },
 	{ "DACR", NULL, "CLK_DSP" },
 
 	{ "Charge pump", NULL, "SYSCLK" },
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 031a1ae71d94..a96eb497a379 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -556,7 +556,7 @@ static struct {
 	{ 22050, 2 },
 	{ 24000, 2 },
 	{ 16000, 3 },
-	{ 11250, 4 },
+	{ 11025, 4 },
 	{ 12000, 4 },
 	{  8000, 5 },
 };
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 91a550f4a10d..5e793bbb6b02 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -302,7 +302,7 @@
 #define ESAI_xCCR_xFP_MASK	(((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT)
 #define ESAI_xCCR_xFP(v)	((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK)
 #define ESAI_xCCR_xDC_SHIFT     9
-#define ESAI_xCCR_xDC_WIDTH	4
+#define ESAI_xCCR_xDC_WIDTH	5
 #define ESAI_xCCR_xDC_MASK	(((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT)
 #define ESAI_xCCR_xDC(v)	((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK)
 #define ESAI_xCCR_xPSR_SHIFT	8
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index a65f17d57ffb..059496ed9ad7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1362,9 +1362,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi_private->irq = platform_get_irq(pdev, 0);
-	if (!ssi_private->irq) {
+	if (ssi_private->irq < 0) {
 		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
-		return -ENXIO;
+		return ssi_private->irq;
 	}
 
 	/* Are the RX and the TX clocks locked? */
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 4caacb05a623..cd146d4fa805 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	if (ret)
 		goto clk_fail;
 	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
 	data->card.dai_link = &data->dai;
 	data->card.dapm_widgets = imx_wm8962_dapm_widgets;
 	data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index fb9240fdc9b7..7fe3009b1c43 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 }
 
 /* Decrease the reference count of the device nodes */
-static int asoc_simple_card_unref(struct platform_device *pdev)
+static int asoc_simple_card_unref(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct snd_soc_dai_link *dai_link;
 	int num_links;
 
@@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 		return ret;
 
 err:
-	asoc_simple_card_unref(pdev);
+	asoc_simple_card_unref(&priv->snd_card);
 	return ret;
 }
 
@@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
 		snd_soc_jack_free_gpios(&simple_card_mic_jack, 1,
 					&simple_card_mic_jack_gpio);
 
-	return asoc_simple_card_unref(pdev);
+	return asoc_simple_card_unref(card);
 }
 
 static const struct of_device_id asoc_simple_of_match[] = {
diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c
index 50d6925893ff..a2ae2c5f2e9f 100644
--- a/sound/soc/intel/sst-firmware.c
+++ b/sound/soc/intel/sst-firmware.c
@@ -707,6 +707,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
 	struct list_head *block_list)
 {
 	struct sst_mem_block *block, *tmp;
+	struct sst_block_allocator ba_tmp = *ba;
 	u32 end = ba->offset + ba->size, block_end;
 	int err;
 
@@ -731,9 +732,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
 		if (ba->offset >= block->offset && ba->offset < block_end) {
 
 			/* align ba to block boundary */
-			ba->size -= block_end - ba->offset;
-			ba->offset = block_end;
-			err = block_alloc_contiguous(dsp, ba, block_list);
+			ba_tmp.size -= block_end - ba->offset;
+			ba_tmp.offset = block_end;
+			err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
 			if (err < 0)
 				return -ENOMEM;
 
@@ -768,10 +769,10 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
 			list_move(&block->list, &dsp->used_block_list);
 			list_add(&block->module_list, block_list);
 			/* align ba to block boundary */
-			ba->size -= block_end - ba->offset;
-			ba->offset = block_end;
+			ba_tmp.size -= block_end - ba->offset;
+			ba_tmp.offset = block_end;
 
-			err = block_alloc_contiguous(dsp, ba, block_list);
+			err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
 			if (err < 0)
 				return -ENOMEM;
 
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 3f8c48231364..5bf14040c24a 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -1228,6 +1228,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 	struct sst_dsp *sst = hsw->dsp;
 	unsigned long flags;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n");
+		return 0;
+	}
+
 	/* dont free DSP streams that are not commited */
 	if (!stream->commited)
 		goto out;
@@ -1415,6 +1420,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 	u32 header;
 	int ret;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n");
+		return 0;
+	}
+
+	if (stream->commited) {
+		dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n");
+		return 0;
+	}
+
 	trace_ipc_request("stream alloc", stream->host_id);
 
 	header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
@@ -1519,6 +1534,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
 {
 	int ret;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n");
+		return 0;
+	}
+
 	trace_ipc_request("stream pause", stream->reply.stream_hw_id);
 
 	ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE,
@@ -1535,6 +1555,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
 {
 	int ret;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n");
+		return 0;
+	}
+
 	trace_ipc_request("stream resume", stream->reply.stream_hw_id);
 
 	ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME,
@@ -1550,6 +1575,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 {
 	int ret, tries = 10;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n");
+		return 0;
+	}
+
 	/* dont reset streams that are not commited */
 	if (!stream->commited)
 		return 0;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8b79cafab1e2..c7eb9dd67f60 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_CBM_CFS:
 		/* McBSP slave. FS clock as output */
 		regs->srgr2	|= FSGM;
-		regs->pcr0	|= FSXM;
+		regs->pcr0	|= FSXM | FSRM;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		/* McBSP slave */
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 0a98076333ff..acb5be53bfb4 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -339,6 +339,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
 			    SNDRV_PCM_FMTBIT_S24_LE),
 	},
 	.ops = &rockchip_i2s_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static const struct snd_soc_component_driver rockchip_i2s_component = {
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 590a82f01d0b..025c38fbe3c0 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 			rtd->dai_link->stream_name);
 
 		ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
-				1, 0, &be_pcm);
+				rtd->dai_link->dpcm_playback,
+				rtd->dai_link->dpcm_capture, &be_pcm);
 		if (ret < 0) {
 			dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n",
 				rtd->dai_link->name);
@@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 
 		rtd->pcm = be_pcm;
 		rtd->fe_compr = 1;
-		be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
-		be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
+		if (rtd->dai_link->dpcm_playback)
+			be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
+		else if (rtd->dai_link->dpcm_capture)
+			be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
 		memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops));
 	} else
 		memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops));
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 41650d5b93b7..3e2ef61c627b 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -913,6 +913,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
 	case USB_ID(0x046d, 0x0807): /* Logitech Webcam C500 */
 	case USB_ID(0x046d, 0x0808):
 	case USB_ID(0x046d, 0x0809):
+	case USB_ID(0x046d, 0x0819): /* Logitech Webcam C210 */
 	case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */
 	case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
 	case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */