Merge tag 'v5.3.6' into toradex_5.3.y

This is the 5.3.6 stable release

31 files changed, 511 insertions, 141 deletions

diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c
index 03cda2166ea3..72908b4de77c 100644
--- a/sound/firewire/motu/motu.c
+++ b/sound/firewire/motu/motu.c
@@ -247,6 +247,17 @@ static const struct snd_motu_spec motu_audio_express = {
 	.analog_out_ports = 4,
 };
 
+static const struct snd_motu_spec motu_4pre = {
+	.name = "4pre",
+	.protocol = &snd_motu_protocol_v3,
+	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
+		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
+		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
+		 SND_MOTU_SPEC_RX_SEPARETED_MAIN,
+	.analog_in_ports = 2,
+	.analog_out_ports = 2,
+};
+
 #define SND_MOTU_DEV_ENTRY(model, data)			\
 {							\
 	.match_flags	= IEEE1394_MATCH_VENDOR_ID |	\
@@ -265,6 +276,7 @@ static const struct ieee1394_device_id motu_id_table[] = {
 	SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3),	/* FireWire only. */
 	SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3),	/* Hybrid. */
 	SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express),
+	SND_MOTU_DEV_ENTRY(0x000045, &motu_4pre),
 	{ }
 };
 MODULE_DEVICE_TABLE(ieee1394, motu_id_table);
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index b5ced5415e40..2377732caa52 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -56,6 +56,9 @@ static int pcm_open(struct snd_pcm_substream *substream)
 		goto err_locked;
 
 	err = snd_tscm_stream_get_clock(tscm, &clock);
+	if (err < 0)
+		goto err_locked;
+
 	if (clock != SND_TSCM_CLOCK_INTERNAL ||
 	    amdtp_stream_pcm_running(&tscm->rx_stream) ||
 	    amdtp_stream_pcm_running(&tscm->tx_stream)) {
diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
index e852e46ebe6f..ccfa92fbc145 100644
--- a/sound/firewire/tascam/tascam-stream.c
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -8,20 +8,37 @@
 #include <linux/delay.h>
 #include "tascam.h"
 
+#define CLOCK_STATUS_MASK      0xffff0000
+#define CLOCK_CONFIG_MASK      0x0000ffff
+
 #define CALLBACK_TIMEOUT 500
 
 static int get_clock(struct snd_tscm *tscm, u32 *data)
 {
+	int trial = 0;
 	__be32 reg;
 	int err;
 
-	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
-				 TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
-				 &reg, sizeof(reg), 0);
-	if (err >= 0)
+	while (trial++ < 5) {
+		err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+				TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+				&reg, sizeof(reg), 0);
+		if (err < 0)
+			return err;
+
 		*data = be32_to_cpu(reg);
+		if (*data & CLOCK_STATUS_MASK)
+			break;
 
-	return err;
+		// In intermediate state after changing clock status.
+		msleep(50);
+	}
+
+	// Still in the intermediate state.
+	if (trial >= 5)
+		return -EAGAIN;
+
+	return 0;
 }
 
 static int set_clock(struct snd_tscm *tscm, unsigned int rate,
@@ -34,7 +51,7 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate,
 	err = get_clock(tscm, &data);
 	if (err < 0)
 		return err;
-	data &= 0x0000ffff;
+	data &= CLOCK_CONFIG_MASK;
 
 	if (rate > 0) {
 		data &= 0x000000ff;
@@ -79,17 +96,14 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate,
 
 int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate)
 {
-	u32 data = 0x0;
-	unsigned int trials = 0;
+	u32 data;
 	int err;
 
-	while (data == 0x0 || trials++ < 5) {
-		err = get_clock(tscm, &data);
-		if (err < 0)
-			return err;
+	err = get_clock(tscm, &data);
+	if (err < 0)
+		return err;
 
-		data = (data & 0xff000000) >> 24;
-	}
+	data = (data & 0xff000000) >> 24;
 
 	/* Check base rate. */
 	if ((data & 0x0f) == 0x01)
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 3b0110545070..196bbc85699e 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -447,6 +447,8 @@ static void azx_int_disable(struct hdac_bus *bus)
 	list_for_each_entry(azx_dev, &bus->stream_list, list)
 		snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0);
 
+	synchronize_irq(bus->irq);
+
 	/* disable SIE for all streams */
 	snd_hdac_chip_writeb(bus, INTCTL, 0);
 
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index 5f59316f982a..7d15093844b9 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -775,11 +775,12 @@ static int build_adc_controls(struct snd_akm4xxx *ak)
 				return err;
 
 			memset(&knew, 0, sizeof(knew));
-			knew.name = ak->adc_info[mixer_ch].selector_name;
-			if (!knew.name) {
+			if (!ak->adc_info ||
+				!ak->adc_info[mixer_ch].selector_name) {
 				knew.name = "Capture Channel";
 				knew.index = mixer_ch + ak->idx_offset * 2;
-			}
+			} else
+				knew.name = ak->adc_info[mixer_ch].selector_name;
 
 			knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 			knew.info = ak4xxx_capture_source_info;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 51f10ed9bc43..a2fb19129219 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -846,7 +846,13 @@ static void snd_hda_codec_dev_release(struct device *dev)
 	snd_hda_sysfs_clear(codec);
 	kfree(codec->modelname);
 	kfree(codec->wcaps);
-	kfree(codec);
+
+	/*
+	 * In the case of ASoC HD-audio, hda_codec is device managed.
+	 * It will be freed when the ASoC device is removed.
+	 */
+	if (codec->core.type == HDA_DEV_LEGACY)
+		kfree(codec);
 }
 
 #define DEV_NAME_LEN 31
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 48d863736b3c..a5a2e9fe7785 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -869,10 +869,13 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
 	 */
 	if (hbus->allow_bus_reset && !hbus->response_reset && !hbus->in_reset) {
 		hbus->response_reset = 1;
+		dev_err(chip->card->dev,
+			"No response from codec, resetting bus: last cmd=0x%08x\n",
+			bus->last_cmd[addr]);
 		return -EAGAIN; /* give a chance to retry */
 	}
 
-	dev_err(chip->card->dev,
+	dev_WARN(chip->card->dev,
 		"azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n",
 		bus->last_cmd[addr]);
 	chip->single_cmd = 1;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b0de3e3b33e5..783f9a9c40ec 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1349,9 +1349,9 @@ static int azx_free(struct azx *chip)
 	}
 
 	if (bus->chip_init) {
+		azx_stop_chip(chip);
 		azx_clear_irq_pending(chip);
 		azx_stop_all_streams(chip);
-		azx_stop_chip(chip);
 	}
 
 	if (bus->irq >= 0)
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bea7b0961080..36240def9bf5 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1421,7 +1421,7 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec,
 /* update per_pin ELD from the given new ELD;
  * setup info frame and notification accordingly
  */
-static void update_eld(struct hda_codec *codec,
+static bool update_eld(struct hda_codec *codec,
 		       struct hdmi_spec_per_pin *per_pin,
 		       struct hdmi_eld *eld)
 {
@@ -1452,18 +1452,22 @@ static void update_eld(struct hda_codec *codec,
 		snd_hdmi_show_eld(codec, &eld->info);
 
 	eld_changed = (pin_eld->eld_valid != eld->eld_valid);
-	if (eld->eld_valid && pin_eld->eld_valid)
+	eld_changed |= (pin_eld->monitor_present != eld->monitor_present);
+	if (!eld_changed && eld->eld_valid && pin_eld->eld_valid)
 		if (pin_eld->eld_size != eld->eld_size ||
 		    memcmp(pin_eld->eld_buffer, eld->eld_buffer,
 			   eld->eld_size) != 0)
 			eld_changed = true;
 
-	pin_eld->monitor_present = eld->monitor_present;
-	pin_eld->eld_valid = eld->eld_valid;
-	pin_eld->eld_size = eld->eld_size;
-	if (eld->eld_valid)
-		memcpy(pin_eld->eld_buffer, eld->eld_buffer, eld->eld_size);
-	pin_eld->info = eld->info;
+	if (eld_changed) {
+		pin_eld->monitor_present = eld->monitor_present;
+		pin_eld->eld_valid = eld->eld_valid;
+		pin_eld->eld_size = eld->eld_size;
+		if (eld->eld_valid)
+			memcpy(pin_eld->eld_buffer, eld->eld_buffer,
+			       eld->eld_size);
+		pin_eld->info = eld->info;
+	}
 
 	/*
 	 * Re-setup pin and infoframe. This is needed e.g. when
@@ -1481,6 +1485,7 @@ static void update_eld(struct hda_codec *codec,
 			       SNDRV_CTL_EVENT_MASK_VALUE |
 			       SNDRV_CTL_EVENT_MASK_INFO,
 			       &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id);
+	return eld_changed;
 }
 
 /* update ELD and jack state via HD-audio verbs */
@@ -1582,6 +1587,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->temp_eld;
 	struct snd_jack *jack = NULL;
+	bool changed;
 	int size;
 
 	mutex_lock(&per_pin->lock);
@@ -1608,15 +1614,13 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
 	 * disconnected event. Jack must be fetched before update_eld()
 	 */
 	jack = pin_idx_to_jack(codec, per_pin);
-	update_eld(codec, per_pin, eld);
+	changed = update_eld(codec, per_pin, eld);
 	if (jack == NULL)
 		jack = pin_idx_to_jack(codec, per_pin);
-	if (jack == NULL)
-		goto unlock;
-	snd_jack_report(jack,
-			(eld->monitor_present && eld->eld_valid) ?
+	if (changed && jack)
+		snd_jack_report(jack,
+				(eld->monitor_present && eld->eld_valid) ?
 				SND_JACK_AVOUT : 0);
- unlock:
 	mutex_unlock(&per_pin->lock);
 }
 
@@ -2612,6 +2616,8 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
 /* precondition and allocation for Intel codecs */
 static int alloc_intel_hdmi(struct hda_codec *codec)
 {
+	int err;
+
 	/* requires i915 binding */
 	if (!codec->bus->core.audio_component) {
 		codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
@@ -2620,7 +2626,12 @@ static int alloc_intel_hdmi(struct hda_codec *codec)
 		return -ENODEV;
 	}
 
-	return alloc_generic_hdmi(codec);
+	err = alloc_generic_hdmi(codec);
+	if (err < 0)
+		return err;
+	/* no need to handle unsol events */
+	codec->patch_ops.unsol_event = NULL;
+	return 0;
 }
 
 /* parse and post-process for Intel codecs */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c1ddfd2fac52..36aee8ad2054 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1058,6 +1058,9 @@ static const struct snd_pci_quirk beep_white_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
 	SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1),
 	SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
+	/* blacklist -- no beep available */
+	SND_PCI_QUIRK(0x17aa, 0x309e, "Lenovo ThinkCentre M73", 0),
+	SND_PCI_QUIRK(0x17aa, 0x30a3, "Lenovo ThinkCentre M93", 0),
 	{}
 };
 
@@ -3755,6 +3758,72 @@ static void alc269_x101_hp_automute_hook(struct hda_codec *codec,
 			    vref);
 }
 
+/*
+ * Magic sequence to make Huawei Matebook X right speaker working (bko#197801)
+ */
+struct hda_alc298_mbxinit {
+	unsigned char value_0x23;
+	unsigned char value_0x25;
+};
+
+static void alc298_huawei_mbx_stereo_seq(struct hda_codec *codec,
+					 const struct hda_alc298_mbxinit *initval,
+					 bool first)
+{
+	snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x0);
+	alc_write_coef_idx(codec, 0x26, 0xb000);
+
+	if (first)
+		snd_hda_codec_write(codec, 0x21, 0, AC_VERB_GET_PIN_SENSE, 0x0);
+
+	snd_hda_codec_write(codec, 0x6, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80);
+	alc_write_coef_idx(codec, 0x26, 0xf000);
+	alc_write_coef_idx(codec, 0x23, initval->value_0x23);
+
+	if (initval->value_0x23 != 0x1e)
+		alc_write_coef_idx(codec, 0x25, initval->value_0x25);
+
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010);
+}
+
+static void alc298_fixup_huawei_mbx_stereo(struct hda_codec *codec,
+					   const struct hda_fixup *fix,
+					   int action)
+{
+	/* Initialization magic */
+	static const struct hda_alc298_mbxinit dac_init[] = {
+		{0x0c, 0x00}, {0x0d, 0x00}, {0x0e, 0x00}, {0x0f, 0x00},
+		{0x10, 0x00}, {0x1a, 0x40}, {0x1b, 0x82}, {0x1c, 0x00},
+		{0x1d, 0x00}, {0x1e, 0x00}, {0x1f, 0x00},
+		{0x20, 0xc2}, {0x21, 0xc8}, {0x22, 0x26}, {0x23, 0x24},
+		{0x27, 0xff}, {0x28, 0xff}, {0x29, 0xff}, {0x2a, 0x8f},
+		{0x2b, 0x02}, {0x2c, 0x48}, {0x2d, 0x34}, {0x2e, 0x00},
+		{0x2f, 0x00},
+		{0x30, 0x00}, {0x31, 0x00}, {0x32, 0x00}, {0x33, 0x00},
+		{0x34, 0x00}, {0x35, 0x01}, {0x36, 0x93}, {0x37, 0x0c},
+		{0x38, 0x00}, {0x39, 0x00}, {0x3a, 0xf8}, {0x38, 0x80},
+		{}
+	};
+	const struct hda_alc298_mbxinit *seq;
+
+	if (action != HDA_FIXUP_ACT_INIT)
+		return;
+
+	/* Start */
+	snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x00);
+	snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80);
+	alc_write_coef_idx(codec, 0x26, 0xf000);
+	alc_write_coef_idx(codec, 0x22, 0x31);
+	alc_write_coef_idx(codec, 0x23, 0x0b);
+	alc_write_coef_idx(codec, 0x25, 0x00);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010);
+
+	for (seq = dac_init; seq->value_0x23; seq++)
+		alc298_huawei_mbx_stereo_seq(codec, seq, seq == dac_init);
+}
+
 static void alc269_fixup_x101_headset_mic(struct hda_codec *codec,
 				     const struct hda_fixup *fix, int action)
 {
@@ -5780,6 +5849,7 @@ enum {
 	ALC255_FIXUP_DUMMY_LINEOUT_VERB,
 	ALC255_FIXUP_DELL_HEADSET_MIC,
 	ALC256_FIXUP_HUAWEI_MACH_WX9_PINS,
+	ALC298_FIXUP_HUAWEI_MBX_STEREO,
 	ALC295_FIXUP_HP_X360,
 	ALC221_FIXUP_HP_HEADSET_MIC,
 	ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
@@ -5800,6 +5870,7 @@ enum {
 	ALC256_FIXUP_ASUS_MIC_NO_PRESENCE,
 	ALC299_FIXUP_PREDATOR_SPK,
 	ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC,
+	ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -6089,6 +6160,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_MIC_MUTE_LED
 	},
+	[ALC298_FIXUP_HUAWEI_MBX_STEREO] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc298_fixup_huawei_mbx_stereo,
+		.chained = true,
+		.chain_id = ALC255_FIXUP_MIC_MUTE_LED
+	},
 	[ALC269_FIXUP_ASUS_X101_FUNC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_x101_headset_mic,
@@ -6850,6 +6927,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
 	},
+	[ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x04a11040 },
+			{ 0x21, 0x04211020 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7113,6 +7200,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
+	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
 
 #if 0
 	/* Below is a quirk table taken from the old code.
@@ -7280,6 +7368,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"},
 	{.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"},
 	{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
+	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
+	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 86495883ca3f..ab7d5f98e759 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -670,8 +670,13 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	ret = regmap_write(dev->regmap, MCHP_I2SMCC_MRA, mra);
-	if (ret < 0)
+	if (ret < 0) {
+		if (dev->gclk_use) {
+			clk_unprepare(dev->gclk);
+			dev->gclk_use = 0;
+		}
 		return ret;
+	}
 	return regmap_write(dev->regmap, MCHP_I2SMCC_MRB, mrb);
 }
 
@@ -686,31 +691,37 @@ static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream,
 		err = wait_event_interruptible_timeout(dev->wq_txrdy,
 						       dev->tx_rdy,
 						       msecs_to_jiffies(500));
+		if (err == 0) {
+			dev_warn_once(dev->dev,
+				      "Timeout waiting for Tx ready\n");
+			regmap_write(dev->regmap, MCHP_I2SMCC_IDRA,
+				     MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels));
+			dev->tx_rdy = 1;
+		}
 	} else {
 		err = wait_event_interruptible_timeout(dev->wq_rxrdy,
 						       dev->rx_rdy,
 						       msecs_to_jiffies(500));
-	}
-
-	if (err == 0) {
-		u32 idra;
-
-		dev_warn_once(dev->dev, "Timeout waiting for %s\n",
-			      is_playback ? "Tx ready" : "Rx ready");
-		if (is_playback)
-			idra = MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels);
-		else
-			idra = MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels);
-		regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, idra);
+		if (err == 0) {
+			dev_warn_once(dev->dev,
+				      "Timeout waiting for Rx ready\n");
+			regmap_write(dev->regmap, MCHP_I2SMCC_IDRA,
+				     MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels));
+			dev->rx_rdy = 1;
+		}
 	}
 
 	if (!mchp_i2s_mcc_is_running(dev)) {
 		regmap_write(dev->regmap, MCHP_I2SMCC_CR, MCHP_I2SMCC_CR_CKDIS);
 
 		if (dev->gclk_running) {
-			clk_disable_unprepare(dev->gclk);
+			clk_disable(dev->gclk);
 			dev->gclk_running = 0;
 		}
+		if (dev->gclk_use) {
+			clk_unprepare(dev->gclk);
+			dev->gclk_use = 0;
+		}
 	}
 
 	return 0;
@@ -809,6 +820,8 @@ static int mchp_i2s_mcc_dai_probe(struct snd_soc_dai *dai)
 
 	init_waitqueue_head(&dev->wq_txrdy);
 	init_waitqueue_head(&dev->wq_rxrdy);
+	dev->tx_rdy = 1;
+	dev->rx_rdy = 1;
 
 	snd_soc_dai_init_dma_data(dai, &dev->playback, &dev->capture);
 
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 6db002cc2058..96d04896193f 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -51,7 +51,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0);
-static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv,
+	0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0),
+	8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0),
+);
 
 static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
 	0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0),
@@ -89,7 +92,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 	SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL,
 		       4, 0, 3, 1, hpout_vol_tlv),
 	SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL,
-		       0, 4, 7, 0, hpmixer_gain_tlv),
+		       0, 4, 11, 0, hpmixer_gain_tlv),
 
 	SOC_ENUM("Playback Polarity", dacpol),
 	SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index 7d4940256914..91242b6f8ea7 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -495,6 +495,10 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev)
 
 static int hdac_hda_dev_remove(struct hdac_device *hdev)
 {
+	struct hdac_hda_priv *hda_pvt;
+
+	hda_pvt = dev_get_drvdata(&hdev->dev);
+	cancel_delayed_work_sync(&hda_pvt->codec.jackpoll_work);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index a6a4748c97f9..8e5e48f6a24b 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -31,6 +31,13 @@
 #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 },
@@ -123,6 +130,13 @@ enum  {
 	I2S_SCLK_STRENGTH_HIGH,
 };
 
+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 */
@@ -137,8 +151,109 @@ struct sgtl5000_priv {
 	u8 micbias_voltage;
 	u8 lrclk_strength;
 	u8 sclk_strength;
+	u16 mute_state[LAST_POWER_EVENT + 1];
 };
 
+static inline int hp_sel_input(struct snd_soc_component *component)
+{
+	return (snd_soc_component_read32(component, SGTL5000_CHIP_ANA_CTRL) &
+		SGTL5000_HP_SEL_MASK) >> SGTL5000_HP_SEL_SHIFT;
+}
+
+static inline u16 mute_output(struct snd_soc_component *component,
+			      u16 mute_mask)
+{
+	u16 mute_reg = snd_soc_component_read32(component,
+					      SGTL5000_CHIP_ANA_CTRL);
+
+	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)
+{
+	if (snd_soc_component_read32(component, SGTL5000_CHIP_ANA_POWER) &
+	    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)
+{
+	u16 ana_pwr = snd_soc_component_read32(component,
+					     SGTL5000_CHIP_ANA_POWER);
+
+	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
@@ -170,36 +285,46 @@ 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_component *component = snd_soc_dapm_to_component(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 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_component_update_bits(component, 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_component_read32(component, SGTL5000_CHIP_ANA_POWER) &
-				mask) != mask) {
-			snd_soc_component_update_bits(component, 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;
@@ -208,6 +333,41 @@ 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"
@@ -280,7 +440,10 @@ 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),
@@ -301,11 +464,12 @@ 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 */
@@ -1173,12 +1337,17 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component)
 					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_component_write(component, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
@@ -1288,6 +1457,7 @@ static int sgtl5000_probe(struct snd_soc_component *component)
 	int ret;
 	u16 reg;
 	struct sgtl5000_priv *sgtl5000 = snd_soc_component_get_drvdata(component);
+	unsigned int zcd_mask = SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN;
 
 	/* power up sgtl5000 */
 	ret = sgtl5000_set_power_regs(component);
@@ -1315,9 +1485,8 @@ static int sgtl5000_probe(struct snd_soc_component *component)
 	       0x1f);
 	snd_soc_component_write(component, SGTL5000_CHIP_PAD_STRENGTH, reg);
 
-	snd_soc_component_write(component, SGTL5000_CHIP_ANA_CTRL,
-			SGTL5000_HP_ZCD_EN |
-			SGTL5000_ADC_ZCD_EN);
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+		zcd_mask, zcd_mask);
 
 	snd_soc_component_update_bits(component, SGTL5000_CHIP_MIC_CTRL,
 			SGTL5000_BIAS_R_MASK,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 9b37e98da0db..26a4f6cd3288 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1553,7 +1553,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 	aic31xx->gpio_reset = devm_gpiod_get_optional(aic31xx->dev, "reset",
 						      GPIOD_OUT_LOW);
 	if (IS_ERR(aic31xx->gpio_reset)) {
-		dev_err(aic31xx->dev, "not able to acquire gpio\n");
+		if (PTR_ERR(aic31xx->gpio_reset) != -EPROBE_DEFER)
+			dev_err(aic31xx->dev, "not able to acquire gpio\n");
 		return PTR_ERR(aic31xx->gpio_reset);
 	}
 
@@ -1564,7 +1565,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 				      ARRAY_SIZE(aic31xx->supplies),
 				      aic31xx->supplies);
 	if (ret) {
-		dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
+		if (ret != -EPROBE_DEFER)
+			dev_err(aic31xx->dev,
+				"Failed to request supplies: %d\n", ret);
 		return ret;
 	}
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index fa862af25c1a..085855f9b08d 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	u32 wl = SSI_SxCCR_WL(sample_size);
 	int ret;
 
-	/*
-	 * SSI is properly configured if it is enabled and running in
-	 * the synchronous mode; Note that AC97 mode is an exception
-	 * that should set separate configurations for STCCR and SRCCR
-	 * despite running in the synchronous mode.
-	 */
-	if (ssi->streams && ssi->synchronous)
-		return 0;
-
 	if (fsl_ssi_is_i2s_master(ssi)) {
 		ret = fsl_ssi_set_bclk(substream, dai, hw_params);
 		if (ret)
@@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
+	/*
+	 * SSI is properly configured if it is enabled and running in
+	 * the synchronous mode; Note that AC97 mode is an exception
+	 * that should set separate configurations for STCCR and SRCCR
+	 * despite running in the synchronous mode.
+	 */
+	if (ssi->streams && ssi->synchronous)
+		return 0;
+
 	if (!fsl_ssi_is_ac97(ssi)) {
 		/*
 		 * Keep the ssi->i2s_net intact while having a local variable
diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c
index 0e8e0a7a11df..5854868650b9 100644
--- a/sound/soc/intel/common/sst-acpi.c
+++ b/sound/soc/intel/common/sst-acpi.c
@@ -141,11 +141,12 @@ static int sst_acpi_probe(struct platform_device *pdev)
 	}
 
 	platform_set_drvdata(pdev, sst_acpi);
+	mach->pdata = sst_pdata;
 
 	/* register machine driver */
 	sst_acpi->pdev_mach =
 		platform_device_register_data(dev, mach->drv_name, -1,
-					      sst_pdata, sizeof(*sst_pdata));
+					      mach, sizeof(*mach));
 	if (IS_ERR(sst_acpi->pdev_mach))
 		return PTR_ERR(sst_acpi->pdev_mach);
 
diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index ef5b66af1cd2..3a66121ee9bb 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -222,6 +222,8 @@ 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/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index b9b4a72a4334..b28a9c2b0380 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -188,7 +188,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf,
 	memset(d->fw_read_buff, 0, FW_REG_BUF);
 
 	if (w0_stat_sz > 0)
-		__iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
+		__ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
 
 	for (offset = 0; offset < FW_REG_SIZE; offset += 16) {
 		ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 1132109cb992..e01815cec6fd 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -225,7 +225,7 @@ int skl_nhlt_update_topology_bin(struct skl *skl)
 	struct hdac_bus *bus = skl_to_bus(skl);
 	struct device *dev = bus->dev;
 
-	dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n",
+	dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n",
 		nhlt->header.oem_id, nhlt->header.oem_table_id,
 		nhlt->header.oem_revision);
 
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index fce4e050a9b7..b9aacf3d3b29 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -30,6 +30,7 @@ struct rsnd_adg {
 	struct clk *clkout[CLKOUTMAX];
 	struct clk_onecell_data onecell;
 	struct rsnd_mod mod;
+	int clk_rate[CLKMAX];
 	u32 flags;
 	u32 ckr;
 	u32 rbga;
@@ -114,9 +115,9 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv,
 	unsigned int val, en;
 	unsigned int min, diff;
 	unsigned int sel_rate[] = {
-		clk_get_rate(adg->clk[CLKA]),	/* 0000: CLKA */
-		clk_get_rate(adg->clk[CLKB]),	/* 0001: CLKB */
-		clk_get_rate(adg->clk[CLKC]),	/* 0010: CLKC */
+		adg->clk_rate[CLKA],	/* 0000: CLKA */
+		adg->clk_rate[CLKB],	/* 0001: CLKB */
+		adg->clk_rate[CLKC],	/* 0010: CLKC */
 		adg->rbga_rate_for_441khz,	/* 0011: RBGA */
 		adg->rbgb_rate_for_48khz,	/* 0100: RBGB */
 	};
@@ -302,7 +303,7 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate)
 	 * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI.
 	 */
 	for_each_rsnd_clk(clk, adg, i) {
-		if (rate == clk_get_rate(clk))
+		if (rate == adg->clk_rate[i])
 			return sel_table[i];
 	}
 
@@ -369,10 +370,18 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 
 	for_each_rsnd_clk(clk, adg, i) {
 		ret = 0;
-		if (enable)
+		if (enable) {
 			ret = clk_prepare_enable(clk);
-		else
+
+			/*
+			 * We shouldn't use clk_get_rate() under
+			 * atomic context. Let's keep it when
+			 * rsnd_adg_clk_enable() was called
+			 */
+			adg->clk_rate[i] = clk_get_rate(adg->clk[i]);
+		} else {
 			clk_disable_unprepare(clk);
+		}
 
 		if (ret < 0)
 			dev_warn(dev, "can't use clk %d\n", i);
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 748f5f641002..d93db2c2b527 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -306,6 +306,12 @@ 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/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index b8b37f082309..0d8437b080bf 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -62,8 +62,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address)
 		address, resp);
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
-	/* snd_hdac_ext_bus_device_exit will use kfree to free hdev */
-	hda_priv = kzalloc(sizeof(*hda_priv), GFP_KERNEL);
+	hda_priv = devm_kzalloc(sdev->dev, sizeof(*hda_priv), GFP_KERNEL);
 	if (!hda_priv)
 		return -ENOMEM;
 
@@ -82,8 +81,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address)
 
 	return 0;
 #else
-	/* snd_hdac_ext_bus_device_exit will use kfree to free hdev */
-	hdev = kzalloc(sizeof(*hdev), GFP_KERNEL);
+	hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL);
 	if (!hdev)
 		return -ENOMEM;
 
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 334e9d59b1ba..3b8955e755b2 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -208,12 +208,11 @@ static int sof_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
+	spcm->prepared[substream->stream] = true;
+
 	/* save pcm hw_params */
 	memcpy(&spcm->params[substream->stream], params, sizeof(*params));
 
-	/* clear hw_params_upon_resume flag */
-	spcm->hw_params_upon_resume[substream->stream] = 0;
-
 	return ret;
 }
 
@@ -236,6 +235,9 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (!spcm)
 		return -EINVAL;
 
+	if (!spcm->prepared[substream->stream])
+		return 0;
+
 	dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id,
 		substream->stream);
 
@@ -258,6 +260,8 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (ret < 0)
 		dev_err(sdev->dev, "error: platform hw free failed\n");
 
+	spcm->prepared[substream->stream] = false;
+
 	return ret;
 }
 
@@ -278,11 +282,7 @@ static int sof_pcm_prepare(struct snd_pcm_substream *substream)
 	if (!spcm)
 		return -EINVAL;
 
-	/*
-	 * check if hw_params needs to be set-up again.
-	 * This is only needed when resuming from system sleep.
-	 */
-	if (!spcm->hw_params_upon_resume[substream->stream])
+	if (spcm->prepared[substream->stream])
 		return 0;
 
 	dev_dbg(sdev->dev, "pcm: prepare stream %d dir %d\n", spcm->pcm.pcm_id,
@@ -311,6 +311,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_sof_pcm *spcm;
 	struct sof_ipc_stream stream;
 	struct sof_ipc_reply reply;
+	bool reset_hw_params = false;
 	int ret;
 
 	/* nothing to do for BE */
@@ -351,6 +352,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
+		reset_hw_params = true;
 		break;
 	default:
 		dev_err(sdev->dev, "error: unhandled trigger cmd %d\n", cmd);
@@ -363,17 +365,17 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
 				 sizeof(stream), &reply, sizeof(reply));
 
-	if (ret < 0 || cmd != SNDRV_PCM_TRIGGER_SUSPEND)
+	if (ret < 0 || !reset_hw_params)
 		return ret;
 
 	/*
-	 * The hw_free op is usually called when the pcm stream is closed.
-	 * Since the stream is not closed during suspend, the DSP needs to be
-	 * notified explicitly to free pcm to prevent errors upon resume.
+	 * In case of stream is stopped, DSP must be reprogrammed upon
+	 * restart, so free PCM here.
 	 */
 	stream.hdr.size = sizeof(stream);
 	stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE;
 	stream.comp_id = spcm->stream[substream->stream].comp_id;
+	spcm->prepared[substream->stream] = false;
 
 	/* send IPC to the DSP */
 	return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
@@ -481,6 +483,7 @@ static int sof_pcm_open(struct snd_pcm_substream *substream)
 	spcm->stream[substream->stream].posn.host_posn = 0;
 	spcm->stream[substream->stream].posn.dai_posn = 0;
 	spcm->stream[substream->stream].substream = substream;
+	spcm->prepared[substream->stream] = false;
 
 	ret = snd_sof_pcm_platform_open(sdev, substream);
 	if (ret < 0)
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 278abfd10490..48c6d78d72e2 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -233,7 +233,7 @@ static int sof_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 
 			state = substream->runtime->status->state;
 			if (state == SNDRV_PCM_STATE_SUSPENDED)
-				spcm->hw_params_upon_resume[dir] = 1;
+				spcm->prepared[dir] = false;
 		}
 	}
 
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index 65d1bac4c6b8..6fd3df7c57a3 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -223,6 +223,9 @@ static void sof_pci_probe_complete(struct device *dev)
 	 */
 	pm_runtime_allow(dev);
 
+	/* mark last_busy for pm_runtime to make sure not suspend immediately */
+	pm_runtime_mark_last_busy(dev);
+
 	/* follow recommendation in pci-driver.c to decrement usage counter */
 	pm_runtime_put_noidle(dev);
 }
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index b8c0b2a22684..fa5cb7d2a660 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -297,7 +297,7 @@ struct snd_sof_pcm {
 	struct snd_sof_pcm_stream stream[2];
 	struct list_head list;	/* list in sdev pcm list */
 	struct snd_pcm_hw_params params[2];
-	int hw_params_upon_resume[2]; /* set up hw_params upon resume */
+	bool prepared[2]; /* PCM_PARAMS set successfully */
 };
 
 /* ALSA SOF Kcontrol device */
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 7fa5c61169db..ab8cb83c8b1a 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -222,10 +222,11 @@ static const struct sun4i_i2s_clk_div sun4i_i2s_mclk_div[] = {
 };
 
 static int sun4i_i2s_get_bclk_div(struct sun4i_i2s *i2s,
-				  unsigned int oversample_rate,
+				  unsigned long parent_rate,
+				  unsigned int sampling_rate,
 				  unsigned int word_size)
 {
-	int div = oversample_rate / word_size / 2;
+	int div = parent_rate / sampling_rate / word_size / 2;
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(sun4i_i2s_bclk_div); i++) {
@@ -315,8 +316,8 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	bclk_div = sun4i_i2s_get_bclk_div(i2s, oversample_rate,
-					  word_size);
+	bclk_div = sun4i_i2s_get_bclk_div(i2s, i2s->mclk_freq,
+					  rate, word_size);
 	if (bclk_div < 0) {
 		dev_err(dai->dev, "Unsupported BCLK divider: %d\n", bclk_div);
 		return -EINVAL;
diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c
index ee90e6c3937c..2ae582a99b63 100644
--- a/sound/soc/uniphier/aio-cpu.c
+++ b/sound/soc/uniphier/aio-cpu.c
@@ -424,8 +424,11 @@ int uniphier_aio_dai_suspend(struct snd_soc_dai *dai)
 {
 	struct uniphier_aio *aio = uniphier_priv(dai);
 
-	reset_control_assert(aio->chip->rst);
-	clk_disable_unprepare(aio->chip->clk);
+	aio->chip->num_wup_aios--;
+	if (!aio->chip->num_wup_aios) {
+		reset_control_assert(aio->chip->rst);
+		clk_disable_unprepare(aio->chip->clk);
+	}
 
 	return 0;
 }
@@ -439,13 +442,15 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai)
 	if (!aio->chip->active)
 		return 0;
 
-	ret = clk_prepare_enable(aio->chip->clk);
-	if (ret)
-		return ret;
+	if (!aio->chip->num_wup_aios) {
+		ret = clk_prepare_enable(aio->chip->clk);
+		if (ret)
+			return ret;
 
-	ret = reset_control_deassert(aio->chip->rst);
-	if (ret)
-		goto err_out_clock;
+		ret = reset_control_deassert(aio->chip->rst);
+		if (ret)
+			goto err_out_clock;
+	}
 
 	aio_iecout_set_enable(aio->chip, true);
 	aio_chip_init(aio->chip);
@@ -458,7 +463,7 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai)
 
 		ret = aio_init(sub);
 		if (ret)
-			goto err_out_clock;
+			goto err_out_reset;
 
 		if (!sub->setting)
 			continue;
@@ -466,11 +471,16 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai)
 		aio_port_reset(sub);
 		aio_src_reset(sub);
 	}
+	aio->chip->num_wup_aios++;
 
 	return 0;
 
+err_out_reset:
+	if (!aio->chip->num_wup_aios)
+		reset_control_assert(aio->chip->rst);
 err_out_clock:
-	clk_disable_unprepare(aio->chip->clk);
+	if (!aio->chip->num_wup_aios)
+		clk_disable_unprepare(aio->chip->clk);
 
 	return ret;
 }
@@ -619,6 +629,7 @@ int uniphier_aio_probe(struct platform_device *pdev)
 		return PTR_ERR(chip->rst);
 
 	chip->num_aios = chip->chip_spec->num_dais;
+	chip->num_wup_aios = chip->num_aios;
 	chip->aios = devm_kcalloc(dev,
 				  chip->num_aios, sizeof(struct uniphier_aio),
 				  GFP_KERNEL);
diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h
index ca6ccbae0ee8..a7ff7e556429 100644
--- a/sound/soc/uniphier/aio.h
+++ b/sound/soc/uniphier/aio.h
@@ -285,6 +285,7 @@ struct uniphier_aio_chip {
 
 	struct uniphier_aio *aios;
 	int num_aios;
+	int num_wup_aios;
 	struct uniphier_aio_pll *plls;
 	int num_plls;
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index e4bbf79de956..33cd26763c0e 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -457,6 +457,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 	}
 	ep = get_endpoint(alts, 1)->bEndpointAddress;
 	if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	    get_endpoint(alts, 0)->bSynchAddress != 0 &&
 	    ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
 	     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
 		dev_err(&dev->dev,