Merge branch 'locking/urgent' into locking/core, to pick up fixes

Signed-off-by: Ingo Molnar <mingo@kernel.org>

77 files changed, 563 insertions, 263 deletions

diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 3788906421a7..fe27034f2846 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
 		snd_BUG();
 		return -EINVAL;
 	}
-	if (snd_BUG_ON(!snd_pcm_format_linear(format->format)))
-		return -ENXIO;
+	if (!snd_pcm_format_linear(format->format))
+		return -EINVAL;
 
 	err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion",
 				   src_format, dst_format,
diff --git a/sound/core/timer.c b/sound/core/timer.c
index d9f85f2d66a3..6e27d87b18ed 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -816,9 +816,9 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer,
  * timer tasklet
  *
  */
-static void snd_timer_tasklet(unsigned long arg)
+static void snd_timer_tasklet(struct tasklet_struct *t)
 {
-	struct snd_timer *timer = (struct snd_timer *) arg;
+	struct snd_timer *timer = from_tasklet(timer, t, task_queue);
 	unsigned long flags;
 
 	if (timer->card && timer->card->shutdown) {
@@ -967,8 +967,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid,
 	INIT_LIST_HEAD(&timer->ack_list_head);
 	INIT_LIST_HEAD(&timer->sack_list_head);
 	spin_lock_init(&timer->lock);
-	tasklet_init(&timer->task_queue, snd_timer_tasklet,
-		     (unsigned long)timer);
+	tasklet_setup(&timer->task_queue, snd_timer_tasklet);
 	timer->max_instances = 1000; /* default limit per timer */
 	if (card != NULL) {
 		timer->module = card->module;
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index f8586f75441d..ee1c428b1fd3 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -64,7 +64,7 @@
 #define IT_PKT_HEADER_SIZE_CIP		8 // For 2 CIP header.
 #define IT_PKT_HEADER_SIZE_NO_CIP	0 // Nothing.
 
-static void pcm_period_tasklet(unsigned long data);
+static void pcm_period_tasklet(struct tasklet_struct *t);
 
 /**
  * amdtp_stream_init - initialize an AMDTP stream structure
@@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->flags = flags;
 	s->context = ERR_PTR(-1);
 	mutex_init(&s->mutex);
-	tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s);
+	tasklet_setup(&s->period_tasklet, pcm_period_tasklet);
 	s->packet_index = 0;
 
 	init_waitqueue_head(&s->callback_wait);
@@ -441,9 +441,9 @@ static void update_pcm_pointers(struct amdtp_stream *s,
 	}
 }
 
-static void pcm_period_tasklet(unsigned long data)
+static void pcm_period_tasklet(struct tasklet_struct *t)
 {
-	struct amdtp_stream *s = (void *)data;
+	struct amdtp_stream *s = from_tasklet(s, t, period_tasklet);
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 
 	if (pcm)
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index c84b913a9fe0..ab8408966ec3 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2");
 #define VENDOR_DIGIDESIGN	0x00a07e
 #define MODEL_CONSOLE		0x000001
 #define MODEL_RACK		0x000002
+#define SPEC_VERSION		0x000001
 
 static int name_card(struct snd_dg00x *dg00x)
 {
@@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = {
 	/* Both of 002/003 use the same ID. */
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_VERSION |
 			       IEEE1394_MATCH_MODEL_ID,
 		.vendor_id = VENDOR_DIGIDESIGN,
+		.version = SPEC_VERSION,
 		.model_id = MODEL_CONSOLE,
 	},
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_VERSION |
 			       IEEE1394_MATCH_MODEL_ID,
 		.vendor_id = VENDOR_DIGIDESIGN,
+		.version = SPEC_VERSION,
 		.model_id = MODEL_RACK,
 	},
 	{}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index 5dac0d9fc58e..75f2edd8e78f 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = {
 		.midi_capture_ports = 2,
 		.midi_playback_ports = 4,
 	},
-	// This kernel module doesn't support FE-8 because the most of features
-	// can be implemented in userspace without any specific support of this
-	// module.
 };
 
 static int identify_model(struct snd_tscm *tscm)
@@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit)
 }
 
 static const struct ieee1394_device_id snd_tscm_id_table[] = {
+	// Tascam, FW-1884.
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
+		.vendor_id = 0x00022e,
+		.specifier_id = 0x00022e,
+		.version = 0x800000,
+	},
+	// Tascam, FE-8 (.version = 0x800001)
+	// This kernel module doesn't support FE-8 because the most of features
+	// can be implemented in userspace without any specific support of this
+	// module.
+	//
+	// .version = 0x800002 is unknown.
+	//
+	// Tascam, FW-1082.
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
+		.vendor_id = 0x00022e,
+		.specifier_id = 0x00022e,
+		.version = 0x800003,
+	},
+	// Tascam, FW-1804.
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
-			       IEEE1394_MATCH_SPECIFIER_ID,
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
 		.vendor_id = 0x00022e,
 		.specifier_id = 0x00022e,
+		.version = 0x800004,
 	},
 	{}
 };
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 333220f0f8af..3e9e9ac804f6 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init);
 void snd_hdac_device_exit(struct hdac_device *codec)
 {
 	pm_runtime_put_noidle(&codec->dev);
+	/* keep balance of runtime PM child_count in parent device */
+	pm_runtime_set_suspended(&codec->dev);
 	snd_hdac_bus_remove_device(codec->bus, codec);
 	kfree(codec->vendor_name);
 	kfree(codec->chip_name);
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index 99aec7349167..1c5114dedda9 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -54,7 +54,7 @@ static const struct config_entry config_table[] = {
 #endif
 /*
  * Apollolake (Broxton-P)
- * the legacy HDaudio driver is used except on Up Squared (SOF) and
+ * the legacy HDAudio driver is used except on Up Squared (SOF) and
  * Chromebooks (SST)
  */
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE)
@@ -89,7 +89,7 @@ static const struct config_entry config_table[] = {
 	},
 #endif
 /*
- * Skylake and Kabylake use legacy HDaudio driver except for Google
+ * Skylake and Kabylake use legacy HDAudio driver except for Google
  * Chromebooks (SST)
  */
 
@@ -135,7 +135,7 @@ static const struct config_entry config_table[] = {
 #endif
 
 /*
- * Geminilake uses legacy HDaudio driver except for Google
+ * Geminilake uses legacy HDAudio driver except for Google
  * Chromebooks
  */
 /* Geminilake */
@@ -157,7 +157,7 @@ static const struct config_entry config_table[] = {
 
 /*
  * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy
- * HDaudio driver except for Google Chromebooks and when DMICs are
+ * HDAudio driver except for Google Chromebooks and when DMICs are
  * present. Two cases are required since Coreboot does not expose NHLT
  * tables.
  *
@@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
 	if (pci->class == 0x040300)
 		return SND_INTEL_DSP_DRIVER_LEGACY;
 	if (pci->class != 0x040100 && pci->class != 0x040380) {
-		dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class);
+		dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class);
 		return SND_INTEL_DSP_DRIVER_LEGACY;
 	}
 
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 023c35a2a951..35e76480306e 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -921,10 +921,10 @@ static void snd_card_asihpi_timer_function(struct timer_list *t)
 		add_timer(&dpcm->timer);
 }
 
-static void snd_card_asihpi_int_task(unsigned long data)
+static void snd_card_asihpi_int_task(struct tasklet_struct *t)
 {
-	struct hpi_adapter *a = (struct hpi_adapter *)data;
-	struct snd_card_asihpi *asihpi;
+	struct snd_card_asihpi *asihpi = from_tasklet(asihpi, t, t);
+	struct hpi_adapter *a = asihpi->hpi;
 
 	WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
 	asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
@@ -2871,8 +2871,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	if (hpi->interrupt_mode) {
 		asihpi->pcm_start = snd_card_asihpi_pcm_int_start;
 		asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop;
-		tasklet_init(&asihpi->t, snd_card_asihpi_int_task,
-			(unsigned long)hpi);
+		tasklet_setup(&asihpi->t, snd_card_asihpi_int_task);
 		hpi->interrupt_callback = snd_card_asihpi_isr;
 	} else {
 		asihpi->pcm_start = snd_card_asihpi_pcm_timer_start;
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 496dcde9715d..9790f5108a16 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -343,7 +343,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 	struct hpi_message hm;
 	struct hpi_response hr;
 	struct hpi_adapter adapter;
-	struct hpi_pci pci;
+	struct hpi_pci pci = { 0 };
 
 	memset(&adapter, 0, sizeof(adapter));
 
@@ -499,7 +499,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 	return 0;
 
 err:
-	for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) {
+	while (--idx >= 0) {
 		if (pci.ap_mem_base[idx]) {
 			iounmap(pci.ap_mem_base[idx]);
 			pci.ap_mem_base[idx] = NULL;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 70d775ff967e..c189f70c82cb 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id,
 		else
 			/* Power down */
 			chip->spi_dac_reg[reg] |= bit;
-		return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+		if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0)
+			return -ENXIO;
 	}
 	return 0;
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e34a4d5d047c..36a9dbc33aa0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2127,9 +2127,10 @@ static int azx_probe(struct pci_dev *pci,
 	 */
 	if (dmic_detect) {
 		err = snd_intel_dsp_driver_probe(pci);
-		if (err != SND_INTEL_DSP_DRIVER_ANY &&
-		    err != SND_INTEL_DSP_DRIVER_LEGACY)
+		if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) {
+			dev_dbg(&pci->dev, "HDAudio driver not selected, aborting probe\n");
 			return -ENODEV;
+		}
 	} else {
 		dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n");
 	}
@@ -2745,8 +2746,6 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
 	/* Zhaoxin */
 	{ PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN },
-	/* Loongson */
-	{ PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index c94553bcca88..70164d1428d4 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -179,6 +179,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev)
 	struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
 
 	if (chip && chip->running) {
+		/* enable controller wake up event */
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+			   STATESTS_INT_MASK);
+
 		azx_stop_chip(chip);
 		azx_enter_link_reset(chip);
 	}
@@ -200,6 +204,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev)
 	if (chip && chip->running) {
 		hda_tegra_init(hda);
 		azx_init_chip(chip, 1);
+		/* disable controller wake up event*/
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			   ~STATESTS_INT_MASK);
 	}
 
 	return 0;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index b8c8490e568b..402050088090 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
 			       hda_nid_t cvt_nid)
 {
 	if (per_pin) {
+		haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid);
 		snd_hda_set_dev_select(codec, per_pin->pin_nid,
 			       per_pin->dev_id);
 		intel_verify_pin_cvt_connect(codec, per_pin);
@@ -3734,6 +3735,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec)
 
 static int patch_tegra_hdmi(struct hda_codec *codec)
 {
+	struct hdmi_spec *spec;
 	int err;
 
 	err = patch_generic_hdmi(codec);
@@ -3741,6 +3743,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec)
 		return err;
 
 	codec->patch_ops.build_pcms = tegra_hdmi_build_pcms;
+	spec = codec->spec;
+	spec->chmap.ops.chmap_cea_alloc_validate_get_type =
+		nvhdmi_chmap_cea_alloc_validate_get_type;
+	spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
 
 	return 0;
 }
@@ -4263,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI",	patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI",	patch_generic_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a1fa983d2a94..d4f17b465892 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3427,7 +3427,11 @@ static void alc256_shutup(struct hda_codec *codec)
 
 	/* 3k pull low control for Headset jack. */
 	/* NOTE: call this before clearing the pin, otherwise codec stalls */
-	alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
+	/* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
+	 * when booting with headset plugged. So skip setting it for the codec alc257
+	 */
+	if (codec->core.vendor_id != 0x10ec0257)
+		alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
 
 	if (!spec->no_shutup_pins)
 		snd_hda_codec_write(codec, hp_pin, 0,
@@ -5867,6 +5871,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec,
 	}
 }
 
+/* Quirk for Thinkpad X1 7th and 8th Gen
+ * The following fixed routing needed
+ * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly
+ * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC
+ * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp
+ */
+static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec,
+					  const struct hda_fixup *fix, int action)
+{
+	static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+	static const hda_nid_t preferred_pairs[] = {
+		0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0
+	};
+	struct alc_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+		spec->gen.preferred_dacs = preferred_pairs;
+		break;
+	case HDA_FIXUP_ACT_BUILD:
+		/* The generic parser creates somewhat unintuitive volume ctls
+		 * with the fixed routing above, and the shared DAC2 may be
+		 * confusing for PA.
+		 * Rename those to unique names so that PA doesn't touch them
+		 * and use only Master volume.
+		 */
+		rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume");
+		rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume");
+		break;
+	}
+}
+
 static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec,
 					 const struct hda_fixup *fix,
 					 int action)
@@ -5959,6 +5996,40 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
 		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
 }
 
+
+static void alc294_gx502_toggle_output(struct hda_codec *codec,
+					struct hda_jack_callback *cb)
+{
+	/* The Windows driver sets the codec up in a very different way where
+	 * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it
+	 */
+	if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT)
+		alc_write_coef_idx(codec, 0x10, 0x8a20);
+	else
+		alc_write_coef_idx(codec, 0x10, 0x0a20);
+}
+
+static void alc294_fixup_gx502_hp(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+	/* Pin 0x21: headphones/headset mic */
+	if (!is_jack_detectable(codec, 0x21))
+		return;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_jack_detect_enable_callback(codec, 0x21,
+				alc294_gx502_toggle_output);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		/* Make sure to start in a correct state, i.e. if
+		 * headphones have been plugged in before powering up the system
+		 */
+		alc294_gx502_toggle_output(codec, NULL);
+		break;
+	}
+}
+
 static void  alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
 			      const struct hda_fixup *fix, int action)
 {
@@ -5983,6 +6054,7 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
 #include "hp_x360_helper.c"
 
 enum {
+	ALC269_FIXUP_GPIO2,
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
 	ALC269_FIXUP_DELL_M101Z,
@@ -6135,9 +6207,13 @@ enum {
 	ALC289_FIXUP_DUAL_SPK,
 	ALC294_FIXUP_SPK2_TO_DAC1,
 	ALC294_FIXUP_ASUS_DUAL_SPK,
+	ALC285_FIXUP_THINKPAD_X1_GEN7,
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_HPE,
 	ALC294_FIXUP_ASUS_COEF_1B,
+	ALC294_FIXUP_ASUS_GX502_HP,
+	ALC294_FIXUP_ASUS_GX502_PINS,
+	ALC294_FIXUP_ASUS_GX502_VERBS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
@@ -6156,9 +6232,14 @@ enum {
 	ALC269_FIXUP_LEMOTE_A1802,
 	ALC269_FIXUP_LEMOTE_A190X,
 	ALC256_FIXUP_INTEL_NUC8_RUGGED,
+	ALC255_FIXUP_XIAOMI_HEADSET_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
+	[ALC269_FIXUP_GPIO2] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio2,
+	},
 	[ALC269_FIXUP_SONY_VAIO] = {
 		.type = HDA_FIXUP_PINCTLS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -6978,6 +7059,8 @@ static const struct hda_fixup alc269_fixups[] = {
 	[ALC233_FIXUP_LENOVO_MULTI_CODECS] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc233_alc662_fixup_lenovo_dual_codecs,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_GPIO2
 	},
 	[ALC233_FIXUP_ACER_HEADSET_MIC] = {
 		.type = HDA_FIXUP_VERBS,
@@ -7280,11 +7363,17 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_SPK2_TO_DAC1
 	},
+	[ALC285_FIXUP_THINKPAD_X1_GEN7] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_thinkpad_x1_gen7,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_THINKPAD_ACPI
+	},
 	[ALC285_FIXUP_THINKPAD_HEADSET_JACK] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_headset_jack,
 		.chained = true,
-		.chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1
+		.chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7
 	},
 	[ALC294_FIXUP_ASUS_HPE] = {
 		.type = HDA_FIXUP_VERBS,
@@ -7297,6 +7386,33 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
 	},
+	[ALC294_FIXUP_ASUS_GX502_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11050 }, /* front HP mic */
+			{ 0x1a, 0x01a11830 }, /* rear external mic */
+			{ 0x21, 0x03211020 }, /* front HP out */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_VERBS
+	},
+	[ALC294_FIXUP_ASUS_GX502_VERBS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* set 0x15 to HP-OUT ctrl */
+			{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+			/* unmute the 0x15 amp */
+			{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_HP
+	},
+	[ALC294_FIXUP_ASUS_GX502_HP] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc294_fixup_gx502_hp,
+	},
 	[ALC294_FIXUP_ASUS_COEF_1B] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -7486,6 +7602,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE
 	},
+	[ALC255_FIXUP_XIAOMI_HEADSET_MIC] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC289_FIXUP_ASUS_GA401
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7670,6 +7796,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
 	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
+	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -7695,7 +7822,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
-	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
 	SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
@@ -7781,6 +7909,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
 	SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
+	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
@@ -7958,6 +8087,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
 	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
 	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
+	{.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index b4f300281822..098c69b3b7aa 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1070,9 +1070,9 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval,
 	return 0;
 }
 
-static void riptide_handleirq(unsigned long dev_id)
+static void riptide_handleirq(struct tasklet_struct *t)
 {
-	struct snd_riptide *chip = (void *)dev_id;
+	struct snd_riptide *chip = from_tasklet(chip, t, riptide_tq);
 	struct cmdif *cif = chip->cif;
 	struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1];
 	struct snd_pcm_runtime *runtime;
@@ -1843,7 +1843,7 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci,
 	chip->received_irqs = 0;
 	chip->handled_irqs = 0;
 	chip->cif = NULL;
-	tasklet_init(&chip->riptide_tq, riptide_handleirq, (unsigned long)chip);
+	tasklet_setup(&chip->riptide_tq, riptide_handleirq);
 
 	if ((chip->res_port =
 	     request_region(chip->port, 64, "RIPTIDE")) == NULL) {
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 227aece17e39..dda56ecfd33b 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -3791,9 +3791,9 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 	return 0;
 }
 
-static void hdsp_midi_tasklet(unsigned long arg)
+static void hdsp_midi_tasklet(struct tasklet_struct *t)
 {
-	struct hdsp *hdsp = (struct hdsp *)arg;
+	struct hdsp *hdsp = from_tasklet(hdsp, t, midi_tasklet);
 
 	if (hdsp->midi[0].pending)
 		snd_hdsp_midi_input_read (&hdsp->midi[0]);
@@ -5182,7 +5182,7 @@ static int snd_hdsp_create(struct snd_card *card,
 
 	spin_lock_init(&hdsp->lock);
 
-	tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp);
+	tasklet_setup(&hdsp->midi_tasklet, hdsp_midi_tasklet);
 
 	pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev);
 	hdsp->firmware_rev &= 0xff;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0fa49f4d15cf..572350aaf18d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -2169,9 +2169,9 @@ static int snd_hdspm_create_midi(struct snd_card *card,
 }
 
 
-static void hdspm_midi_tasklet(unsigned long arg)
+static void hdspm_midi_tasklet(struct tasklet_struct *t)
 {
-	struct hdspm *hdspm = (struct hdspm *)arg;
+	struct hdspm *hdspm = from_tasklet(hdspm, t, midi_tasklet);
 	int i = 0;
 
 	while (i < hdspm->midiPorts) {
@@ -6836,8 +6836,7 @@ static int snd_hdspm_create(struct snd_card *card,
 
 	}
 
-	tasklet_init(&hdspm->midi_tasklet,
-			hdspm_midi_tasklet, (unsigned long) hdspm);
+	tasklet_setup(&hdspm->midi_tasklet, hdspm_midi_tasklet);
 
 
 	if (hdspm->io_type != MADIface) {
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index b8161a08f2ca..58bb49fff184 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -227,14 +227,14 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
 	switch (filltype) {
 	case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL:
 		silent = 1;
-		/* intentionally fall thru */
+		fallthrough;
 	case SND_PS3_DMA_FILLTYPE_FIRSTFILL:
 		ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS;
 		break;
 
 	case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING:
 		silent = 1;
-		/* intentionally fall thru */
+		fallthrough;
 	case SND_PS3_DMA_FILLTYPE_RUNNING:
 		ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY;
 		break;
diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 3cb63886195f..04acc18f2d72 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -536,7 +536,7 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
 		/* cpu is BCLK master */
 		mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT;
 		set_divs = 1;
-		/* fall through */
+		fallthrough;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		/* cpu is slave */
 		mra |= MCHP_I2SMCC_MRA_MODE_SLAVE;
diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c
index c0a28f06b09a..298689a07168 100644
--- a/sound/soc/codecs/jz4770.c
+++ b/sound/soc/codecs/jz4770.c
@@ -202,7 +202,7 @@ static int jz4770_codec_set_bias_level(struct snd_soc_component *codec,
 				   REG_CR_VIC_SB_SLEEP, REG_CR_VIC_SB_SLEEP);
 		regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC,
 				   REG_CR_VIC_SB, REG_CR_VIC_SB);
-	/* fall-through */
+		fallthrough;
 	default:
 		break;
 	}
diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c
index 5fe724728e84..e4675cfff7b2 100644
--- a/sound/soc/codecs/max98373-sdw.c
+++ b/sound/soc/codecs/max98373-sdw.c
@@ -838,8 +838,8 @@ static int max98373_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	regmap = devm_regmap_init_sdw(slave, &max98373_sdw_regmap);
-	if (!regmap)
-		return -EINVAL;
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
 
 	return max98373_init(slave, regmap);
 }
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index f0da55901dcb..b8845f45549e 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,7 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		priv->tdm_offset += 1;
-		/* fall through */
+		fallthrough;
 		/* DSP_A uses the same basic config as DSP_B
 		 * except we need to shift the TDM output by one BCK cycle
 		 */
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 5e445fee4ef5..821e7395f90f 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -306,6 +306,13 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai,
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component);
 	int ret;
 
+	/*
+	 * Some sound card sets 0 Hz as reset,
+	 * but it is impossible to set. Ignore it here
+	 */
+	if (freq == 0)
+		return 0;
+
 	if (freq > PCM3168A_MAX_SYSCLK)
 		return -EINVAL;
 
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
index b0ba0d2acbdd..56e952a904a3 100644
--- a/sound/soc/codecs/rt1308-sdw.c
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -684,8 +684,8 @@ static int rt1308_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	regmap = devm_regmap_init_sdw(slave, &rt1308_sdw_regmap);
-	if (!regmap)
-		return -EINVAL;
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
 
 	rt1308_sdw_init(&slave->dev, regmap, slave);
 
diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c
index 4d14048d1197..1d24bf040718 100644
--- a/sound/soc/codecs/rt700-sdw.c
+++ b/sound/soc/codecs/rt700-sdw.c
@@ -452,8 +452,8 @@ static int rt700_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	sdw_regmap = devm_regmap_init_sdw(slave, &rt700_sdw_regmap);
-	if (!sdw_regmap)
-		return -EINVAL;
+	if (IS_ERR(sdw_regmap))
+		return PTR_ERR(sdw_regmap);
 
 	regmap = devm_regmap_init(&slave->dev, NULL,
 		&slave->dev, &rt700_regmap);
diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c
index 45b928954b58..7efff130a638 100644
--- a/sound/soc/codecs/rt711-sdw.c
+++ b/sound/soc/codecs/rt711-sdw.c
@@ -452,8 +452,8 @@ static int rt711_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	sdw_regmap = devm_regmap_init_sdw(slave, &rt711_sdw_regmap);
-	if (!sdw_regmap)
-		return -EINVAL;
+	if (IS_ERR(sdw_regmap))
+		return PTR_ERR(sdw_regmap);
 
 	regmap = devm_regmap_init(&slave->dev, NULL,
 		&slave->dev, &rt711_regmap);
diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c
index d11b23d6b240..68a36739f1b0 100644
--- a/sound/soc/codecs/rt715-sdw.c
+++ b/sound/soc/codecs/rt715-sdw.c
@@ -527,8 +527,8 @@ static int rt715_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	sdw_regmap = devm_regmap_init_sdw(slave, &rt715_sdw_regmap);
-	if (!sdw_regmap)
-		return -EINVAL;
+	if (IS_ERR(sdw_regmap))
+		return PTR_ERR(sdw_regmap);
 
 	regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev,
 		&rt715_regmap);
diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
index 5cd50d841177..8efe20605f9b 100644
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -842,6 +842,18 @@ static int adcx140_codec_probe(struct snd_soc_component *component)
 	if (ret)
 		goto out;
 
+	if (adcx140->supply_areg == NULL)
+		sleep_cfg_val |= ADCX140_AREG_INTERNAL;
+
+	ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val);
+	if (ret) {
+		dev_err(adcx140->dev, "setting sleep config failed %d\n", ret);
+		goto out;
+	}
+
+	/* 8.4.3: Wait >= 1ms after entering active mode. */
+	usleep_range(1000, 100000);
+
 	pdm_count = device_property_count_u32(adcx140->dev,
 					      "ti,pdm-edge-select");
 	if (pdm_count <= ADCX140_NUM_PDM_EDGES && pdm_count > 0) {
@@ -889,18 +901,6 @@ static int adcx140_codec_probe(struct snd_soc_component *component)
 	if (ret)
 		goto out;
 
-	if (adcx140->supply_areg == NULL)
-		sleep_cfg_val |= ADCX140_AREG_INTERNAL;
-
-	ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val);
-	if (ret) {
-		dev_err(adcx140->dev, "setting sleep config failed %d\n", ret);
-		goto out;
-	}
-
-	/* 8.4.3: Wait >= 1ms after entering active mode. */
-	usleep_range(1000, 100000);
-
 	ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG,
 				ADCX140_MIC_BIAS_VAL_MSK |
 				ADCX140_MIC_BIAS_VREF_MSK, bias_cfg);
@@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c,
 	if (!adcx140)
 		return -ENOMEM;
 
+	adcx140->dev = &i2c->dev;
+
 	adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev,
 						      "reset", GPIOD_OUT_LOW);
 	if (IS_ERR(adcx140->gpio_reset))
@@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c,
 			ret);
 		return ret;
 	}
-	adcx140->dev = &i2c->dev;
+
 	i2c_set_clientdata(i2c, adcx140);
 
 	return devm_snd_soc_register_component(&i2c->dev,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 038be667c1a6..fc9ea198ac79 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3514,6 +3514,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(component->dev);
+
 	switch (micbias) {
 	case 1:
 		micdet = &wm8994->micdet[0];
@@ -3561,6 +3563,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 
 	snd_soc_dapm_sync(dapm);
 
+	pm_runtime_put(component->dev);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8994_mic_detect);
@@ -3932,6 +3936,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(component->dev);
+
 	if (jack) {
 		snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS");
 		snd_soc_dapm_sync(dapm);
@@ -4000,6 +4006,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		snd_soc_dapm_sync(dapm);
 	}
 
+	pm_runtime_put(component->dev);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8958_mic_detect);
@@ -4193,11 +4201,13 @@ static int wm8994_component_probe(struct snd_soc_component *component)
 			wm8994->hubs.dcs_readback_mode = 2;
 			break;
 		}
+		wm8994->hubs.micd_scthr = true;
 		break;
 
 	case WM8958:
 		wm8994->hubs.dcs_readback_mode = 1;
 		wm8994->hubs.hp_startup_mode = 1;
+		wm8994->hubs.micd_scthr = true;
 
 		switch (control->revision) {
 		case 0:
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 891effe220fe..0c881846f485 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component,
 		snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL,
 				    WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
 
+	if (!hubs->micd_scthr)
+		return 0;
+
 	snd_soc_component_update_bits(component, WM8993_MICBIAS,
 			    WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
 			    WM8993_MICB1_LVL | WM8993_MICB2_LVL,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 4b8e5f0d6e32..988b29e63060 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -27,6 +27,7 @@ struct wm_hubs_data {
 	int hp_startup_mode;
 	int series_startup;
 	int no_series_update;
+	bool micd_scthr;
 
 	bool no_cache_dac_hp_direct;
 	struct list_head dcs_cache;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 4ae36099ae82..79b861afd986 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -708,9 +708,9 @@ static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx)
 			   ESAI_xFCR_xFR, 0);
 }
 
-static void fsl_esai_hw_reset(unsigned long arg)
+static void fsl_esai_hw_reset(struct tasklet_struct *t)
 {
-	struct fsl_esai *esai_priv = (struct fsl_esai *)arg;
+	struct fsl_esai *esai_priv = from_tasklet(esai_priv, t, task);
 	bool tx = true, rx = false, enabled[2];
 	unsigned long lock_flags;
 	u32 tfcr, rfcr;
@@ -1070,8 +1070,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	tasklet_init(&esai_priv->task, fsl_esai_hw_reset,
-		     (unsigned long)esai_priv);
+	tasklet_setup(&esai_priv->task, fsl_esai_hw_reset);
 
 	pm_runtime_enable(&pdev->dev);
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d8b9c6547142..404be27c15fe 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -898,7 +898,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
 					"missing baudclk for master mode\n");
 				return -EINVAL;
 			}
-			/* fall through */
+			fallthrough;
 		case SND_SOC_DAIFMT_CBM_CFS:
 			ssi->i2s_net |= SSI_SCR_I2S_MODE_MASTER;
 			break;
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index fd5dcd6b9f85..907f5f1f7b44 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -261,13 +261,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U16_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bits = HII2S_BITS_16;
 		break;
 	case SNDRV_PCM_FORMAT_U24_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		bits = HII2S_BITS_24;
 		break;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index b1cac7abdc0a..fba2c795ce0d 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -333,6 +333,17 @@ static int sst_media_open(struct snd_pcm_substream *substream,
 	if (ret_val < 0)
 		goto out_power_up;
 
+	/*
+	 * Make sure the period to be multiple of 1ms to align the
+	 * design of firmware. Apply same rule to buffer size to make
+	 * sure alsa could always find a value for period size
+	 * regardless the buffer size given by user space.
+	 */
+	snd_pcm_hw_constraint_step(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 48);
+	snd_pcm_hw_constraint_step(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 48);
+
 	/* Make sure, that the period size is always even */
 	snd_pcm_hw_constraint_step(substream->runtime, 0,
 			   SNDRV_PCM_HW_PARAM_PERIODS, 2);
diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
index 54a66cc6db89..d2cda33b65d5 100644
--- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c
+++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
@@ -181,7 +181,7 @@ static int sst_byt_pcm_trigger(struct snd_soc_component *component,
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		pdata->restore_stream = false;
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		sst_byt_stream_pause(byt, pcm_data->stream);
 		break;
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 414ae4bb5224..7ae34b49815c 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -573,7 +573,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 			break;
 		default:
 			dev_err(dev, "get speaker GPIO failed: %d\n", ret);
-			/* fall through */
+			fallthrough;
 		case -EPROBE_DEFER:
 			return ret;
 		}
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 479992f4e97a..fc202747ba83 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -591,6 +591,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		/* MPMAN MPWIN895CL */
 		.matches = {
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 4e2897596cea..688b5e0a49e3 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -1009,7 +1009,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 			default:
 				dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n",
 					ret_val);
-				/* fall through */
+				fallthrough;
 			case -EPROBE_DEFER:
 				put_device(codec_dev);
 				return ret_val;
@@ -1029,7 +1029,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 			default:
 				dev_err(&pdev->dev, "Failed to get hp-detect GPIO: %d\n",
 					ret_val);
-				/* fall through */
+				fallthrough;
 			case -EPROBE_DEFER:
 				put_device(codec_dev);
 				return ret_val;
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index ca4900036ead..bc50eda297ab 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -181,7 +181,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card)
 	struct snd_soc_dai *dai;
 
 	for_each_card_rtds(card, rtd) {
-		if (!strstr(rtd->dai_link->codecs->name, "ehdaudio"))
+		if (!strstr(rtd->dai_link->codecs->name, "ehdaudio0D0"))
 			continue;
 		dai = asoc_rtd_to_codec(rtd, 0);
 		hda_pvt = snd_soc_component_get_drvdata(dai->component);
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
index 1a6961592029..b6e63ea13d64 100644
--- a/sound/soc/intel/boards/sof_maxim_common.c
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -66,6 +66,10 @@ int max98373_trigger(struct snd_pcm_substream *substream, int cmd)
 	int j;
 	int ret = 0;
 
+	/* set spk pin by playback only */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		return 0;
+
 	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 		struct snd_soc_component *component = codec_dai->component;
 		struct snd_soc_dapm_context *dapm =
@@ -86,9 +90,6 @@ int max98373_trigger(struct snd_pcm_substream *substream, int cmd)
 		case SNDRV_PCM_TRIGGER_STOP:
 		case SNDRV_PCM_TRIGGER_SUSPEND:
 		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-			/* Make sure no streams are active before disable pin */
-			if (snd_soc_dai_active(codec_dai) != 1)
-				break;
 			ret = snd_soc_dapm_disable_pin(dapm, pin_name);
 			if (!ret)
 				snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c
index de80e19454c1..88c3f63bded9 100644
--- a/sound/soc/intel/haswell/sst-haswell-dsp.c
+++ b/sound/soc/intel/haswell/sst-haswell-dsp.c
@@ -243,92 +243,45 @@ static irqreturn_t hsw_irq(int irq, void *context)
 	return ret;
 }
 
-#define CSR_DEFAULT_VALUE 0x8480040E
-#define ISC_DEFAULT_VALUE 0x0
-#define ISD_DEFAULT_VALUE 0x0
-#define IMC_DEFAULT_VALUE 0x7FFF0003
-#define IMD_DEFAULT_VALUE 0x7FFF0003
-#define IPCC_DEFAULT_VALUE 0x0
-#define IPCD_DEFAULT_VALUE 0x0
-#define CLKCTL_DEFAULT_VALUE 0x7FF
-#define CSR2_DEFAULT_VALUE 0x0
-#define LTR_CTRL_DEFAULT_VALUE 0x0
-#define HMD_CTRL_DEFAULT_VALUE 0x0
-
-static void hsw_set_shim_defaults(struct sst_dsp *sst)
-{
-	sst_dsp_shim_write_unlocked(sst, SST_CSR, CSR_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_ISRX, ISC_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_ISRD, ISD_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IMRX, IMC_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IMRD, IMD_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IPCX, IPCC_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IPCD, IPCD_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_CLKCTL, CLKCTL_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_CSR2, CSR2_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_LTRC, LTR_CTRL_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_HMDC, HMD_CTRL_DEFAULT_VALUE);
-}
-
-/* all clock-gating minus DCLCGE and DTCGE */
-#define SST_VDRTCL2_CG_OTHER	0xB7D
-
 static void hsw_set_dsp_D3(struct sst_dsp *sst)
 {
+	u32 val;
 	u32 reg;
 
-	/* disable clock core gating */
+	/* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg &= ~(SST_VDRTCL2_DCLCGE);
+	reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE);
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* stall, reset and set 24MHz XOSC */
-	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
-			SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST,
-			SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST);
-
-	/* DRAM power gating all */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
-	reg |= SST_VDRTCL0_ISRAMPGE_MASK |
-		SST_VDRTCL0_DSRAMPGE_MASK;
-	reg &= ~(SST_VDRTCL0_D3SRAMPGD);
-	reg |= SST_VDRTCL0_D3PGD;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
-	udelay(50);
+	/* enable power gating and switch off DRAM & IRAM blocks */
+	val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+	val |= SST_VDRTCL0_DSRAMPGE_MASK |
+		SST_VDRTCL0_ISRAMPGE_MASK;
+	val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD);
+	writel(val, sst->addr.pci_cfg + SST_VDRTCTL0);
 
-	/* PLL shutdown enable */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_APLLSE_MASK;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+	/* switch off audio PLL */
+	val = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
+	val |= SST_VDRTCL2_APLLSE_MASK;
+	writel(val, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* disable MCLK */
+	/* disable MCLK(clkctl.smos = 0) */
 	sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL,
-			SST_CLKCTL_MASK, 0);
-
-	/* switch clock gating */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_CG_OTHER;
-	reg &= ~(SST_VDRTCL2_DTCGE);
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
-	/* enable DTCGE separatelly */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_DTCGE;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+		SST_CLKCTL_MASK, 0);
 
-	/* set shim defaults */
-	hsw_set_shim_defaults(sst);
-
-	/* set D3 */
-	reg = readl(sst->addr.pci_cfg + SST_PMCS);
-	reg |= SST_PMCS_PS_MASK;
-	writel(reg, sst->addr.pci_cfg + SST_PMCS);
+	/* Set D3 state, delay 50 us */
+	val = readl(sst->addr.pci_cfg + SST_PMCS);
+	val |= SST_PMCS_PS_MASK;
+	writel(val, sst->addr.pci_cfg + SST_PMCS);
 	udelay(50);
 
-	/* enable clock core gating */
+	/* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_DCLCGE;
+	reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE;
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+
 	udelay(50);
+
 }
 
 static void hsw_reset(struct sst_dsp *sst)
@@ -346,62 +299,75 @@ static void hsw_reset(struct sst_dsp *sst)
 		SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL);
 }
 
-/* recommended CSR state for power-up */
-#define SST_CSR_D0_MASK (0x18A09C0C | SST_CSR_DCS_MASK)
-
 static int hsw_set_dsp_D0(struct sst_dsp *sst)
 {
-	u32 reg;
+	int tries = 10;
+	u32 reg, fw_dump_bit;
 
-	/* disable clock core gating */
+	/* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg &= ~(SST_VDRTCL2_DCLCGE);
+	reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE);
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* switch clock gating */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_CG_OTHER;
-	reg &= ~(SST_VDRTCL2_DTCGE);
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+	/* Disable D3PG (VDRTCTL0.D3PGD = 1) */
+	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+	reg |= SST_VDRTCL0_D3PGD;
+	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
 
-	/* set D0 */
+	/* Set D0 state */
 	reg = readl(sst->addr.pci_cfg + SST_PMCS);
-	reg &= ~(SST_PMCS_PS_MASK);
+	reg &= ~SST_PMCS_PS_MASK;
 	writel(reg, sst->addr.pci_cfg + SST_PMCS);
 
-	/* DRAM power gating none */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
-	reg &= ~(SST_VDRTCL0_ISRAMPGE_MASK |
-		SST_VDRTCL0_DSRAMPGE_MASK);
-	reg |= SST_VDRTCL0_D3SRAMPGD;
-	reg |= SST_VDRTCL0_D3PGD;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
-	mdelay(10);
+	/* check that ADSP shim is enabled */
+	while (tries--) {
+		reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK;
+		if (reg == 0)
+			goto finish;
+
+		msleep(1);
+	}
+
+	return -ENODEV;
 
-	/* set shim defaults */
-	hsw_set_shim_defaults(sst);
+finish:
+	/* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */
+	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+		SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0);
+
+	/* stall DSP core, set clk to 192/96Mhz */
+	sst_dsp_shim_update_bits_unlocked(sst,
+		SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK,
+		SST_CSR_STALL | SST_CSR_DCS(4));
 
-	/* restore MCLK */
+	/* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */
 	sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL,
-			SST_CLKCTL_MASK, SST_CLKCTL_MASK);
+		SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0,
+		SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0);
 
-	/* PLL shutdown disable */
+	/* Stall and reset core, set CSR */
+	hsw_reset(sst);
+
+	/* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg &= ~(SST_VDRTCL2_APLLSE_MASK);
+	reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE;
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
-			SST_CSR_D0_MASK, SST_CSR_SBCS0 | SST_CSR_SBCS1 |
-			SST_CSR_STALL | SST_CSR_DCS(4));
 	udelay(50);
 
-	/* enable clock core gating */
+	/* switch on audio PLL */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_DCLCGE;
+	reg &= ~SST_VDRTCL2_APLLSE_MASK;
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* clear reset */
-	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_RST, 0);
+	/* set default power gating control, enable power gating control for all blocks. that is,
+	can't be accessed, please enable each block before accessing. */
+	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+	reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK;
+	/* for D0, always enable the block(DSRAM[0]) used for FW dump */
+	fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT;
+	writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0);
+
 
 	/* disable DMA finish function for SSP0 & SSP1 */
 	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1,
@@ -418,6 +384,12 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst)
 	sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY |
 				SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0);
 
+	/* clear IPC registers */
+	sst_dsp_shim_write(sst, SST_IPCX, 0x0);
+	sst_dsp_shim_write(sst, SST_IPCD, 0x0);
+	sst_dsp_shim_write(sst, 0x80, 0x6);
+	sst_dsp_shim_write(sst, 0xe0, 0x300a);
+
 	return 0;
 }
 
@@ -443,6 +415,11 @@ static void hsw_sleep(struct sst_dsp *sst)
 {
 	dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n");
 
+	/* put DSP into reset and stall */
+	sst_dsp_shim_update_bits(sst, SST_CSR,
+		SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL,
+		SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS);
+
 	hsw_set_dsp_D3(sst);
 	dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n");
 }
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 5dee55e9546b..bbe8d782e0af 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -488,7 +488,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 							stream->lpib);
 			snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
 		}
-		/* fall through */
+		fallthrough;
 
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 36df30915378..c8664ab80d45 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -58,17 +58,17 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
 	switch (slot_width) {
 	case 0:
 		slot_width = 32;
-		/* Fall-through */
+		fallthrough;
 	case 32:
 		fmt |= SNDRV_PCM_FMTBIT_S32_LE;
-		/* Fall-through */
+		fallthrough;
 	case 24:
 		fmt |= SNDRV_PCM_FMTBIT_S24_LE;
 		fmt |= SNDRV_PCM_FMTBIT_S20_LE;
-		/* Fall-through */
+		fallthrough;
 	case 16:
 		fmt |= SNDRV_PCM_FMTBIT_S16_LE;
-		/* Fall-through */
+		fallthrough;
 	case 8:
 		fmt |= SNDRV_PCM_FMTBIT_S8;
 		break;
@@ -133,7 +133,7 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_CBS_CFM:
 	case SND_SOC_DAIFMT_CBM_CFS:
 		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
-		/* Fall-through */
+		fallthrough;
 	default:
 		return -EINVAL;
 	}
diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
index e711abcf8c12..d6adf7edea41 100644
--- a/sound/soc/meson/axg-toddr.c
+++ b/sound/soc/meson/axg-toddr.c
@@ -18,6 +18,7 @@
 #define CTRL0_TODDR_SEL_RESAMPLE	BIT(30)
 #define CTRL0_TODDR_EXT_SIGNED		BIT(29)
 #define CTRL0_TODDR_PP_MODE		BIT(28)
+#define CTRL0_TODDR_SYNC_CH		BIT(27)
 #define CTRL0_TODDR_TYPE_MASK		GENMASK(15, 13)
 #define CTRL0_TODDR_TYPE(x)		((x) << 13)
 #define CTRL0_TODDR_MSB_POS_MASK	GENMASK(12, 8)
@@ -189,10 +190,31 @@ static const struct axg_fifo_match_data axg_toddr_match_data = {
 	.dai_drv		= &axg_toddr_dai_drv
 };
 
+static int g12a_toddr_dai_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = axg_toddr_dai_startup(substream, dai);
+	if (ret)
+		return ret;
+
+	/*
+	 * Make sure the first channel ends up in the at beginning of the output
+	 * As weird as it looks, without this the first channel may be misplaced
+	 * in memory, with a random shift of 2 channels.
+	 */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SYNC_CH,
+			   CTRL0_TODDR_SYNC_CH);
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops g12a_toddr_ops = {
 	.prepare	= g12a_toddr_dai_prepare,
 	.hw_params	= axg_toddr_dai_hw_params,
-	.startup	= axg_toddr_dai_startup,
+	.startup	= g12a_toddr_dai_startup,
 	.shutdown	= axg_toddr_dai_shutdown,
 };
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d1e09ade0190..c4e7307a4437 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -488,7 +488,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
 
 	case SND_SOC_DAIFMT_DSP_A:
 		sspsp |= SSPSP_FSRT;
-		/* fall through */
+		fallthrough;
 	case SND_SOC_DAIFMT_DSP_B:
 		sscr0 |= SSCR0_MOD | SSCR0_PSP;
 		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index 083413abc2f6..575e2aefefe3 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -143,6 +143,7 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev)
 
 	card = &data->card;
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	card->dapm_widgets = apq8016_sbc_dapm_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(apq8016_sbc_dapm_widgets);
 
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 253549600c5a..1a69baefc5ce 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -114,6 +114,7 @@ static int apq8096_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
 	if (ret)
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 5194d90ddb96..fd69cf8b1f23 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -52,8 +52,10 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 
 	for_each_child_of_node(dev->of_node, np) {
 		dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
-		if (!dlc)
-			return -ENOMEM;
+		if (!dlc) {
+			ret = -ENOMEM;
+			goto err;
+		}
 
 		link->cpus	= &dlc[0];
 		link->platforms	= &dlc[1];
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 0d10fba53945..ab1bf23c21a6 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -555,6 +555,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
 	card->dapm_widgets = sdm845_snd_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
 	if (ret)
diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c
index c0c388d4db82..80c9cf2f254a 100644
--- a/sound/soc/qcom/storm.c
+++ b/sound/soc/qcom/storm.c
@@ -96,6 +96,7 @@ static int storm_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
 
 	ret = snd_soc_of_parse_card_name(card, "qcom,model");
 	if (ret) {
diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 1707414cfa92..5adb293d0435 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -229,13 +229,13 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream,
 	switch (params_channels(params)) {
 	case 8:
 		val |= PDM_PATH3_EN;
-		/* fallthrough */
+		fallthrough;
 	case 6:
 		val |= PDM_PATH2_EN;
-		/* fallthrough */
+		fallthrough;
 	case 4:
 		val |= PDM_PATH1_EN;
-		/* fallthrough */
+		fallthrough;
 	case 2:
 		val |= PDM_PATH0_EN;
 		break;
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 80ecb5c7fed0..df53d4ea808f 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -733,7 +733,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_channels(params)) {
 	case 6:
 		val |= MOD_DC2_EN;
-		/* Fall through */
+		fallthrough;
 	case 4:
 		val |= MOD_DC1_EN;
 		break;
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index bd9de77c35f3..50fc7810723e 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -198,9 +198,9 @@ static int siu_pcm_rd_set(struct siu_port *port_info,
 	return 0;
 }
 
-static void siu_io_tasklet(unsigned long data)
+static void siu_io_tasklet(struct tasklet_struct *t)
 {
-	struct siu_stream *siu_stream = (struct siu_stream *)data;
+	struct siu_stream *siu_stream = from_tasklet(siu_stream, t, tasklet);
 	struct snd_pcm_substream *substream = siu_stream->substream;
 	struct device *dev = substream->pcm->card->dev;
 	struct snd_pcm_runtime *rt = substream->runtime;
@@ -520,10 +520,8 @@ static int siu_pcm_new(struct snd_soc_component *component,
 		(*port_info)->pcm = pcm;
 
 		/* IO tasklets */
-		tasklet_init(&(*port_info)->playback.tasklet, siu_io_tasklet,
-			     (unsigned long)&(*port_info)->playback);
-		tasklet_init(&(*port_info)->capture.tasklet, siu_io_tasklet,
-			     (unsigned long)&(*port_info)->capture);
+		tasklet_setup(&(*port_info)->playback.tasklet, siu_io_tasklet);
+		tasklet_setup(&(*port_info)->capture.tasklet, siu_io_tasklet);
 	}
 
 	dev_info(card->dev, "SuperH SIU driver initialized.\n");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2fe1b2ec7c8f..054437660678 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -618,7 +618,7 @@ int snd_soc_suspend(struct device *dev)
 						"ASoC: idle_bias_off CODEC on over suspend\n");
 					break;
 				}
-				/* fall through */
+				fallthrough;
 
 			case SND_SOC_BIAS_OFF:
 				snd_soc_component_suspend(component);
@@ -834,6 +834,19 @@ struct snd_soc_dai *snd_soc_find_dai(
 }
 EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
+struct snd_soc_dai *snd_soc_find_dai_with_mutex(
+	const struct snd_soc_dai_link_component *dlc)
+{
+	struct snd_soc_dai *dai;
+
+	mutex_lock(&client_mutex);
+	dai = snd_soc_find_dai(dlc);
+	mutex_unlock(&client_mutex);
+
+	return dai;
+}
+EXPORT_SYMBOL_GPL(snd_soc_find_dai_with_mutex);
+
 static int soc_dai_link_sanity_check(struct snd_soc_card *card,
 				     struct snd_soc_dai_link *link)
 {
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 91a2551e4cef..0dbd312aad08 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -412,14 +412,14 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
 		supported_codec = false;
 
 		for_each_link_cpus(dai_link, i, cpu) {
-			dai = snd_soc_find_dai(cpu);
+			dai = snd_soc_find_dai_with_mutex(cpu);
 			if (dai && snd_soc_dai_stream_valid(dai, direction)) {
 				supported_cpu = true;
 				break;
 			}
 		}
 		for_each_link_codecs(dai_link, i, codec) {
-			dai = snd_soc_find_dai(codec);
+			dai = snd_soc_find_dai_with_mutex(codec);
 			if (dai && snd_soc_dai_stream_valid(dai, direction)) {
 				supported_codec = true;
 				break;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 00ac1cbf6f88..4c9d4cd8cf0b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -812,7 +812,7 @@ dynamic:
 	return 0;
 
 config_err:
-	for_each_rtd_dais(rtd, i, dai)
+	for_each_rtd_dais_rollback(rtd, i, dai)
 		snd_soc_dai_shutdown(dai, substream);
 
 	snd_soc_link_shutdown(substream);
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index cee998671318..5b60379237bf 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1057,7 +1057,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
 					ec->hdr.name);
 				goto err_denum;
 			}
-			/* fall through */
+			fallthrough;
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1445,7 +1445,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
 					ec->hdr.name);
 				goto err_se;
 			}
-			/* fall through */
+			fallthrough;
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index df1c6997cb4e..c6cb8c212eca 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -310,7 +310,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 			return ret;
 		}
 
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		snd_hdac_ext_link_stream_start(link_dev);
@@ -333,7 +333,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 
 		link_dev->link_prepared = 0;
 
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		snd_hdac_ext_link_stream_clear(link_dev);
 		break;
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index d730e437e4ba..71c3f29057a7 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -361,7 +361,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
 			return ret;
 		}
 
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_START:
 		if (spcm->stream[substream->stream].suspend_ignored) {
 			/*
@@ -386,7 +386,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
 			spcm->stream[substream->stream].suspend_ignored = true;
 			return 0;
 		}
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_STOP:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
 		ipc_first = true;
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index 5c47de96c529..57feb473a579 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -446,12 +446,12 @@ static const struct snd_soc_dai_ops ams_delta_dai_ops = {
 /* Will be used if the codec ever has its own digital_mute function */
 static int ams_delta_startup(struct snd_pcm_substream *substream)
 {
-	return ams_delta_digital_mute(NULL, 0, substream->stream);
+	return ams_delta_mute(NULL, 0, substream->stream);
 }
 
 static void ams_delta_shutdown(struct snd_pcm_substream *substream)
 {
-	ams_delta_digital_mute(NULL, 1, substream->stream);
+	ams_delta_mute(NULL, 1, substream->stream);
 }
 
 
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index d89b5c928c4d..dd34504c09ba 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -289,7 +289,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		 * rate is lowered.
 		 */
 		inv_fs = true;
-		/* fall through */
+		fallthrough;
 	case SND_SOC_DAIFMT_DSP_A:
 		dev->mode = MOD_DSP_A;
 		break;
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 2802a33b9c5f..ed217b34f846 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -46,7 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
 	switch (n810_jack_func) {
 	case N810_JACK_HS:
 		line1l = 1;
-		/* fall through */
+		fallthrough;
 	case N810_JACK_HP:
 		hp = 1;
 		break;
diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c
index 01abf1be5d78..a26588e9c3bc 100644
--- a/sound/soc/ti/omap-dmic.c
+++ b/sound/soc/ti/omap-dmic.c
@@ -203,10 +203,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	switch (channels) {
 	case 6:
 		dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
-		/* fall through */
+		fallthrough;
 	case 4:
 		dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
-		/* fall through */
+		fallthrough;
 	case 2:
 		dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
 		break;
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index d482b62f314a..fafb2998ad0d 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -309,19 +309,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 4;
-		/* fall through */
+		fallthrough;
 	case 4:
 		if (stream == SNDRV_PCM_STREAM_CAPTURE)
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 3;
-		/* fall through */
+		fallthrough;
 	case 3:
 		link_mask |= 1 << 2;
-		/* fall through */
+		fallthrough;
 	case 2:
 		link_mask |= 1 << 1;
-		/* fall through */
+		fallthrough;
 	case 1:
 		link_mask |= 1 << 0;
 		break;
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 2176a95201bf..a2629ccc1dc8 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -55,7 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
 		break;
 	case RX51_JACK_HS:
 		hs = 1;
-		/* fall through */
+		fallthrough;
 	case RX51_JACK_HP:
 		hp = 1;
 		break;
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 4b1cd4da3e36..939b33ec39f5 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -134,9 +134,9 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
 
 #define NR_DMA_CHAIN		2
 
-static void txx9aclc_dma_tasklet(unsigned long data)
+static void txx9aclc_dma_tasklet(struct tasklet_struct *t)
 {
-	struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data;
+	struct txx9aclc_dmadata *dmadata = from_tasklet(dmadata, t, tasklet);
 	struct dma_chan *chan = dmadata->dma_chan;
 	struct dma_async_tx_descriptor *desc;
 	struct snd_pcm_substream *substream = dmadata->substream;
@@ -352,8 +352,7 @@ static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev,
 			"playback" : "capture");
 		return -EBUSY;
 	}
-	tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet,
-		     (unsigned long)dmadata);
+	tasklet_setup(&dmadata->tasklet, txx9aclc_dma_tasklet);
 	return 0;
 }
 
diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c
index 568cde64ff8b..1c1a44e08a67 100644
--- a/sound/soc/zte/zx-i2s.c
+++ b/sound/soc/zte/zx-i2s.c
@@ -294,7 +294,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			zx_i2s_rx_dma_en(zx_i2s->reg_base, true);
 		else
 			zx_i2s_tx_dma_en(zx_i2s->reg_base, true);
-	/* fall thru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (capture)
@@ -308,7 +308,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			zx_i2s_rx_dma_en(zx_i2s->reg_base, false);
 		else
 			zx_i2s_tx_dma_en(zx_i2s->reg_base, false);
-	/* fall thru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (capture)
diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c
index a3a07c0730e6..b4168bd532b7 100644
--- a/sound/soc/zte/zx-spdif.c
+++ b/sound/soc/zte/zx-spdif.c
@@ -218,7 +218,7 @@ static int zx_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
 		val = readl_relaxed(zx_spdif->reg_base + ZX_FIFOCTRL);
 		val |= ZX_FIFOCTRL_TX_FIFO_RST;
 		writel_relaxed(val, zx_spdif->reg_base + ZX_FIFOCTRL);
-	/* fall thru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		zx_spdif_cfg_tx(zx_spdif->reg_base, true);
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index df639fe03118..e8287a05e36b 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -344,10 +344,9 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep)
 	spin_unlock_irqrestore(&ep->buffer_lock, flags);
 }
 
-static void snd_usbmidi_out_tasklet(unsigned long data)
+static void snd_usbmidi_out_tasklet(struct tasklet_struct *t)
 {
-	struct snd_usb_midi_out_endpoint *ep =
-		(struct snd_usb_midi_out_endpoint *) data;
+	struct snd_usb_midi_out_endpoint *ep = from_tasklet(ep, t, tasklet);
 
 	snd_usbmidi_do_output(ep);
 }
@@ -1441,7 +1440,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
 	}
 
 	spin_lock_init(&ep->buffer_lock);
-	tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep);
+	tasklet_setup(&ep->tasklet, snd_usbmidi_out_tasklet);
 	init_waitqueue_head(&ep->drain_wait);
 
 	for (i = 0; i < 0x10; ++i)
diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index 884e740a785c..3b2dce1043f5 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -247,9 +247,9 @@ static inline void add_with_wraparound(struct ua101 *ua,
 		*value -= ua->playback.queue_length;
 }
 
-static void playback_tasklet(unsigned long data)
+static void playback_tasklet(struct tasklet_struct *t)
 {
-	struct ua101 *ua = (void *)data;
+	struct ua101 *ua = from_tasklet(ua, t, playback_tasklet);
 	unsigned long flags;
 	unsigned int frames;
 	struct ua101_urb *urb;
@@ -1218,8 +1218,7 @@ static int ua101_probe(struct usb_interface *interface,
 	spin_lock_init(&ua->lock);
 	mutex_init(&ua->mutex);
 	INIT_LIST_HEAD(&ua->ready_playback_urbs);
-	tasklet_init(&ua->playback_tasklet,
-		     playback_tasklet, (unsigned long)ua);
+	tasklet_setup(&ua->playback_tasklet, playback_tasklet);
 	init_waitqueue_head(&ua->alsa_capture_wait);
 	init_waitqueue_head(&ua->rate_feedback_wait);
 	init_waitqueue_head(&ua->alsa_playback_wait);
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 5b43e9e40e49..c369c81e74c4 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -371,7 +371,6 @@ static const struct usbmix_name_map asus_rog_map[] = {
 };
 
 static const struct usbmix_name_map lenovo_p620_rear_map[] = {
-	{ 19, NULL, 2 }, /* FU, Volume */
 	{ 19, NULL, 12 }, /* FU, Input Gain Pad */
 	{}
 };
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 5600751803cf..b401ee894e1b 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -369,11 +369,13 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
 	case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
 	case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+	case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */
 	case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
 		ep = 0x82;
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index f4fb002e3ef4..23eafd50126f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2827,14 +2827,24 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
 {
 	USB_DEVICE(0x17aa, 0x1046),
-	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear",
-			     "Lenovo-ThinkStation-P620-Rear"),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Lenovo",
+		.product_name = "ThinkStation P620 Rear",
+		.profile_name = "Lenovo-ThinkStation-P620-Rear",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+	}
 },
 /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
 {
 	USB_DEVICE(0x17aa, 0x104d),
-	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main",
-			     "Lenovo-ThinkStation-P620-Main"),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Lenovo",
+		.product_name = "ThinkStation P620 Main",
+		.profile_name = "Lenovo-ThinkStation-P620-Main",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+	}
 },
 
 /* Native Instruments MK2 series */
@@ -3549,14 +3559,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 {
 	/*
 	 * Pioneer DJ DJM-250MK2
-	 * PCM is 8 channels out @ 48 fixed (endpoints 0x01).
-	 * The output from computer to the mixer is usable.
+	 * PCM is 8 channels out @ 48 fixed (endpoint 0x01)
+	 * and 8 channels in @ 48 fixed (endpoint 0x82).
+	 *
+	 * Both playback and recording is working, even simultaneously.
 	 *
-	 * The input (phono or line to computer) is not working.
-	 * It should be at endpoint 0x82 and probably also 8 channels,
-	 * but it seems that it works only with Pioneer proprietary software.
-	 * Even on officially supported OS, the Audacity was unable to record
-	 * and Mixxx to recognize the control vinyls.
+	 * Playback channels could be mapped to:
+	 *  - CH1
+	 *  - CH2
+	 *  - AUX
+	 *
+	 * Recording channels could be mapped to:
+	 *  - Post CH1 Fader
+	 *  - Post CH2 Fader
+	 *  - Cross Fader A
+	 *  - Cross Fader B
+	 *  - MIC
+	 *  - AUX
+	 *  - REC OUT
+	 *
+	 * There is remaining problem with recording directly from PHONO/LINE.
+	 * If we map a channel to:
+	 *  - CH1 Control Tone PHONO
+	 *  - CH1 Control Tone LINE
+	 *  - CH2 Control Tone PHONO
+	 *  - CH2 Control Tone LINE
+	 * it is silent.
+	 * There is no signal even on other operating systems with official drivers.
+	 * The signal appears only when a supported application is started.
+	 * This needs to be investigated yet...
+	 * (there is quite a lot communication on the USB in both directions)
+	 *
+	 * In current version this mixer could be used for playback
+	 * and for recording from vinyls (through Post CH* Fader)
+	 * but not for DVS (Digital Vinyl Systems) like in Mixxx.
 	 */
 	USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
@@ -3580,6 +3616,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.rate_max = 48000,
 					.nr_rates = 1,
 					.rate_table = (unsigned int[]) { 48000 }
+					}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 8, // inputs
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						USB_ENDPOINT_SYNC_ASYNC|
+						USB_ENDPOINT_USAGE_IMPLICIT_FB,
+					.rates = SNDRV_PCM_RATE_48000,
+					.rate_min = 48000,
+					.rate_max = 48000,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 48000 }
 				}
 			},
 			{
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index abf99b814a0f..892296df131d 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -518,6 +518,15 @@ static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip,
 	return 1;	/* Continue with creating streams and mixer */
 }
 
+static int setup_disable_autosuspend(struct snd_usb_audio *chip,
+				       struct usb_interface *iface,
+				       struct usb_driver *driver,
+				       const struct snd_usb_audio_quirk *quirk)
+{
+	driver->supports_autosuspend = 0;
+	return 1;	/* Continue with creating streams and mixer */
+}
+
 /*
  * audio-interface quirks
  *
@@ -557,6 +566,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
 		[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
 		[QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk,
+		[QUIRK_SETUP_DISABLE_AUTOSUSPEND] = setup_disable_autosuspend,
 	};
 
 	if (quirk->type < QUIRK_TYPE_COUNT) {
@@ -1493,6 +1503,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 		set_format_emu_quirk(subs, fmt);
 		break;
 	case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
 		pioneer_djm_set_format_quirk(subs);
 		break;
 	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
@@ -1667,12 +1678,13 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	    && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
 		msleep(20);
 
-	/* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny
-	 * delay here, otherwise requests like get/set frequency return as
-	 * failed despite actually succeeding.
+	/* Zoom R16/24, Logitech H650e/H570e, Jabra 550a, Kingston HyperX
+	 *  needs a tiny delay here, otherwise requests like get/set
+	 *  frequency return as failed despite actually succeeding.
 	 */
 	if ((chip->usb_id == USB_ID(0x1686, 0x00dd) ||
 	     chip->usb_id == USB_ID(0x046d, 0x0a46) ||
+	     chip->usb_id == USB_ID(0x046d, 0x0a56) ||
 	     chip->usb_id == USB_ID(0x0b0e, 0x0349) ||
 	     chip->usb_id == USB_ID(0x0951, 0x16ad)) &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b91c4c0807ec..6839915a0128 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -102,6 +102,7 @@ enum quirk_type {
 	QUIRK_AUDIO_ALIGN_TRANSFER,
 	QUIRK_AUDIO_STANDARD_MIXER,
 	QUIRK_SETUP_FMT_AFTER_RESUME,
+	QUIRK_SETUP_DISABLE_AUTOSUSPEND,
 
 	QUIRK_TYPE_COUNT
 };
diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig
index 77777192f650..4ffcc5e623c2 100644
--- a/sound/x86/Kconfig
+++ b/sound/x86/Kconfig
@@ -9,7 +9,7 @@ menuconfig SND_X86
 if SND_X86
 
 config HDMI_LPE_AUDIO
-	tristate "HDMI audio without HDaudio on Intel Atom platforms"
+	tristate "HDMI audio without HDAudio on Intel Atom platforms"
 	depends on DRM_I915
 	select SND_PCM
 	help