Merge series "Support ROHM BD9576MUF and BD9573MUF PMICs" from Matti Vaittinen <matti.vaittinen@fi.rohmeurope.com>:

Initial support for ROHM BD9576MUF and BD9573MUF PMICs.

These PMICs are primarily intended to be used to power the R-Car family
processors. BD9576MUF includes some additional safety features the
BD9573MUF does not have. This initial version of drivers does not
utilize these features and for now the SW behaviour is identical.

Please note that this version of drivers is only tested on BD9576MUF
but according to the data-sheets the relevant parts of registers should
be same so drivers should also work on BD9573MUF.

This patch series includes MFD, watchdog and regulator drivers with
basic functionality such as:

- Enabling and pinging the watchdog
- configuring watchog timeout / window from device-tree
- reading regulator states/voltages
- enabling/disabling VOUT1 (VD50) when control mode B is used.

This patch series does not bring interrupt support. BD9576MUF and BD9573MUF
are designed to keep the IRQ line low for whole duration of error
condition. IRQ can't be 'acked'. So proper IRQ support would require
some IRQ limiter implementation (delayed unmask?) in order to not hog
the CPU.

---

Matti Vaittinen (6):
  dt_bindings: mfd: Add ROHM BD9576MUF and BD9573MUF PMICs
  dt_bindings: regulator: Add ROHM BD9576MUF and BD9573MUF PMICs
  mfd: Support ROHM BD9576MUF and BD9573MUF
  wdt: Support wdt on ROHM BD9576MUF and BD9573MUF
  regulator: Support ROHM BD9576MUF and BD9573MUF
  MAINTAINERS: Add ROHM BD9576MUF and BD9573MUF drivers

 .../bindings/mfd/rohm,bd9576-pmic.yaml        | 129 +++++++
 .../regulator/rohm,bd9576-regulator.yaml      |  33 ++
 MAINTAINERS                                   |   4 +
 drivers/mfd/Kconfig                           |  11 +
 drivers/mfd/Makefile                          |   1 +
 drivers/mfd/rohm-bd9576.c                     | 130 +++++++
 drivers/regulator/Kconfig                     |  10 +
 drivers/regulator/Makefile                    |   1 +
 drivers/regulator/bd9576-regulator.c          | 337 ++++++++++++++++++
 drivers/watchdog/Kconfig                      |  13 +
 drivers/watchdog/Makefile                     |   1 +
 drivers/watchdog/bd9576_wdt.c                 | 295 +++++++++++++++
 include/linux/mfd/rohm-bd957x.h               |  61 ++++
 include/linux/mfd/rohm-generic.h              |   2 +
 14 files changed, 1028 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/mfd/rohm,bd9576-pmic.yaml
 create mode 100644 Documentation/devicetree/bindings/regulator/rohm,bd9576-regulator.yaml
 create mode 100644 drivers/mfd/rohm-bd9576.c
 create mode 100644 drivers/regulator/bd9576-regulator.c
 create mode 100644 drivers/watchdog/bd9576_wdt.c
 create mode 100644 include/linux/mfd/rohm-bd957x.h

base-commit: f4d51dffc6c01a9e94650d95ce0104964f8ae822
--
2.21.0

--
Matti Vaittinen, Linux device drivers
ROHM Semiconductors, Finland SWDC
Kiviharjunlenkki 1E
90220 OULU
FINLAND

~~~ "I don't think so," said Rene Descartes. Just then he vanished ~~~
Simon says - in Latin please.
~~~ "non cogito me" dixit Rene Descarte, deinde evanescavit ~~~
Thanks to Simon Glass for the translation =]

51 files changed, 266 insertions, 117 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/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/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/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/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/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_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/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/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/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..663e3839f251 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);
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/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/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