Merge tag 'sound-fix-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes gathered since the previous update.

  ALSA core:
   - Regression fix for OSS PCM emulation

  ASoC:
   - Trivial fixes in reg bit mask ops, DAPM, DPCM and topology
   - Lots of fixes for Intel-based devices
   - Minor fixes for AMD, STM32, Qualcomm, Realtek

  Others:
   - Fixes for the bugs in mixer handling in HD-audio and ice1724
     drivers that were caught by the recent kctl validator
   - New quirks for HD-audio and USB-audio

  Also this contains a fix for EDD firmware fix, which slipped from
  anyone's hands"

* tag 'sound-fix-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (35 commits)
  ALSA: hda: Add driver blacklist
  ALSA: usb-audio: Add mixer workaround for TRX40 and co
  ALSA: hda/realtek - Add quirk for MSI GL63
  ALSA: ice1724: Fix invalid access for enumerated ctl items
  ALSA: hda: Fix potential access overflow in beep helper
  ASoC: cs4270: pull reset GPIO low then high
  ALSA: hda/realtek - Add HP new mute led supported for ALC236
  ALSA: hda/realtek - Add supported new mute Led for HP
  ASoC: rt5645: Add platform-data for Medion E1239T
  ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
  ASoC: stm32: sai: Add missing cleanup
  ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Alpha S
  ASoC: Intel: atom: Fix uninitialized variable compiler warning
  ASoC: Intel: atom: Check drv->lock is locked in sst_fill_and_send_cmd_unlocked
  ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
  ASoC: SOF: Turn "firmware boot complete" message into a dbg message
  ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
  ALSA: pcm: oss: Fix regression by buffer overflow fix (again)
  ALSA: pcm: oss: Fix regression by buffer overflow fix
  edd: Use scnprintf() for avoiding potential buffer overflow
  ...

29 files changed, 324 insertions, 39 deletions

diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index fbda4ebf38b3..59d62f05658f 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -197,7 +197,8 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin)
 }
 
 static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug,
-					 snd_pcm_sframes_t frames)
+					 snd_pcm_sframes_t frames,
+					 bool check_size)
 {
 	struct snd_pcm_plugin *plugin, *plugin_next;
 
@@ -209,7 +210,7 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug,
 			if (frames < 0)
 				return frames;
 		}
-		if (frames > plugin->buf_frames)
+		if (check_size && frames > plugin->buf_frames)
 			frames = plugin->buf_frames;
 		plugin = plugin_next;
 	}
@@ -217,13 +218,14 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug,
 }
 
 static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug,
-					 snd_pcm_sframes_t frames)
+					 snd_pcm_sframes_t frames,
+					 bool check_size)
 {
 	struct snd_pcm_plugin *plugin, *plugin_prev;
 
 	plugin = snd_pcm_plug_last(plug);
 	while (plugin && frames > 0) {
-		if (frames > plugin->buf_frames)
+		if (check_size && frames > plugin->buf_frames)
 			frames = plugin->buf_frames;
 		plugin_prev = plugin->prev;
 		if (plugin->src_frames) {
@@ -242,9 +244,9 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p
 		return -ENXIO;
 	switch (snd_pcm_plug_stream(plug)) {
 	case SNDRV_PCM_STREAM_PLAYBACK:
-		return calc_src_frames(plug, drv_frames);
+		return calc_src_frames(plug, drv_frames, false);
 	case SNDRV_PCM_STREAM_CAPTURE:
-		return calc_dst_frames(plug, drv_frames);
+		return calc_dst_frames(plug, drv_frames, false);
 	default:
 		snd_BUG();
 		return -EINVAL;
@@ -257,9 +259,9 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc
 		return -ENXIO;
 	switch (snd_pcm_plug_stream(plug)) {
 	case SNDRV_PCM_STREAM_PLAYBACK:
-		return calc_dst_frames(plug, clt_frames);
+		return calc_dst_frames(plug, clt_frames, false);
 	case SNDRV_PCM_STREAM_CAPTURE:
-		return calc_src_frames(plug, clt_frames);
+		return calc_src_frames(plug, clt_frames, false);
 	default:
 		snd_BUG();
 		return -EINVAL;
@@ -622,7 +624,7 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st
 		src_channels = dst_channels;
 		plugin = next;
 	}
-	return snd_pcm_plug_client_size(plug, frames);
+	return calc_src_frames(plug, frames, true);
 }
 
 snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *dst_channels_final, snd_pcm_uframes_t size)
@@ -632,7 +634,7 @@ snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, str
 	snd_pcm_sframes_t frames = size;
 	int err;
 
-	frames = snd_pcm_plug_slave_size(plug, frames);
+	frames = calc_src_frames(plug, frames, true);
 	if (frames < 0)
 		return frames;
 
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index f5fd62ed4df5..841523f6b88d 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
+	int chs = get_amp_channels(kcontrol);
+
 	if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
-		ucontrol->value.integer.value[0] =
+		if (chs & 1)
+			ucontrol->value.integer.value[0] = beep->enabled;
+		if (chs & 2)
 			ucontrol->value.integer.value[1] = beep->enabled;
 		return 0;
 	}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 92a042e34d3e..bd093593f8fb 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2076,6 +2076,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
 #endif
 }
 
+/* Blacklist for skipping the whole probe:
+ * some HD-audio PCI entries are exposed without any codecs, and such devices
+ * should be ignored from the beginning.
+ */
+static const struct snd_pci_quirk driver_blacklist[] = {
+	SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0),
+	SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0),
+	SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0),
+	{}
+};
+
 static const struct hda_controller_ops pci_hda_ops = {
 	.disable_msi_reset_irq = disable_msi_reset_irq,
 	.pcm_mmap_prepare = pcm_mmap_prepare,
@@ -2092,6 +2103,11 @@ static int azx_probe(struct pci_dev *pci,
 	bool schedule_probe;
 	int err;
 
+	if (snd_pci_quirk_lookup(pci, driver_blacklist)) {
+		dev_info(&pci->dev, "Skipping the blacklisted device\n");
+		return -ENODEV;
+	}
+
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 	if (!enable[dev]) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f66a48154a57..de2826f90d34 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -86,6 +86,14 @@ struct alc_spec {
 
 	unsigned int gpio_mute_led_mask;
 	unsigned int gpio_mic_led_mask;
+	unsigned int mute_led_coef_idx;
+	unsigned int mute_led_coefbit_mask;
+	unsigned int mute_led_coefbit_on;
+	unsigned int mute_led_coefbit_off;
+	unsigned int mic_led_coef_idx;
+	unsigned int mic_led_coefbit_mask;
+	unsigned int mic_led_coefbit_on;
+	unsigned int mic_led_coefbit_off;
 
 	hda_nid_t headset_mic_pin;
 	hda_nid_t headphone_mic_pin;
@@ -2447,6 +2455,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
+	SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950),
 	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),
@@ -4178,6 +4187,111 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec,
 	}
 }
 
+/* update mute-LED according to the speaker mute state via COEF bit */
+static void alc_fixup_mute_led_coefbit_hook(void *private_data, int enabled)
+{
+	struct hda_codec *codec = private_data;
+	struct alc_spec *spec = codec->spec;
+
+	if (spec->mute_led_polarity)
+		enabled = !enabled;
+
+	/* temporarily power up/down for setting COEF bit */
+	enabled ? alc_update_coef_idx(codec, spec->mute_led_coef_idx,
+		spec->mute_led_coefbit_mask, spec->mute_led_coefbit_off) :
+		  alc_update_coef_idx(codec, spec->mute_led_coef_idx,
+		spec->mute_led_coefbit_mask, spec->mute_led_coefbit_on);
+}
+
+static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
+					  const struct hda_fixup *fix,
+					  int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->mute_led_polarity = 0;
+		spec->mute_led_coef_idx = 0x0b;
+		spec->mute_led_coefbit_mask = 1<<3;
+		spec->mute_led_coefbit_on = 1<<3;
+		spec->mute_led_coefbit_off = 0;
+		spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
+		spec->gen.vmaster_mute_enum = 1;
+	}
+}
+
+static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
+					  const struct hda_fixup *fix,
+					  int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->mute_led_polarity = 0;
+		spec->mute_led_coef_idx = 0x34;
+		spec->mute_led_coefbit_mask = 1<<5;
+		spec->mute_led_coefbit_on = 0;
+		spec->mute_led_coefbit_off = 1<<5;
+		spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
+		spec->gen.vmaster_mute_enum = 1;
+	}
+}
+
+/* turn on/off mic-mute LED per capture hook by coef bit */
+static void alc_hp_cap_micmute_update(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (spec->gen.micmute_led.led_value)
+		alc_update_coef_idx(codec, spec->mic_led_coef_idx,
+			spec->mic_led_coefbit_mask, spec->mic_led_coefbit_on);
+	else
+		alc_update_coef_idx(codec, spec->mic_led_coef_idx,
+			spec->mic_led_coefbit_mask, spec->mic_led_coefbit_off);
+}
+
+static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->mic_led_coef_idx = 0x19;
+		spec->mic_led_coefbit_mask = 1<<13;
+		spec->mic_led_coefbit_on = 1<<13;
+		spec->mic_led_coefbit_off = 0;
+		snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+	}
+}
+
+static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->mic_led_coef_idx = 0x35;
+		spec->mic_led_coefbit_mask = 3<<2;
+		spec->mic_led_coefbit_on = 2<<2;
+		spec->mic_led_coefbit_off = 1<<2;
+		snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+	}
+}
+
+static void alc285_fixup_hp_mute_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	alc285_fixup_hp_mute_led_coefbit(codec, fix, action);
+	alc285_fixup_hp_coef_micmute_led(codec, fix, action);
+}
+
+static void alc236_fixup_hp_mute_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	alc236_fixup_hp_mute_led_coefbit(codec, fix, action);
+	alc236_fixup_hp_coef_micmute_led(codec, fix, action);
+}
+
 #if IS_REACHABLE(CONFIG_INPUT)
 static void gpio2_mic_hotkey_event(struct hda_codec *codec,
 				   struct hda_jack_callback *event)
@@ -5964,6 +6078,8 @@ enum {
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_HPE,
 	ALC285_FIXUP_HP_GPIO_LED,
+	ALC285_FIXUP_HP_MUTE_LED,
+	ALC236_FIXUP_HP_MUTE_LED,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -7089,6 +7205,14 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc285_fixup_hp_gpio_led,
 	},
+	[ALC285_FIXUP_HP_MUTE_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_mute_led,
+	},
+	[ALC236_FIXUP_HP_MUTE_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc236_fixup_hp_mute_led,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7234,6 +7358,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -7325,6 +7451,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
 	SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+	SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
 	SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 91f83cef0e56..9aa12a67d370 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol,
 	struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
 
 	mutex_lock(&ice->gpio_mutex);
-	ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f;
+	ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f;
 	mutex_unlock(&ice->gpio_mutex);
 	return 0;
 }
@@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol,
 
 	mutex_lock(&ice->gpio_mutex);
 	oval = wm_get(ice, WM_ADC_MUX);
-	nval = (oval & 0xe0) | ucontrol->value.integer.value[0];
+	nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0];
 	if (nval != oval) {
 		wm_put(ice, WM_ADC_MUX, nval);
 		change = 1;
diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c
index 3a3c47e820ab..f160d35a6832 100644
--- a/sound/soc/amd/raven/acp3x-i2s.c
+++ b/sound/soc/amd/raven/acp3x-i2s.c
@@ -139,6 +139,7 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream,
 		rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg);
 	}
 	val = rv_readl(rtd->acp3x_base + reg_val);
+	val &= ~ACP3x_ITER_IRER_SAMP_LEN_MASK;
 	val = val | (rtd->xfer_resolution  << 3);
 	rv_writel(val, rtd->acp3x_base + reg_val);
 	return 0;
diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h
index 21e7ac017f2b..03fe93913e12 100644
--- a/sound/soc/amd/raven/acp3x.h
+++ b/sound/soc/amd/raven/acp3x.h
@@ -76,6 +76,8 @@
 #define ACP_POWERED_OFF			0x02
 #define ACP_POWER_OFF_IN_PROGRESS	0x03
 
+#define ACP3x_ITER_IRER_SAMP_LEN_MASK	0x38
+
 struct acp3x_platform_info {
 	u16 play_i2s_instance;
 	u16 cap_i2s_instance;
diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c
index e46c390683e7..b7a1efc7406e 100644
--- a/sound/soc/bcm/bcm63xx-pcm-whistler.c
+++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c
@@ -181,7 +181,7 @@ bcm63xx_pcm_pointer(struct snd_soc_component *component,
 	snd_pcm_uframes_t x;
 	struct bcm63xx_runtime_data *prtd = substream->runtime->private_data;
 
-	if ((void *)prtd->dma_addr_next == NULL)
+	if (!prtd->dma_addr_next)
 		prtd->dma_addr_next = substream->runtime->dma_addr;
 
 	x = bytes_to_frames(substream->runtime,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 5f25b9f872bd..8a02791e44ad 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -137,6 +137,9 @@ struct cs4270_private {
 
 	/* power domain regulators */
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+
+	/* reset gpio */
+	struct gpio_desc *reset_gpio;
 };
 
 static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = {
@@ -649,6 +652,22 @@ static const struct regmap_config cs4270_regmap = {
 };
 
 /**
+ * cs4270_i2c_remove - deinitialize the I2C interface of the CS4270
+ * @i2c_client: the I2C client object
+ *
+ * This function puts the chip into low power mode when the i2c device
+ * is removed.
+ */
+static int cs4270_i2c_remove(struct i2c_client *i2c_client)
+{
+	struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client);
+
+	gpiod_set_value_cansleep(cs4270->reset_gpio, 0);
+
+	return 0;
+}
+
+/**
  * cs4270_i2c_probe - initialize the I2C interface of the CS4270
  * @i2c_client: the I2C client object
  * @id: the I2C device ID (ignored)
@@ -660,7 +679,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	const struct i2c_device_id *id)
 {
 	struct cs4270_private *cs4270;
-	struct gpio_desc *reset_gpiod;
 	unsigned int val;
 	int ret, i;
 
@@ -679,10 +697,21 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	if (ret < 0)
 		return ret;
 
-	reset_gpiod = devm_gpiod_get_optional(&i2c_client->dev, "reset",
-					      GPIOD_OUT_HIGH);
-	if (PTR_ERR(reset_gpiod) == -EPROBE_DEFER)
-		return -EPROBE_DEFER;
+	/* reset the device */
+	cs4270->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, "reset",
+						     GPIOD_OUT_LOW);
+	if (IS_ERR(cs4270->reset_gpio)) {
+		dev_dbg(&i2c_client->dev, "Error getting CS4270 reset GPIO\n");
+		return PTR_ERR(cs4270->reset_gpio);
+	}
+
+	if (cs4270->reset_gpio) {
+		dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
+		gpiod_set_value_cansleep(cs4270->reset_gpio, 1);
+	}
+
+	/* Sleep 500ns before i2c communications */
+	ndelay(500);
 
 	cs4270->regmap = devm_regmap_init_i2c(i2c_client, &cs4270_regmap);
 	if (IS_ERR(cs4270->regmap))
@@ -735,6 +764,7 @@ static struct i2c_driver cs4270_i2c_driver = {
 	},
 	.id_table = cs4270_id,
 	.probe = cs4270_i2c_probe,
+	.remove = cs4270_i2c_remove,
 };
 
 module_i2c_driver(cs4270_i2c_driver);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 92d67010aeed..6ba1849a77b0 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3758,6 +3758,14 @@ static const struct dmi_system_id dmi_platform_data[] = {
 		},
 		.driver_data = (void *)&kahlee_platform_data,
 	},
+	{
+		.ident = "Medion E1239T",
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MEDION"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "E1239T MD60568"),
+		},
+		.driver_data = (void *)&intel_braswell_platform_data,
+	},
 	{ }
 };
 
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index c9268a230daa..d36f560ad7a8 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -3703,7 +3703,7 @@ static const struct acpi_device_id rt5682_acpi_match[] = {
 MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match);
 #endif
 
-static struct i2c_driver rt5682_i2c_driver = {
+static struct i2c_driver __maybe_unused rt5682_i2c_driver = {
 	.driver = {
 		.name = "rt5682",
 		.of_match_table = of_match_ptr(rt5682_of_match),
@@ -3713,7 +3713,10 @@ static struct i2c_driver rt5682_i2c_driver = {
 	.shutdown = rt5682_i2c_shutdown,
 	.id_table = rt5682_i2c_id,
 };
+
+#ifdef CONFIG_I2C
 module_i2c_driver(rt5682_i2c_driver);
+#endif
 
 MODULE_DESCRIPTION("ASoC RT5682 driver");
 MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index f883c9340eee..69f3af4524ab 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -50,6 +50,8 @@ static int sst_fill_and_send_cmd_unlocked(struct sst_data *drv,
 {
 	int ret = 0;
 
+	WARN_ON(!mutex_is_locked(&drv->lock));
+
 	ret = sst_fill_byte_control(drv, ipc_msg,
 				block, task_id, pipe_id, len, cmd_data);
 	if (ret < 0)
@@ -966,7 +968,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w,
 	dev_dbg(c->dev, "Enter: widget=%s\n", w->name);
 
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		mutex_lock(&drv->lock);
 		ret = sst_send_slot_map(drv);
+		mutex_unlock(&drv->lock);
 		if (ret)
 			return ret;
 		ret = sst_send_pipe_module_params(w, k);
diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c
index 13db2854db3e..053c27707147 100644
--- a/sound/soc/intel/atom/sst/sst_pvt.c
+++ b/sound/soc/intel/atom/sst/sst_pvt.c
@@ -223,9 +223,9 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst,
 		size_t mbox_data_len, const void *mbox_data, void **data,
 		bool large, bool fill_dsp, bool sync, bool response)
 {
+	struct sst_block *block = NULL;
 	struct ipc_post *msg = NULL;
 	struct ipc_dsp_hdr dsp_hdr;
-	struct sst_block *block;
 	int ret = 0, pvt_id;
 
 	pvt_id = sst_assign_pvt_id(sst);
diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c
index 6c2fdb5659ed..af2f50293208 100644
--- a/sound/soc/intel/boards/bdw-rt5650.c
+++ b/sound/soc/intel/boards/bdw-rt5650.c
@@ -254,7 +254,6 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = {
 		.no_pcm = 1,
 		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
-		.ignore_suspend = 1,
 		.ignore_pmdown_time = 1,
 		.be_hw_params_fixup = broadwell_ssp0_fixup,
 		.ops = &bdw_rt5650_ops,
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 6b4b64098d36..cc41a348295e 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -340,7 +340,6 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
 		.no_pcm = 1,
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
-		.ignore_suspend = 1,
 		.ignore_pmdown_time = 1,
 		.be_hw_params_fixup = broadwell_ssp0_fixup,
 		.ops = &bdw_rt5677_ops,
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index acb4e36682cb..f9a8336a0541 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -217,7 +217,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 		.init = broadwell_rt286_codec_init,
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
-		.ignore_suspend = 1,
 		.ignore_pmdown_time = 1,
 		.be_hw_params_fixup = broadwell_ssp0_fixup,
 		.ops = &broadwell_rt286_ops,
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 33fb8ea4e5cb..08f4ae964b02 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{
+		/* MPMAN MPWIN895CL */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* MSI S100 tablet */
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."),
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 3ed53d7db4e6..74af090f2657 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -162,7 +162,6 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 		.no_pcm = 1,
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
-		.ignore_suspend = 1,
 		.ignore_pmdown_time = 1,
 		.be_hw_params_fixup = haswell_ssp0_fixup,
 		.ops = &haswell_rt5640_ops,
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index f6c7cddf08e8..125af00bba53 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -78,7 +78,7 @@ struct q6asm_dai_data {
 };
 
 static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
-	.info =                 (SNDRV_PCM_INFO_MMAP |
+	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
 				SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				SNDRV_PCM_INFO_MMAP_VALID |
 				SNDRV_PCM_INFO_INTERLEAVED |
@@ -100,7 +100,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
 };
 
 static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
-	.info =                 (SNDRV_PCM_INFO_MMAP |
+	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
 				SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				SNDRV_PCM_INFO_MMAP_VALID |
 				SNDRV_PCM_INFO_INTERLEAVED |
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 19142f6e533c..8f3cad8db89a 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -295,12 +295,12 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
 {
 	int ret = 0;
 
-	if (!dai->started &&
+	if (!dai->started[substream->stream] &&
 	    dai->driver->ops->startup)
 		ret = dai->driver->ops->startup(substream, dai);
 
 	if (ret == 0)
-		dai->started = 1;
+		dai->started[substream->stream] = 1;
 
 	return ret;
 }
@@ -308,11 +308,11 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
 void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			 struct snd_pcm_substream *substream)
 {
-	if (dai->started &&
+	if (dai->started[substream->stream] &&
 	    dai->driver->ops->shutdown)
 		dai->driver->ops->shutdown(substream, dai);
 
-	dai->started = 0;
+	dai->started[substream->stream] = 0;
 }
 
 int snd_soc_dai_prepare(struct snd_soc_dai *dai,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 04da7928c873..679ed60d850e 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -802,7 +802,13 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i,
 			val = max - val;
 		p->connect = !!val;
 	} else {
-		p->connect = 0;
+		/* since a virtual mixer has no backing registers to
+		 * decide which path to connect, it will try to match
+		 * with initial state.  This is to ensure
+		 * that the default mixer choice will be
+		 * correctly powered up during initialization.
+		 */
+		p->connect = invert;
 	}
 }
 
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 652657dc6809..55ffb34be95e 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -825,7 +825,7 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
 	unsigned int regbase = mc->regbase;
 	unsigned int regcount = mc->regcount;
 	unsigned int regwshift = component->val_bytes * BITS_PER_BYTE;
-	unsigned int regwmask = (1<<regwshift)-1;
+	unsigned int regwmask = (1UL<<regwshift)-1;
 	unsigned int invert = mc->invert;
 	unsigned long mask = (1UL<<mc->nbits)-1;
 	long min = mc->min;
@@ -874,7 +874,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
 	unsigned int regbase = mc->regbase;
 	unsigned int regcount = mc->regcount;
 	unsigned int regwshift = component->val_bytes * BITS_PER_BYTE;
-	unsigned int regwmask = (1<<regwshift)-1;
+	unsigned int regwmask = (1UL<<regwshift)-1;
 	unsigned int invert = mc->invert;
 	unsigned long mask = (1UL<<mc->nbits)-1;
 	long max = mc->max;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e256d438ee68..289aebc15529 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2324,7 +2324,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 		switch (cmd) {
 		case SNDRV_PCM_TRIGGER_START:
 			if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
-			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
+			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
 				continue;
 
 			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
@@ -2354,7 +2355,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 			be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
 			break;
 		case SNDRV_PCM_TRIGGER_STOP:
-			if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+			if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) &&
+			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
 				continue;
 
 			if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 1f81cd2d29cf..87f75edba3dc 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg,
 	struct snd_soc_component *comp = tplg->comp;
 
 	return soc_tplg_add_dcontrol(comp->card->snd_card,
-				comp->dev, k, NULL, comp, kcontrol);
+				comp->dev, k, comp->name_prefix, comp, kcontrol);
 }
 
 /* remove a mixer kcontrol */
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 1f2e0be812bd..64af08293daa 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -597,7 +597,7 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev)
 	}
 
 	if (sdev->fw_state == SOF_FW_BOOT_COMPLETE)
-		dev_info(sdev->dev, "firmware boot complete\n");
+		dev_dbg(sdev->dev, "firmware boot complete\n");
 	else
 		return -EIO; /* FW boots but fw_ready op failed */
 
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 2bd280c01c33..0d0c9afd8791 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -1556,8 +1556,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
 
 	ret = snd_soc_register_component(&pdev->dev, &stm32_component,
 					 &sai->cpu_dai_drv, 1);
-	if (ret)
+	if (ret) {
+		snd_dmaengine_pcm_unregister(&pdev->dev);
 		return ret;
+	}
 
 	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
 		conf = &stm32_sai_pcm_config_spdif;
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 5ebca8013840..72b575c34860 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -359,6 +359,14 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = {
 	{ 0 }
 };
 
+/* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX
+ * response for Input Gain Pad (id=19, control=12).  Skip it.
+ */
+static const struct usbmix_name_map asus_rog_map[] = {
+	{ 19, NULL, 12 }, /* FU, Input Gain Pad */
+	{}
+};
+
 /*
  * Control map entries
  */
@@ -488,6 +496,26 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.id = USB_ID(0x1b1c, 0x0a42),
 		.map = corsair_virtuoso_map,
 	},
+	{	/* Gigabyte TRX40 Aorus Pro WiFi */
+		.id = USB_ID(0x0414, 0xa002),
+		.map = asus_rog_map,
+	},
+	{	/* ASUS ROG Zenith II */
+		.id = USB_ID(0x0b05, 0x1916),
+		.map = asus_rog_map,
+	},
+	{	/* ASUS ROG Strix */
+		.id = USB_ID(0x0b05, 0x1917),
+		.map = asus_rog_map,
+	},
+	{	/* MSI TRX40 Creator */
+		.id = USB_ID(0x0db0, 0x0d64),
+		.map = asus_rog_map,
+	},
+	{	/* MSI TRX40 */
+		.id = USB_ID(0x0db0, 0x543d),
+		.map = asus_rog_map,
+	},
 	{ 0 } /* terminator */
 };
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 1c8719292eee..e009d584e7d0 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3592,5 +3592,47 @@ 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.
+	 *
+	 * 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.
+	 */
+	USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 8, // outputs
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x01,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_48000,
+					.rate_min = 48000,
+					.rate_max = 48000,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 48000 }
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 #undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 86f192a3043d..a8ece1701068 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1827,6 +1827,7 @@ struct registration_quirk {
 
 static const struct registration_quirk registration_quirks[] = {
 	REG_QUIRK_ENTRY(0x0951, 0x16d8, 2),	/* Kingston HyperX AMP */
+	REG_QUIRK_ENTRY(0x0951, 0x16ed, 2),	/* Kingston HyperX Cloud Alpha S */
 	{ 0 }					/* terminator */
 };