Merge tag 'sound-5.9-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small changes, nothing intrusive:

   - remaining tasklet API conversions, now all sound stuff have been
     converted

   - a few HD-audio and USB-audio quirks and minor fixes

   - FireWire Tascam and Digi00xx fixes

   - drop a kernel WARNING from PCM OSS for syzkaller"

* tag 'sound-5.9-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits)
  ALSA: hda/realtek - Improved routing for Thinkpad X1 7th/8th Gen
  ALSA: hda: use consistent HDAudio spelling in comments/docs
  ALSA: hda: add dev_dbg log when driver is not selected
  ALSA: hda: fix a runtime pm issue in SOF when integrated GPU is disabled
  ALSA: hda: hdmi - add Rocketlake support
  ALSA: ua101: convert tasklets to use new tasklet_setup() API
  ALSA: usb-audio: convert tasklets to use new tasklet_setup() API
  ASoC: txx9: convert tasklets to use new tasklet_setup() API
  ASoC: siu: convert tasklets to use new tasklet_setup() API
  ASoC: fsl_esai: convert tasklets to use new tasklet_setup() API
  ALSA: hdsp: convert tasklets to use new tasklet_setup() API
  ALSA: riptide: convert tasklets to use new tasklet_setup() API
  ALSA: pci/asihpi: convert tasklets to use new tasklet_setup() API
  ALSA: firewire: convert tasklets to use new tasklet_setup() API
  ALSA: core: convert tasklets to use new tasklet_setup() API
  ALSA: pcm: oss: Remove superfluous WARN_ON() for mulaw sanity check
  ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO
  ALSA: hda/hdmi: always check pin power status in i915 pin fixup
  ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion NT950XCJ-X716A
  ALSA: usb-audio: Add basic capture support for Pioneer DJ DJM-250MK2
  ...

29 files changed, 227 insertions, 78 deletions

diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst
index 998f76e19cdd..f3f4640ee2af 100644
--- a/Documentation/sound/cards/audigy-mixer.rst
+++ b/Documentation/sound/cards/audigy-mixer.rst
@@ -332,7 +332,7 @@ WO 9901953 (A1)
 
 
 US Patents (https://www.uspto.gov/)
-----------------------------------
+-----------------------------------
 
 US 5925841
 	Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst
index eccb0f0ffd0f..2ce41d3822d8 100644
--- a/Documentation/sound/cards/sb-live-mixer.rst
+++ b/Documentation/sound/cards/sb-live-mixer.rst
@@ -337,7 +337,7 @@ WO 9901953 (A1)
 
 
 US Patents (https://www.uspto.gov/)
-----------------------------------
+-----------------------------------
 
 US 5925841
 	Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
diff --git a/Documentation/sound/designs/timestamping.rst b/Documentation/sound/designs/timestamping.rst
index 2b0fff503415..7c7ecf5dbc4b 100644
--- a/Documentation/sound/designs/timestamping.rst
+++ b/Documentation/sound/designs/timestamping.rst
@@ -143,7 +143,7 @@ timestamp shows when the information is put together by the driver
 before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases
 this driver_timestamp will be identical to the regular system tstamp.
 
-Examples of typestamping with HDaudio:
+Examples of timestamping with HDAudio:
 
 1. DMA timestamp, no compensation for DMA+analog delay
 ::
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/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..c521a1f17096 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2475,6 +2475,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
+	SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
@@ -5867,6 +5868,39 @@ 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)
@@ -6135,6 +6169,7 @@ 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,
@@ -7280,11 +7315,17 @@ 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,
@@ -7695,7 +7736,8 @@ 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),
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/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/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/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/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/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..75bbdc691243 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 */
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