Merge tag 'v4.4.214' into toradex_vf_4.4-next

This is the 4.4.214 stable release

30 files changed, 474 insertions, 147 deletions

diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 93b02be3a90e..6edec2387861 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = {
 static bool cs4265_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
-	case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2:
+	case CS4265_CHIP_ID ... CS4265_MAX_REGISTER:
 		return true;
 	default:
 		return false;
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 3670086b9227..f273533c6653 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -641,6 +641,7 @@ static const struct regmap_config cs4270_regmap = {
 	.reg_defaults =		cs4270_reg_defaults,
 	.num_reg_defaults =	ARRAY_SIZE(cs4270_reg_defaults),
 	.cache_type =		REGCACHE_RBTREE,
+	.write_flag_mask =	CS4270_I2C_INCR,
 
 	.readable_reg =		cs4270_reg_is_readable,
 	.volatile_reg =		cs4270_reg_is_volatile,
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index d562e1b9a5d1..5b079709ec8a 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -561,6 +561,7 @@ static int cs42xx8_runtime_resume(struct device *dev)
 	msleep(5);
 
 	regcache_cache_only(cs42xx8->regmap, false);
+	regcache_mark_dirty(cs42xx8->regmap);
 
 	ret = regcache_sync(cs42xx8->regmap);
 	if (ret) {
diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index 0ac8fc5ed4ae..9ebd500ecf38 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -379,6 +379,7 @@ static struct i2c_driver cs4349_i2c_driver = {
 	.driver = {
 		.name		= "cs4349",
 		.of_match_table	= cs4349_of_match,
+		.pm = &cs4349_runtime_pm,
 	},
 	.id_table	= cs4349_i2c_id,
 	.probe		= cs4349_i2c_probe,
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index afa6c5db9dcc..2bf30d0eb82f 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -210,7 +210,7 @@ static const struct soc_enum es8328_rline_enum =
 			      ARRAY_SIZE(es8328_line_texts),
 			      es8328_line_texts);
 static const struct snd_kcontrol_new es8328_right_line_controls =
-	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+	SOC_DAPM_ENUM("Route", es8328_rline_enum);
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 584aab83e478..e7aef841f87d 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1209,14 +1209,14 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
 		&max98090_right_rcv_mixer_controls[0],
 		ARRAY_SIZE(max98090_right_rcv_mixer_controls)),
 
-	SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER,
-		M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux),
+	SND_SOC_DAPM_MUX("LINMOD Mux", SND_SOC_NOPM, 0, 0,
+		&max98090_linmod_mux),
 
-	SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL,
-		M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux),
+	SND_SOC_DAPM_MUX("MIXHPLSEL Mux", SND_SOC_NOPM, 0, 0,
+		&max98090_mixhplsel_mux),
 
-	SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL,
-		M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux),
+	SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0,
+		&max98090_mixhprsel_mux),
 
 	SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE,
 		M98090_HPLEN_SHIFT, 0, NULL, 0),
@@ -1924,6 +1924,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090,
 	return 0;
 }
 
+static int max98090_dai_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component);
+	unsigned int fmt = max98090->dai_fmt;
+
+	/* Remove 24-bit format support if it is not in right justified mode. */
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) {
+		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+		snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16);
+	}
+	return 0;
+}
+
 static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params,
 				   struct snd_soc_dai *dai)
@@ -2331,6 +2346,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect);
 #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct snd_soc_dai_ops max98090_dai_ops = {
+	.startup = max98090_dai_startup,
 	.set_sysclk = max98090_dai_set_sysclk,
 	.set_fmt = max98090_dai_set_fmt,
 	.set_tdm_slot = max98090_set_tdm_slot,
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index 91879ea95415..01aa75cde571 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -60,13 +60,15 @@ static DEFINE_MUTEX(spi_mutex);
  * RT5677_SPI_READ/WRITE_32:	Transfer 4 bytes
  * RT5677_SPI_READ/WRITE_BURST:	Transfer any multiples of 8 bytes
  *
- * For example, reading 260 bytes at 0x60030002 uses the following commands:
- * 0x60030002 RT5677_SPI_READ_16	2 bytes
+ * Note:
+ * 16 Bit writes and reads are restricted to the address range
+ * 0x18020000 ~ 0x18021000
+ *
+ * For example, reading 256 bytes at 0x60030004 uses the following commands:
  * 0x60030004 RT5677_SPI_READ_32	4 bytes
  * 0x60030008 RT5677_SPI_READ_BURST	240 bytes
  * 0x600300F8 RT5677_SPI_READ_BURST	8 bytes
  * 0x60030100 RT5677_SPI_READ_32	4 bytes
- * 0x60030104 RT5677_SPI_READ_16	2 bytes
  *
  * Input:
  * @read: true for read commands; false for write commands
@@ -81,15 +83,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len)
 {
 	u8 cmd;
 
-	if (align == 2 || align == 6 || remain == 2) {
-		cmd = RT5677_SPI_READ_16;
-		*len = 2;
-	} else if (align == 4 || remain <= 6) {
+	if (align == 4 || remain <= 4) {
 		cmd = RT5677_SPI_READ_32;
 		*len = 4;
 	} else {
 		cmd = RT5677_SPI_READ_BURST;
-		*len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN);
+		*len = (((remain - 1) >> 3) + 1) << 3;
+		*len = min_t(u32, *len, RT5677_SPI_BURST_LEN);
 	}
 	return read ? cmd : cmd + 1;
 }
@@ -110,7 +110,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen)
 	}
 }
 
-/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */
+/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */
 int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
 {
 	u32 offset;
@@ -126,7 +126,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
 	if (!g_spi)
 		return -ENODEV;
 
-	if ((addr & 1) || (len & 1)) {
+	if ((addr & 3) || (len & 3)) {
 		dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len);
 		return -EACCES;
 	}
@@ -161,13 +161,13 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
 }
 EXPORT_SYMBOL_GPL(rt5677_spi_read);
 
-/* Write DSP address space using SPI. addr has to be 2-byte aligned.
- * If len is not 2-byte aligned, an extra byte of zero is written at the end
+/* Write DSP address space using SPI. addr has to be 4-byte aligned.
+ * If len is not 4-byte aligned, then extra zeros are written at the end
  * as padding.
  */
 int rt5677_spi_write(u32 addr, const void *txbuf, size_t len)
 {
-	u32 offset, len_with_pad = len;
+	u32 offset;
 	int status = 0;
 	struct spi_transfer t;
 	struct spi_message m;
@@ -180,22 +180,19 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len)
 	if (!g_spi)
 		return -ENODEV;
 
-	if (addr & 1) {
+	if (addr & 3) {
 		dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len);
 		return -EACCES;
 	}
 
-	if (len & 1)
-		len_with_pad = len + 1;
-
 	memset(&t, 0, sizeof(t));
 	t.tx_buf = buf;
 	t.speed_hz = RT5677_SPI_FREQ;
 	spi_message_init_with_transfers(&m, &t, 1);
 
-	for (offset = 0; offset < len_with_pad;) {
+	for (offset = 0; offset < len;) {
 		spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7,
-				len_with_pad - offset, &t.len);
+				len - offset, &t.len);
 
 		/* Construct SPI message header */
 		buf[0] = spi_cmd;
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 69d987a9935c..90f8173123f6 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -295,6 +295,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg)
 	case RT5677_I2C_MASTER_CTRL7:
 	case RT5677_I2C_MASTER_CTRL8:
 	case RT5677_HAP_GENE_CTRL2:
+	case RT5677_PWR_ANLG2: /* Modified by DSP firmware */
 	case RT5677_PWR_DSP_ST:
 	case RT5677_PRIV_DATA:
 	case RT5677_PLL1_CTRL2:
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 08b40460663c..a3dd7030f629 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -35,6 +35,13 @@
 #define SGTL5000_DAP_REG_OFFSET	0x0100
 #define SGTL5000_MAX_REG_OFFSET	0x013A
 
+/* Delay for the VAG ramp up */
+#define SGTL5000_VAG_POWERUP_DELAY 500 /* ms */
+/* Delay for the VAG ramp down */
+#define SGTL5000_VAG_POWERDOWN_DELAY 500 /* ms */
+
+#define SGTL5000_OUTPUTS_MUTE (SGTL5000_HP_MUTE | SGTL5000_LINE_OUT_MUTE)
+
 /* default value of sgtl5000 registers */
 static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_CHIP_DIG_POWER,		0x0000 },
@@ -129,6 +136,13 @@ enum sgtl5000_micbias_resistor {
 	SGTL5000_MICBIAS_8K = 8,
 };
 
+enum {
+	HP_POWER_EVENT,
+	DAC_POWER_EVENT,
+	ADC_POWER_EVENT,
+	LAST_POWER_EVENT = ADC_POWER_EVENT
+};
+
 /* sgtl5000 private structure in codec */
 struct sgtl5000_priv {
 	int sysclk;	/* sysclk rate */
@@ -141,8 +155,117 @@ struct sgtl5000_priv {
 	int revision;
 	u8 micbias_resistor;
 	u8 micbias_voltage;
+	u16 mute_state[LAST_POWER_EVENT + 1];
 };
 
+static inline int hp_sel_input(struct snd_soc_component *component)
+{
+	unsigned int ana_reg = 0;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &ana_reg);
+
+	return (ana_reg & SGTL5000_HP_SEL_MASK) >> SGTL5000_HP_SEL_SHIFT;
+}
+
+static inline u16 mute_output(struct snd_soc_component *component,
+			      u16 mute_mask)
+{
+	unsigned int mute_reg = 0;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &mute_reg);
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+			    mute_mask, mute_mask);
+	return mute_reg;
+}
+
+static inline void restore_output(struct snd_soc_component *component,
+				  u16 mute_mask, u16 mute_reg)
+{
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+		mute_mask, mute_reg);
+}
+
+static void vag_power_on(struct snd_soc_component *component, u32 source)
+{
+	unsigned int ana_reg = 0;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_reg);
+
+	if (ana_reg & SGTL5000_VAG_POWERUP)
+		return;
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+			    SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+
+	/* When VAG powering on to get local loop from Line-In, the sleep
+	 * is required to avoid loud pop.
+	 */
+	if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN &&
+	    source == HP_POWER_EVENT)
+		msleep(SGTL5000_VAG_POWERUP_DELAY);
+}
+
+static int vag_power_consumers(struct snd_soc_component *component,
+			       u16 ana_pwr_reg, u32 source)
+{
+	int consumers = 0;
+
+	/* count dac/adc consumers unconditional */
+	if (ana_pwr_reg & SGTL5000_DAC_POWERUP)
+		consumers++;
+	if (ana_pwr_reg & SGTL5000_ADC_POWERUP)
+		consumers++;
+
+	/*
+	 * If the event comes from HP and Line-In is selected,
+	 * current action is 'DAC to be powered down'.
+	 * As HP_POWERUP is not set when HP muxed to line-in,
+	 * we need to keep VAG power ON.
+	 */
+	if (source == HP_POWER_EVENT) {
+		if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN)
+			consumers++;
+	} else {
+		if (ana_pwr_reg & SGTL5000_HP_POWERUP)
+			consumers++;
+	}
+
+	return consumers;
+}
+
+static void vag_power_off(struct snd_soc_component *component, u32 source)
+{
+	unsigned int ana_pwr = SGTL5000_VAG_POWERUP;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_pwr);
+
+	if (!(ana_pwr & SGTL5000_VAG_POWERUP))
+		return;
+
+	/*
+	 * This function calls when any of VAG power consumers is disappearing.
+	 * Thus, if there is more than one consumer at the moment, as minimum
+	 * one consumer will definitely stay after the end of the current
+	 * event.
+	 * Don't clear VAG_POWERUP if 2 or more consumers of VAG present:
+	 * - LINE_IN (for HP events) / HP (for DAC/ADC events)
+	 * - DAC
+	 * - ADC
+	 * (the current consumer is disappearing right now)
+	 */
+	if (vag_power_consumers(component, ana_pwr, source) >= 2)
+		return;
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+		SGTL5000_VAG_POWERUP, 0);
+	/* In power down case, we need wait 400-1000 ms
+	 * when VAG fully ramped down.
+	 * As longer we wait, as smaller pop we've got.
+	 */
+	msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+}
+
 /*
  * mic_bias power on/off share the same register bits with
  * output impedance of mic bias, when power on mic bias, we
@@ -174,36 +297,46 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-/*
- * As manual described, ADC/DAC only works when VAG powerup,
- * So enabled VAG before ADC/DAC up.
- * In power down case, we need wait 400ms when vag fully ramped down.
- */
-static int power_vag_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
+static int vag_and_mute_control(struct snd_soc_component *component,
+				 int event, int event_source)
 {
-	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-	const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+	static const u16 mute_mask[] = {
+		/*
+		 * Mask for HP_POWER_EVENT.
+		 * Muxing Headphones have to be wrapped with mute/unmute
+		 * headphones only.
+		 */
+		SGTL5000_HP_MUTE,
+		/*
+		 * Masks for DAC_POWER_EVENT/ADC_POWER_EVENT.
+		 * Muxing DAC or ADC block have to be wrapped with mute/unmute
+		 * both headphones and line-out.
+		 */
+		SGTL5000_OUTPUTS_MUTE,
+		SGTL5000_OUTPUTS_MUTE
+	};
+
+	struct sgtl5000_priv *sgtl5000 =
+		snd_soc_component_get_drvdata(component);
 
 	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		sgtl5000->mute_state[event_source] =
+			mute_output(component, mute_mask[event_source]);
+		break;
 	case SND_SOC_DAPM_POST_PMU:
-		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
-			SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
-		msleep(400);
+		vag_power_on(component, event_source);
+		restore_output(component, mute_mask[event_source],
+			       sgtl5000->mute_state[event_source]);
 		break;
-
 	case SND_SOC_DAPM_PRE_PMD:
-		/*
-		 * Don't clear VAG_POWERUP, when both DAC and ADC are
-		 * operational to prevent inadvertently starving the
-		 * other one of them.
-		 */
-		if ((snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER) &
-				mask) != mask) {
-			snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
-				SGTL5000_VAG_POWERUP, 0);
-			msleep(400);
-		}
+		sgtl5000->mute_state[event_source] =
+			mute_output(component, mute_mask[event_source]);
+		vag_power_off(component, event_source);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		restore_output(component, mute_mask[event_source],
+			       sgtl5000->mute_state[event_source]);
 		break;
 	default:
 		break;
@@ -212,6 +345,41 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+/*
+ * Mute Headphone when power it up/down.
+ * Control VAG power on HP power path.
+ */
+static int headphone_pga_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, HP_POWER_EVENT);
+}
+
+/* As manual describes, ADC/DAC powering up/down requires
+ * to mute outputs to avoid pops.
+ * Control VAG power on ADC/DAC power path.
+ */
+static int adc_updown_depop(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, ADC_POWER_EVENT);
+}
+
+static int dac_updown_depop(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, DAC_POWER_EVENT);
+}
+
 /* input sources for ADC */
 static const char *adc_mux_text[] = {
 	"MIC_IN", "LINE_IN"
@@ -247,7 +415,10 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 			    mic_bias_event,
 			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
-	SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
+			   headphone_pga_event,
+			   SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
 	SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
@@ -263,11 +434,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 				0, SGTL5000_CHIP_DIG_POWER,
 				1, 0),
 
-	SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
-	SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
-
-	SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event),
-	SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event),
+	SND_SOC_DAPM_ADC_E("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0,
+			   adc_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_DAC_E("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0,
+			   dac_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
 };
 
 /* routes for sgtl5000 */
@@ -1166,12 +1338,17 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 					SGTL5000_INT_OSC_EN);
 		/* Enable VDDC charge pump */
 		ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
-	} else if (vddio >= 3100 && vdda >= 3100) {
+	} else {
 		ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP;
-		/* VDDC use VDDIO rail */
-		lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
-		lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
-			    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+		/*
+		 * if vddio == vdda the source of charge pump should be
+		 * assigned manually to VDDIO
+		 */
+		if (vddio == vdda) {
+			lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
+			lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
+				    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+		}
 	}
 
 	snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
@@ -1238,7 +1415,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 	 * Searching for a suitable index solving this formula:
 	 * idx = 40 * log10(vag_val / lo_cagcntrl) + 15
 	 */
-	vol_quot = (vag * 100) / lo_vag;
+	vol_quot = lo_vag ? (vag * 100) / lo_vag : 0;
 	lo_vol = 0;
 	for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) {
 		if (vol_quot >= vol_quot_table[i])
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f2d3191961e1..714bd0e3fc71 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -234,6 +234,8 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("IN2_R"),
 	SND_SOC_DAPM_INPUT("IN3_L"),
 	SND_SOC_DAPM_INPUT("IN3_R"),
+	SND_SOC_DAPM_INPUT("CM_L"),
+	SND_SOC_DAPM_INPUT("CM_R"),
 };
 
 static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index e7807601e675..ae69cb790ac3 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0),
 SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0),
 SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0),
 SOC_ENUM("3D Low Cut-off", low_3d),
-SOC_ENUM("3D High Cut-off", low_3d),
+SOC_ENUM("3D High Cut-off", high_3d),
 SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv),
 
 SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index a7e79784fc16..4a3ce9b85253 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2792,7 +2792,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	if (target % Fref == 0) {
 		fll_div->theta = 0;
-		fll_div->lambda = 0;
+		fll_div->lambda = 1;
 	} else {
 		gcd_fll = gcd(target, fratio * Fref);
 
@@ -2862,7 +2862,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 		return -EINVAL;
 	}
 
-	if (fll_div.theta || fll_div.lambda)
+	if (fll_div.theta)
 		fll1 |= WM8962_FLL_FRAC;
 
 	/* Stop the FLL while we reconfigure */
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 2ccb8bccc9d4..fc0a73227b02 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -43,6 +43,7 @@
 
 #define MCASP_MAX_AFIFO_DEPTH	64
 
+#ifdef CONFIG_PM
 static u32 context_regs[] = {
 	DAVINCI_MCASP_TXFMCTL_REG,
 	DAVINCI_MCASP_RXFMCTL_REG,
@@ -65,6 +66,7 @@ struct davinci_mcasp_context {
 	u32	*xrsr_regs; /* for serializer configuration */
 	bool	pm_state;
 };
+#endif
 
 struct davinci_mcasp_ruledata {
 	struct davinci_mcasp *mcasp;
@@ -873,14 +875,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
 		active_slots = hweight32(mcasp->tdm_mask[stream]);
 		active_serializers = (channels + active_slots - 1) /
 			active_slots;
-		if (active_serializers == 1) {
+		if (active_serializers == 1)
 			active_slots = channels;
-			for (i = 0; i < total_slots; i++) {
-				if ((1 << i) & mcasp->tdm_mask[stream]) {
-					mask |= (1 << i);
-					if (--active_slots <= 0)
-						break;
-				}
+		for (i = 0; i < total_slots; i++) {
+			if ((1 << i) & mcasp->tdm_mask[stream]) {
+				mask |= (1 << i);
+				if (--active_slots <= 0)
+					break;
 			}
 		}
 	} else {
@@ -1126,6 +1127,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+					    struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, slot_width;
+
+	snd_mask_none(&nfmt);
+	slot_width = rd->mcasp->slot_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) <= slot_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
 static const unsigned int davinci_mcasp_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000,
@@ -1217,7 +1240,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	struct davinci_mcasp_ruledata *ruledata =
 					&mcasp->ruledata[substream->stream];
 	u32 max_channels = 0;
-	int i, dir;
+	int i, dir, ret;
 	int tdm_slots = mcasp->tdm_slots;
 
 	if (mcasp->tdm_mask[substream->stream])
@@ -1242,6 +1265,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 			max_channels++;
 	}
 	ruledata->serializers = max_channels;
+	ruledata->mcasp = mcasp;
 	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
@@ -1267,20 +1291,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				   &mcasp->chconstr[substream->stream]);
 
-	if (mcasp->slot_width)
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     8, mcasp->slot_width);
+	if (mcasp->slot_width) {
+		/* Only allow formats require <= slot_width bits on the bus */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_slot_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
 
 	/*
 	 * If we rely on implicit BCLK divider setting we should
 	 * set constraints based on what we can provide.
 	 */
 	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
-		int ret;
-
-		ruledata->mcasp = mcasp;
-
 		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_RATE,
 					  davinci_mcasp_hw_rule_rate,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index fbb5b979f910..74508964b0ae 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -172,16 +172,17 @@ config SND_MPC52xx_SOC_EFIKA
 
 endif # SND_POWERPC_SOC
 
+config SND_SOC_IMX_PCM_FIQ
+	tristate
+	default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC)
+	select FIQ
+
 if SND_IMX_SOC
 
 config SND_SOC_IMX_SSI
 	tristate
 	select SND_SOC_FSL_UTILS
 
-config SND_SOC_IMX_PCM_FIQ
-	tristate
-	select FIQ
-
 comment "SoC Audio support for Freescale i.MX boards:"
 
 config SND_MXC_SOC_WM1133_EV1
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 883087f2b092..38132143b7d5 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -119,13 +119,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
 		if (ret) {
 			dev_err(&pdev->dev,
 				"fsl,mux-int-port node missing or invalid.\n");
-			return ret;
+			goto err;
 		}
 		ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port);
 		if (ret) {
 			dev_err(&pdev->dev,
 				"fsl,mux-ext-port node missing or invalid.\n");
-			return ret;
+			goto err;
 		}
 
 		/*
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 1b05d1c5d9fd..a32fe14b4687 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -659,6 +659,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 asrc_fail:
 	of_node_put(asrc_np);
 	of_node_put(codec_np);
+	put_device(&cpu_pdev->dev);
 fail:
 	of_node_put(cpu_np);
 
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a87836d4de15..40075b9afb79 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -57,6 +57,8 @@ struct fsl_esai {
 	u32 fifo_depth;
 	u32 slot_width;
 	u32 slots;
+	u32 tx_mask;
+	u32 rx_mask;
 	u32 hck_rate[2];
 	u32 sck_rate[2];
 	bool hck_dir[2];
@@ -357,21 +359,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
 			   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
 
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA,
-			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask));
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB,
-			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask));
-
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
 			   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
 
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA,
-			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask));
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB,
-			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
-
 	esai_priv->slot_width = slot_width;
 	esai_priv->slots = slots;
+	esai_priv->tx_mask = tx_mask;
+	esai_priv->rx_mask = rx_mask;
 
 	return 0;
 }
@@ -582,6 +576,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u8 i, channels = substream->runtime->channels;
 	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
+	u32 mask;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -594,15 +589,38 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 		for (i = 0; tx && i < channels; i++)
 			regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0);
 
+		/*
+		 * When set the TE/RE in the end of enablement flow, there
+		 * will be channel swap issue for multi data line case.
+		 * In order to workaround this issue, we switch the bit
+		 * enablement sequence to below sequence
+		 * 1) clear the xSMB & xSMA: which is done in probe and
+		 *                           stop state.
+		 * 2) set TE/RE
+		 * 3) set xSMB
+		 * 4) set xSMA:  xSMA is the last one in this flow, which
+		 *               will trigger esai to start.
+		 */
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
 				   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
+		mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask;
+
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
+				   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask));
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
+				   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask));
+
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0);
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
+				   ESAI_xSMA_xS_MASK, 0);
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
+				   ESAI_xSMB_xS_MASK, 0);
 
 		/* Disable and reset FIFO */
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
@@ -887,6 +905,15 @@ static int fsl_esai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	esai_priv->tx_mask = 0xFFFFFFFF;
+	esai_priv->rx_mask = 0xFFFFFFFF;
+
+	/* Clear the TSMA, TSMB, RSMA, RSMB */
+	regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0);
+	regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0);
+	regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0);
+	regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0);
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component,
 					      &fsl_esai_dai, 1);
 	if (ret) {
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 08b460ba06ef..61d2d955f26a 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -260,12 +260,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_CBS_CFS:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+		sai->is_slave_mode = false;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		sai->is_slave_mode = true;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFM:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+		sai->is_slave_mode = false;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 7ca67613e0d4..d46e9ad600b4 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1374,6 +1374,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	struct fsl_ssi_private *ssi_private;
 	int ret = 0;
 	struct device_node *np = pdev->dev.of_node;
+	struct device_node *root;
 	const struct of_device_id *of_id;
 	const char *p, *sprop;
 	const uint32_t *iprop;
@@ -1510,7 +1511,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	 * device tree.  We also pass the address of the CPU DAI driver
 	 * structure.
 	 */
-	sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL);
+	root = of_find_node_by_path("/");
+	sprop = of_get_property(root, "compatible", NULL);
+	of_node_put(root);
 	/* Sometimes the compatible name has a "fsl," prefix, so we strip it. */
 	p = strrchr(sprop, ',');
 	if (p)
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index b9e42b503a37..4f8bdb7650e8 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -75,6 +75,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
 	iprop = of_get_property(dma_np, "cell-index", NULL);
 	if (!iprop) {
 		of_node_put(dma_np);
+		of_node_put(dma_channel_np);
 		return -EINVAL;
 	}
 	*dma_id = be32_to_cpup(iprop);
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index b99e0b5e00e9..3d99a8579c99 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -115,10 +115,12 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 		ret = -EPROBE_DEFER;
 		goto fail;
 	}
+	put_device(&ssi_pdev->dev);
 	codec_dev = of_find_i2c_device_by_node(codec_np);
 	if (!codec_dev) {
 		dev_err(&pdev->dev, "failed to find codec platform device\n");
-		return -EPROBE_DEFER;
+		ret = -EPROBE_DEFER;
+		goto fail;
 	}
 
 	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c
index c9452e02e0dd..c0a50ecb6dbd 100644
--- a/sound/soc/intel/common/sst-dsp.c
+++ b/sound/soc/intel/common/sst-dsp.c
@@ -463,11 +463,15 @@ struct sst_dsp *sst_dsp_new(struct device *dev,
 		goto irq_err;
 
 	err = sst_dma_new(sst);
-	if (err)
-		dev_warn(dev, "sst_dma_new failed %d\n", err);
+	if (err)  {
+		dev_err(dev, "sst_dma_new failed %d\n", err);
+		goto dma_err;
+	}
 
 	return sst;
 
+dma_err:
+	free_irq(sst->irq, sst);
 irq_err:
 	if (sst->ops->free)
 		sst->ops->free(sst);
diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index a12c7bb08d3b..b96bf44be2d5 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -211,6 +211,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc,
 
 	if (ipc->ops.reply_msg_match != NULL)
 		header = ipc->ops.reply_msg_match(header, &mask);
+	else
+		mask = (u64)-1;
 
 	if (list_empty(&ipc->rx_list)) {
 		dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n",
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3a36d60e1785..0a5d9fb6fc84 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -570,10 +570,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return PTR_ERR(priv->clk);
 	}
 
-	err = clk_prepare_enable(priv->clk);
-	if (err < 0)
-		return err;
-
 	priv->extclk = devm_clk_get(&pdev->dev, "extclk");
 	if (IS_ERR(priv->extclk)) {
 		if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
@@ -589,6 +585,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		}
 	}
 
+	err = clk_prepare_enable(priv->clk);
+	if (err < 0)
+		return err;
+
 	/* Some sensible defaults - this reflects the powerup values */
 	priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24;
 	priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index 1efdf0088ecd..f2c71bcd06fa 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -98,31 +98,34 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
 
 		if (!cpu || !codec) {
 			dev_err(dev, "Can't find cpu/codec DT node\n");
-			return ERR_PTR(-EINVAL);
+			ret = -EINVAL;
+			goto error;
 		}
 
 		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
 		if (!link->cpu_of_node) {
 			dev_err(card->dev, "error getting cpu phandle\n");
-			return ERR_PTR(-EINVAL);
+			ret = -EINVAL;
+			goto error;
 		}
 
 		link->codec_of_node = of_parse_phandle(codec, "sound-dai", 0);
 		if (!link->codec_of_node) {
 			dev_err(card->dev, "error getting codec phandle\n");
-			return ERR_PTR(-EINVAL);
+			ret = -EINVAL;
+			goto error;
 		}
 
 		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
 		if (ret) {
 			dev_err(card->dev, "error getting cpu dai name\n");
-			return ERR_PTR(ret);
+			goto error;
 		}
 
 		ret = snd_soc_of_get_dai_name(codec, &link->codec_dai_name);
 		if (ret) {
 			dev_err(card->dev, "error getting codec dai name\n");
-			return ERR_PTR(ret);
+			goto error;
 		}
 
 		link->platform_of_node = link->cpu_of_node;
@@ -132,15 +135,24 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
 		ret = of_property_read_string(np, "link-name", &link->name);
 		if (ret) {
 			dev_err(card->dev, "error getting codec dai_link name\n");
-			return ERR_PTR(ret);
+			goto error;
 		}
 
 		link->stream_name = link->name;
 		link->init = apq8016_sbc_dai_init;
 		link++;
+
+		of_node_put(cpu);
+		of_node_put(codec);
 	}
 
 	return data;
+
+ error:
+	of_node_put(np);
+	of_node_put(cpu);
+	of_node_put(codec);
+	return ERR_PTR(ret);
 }
 
 static int apq8016_sbc_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 58ee64594f07..f583f317644a 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -530,7 +530,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register PCM\n");
-		return ret;
+		goto err_suspend;
 	}
 
 	return 0;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index e00dfbec22c5..f18485c6a5d8 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -524,6 +524,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	}
 
 	/* set format */
+	rdai->bit_clk_inv = 0;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		rdai->sys_delay = 0;
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 6fd1906af387..fe65754c2e50 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -301,6 +301,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
 
 		if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i]))
 			pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE;
+
+		if (rtd->pcm->streams[i].pcm->name[0] == '\0') {
+			strncpy(rtd->pcm->streams[i].pcm->name,
+				rtd->pcm->streams[i].pcm->id,
+				sizeof(rtd->pcm->streams[i].pcm->name));
+		}
 	}
 
 	return 0;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index fbaa1bb41102..00d7902ad427 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -80,10 +80,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	unsigned int sync = 0;
 	int enable;
 
-	trace_snd_soc_jack_report(jack, mask, status);
-
 	if (!jack)
 		return;
+	trace_snd_soc_jack_report(jack, mask, status);
 
 	dapm = &jack->card->dapm;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index f99eb8f44282..81bedd9bb922 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -48,8 +48,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
 	else
 		codec_stream = &dai->driver->capture;
 
-	/* If the codec specifies any rate at all, it supports the stream. */
-	return codec_stream->rates;
+	/* If the codec specifies any channels at all, it supports the stream */
+	return codec_stream->channels_min;
 }
 
 /**
@@ -882,10 +882,13 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		codec_params = *params;
 
 		/* fixup params based on TDM slot masks */
-		if (codec_dai->tx_mask)
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+		    codec_dai->tx_mask)
 			soc_pcm_codec_params_fixup(&codec_params,
 						   codec_dai->tx_mask);
-		if (codec_dai->rx_mask)
+
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
+		    codec_dai->rx_mask)
 			soc_pcm_codec_params_fixup(&codec_params,
 						   codec_dai->rx_mask);
 
@@ -1538,7 +1541,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 				 u64 formats)
 {
 	runtime->hw.rate_min = stream->rate_min;
-	runtime->hw.rate_max = stream->rate_max;
+	runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX);
 	runtime->hw.channels_min = stream->channels_min;
 	runtime->hw.channels_max = stream->channels_max;
 	if (runtime->hw.formats)
@@ -2023,42 +2026,81 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 }
 EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger);
 
+static int dpcm_dai_trigger_fe_be(struct snd_pcm_substream *substream,
+				  int cmd, bool fe_first)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	int ret;
+
+	/* call trigger on the frontend before the backend. */
+	if (fe_first) {
+		dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n",
+			fe->dai_link->name, cmd);
+
+		ret = soc_pcm_trigger(substream, cmd);
+		if (ret < 0)
+			return ret;
+
+		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+		return ret;
+	}
+
+	/* call trigger on the frontend after the backend. */
+	ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+	if (ret < 0)
+		return ret;
+
+	dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n",
+		fe->dai_link->name, cmd);
+
+	ret = soc_pcm_trigger(substream, cmd);
+
+	return ret;
+}
+
 static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *fe = substream->private_data;
-	int stream = substream->stream, ret;
+	int stream = substream->stream;
+	int ret = 0;
 	enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
 
 	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
 
 	switch (trigger) {
 	case SND_SOC_DPCM_TRIGGER_PRE:
-		/* call trigger on the frontend before the backend. */
-
-		dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n",
-				fe->dai_link->name, cmd);
-
-		ret = soc_pcm_trigger(substream, cmd);
-		if (ret < 0) {
-			dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
-			goto out;
+		switch (cmd) {
+		case SNDRV_PCM_TRIGGER_START:
+		case SNDRV_PCM_TRIGGER_RESUME:
+		case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, true);
+			break;
+		case SNDRV_PCM_TRIGGER_STOP:
+		case SNDRV_PCM_TRIGGER_SUSPEND:
+		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, false);
+			break;
+		default:
+			ret = -EINVAL;
+			break;
 		}
-
-		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
 		break;
 	case SND_SOC_DPCM_TRIGGER_POST:
-		/* call trigger on the frontend after the backend. */
-
-		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
-		if (ret < 0) {
-			dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
-			goto out;
+		switch (cmd) {
+		case SNDRV_PCM_TRIGGER_START:
+		case SNDRV_PCM_TRIGGER_RESUME:
+		case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, false);
+			break;
+		case SNDRV_PCM_TRIGGER_STOP:
+		case SNDRV_PCM_TRIGGER_SUSPEND:
+		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, true);
+			break;
+		default:
+			ret = -EINVAL;
+			break;
 		}
-
-		dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n",
-				fe->dai_link->name, cmd);
-
-		ret = soc_pcm_trigger(substream, cmd);
 		break;
 	case SND_SOC_DPCM_TRIGGER_BESPOKE:
 		/* bespoke trigger() - handles both FE and BEs */
@@ -2067,10 +2109,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
 				fe->dai_link->name, cmd);
 
 		ret = soc_pcm_bespoke_trigger(substream, cmd);
-		if (ret < 0) {
-			dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
-			goto out;
-		}
 		break;
 	default:
 		dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd,
@@ -2079,6 +2117,12 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
 		goto out;
 	}
 
+	if (ret < 0) {
+		dev_err(fe->dev, "ASoC: trigger FE cmd: %d failed: %d\n",
+			cmd, ret);
+		goto out;
+	}
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME: