Merge branch 'for-linus' into for-next

Back-merge the development process for catching up the HD-audio fix
(and apply a new one on top of that).

Signed-off-by: Takashi Iwai <tiwai@suse.de>

34 files changed, 382 insertions, 102 deletions

diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 7ac0d9f495c4..f7f0db5aa811 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -252,8 +252,7 @@ end:
 	return err;
 }
 
-static unsigned int
-map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
+static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
 {
 	unsigned int sec, sections, ch, channels;
 	unsigned int pcm, midi, location;
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 211ca85acd8c..cfab60d88c92 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -271,6 +271,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus,
 		ret = snd_hdac_ext_bus_link_power_up(link);
 
 		/*
+		 * clear the register to invalidate all the output streams
+		 */
+		snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV,
+				 ML_LOSIDV_STREAM_MASK, 0);
+		/*
 		 *  wait for 521usec for codec to report status
 		 *  HDA spec section 4.3 - Codec Discovery
 		 */
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index d3999e7b0705..7e7be8e4dcf9 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -447,8 +447,6 @@ static void azx_int_disable(struct hdac_bus *bus)
 	list_for_each_entry(azx_dev, &bus->stream_list, list)
 		snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0);
 
-	synchronize_irq(bus->irq);
-
 	/* disable SIE for all streams */
 	snd_hdac_chip_writeb(bus, INTCTL, 0);
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ca462dd39a48..2a9d87ff2e1c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1355,9 +1355,9 @@ static int azx_free(struct azx *chip)
 	}
 
 	if (bus->chip_init) {
-		azx_stop_chip(chip);
 		azx_clear_irq_pending(chip);
 		azx_stop_all_streams(chip);
+		azx_stop_chip(chip);
 	}
 
 	if (bus->irq >= 0)
@@ -2386,6 +2386,12 @@ static const struct pci_device_id azx_ids[] = {
 	/* Icelake */
 	{ PCI_DEVICE(0x8086, 0x34c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Jasperlake */
+	{ PCI_DEVICE(0x8086, 0x38c8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Tigerlake */
+	{ PCI_DEVICE(0x8086, 0xa0c8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bca5de78e9ad..795cbda32cbb 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3474,6 +3474,8 @@ static int patch_nvhdmi(struct hda_codec *codec)
 		nvhdmi_chmap_cea_alloc_validate_get_type;
 	spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
 
+	codec->link_down_at_suspend = 1;
+
 	generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin);
 
 	return 0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b000b36ac3c6..80f66ba85f87 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -393,6 +393,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0700:
 	case 0x10ec0701:
 	case 0x10ec0703:
+	case 0x10ec0711:
 		alc_update_coef_idx(codec, 0x10, 1<<15, 0);
 		break;
 	case 0x10ec0662:
@@ -408,6 +409,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0672:
 		alc_update_coef_idx(codec, 0xd, 0, 1<<14); /* EAPD Ctrl */
 		break;
+	case 0x10ec0623:
+		alc_update_coef_idx(codec, 0x19, 1<<13, 0);
+		break;
 	case 0x10ec0668:
 		alc_update_coef_idx(codec, 0x7, 3<<13, 0);
 		break;
@@ -2919,6 +2923,7 @@ enum {
 	ALC269_TYPE_ALC225,
 	ALC269_TYPE_ALC294,
 	ALC269_TYPE_ALC300,
+	ALC269_TYPE_ALC623,
 	ALC269_TYPE_ALC700,
 };
 
@@ -2954,6 +2959,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC225:
 	case ALC269_TYPE_ALC294:
 	case ALC269_TYPE_ALC300:
+	case ALC269_TYPE_ALC623:
 	case ALC269_TYPE_ALC700:
 		ssids = alc269_ssids;
 		break;
@@ -5358,6 +5364,17 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 	}
 }
 
+static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec,
+						      const struct hda_fixup *fix,
+						      int action)
+{
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+
+	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1);
+	snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP);
+}
+
 static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
 					     const struct hda_fixup *fix,
 					     int action)
@@ -5822,6 +5839,7 @@ enum {
 	ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
 	ALC275_FIXUP_DELL_XPS,
 	ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
+	ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2,
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
 	ALC255_FIXUP_DELL_SPK_NOISE,
@@ -5869,6 +5887,7 @@ enum {
 	ALC225_FIXUP_WYSE_AUTO_MUTE,
 	ALC225_FIXUP_WYSE_DISABLE_MIC_VREF,
 	ALC286_FIXUP_ACER_AIO_HEADSET_MIC,
+	ALC256_FIXUP_ASUS_HEADSET_MIC,
 	ALC256_FIXUP_ASUS_MIC_NO_PRESENCE,
 	ALC299_FIXUP_PREDATOR_SPK,
 	ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC,
@@ -6558,6 +6577,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
 	},
+	[ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc256_fixup_dell_xps_13_headphone_noise2,
+		.chained = true,
+		.chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE
+	},
 	[ALC293_FIXUP_LENOVO_SPK_NOISE] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_disable_aamix,
@@ -6912,6 +6937,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE
 	},
+	[ALC256_FIXUP_ASUS_HEADSET_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11020 }, /* headset mic with jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+	},
 	[ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -7001,17 +7035,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
 	SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
 	SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
-	SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
+	SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
 	SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
 	SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE),
 	SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP),
-	SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
+	SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
 	SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME),
 	SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME),
 	SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3),
 	SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
 	SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
+	SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
 	SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
 	SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
 	SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC),
@@ -7108,6 +7142,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
@@ -7186,6 +7221,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
@@ -7987,9 +8024,13 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->codec_variant = ALC269_TYPE_ALC300;
 		spec->gen.mixer_nid = 0; /* no loopback on ALC300 */
 		break;
+	case 0x10ec0623:
+		spec->codec_variant = ALC269_TYPE_ALC623;
+		break;
 	case 0x10ec0700:
 	case 0x10ec0701:
 	case 0x10ec0703:
+	case 0x10ec0711:
 		spec->codec_variant = ALC269_TYPE_ALC700;
 		spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */
 		alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */
@@ -9187,6 +9228,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269),
 	HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
 	HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
 	HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
@@ -9204,6 +9246,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662),
 	HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
 	HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index e609abcf3220..eb709d528259 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c,
 		max98373->i_slot = value & 0xF;
 	else
 		max98373->i_slot = 1;
-
-	max98373->reset_gpio = of_get_named_gpio(dev->of_node,
+	if (dev->of_node) {
+		max98373->reset_gpio = of_get_named_gpio(dev->of_node,
 						"maxim,reset-gpio", 0);
-	if (!gpio_is_valid(max98373->reset_gpio)) {
-		dev_err(dev, "Looking up %s property in node %s failed %d\n",
-			"maxim,reset-gpio", dev->of_node->full_name,
-			max98373->reset_gpio);
+		if (!gpio_is_valid(max98373->reset_gpio)) {
+			dev_err(dev, "Looking up %s property in node %s failed %d\n",
+				"maxim,reset-gpio", dev->of_node->full_name,
+				max98373->reset_gpio);
+		} else {
+			dev_dbg(dev, "maxim,reset-gpio=%d",
+				max98373->reset_gpio);
+		}
 	} else {
-		dev_dbg(dev, "maxim,reset-gpio=%d",
-			max98373->reset_gpio);
+		/* this makes reset_gpio as invalid */
+		max98373->reset_gpio = -1;
 	}
 
 	if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value))
diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 9fa5d44fdc79..58b2468fb2a7 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -243,6 +243,10 @@ static const char *const rx_mix1_text[] = {
 	"ZERO", "IIR1", "IIR2", "RX1", "RX2", "RX3"
 };
 
+static const char * const rx_mix2_text[] = {
+	"ZERO", "IIR1", "IIR2"
+};
+
 static const char *const dec_mux_text[] = {
 	"ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2"
 };
@@ -270,6 +274,16 @@ static const struct soc_enum rx3_mix1_inp_enum[] = {
 	SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text),
 };
 
+/* RX1 MIX2 */
+static const struct soc_enum rx_mix2_inp1_chain_enum =
+	SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX1_B3_CTL,
+		0, 3, rx_mix2_text);
+
+/* RX2 MIX2 */
+static const struct soc_enum rx2_mix2_inp1_chain_enum =
+	SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B3_CTL,
+		0, 3, rx_mix2_text);
+
 /* DEC */
 static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE(
 				LPASS_CDC_CONN_TX_B1_CTL, 0, 6, dec_mux_text);
@@ -309,6 +323,10 @@ static const struct snd_kcontrol_new rx3_mix1_inp2_mux = SOC_DAPM_ENUM(
 				"RX3 MIX1 INP2 Mux", rx3_mix1_inp_enum[1]);
 static const struct snd_kcontrol_new rx3_mix1_inp3_mux = SOC_DAPM_ENUM(
 				"RX3 MIX1 INP3 Mux", rx3_mix1_inp_enum[2]);
+static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM(
+				"RX1 MIX2 INP1 Mux", rx_mix2_inp1_chain_enum);
+static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM(
+				"RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum);
 
 /* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */
 static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0);
@@ -740,6 +758,10 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = {
 			 &rx3_mix1_inp2_mux),
 	SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0,
 			 &rx3_mix1_inp3_mux),
+	SND_SOC_DAPM_MUX("RX1 MIX2 INP1", SND_SOC_NOPM, 0, 0,
+			 &rx1_mix2_inp1_mux),
+	SND_SOC_DAPM_MUX("RX2 MIX2 INP1", SND_SOC_NOPM, 0, 0,
+			 &rx2_mix2_inp1_mux),
 
 	SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux),
 	SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux),
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 762595de956c..c506c9305043 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1770,6 +1770,9 @@ static int rt5651_detect_headset(struct snd_soc_component *component)
 
 static bool rt5651_support_button_press(struct rt5651_priv *rt5651)
 {
+	if (!rt5651->hp_jack)
+		return false;
+
 	/* Button press support only works with internal jack-detection */
 	return (rt5651->hp_jack->status & SND_JACK_MICROPHONE) &&
 		rt5651->gpiod_hp_det == NULL;
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 1ef470700ed5..c50b75ce82e0 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -995,6 +995,16 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
 {
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
 
+	rt5682->hs_jack = hs_jack;
+
+	if (!hs_jack) {
+		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+				   RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+				   RT5682_POW_JDH | RT5682_POW_JDL, 0);
+		return 0;
+	}
+
 	switch (rt5682->pdata.jd_src) {
 	case RT5682_JD1:
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
@@ -1032,8 +1042,6 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
 		break;
 	}
 
-	rt5682->hs_jack = hs_jack;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index c3d06e8bc54f..d5fb7f5dd551 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -533,13 +533,10 @@ static SOC_ENUM_SINGLE_DECL(dac_osr,
 static SOC_ENUM_SINGLE_DECL(adc_osr,
 			    WM8994_OVERSAMPLING, 1, osr_text);
 
-static const struct snd_kcontrol_new wm8994_snd_controls[] = {
+static const struct snd_kcontrol_new wm8994_common_snd_controls[] = {
 SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME,
 		 WM8994_AIF1_ADC1_RIGHT_VOLUME,
 		 1, 119, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME,
-		 WM8994_AIF1_ADC2_RIGHT_VOLUME,
-		 1, 119, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME,
 		 WM8994_AIF2_ADC_RIGHT_VOLUME,
 		 1, 119, 0, digital_tlv),
@@ -556,8 +553,6 @@ SOC_ENUM("AIF2DACR Source", aif2dacr_src),
 
 SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME,
 		 WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME,
-		 WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME,
 		 WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
 
@@ -565,17 +560,12 @@ SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv),
 SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv),
 
 SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0),
-SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0),
 SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0),
 
 WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2),
 WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1),
 WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0),
 
-WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2),
-WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1),
-WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0),
-
 WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2),
 WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1),
 WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0),
@@ -594,9 +584,6 @@ SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0),
 SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf),
 SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0),
 
-SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
-SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
-
 SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf),
 SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0),
 
@@ -637,6 +624,24 @@ SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF2_DAC_FILTERS_2,
 	   8, 1, 0),
 };
 
+/* Controls not available on WM1811 */
+static const struct snd_kcontrol_new wm8994_snd_controls[] = {
+SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME,
+		 WM8994_AIF1_ADC2_RIGHT_VOLUME,
+		 1, 119, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME,
+		 WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
+
+SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0),
+
+WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2),
+WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1),
+WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0),
+
+SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
+SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
+};
+
 static const struct snd_kcontrol_new wm8994_eq_controls[] = {
 SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0,
 	       eq_tlv),
@@ -4258,13 +4263,15 @@ static int wm8994_component_probe(struct snd_soc_component *component)
 	wm8994_handle_pdata(wm8994);
 
 	wm_hubs_add_analogue_controls(component);
-	snd_soc_add_component_controls(component, wm8994_snd_controls,
-			     ARRAY_SIZE(wm8994_snd_controls));
+	snd_soc_add_component_controls(component, wm8994_common_snd_controls,
+				       ARRAY_SIZE(wm8994_common_snd_controls));
 	snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets,
 				  ARRAY_SIZE(wm8994_dapm_widgets));
 
 	switch (control->type) {
 	case WM8994:
+		snd_soc_add_component_controls(component, wm8994_snd_controls,
+					       ARRAY_SIZE(wm8994_snd_controls));
 		snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
 					  ARRAY_SIZE(wm8994_specific_dapm_widgets));
 		if (control->revision < 4) {
@@ -4284,8 +4291,10 @@ static int wm8994_component_probe(struct snd_soc_component *component)
 		}
 		break;
 	case WM8958:
+		snd_soc_add_component_controls(component, wm8994_snd_controls,
+					       ARRAY_SIZE(wm8994_snd_controls));
 		snd_soc_add_component_controls(component, wm8958_snd_controls,
-				     ARRAY_SIZE(wm8958_snd_controls));
+					       ARRAY_SIZE(wm8958_snd_controls));
 		snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
 					  ARRAY_SIZE(wm8958_dapm_widgets));
 		if (control->revision < 1) {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ae28d9907c30..9b8bb7bbe945 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1259,8 +1259,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len)
 	}
 
 	if (in) {
-		if (in & WMFW_CTL_FLAG_READABLE)
-			out |= rd;
+		out |= rd;
 		if (in & WMFW_CTL_FLAG_WRITEABLE)
 			out |= wr;
 		if (in & WMFW_CTL_FLAG_VOLATILE)
@@ -3697,11 +3696,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp)
 	u32 xmalg, addr, magic;
 	int i, ret;
 
+	alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id);
+	if (!alg_region) {
+		adsp_err(dsp, "No algorithm region found\n");
+		return -EINVAL;
+	}
+
 	buf = wm_adsp_buffer_alloc(dsp);
 	if (!buf)
 		return -ENOMEM;
 
-	alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id);
 	xmalg = dsp->ops->sys_config_size / sizeof(__be32);
 
 	addr = alg_region->base + xmalg + ALG_XM_FIELD(magic);
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index a437567b8cee..4f6e58c3954a 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -308,6 +308,9 @@ static const struct snd_soc_dapm_widget sof_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_SPK("Spk", NULL),
+};
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
 	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
 };
 
@@ -318,10 +321,6 @@ static const struct snd_soc_dapm_route sof_map[] = {
 
 	/* other jacks */
 	{ "IN1P", NULL, "Headset Mic" },
-
-	/* digital mics */
-	{"DMic", NULL, "SoC DMIC"},
-
 };
 
 static const struct snd_soc_dapm_route speaker_map[] = {
@@ -329,6 +328,11 @@ static const struct snd_soc_dapm_route speaker_map[] = {
 	{ "Spk", NULL, "Speaker" },
 };
 
+static const struct snd_soc_dapm_route dmic_map[] = {
+	/* digital mics */
+	{"DMic", NULL, "SoC DMIC"},
+};
+
 static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_card *card = rtd->card;
@@ -342,6 +346,28 @@ static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
+static int dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+					ARRAY_SIZE(dmic_widgets));
+	if (ret) {
+		dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+		/* Don't need to add routes if widget addition failed */
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+				      ARRAY_SIZE(dmic_map));
+
+	if (ret)
+		dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+	return ret;
+}
+
 /* sof audio machine driver for rt5682 codec */
 static struct snd_soc_card sof_audio_card_rt5682 = {
 	.name = "sof_rt5682",
@@ -445,6 +471,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
 		links[id].name = "dmic01";
 		links[id].cpus = &cpus[id];
 		links[id].cpus->dai_name = "DMIC01 Pin";
+		links[id].init = dmic_init;
 		if (dmic_be_num > 1) {
 			/* set up 2 BE links at most */
 			links[id + 1].name = "dmic16k";
@@ -576,6 +603,15 @@ static int sof_audio_probe(struct platform_device *pdev)
 	/* need to get main clock from pmc */
 	if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
 		ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
+		if (IS_ERR(ctx->mclk)) {
+			ret = PTR_ERR(ctx->mclk);
+
+			dev_err(&pdev->dev,
+				"Failed to get MCLK from pmc_plt_clk_3: %d\n",
+				ret);
+			return ret;
+		}
+
 		ret = clk_prepare_enable(ctx->mclk);
 		if (ret < 0) {
 			dev_err(&pdev->dev,
@@ -621,8 +657,24 @@ static int sof_audio_probe(struct platform_device *pdev)
 					  &sof_audio_card_rt5682);
 }
 
+static int sof_rt5682_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_component *component = NULL;
+
+	for_each_card_components(card, component) {
+		if (!strcmp(component->name, rt5682_component[0].name)) {
+			snd_soc_component_set_jack(component, NULL, NULL);
+			break;
+		}
+	}
+
+	return 0;
+}
+
 static struct platform_driver sof_audio = {
 	.probe = sof_audio_probe,
+	.remove = sof_rt5682_remove,
 	.driver = {
 		.name = "sof_rt5682",
 		.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index af2d5a6124c8..61c984f10d8e 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -677,7 +677,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 	ret = rockchip_pcm_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register PCM\n");
-		return ret;
+		goto err_suspend;
 	}
 
 	return 0;
diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c
index c213913eb984..fd8c6642fb0d 100644
--- a/sound/soc/samsung/arndale_rt5631.c
+++ b/sound/soc/samsung/arndale_rt5631.c
@@ -5,6 +5,7 @@
 //  Author: Claude <claude@insginal.co.kr>
 
 #include <linux/module.h>
+#include <linux/of_device.h>
 #include <linux/platform_device.h>
 #include <linux/clk.h>
 
@@ -74,6 +75,17 @@ static struct snd_soc_card arndale_rt5631 = {
 	.num_links = ARRAY_SIZE(arndale_rt5631_dai),
 };
 
+static void arndale_put_of_nodes(struct snd_soc_card *card)
+{
+	struct snd_soc_dai_link *dai_link;
+	int i;
+
+	for_each_card_prelinks(card, i, dai_link) {
+		of_node_put(dai_link->cpus->of_node);
+		of_node_put(dai_link->codecs->of_node);
+	}
+}
+
 static int arndale_audio_probe(struct platform_device *pdev)
 {
 	int n, ret;
@@ -103,18 +115,31 @@ static int arndale_audio_probe(struct platform_device *pdev)
 		if (!arndale_rt5631_dai[0].codecs->of_node) {
 			dev_err(&pdev->dev,
 			"Property 'samsung,audio-codec' missing or invalid\n");
-			return -EINVAL;
+			ret = -EINVAL;
+			goto err_put_of_nodes;
 		}
 	}
 
 	ret = devm_snd_soc_register_card(card->dev, card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret);
+		goto err_put_of_nodes;
+	}
+	return 0;
 
-	if (ret)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
-
+err_put_of_nodes:
+	arndale_put_of_nodes(card);
 	return ret;
 }
 
+static int arndale_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	arndale_put_of_nodes(card);
+	return 0;
+}
+
 static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = {
 	{ .compatible = "samsung,arndale-rt5631", },
 	{ .compatible = "samsung,arndale-alc5631", },
@@ -129,6 +154,7 @@ static struct platform_driver arndale_audio_driver = {
 		.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
 	},
 	.probe = arndale_audio_probe,
+	.remove = arndale_audio_remove,
 };
 
 module_platform_driver(arndale_audio_driver);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index bda5b958d0dc..e9596c2096cd 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -761,6 +761,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	}
 
 	/* set format */
+	rdai->bit_clk_inv = 0;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		rdai->sys_delay = 0;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e163dde5eab1..b600d3eaaf5c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1070,7 +1070,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 			return ret;
 	}
 
-	snd_soc_dai_trigger(cpu_dai, substream, cmd);
+	ret = snd_soc_dai_trigger(cpu_dai, substream, cmd);
 	if (ret < 0)
 		return ret;
 
@@ -1097,7 +1097,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
 			return ret;
 	}
 
-	snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
+	ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
 	if (ret < 0)
 		return ret;
 
@@ -1146,6 +1146,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
 {
 	struct snd_soc_dpcm *dpcm;
 	unsigned long flags;
+	char *name;
 
 	/* only add new dpcms */
 	for_each_dpcm_be(fe, stream, dpcm) {
@@ -1171,9 +1172,15 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
 			stream ? "<-" : "->", be->dai_link->name);
 
 #ifdef CONFIG_DEBUG_FS
-	dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name,
-						 fe->debugfs_dpcm_root);
-	debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state);
+	name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name,
+			 stream ? "capture" : "playback");
+	if (name) {
+		dpcm->debugfs_state = debugfs_create_dir(name,
+							 fe->debugfs_dpcm_root);
+		debugfs_create_u32("state", 0644, dpcm->debugfs_state,
+				   &dpcm->state);
+		kfree(name);
+	}
 #endif
 	return 1;
 }
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index aa9a1fca46fa..0fd032914a31 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1582,7 +1582,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg,
 
 	/* map user to kernel widget ID */
 	template.id = get_widget_id(le32_to_cpu(w->id));
-	if (template.id < 0)
+	if ((int)template.id < 0)
 		return template.id;
 
 	/* strings are allocated here, but used and freed by the widget */
diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c
index a4983f90ff5b..2b8711eda362 100644
--- a/sound/soc/sof/control.c
+++ b/sound/soc/sof/control.c
@@ -60,13 +60,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol,
 	struct snd_sof_dev *sdev = scontrol->sdev;
 	struct sof_ipc_ctrl_data *cdata = scontrol->control_data;
 	unsigned int i, channels = scontrol->num_channels;
+	bool change = false;
+	u32 value;
 
 	/* update each channel */
 	for (i = 0; i < channels; i++) {
-		cdata->chanv[i].value =
-			mixer_to_ipc(ucontrol->value.integer.value[i],
+		value = mixer_to_ipc(ucontrol->value.integer.value[i],
 				     scontrol->volume_table, sm->max + 1);
+		change = change || (value != cdata->chanv[i].value);
 		cdata->chanv[i].channel = i;
+		cdata->chanv[i].value = value;
 	}
 
 	/* notify DSP of mixer updates */
@@ -76,8 +79,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol,
 					      SOF_CTRL_TYPE_VALUE_CHAN_GET,
 					      SOF_CTRL_CMD_VOLUME,
 					      true);
-
-	return 0;
+	return change;
 }
 
 int snd_sof_switch_get(struct snd_kcontrol *kcontrol,
@@ -105,11 +107,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol,
 	struct snd_sof_dev *sdev = scontrol->sdev;
 	struct sof_ipc_ctrl_data *cdata = scontrol->control_data;
 	unsigned int i, channels = scontrol->num_channels;
+	bool change = false;
+	u32 value;
 
 	/* update each channel */
 	for (i = 0; i < channels; i++) {
-		cdata->chanv[i].value = ucontrol->value.integer.value[i];
+		value = ucontrol->value.integer.value[i];
+		change = change || (value != cdata->chanv[i].value);
 		cdata->chanv[i].channel = i;
+		cdata->chanv[i].value = value;
 	}
 
 	/* notify DSP of mixer updates */
@@ -120,7 +126,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol,
 					      SOF_CTRL_CMD_SWITCH,
 					      true);
 
-	return 0;
+	return change;
 }
 
 int snd_sof_enum_get(struct snd_kcontrol *kcontrol,
@@ -148,11 +154,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol,
 	struct snd_sof_dev *sdev = scontrol->sdev;
 	struct sof_ipc_ctrl_data *cdata = scontrol->control_data;
 	unsigned int i, channels = scontrol->num_channels;
+	bool change = false;
+	u32 value;
 
 	/* update each channel */
 	for (i = 0; i < channels; i++) {
-		cdata->chanv[i].value = ucontrol->value.enumerated.item[i];
+		value = ucontrol->value.enumerated.item[i];
+		change = change || (value != cdata->chanv[i].value);
 		cdata->chanv[i].channel = i;
+		cdata->chanv[i].value = value;
 	}
 
 	/* notify DSP of enum updates */
@@ -163,7 +173,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol,
 					      SOF_CTRL_CMD_ENUM,
 					      true);
 
-	return 0;
+	return change;
 }
 
 int snd_sof_bytes_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 806dfa0e5eae..05f4aed13af9 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -273,6 +273,16 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC
 	  Say Y if you want to enable HDAudio codecs with SOF.
 	  If unsure select "N".
 
+config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1
+	bool "SOF enable DMI Link L1"
+	help
+	  This option enables DMI L1 for both playback and capture
+	  and disables known workarounds for specific HDaudio platforms.
+	  Only use to look into power optimizations on platforms not
+	  affected by DMI L1 issues. This option is not recommended.
+	  Say Y if you want to enable DMI Link L1
+	  If unsure, select "N".
+
 endif ## SND_SOC_SOF_HDA_COMMON
 
 config SND_SOC_SOF_HDA_LINK_BASELINE
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index e282179263e8..80e2826fb447 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -37,6 +37,7 @@
 #define MBOX_SIZE       0x1000
 #define MBOX_DUMP_SIZE 0x30
 #define EXCEPT_OFFSET	0x800
+#define EXCEPT_MAX_HDR_SIZE	0x400
 
 /* DSP peripherals */
 #define DMAC0_OFFSET    0xFE000
@@ -228,6 +229,11 @@ static void bdw_get_registers(struct snd_sof_dev *sdev,
 	/* note: variable AR register array is not read */
 
 	/* then get panic info */
+	if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+		dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+			xoops->arch_hdr.totalsize);
+		return;
+	}
 	offset += xoops->arch_hdr.totalsize;
 	sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info));
 
@@ -451,6 +457,7 @@ static int bdw_probe(struct snd_sof_dev *sdev)
 	/* TODO: add offsets */
 	sdev->mmio_bar = BDW_DSP_BAR;
 	sdev->mailbox_bar = BDW_DSP_BAR;
+	sdev->dsp_oops_offset = MBOX_OFFSET;
 
 	/* PCI base */
 	mmio = platform_get_resource(pdev, IORESOURCE_MEM,
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 5e7a6aaa627a..a1e514f71739 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -28,6 +28,7 @@
 #define MBOX_OFFSET		0x144000
 #define MBOX_SIZE		0x1000
 #define EXCEPT_OFFSET		0x800
+#define EXCEPT_MAX_HDR_SIZE	0x400
 
 /* DSP peripherals */
 #define DMAC0_OFFSET		0x098000
@@ -126,6 +127,11 @@ static void byt_get_registers(struct snd_sof_dev *sdev,
 	/* note: variable AR register array is not read */
 
 	/* then get panic info */
+	if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+		dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+			xoops->arch_hdr.totalsize);
+		return;
+	}
 	offset += xoops->arch_hdr.totalsize;
 	sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info));
 
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index bc41028a7a01..df1909e1d950 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -139,20 +139,16 @@ void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable)
  */
 int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable)
 {
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-	struct hdac_bus *bus = sof_to_bus(sdev);
-#endif
 	u32 val;
 
 	/* enable/disable audio dsp clock gating */
 	val = enable ? PCI_CGCTL_ADSPDCGE : 0;
 	snd_sof_pci_update_bits(sdev, PCI_CGCTL, PCI_CGCTL_ADSPDCGE, val);
 
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-	/* enable/disable L1 support */
-	val = enable ? SOF_HDA_VS_EM2_L1SEN : 0;
-	snd_hdac_chip_updatel(bus, VS_EM2, SOF_HDA_VS_EM2_L1SEN, val);
-#endif
+	/* enable/disable DMI Link L1 support */
+	val = enable ? HDA_VS_INTEL_EM2_L1SEN : 0;
+	snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2,
+				HDA_VS_INTEL_EM2_L1SEN, val);
 
 	/* enable/disable audio dsp power gating */
 	val = enable ? 0 : PCI_PGCTL_ADSPPGD;
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 6427f0b3a2f1..65c2af3fcaab 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -44,6 +44,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
 		return -ENODEV;
 	}
 	hstream = &dsp_stream->hstream;
+	hstream->substream = NULL;
 
 	/* allocate DMA buffer */
 	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab);
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index ad8d41f22e92..2c7447188402 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -185,6 +185,17 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction)
 			direction == SNDRV_PCM_STREAM_PLAYBACK ?
 			"playback" : "capture");
 
+	/*
+	 * Disable DMI Link L1 entry when capture stream is opened.
+	 * Workaround to address a known issue with host DMA that results
+	 * in xruns during pause/release in capture scenarios.
+	 */
+	if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1))
+		if (stream && direction == SNDRV_PCM_STREAM_CAPTURE)
+			snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
+						HDA_VS_INTEL_EM2,
+						HDA_VS_INTEL_EM2_L1SEN, 0);
+
 	return stream;
 }
 
@@ -193,23 +204,43 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag)
 {
 	struct hdac_bus *bus = sof_to_bus(sdev);
 	struct hdac_stream *s;
+	bool active_capture_stream = false;
+	bool found = false;
 
 	spin_lock_irq(&bus->reg_lock);
 
-	/* find used stream */
+	/*
+	 * close stream matching the stream tag
+	 * and check if there are any open capture streams.
+	 */
 	list_for_each_entry(s, &bus->stream_list, list) {
-		if (s->direction == direction &&
-		    s->opened && s->stream_tag == stream_tag) {
+		if (!s->opened)
+			continue;
+
+		if (s->direction == direction && s->stream_tag == stream_tag) {
 			s->opened = false;
-			spin_unlock_irq(&bus->reg_lock);
-			return 0;
+			found = true;
+		} else if (s->direction == SNDRV_PCM_STREAM_CAPTURE) {
+			active_capture_stream = true;
 		}
 	}
 
 	spin_unlock_irq(&bus->reg_lock);
 
-	dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag);
-	return -ENODEV;
+	/* Enable DMI L1 entry if there are no capture streams open */
+	if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1))
+		if (!active_capture_stream)
+			snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
+						HDA_VS_INTEL_EM2,
+						HDA_VS_INTEL_EM2_L1SEN,
+						HDA_VS_INTEL_EM2_L1SEN);
+
+	if (!found) {
+		dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag);
+		return -ENODEV;
+	}
+
+	return 0;
 }
 
 int hda_dsp_stream_trigger(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index c72e9a09eee1..06e84679087b 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -35,6 +35,8 @@
 #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348)
 #define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8)
 
+#define EXCEPT_MAX_HDR_SIZE	0x400
+
 /*
  * Debug
  */
@@ -131,6 +133,11 @@ static void hda_dsp_get_registers(struct snd_sof_dev *sdev,
 	/* note: variable AR register array is not read */
 
 	/* then get panic info */
+	if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+		dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+			xoops->arch_hdr.totalsize);
+		return;
+	}
 	offset += xoops->arch_hdr.totalsize;
 	sof_block_read(sdev, sdev->mmio_bar, offset,
 		       panic_info, sizeof(*panic_info));
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 5591841a1b6f..23e430d3e056 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -39,7 +39,6 @@
 #define SOF_HDA_WAKESTS			0x0E
 #define SOF_HDA_WAKESTS_INT_MASK	((1 << 8) - 1)
 #define SOF_HDA_RIRBSTS			0x5d
-#define SOF_HDA_VS_EM2_L1SEN            BIT(13)
 
 /* SOF_HDA_GCTL register bist */
 #define SOF_HDA_GCTL_RESET		BIT(0)
@@ -228,6 +227,10 @@
 #define HDA_DSP_REG_HIPCIE		(HDA_DSP_IPC_BASE + 0x0C)
 #define HDA_DSP_REG_HIPCCTL		(HDA_DSP_IPC_BASE + 0x10)
 
+/* Intel Vendor Specific Registers */
+#define HDA_VS_INTEL_EM2		0x1030
+#define HDA_VS_INTEL_EM2_L1SEN		BIT(13)
+
 /*  HIPCI */
 #define HDA_DSP_REG_HIPCI_BUSY		BIT(31)
 #define HDA_DSP_REG_HIPCI_MSG_MASK	0x7FFFFFFF
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index d7f32745fefe..9a9a381a908d 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -546,10 +546,10 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev)
 				 msecs_to_jiffies(sdev->boot_timeout));
 	if (ret == 0) {
 		dev_err(sdev->dev, "error: firmware boot failure\n");
-		/* after this point FW_READY msg should be ignored */
-		sdev->boot_complete = true;
 		snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX |
 			SOF_DBG_TEXT | SOF_DBG_PCI);
+		/* after this point FW_READY msg should be ignored */
+		sdev->boot_complete = true;
 		return -EIO;
 	}
 
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index e3f6a6dc0f36..2b876d497447 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -244,7 +244,7 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
 	struct snd_sof_pcm *spcm;
-	int ret;
+	int ret, err = 0;
 
 	/* nothing to do for BE */
 	if (rtd->dai_link->no_pcm)
@@ -254,26 +254,26 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (!spcm)
 		return -EINVAL;
 
-	if (!spcm->prepared[substream->stream])
-		return 0;
-
 	dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id,
 		substream->stream);
 
-	ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
+	if (spcm->prepared[substream->stream]) {
+		ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
+		if (ret < 0)
+			err = ret;
+	}
 
 	snd_pcm_lib_free_pages(substream);
 
 	cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work);
 
-	if (ret < 0)
-		return ret;
-
 	ret = snd_sof_pcm_platform_hw_free(sdev, substream);
-	if (ret < 0)
+	if (ret < 0) {
 		dev_err(sdev->dev, "error: platform hw free failed\n");
+		err = ret;
+	}
 
-	return ret;
+	return err;
 }
 
 static int sof_pcm_prepare(struct snd_pcm_substream *substream)
@@ -323,6 +323,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct sof_ipc_stream stream;
 	struct sof_ipc_reply reply;
 	bool reset_hw_params = false;
+	bool ipc_first = false;
 	int ret;
 
 	/* nothing to do for BE */
@@ -343,6 +344,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE;
+		ipc_first = true;
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE;
@@ -363,6 +365,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
+		ipc_first = true;
 		reset_hw_params = true;
 		break;
 	default:
@@ -370,12 +373,22 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		return -EINVAL;
 	}
 
-	snd_sof_pcm_platform_trigger(sdev, substream, cmd);
+	/*
+	 * DMA and IPC sequence is different for start and stop. Need to send
+	 * STOP IPC before stop DMA
+	 */
+	if (!ipc_first)
+		snd_sof_pcm_platform_trigger(sdev, substream, cmd);
 
 	/* send IPC to the DSP */
 	ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
 				 sizeof(stream), &reply, sizeof(reply));
 
+	/* need to STOP DMA even if STOP IPC failed */
+	if (ipc_first)
+		snd_sof_pcm_platform_trigger(sdev, substream, cmd);
+
+	/* free PCM if reset_hw_params is set and the STOP IPC is successful */
 	if (!ret && reset_hw_params)
 		ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
 
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index fc85efbad378..0aabb3190ddc 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -920,7 +920,9 @@ static void sof_parse_word_tokens(struct snd_soc_component *scomp,
 		for (j = 0; j < count; j++) {
 			/* match token type */
 			if (!(tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_WORD ||
-			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT))
+			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT ||
+			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BYTE ||
+			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BOOL))
 				continue;
 
 			/* match token id */
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index d7501f88aaa6..a4060813bc74 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -505,10 +505,20 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai,
 	if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) {
 		ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
 					 SAI_XCR1_NODIV,
-					 (unsigned int)~SAI_XCR1_NODIV);
+					 freq ? 0 : SAI_XCR1_NODIV);
 		if (ret < 0)
 			return ret;
 
+		/* Assume shutdown if requested frequency is 0Hz */
+		if (!freq) {
+			/* Release mclk rate only if rate was actually set */
+			if (sai->mclk_rate) {
+				clk_rate_exclusive_put(sai->sai_mclk);
+				sai->mclk_rate = 0;
+			}
+			return 0;
+		}
+
 		/* If master clock is used, set parent clock now */
 		ret = stm32_sai_set_parent_clock(sai, freq);
 		if (ret)
@@ -1093,15 +1103,6 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
 
 	regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0);
 
-	regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV,
-			   SAI_XCR1_NODIV);
-
-	/* Release mclk rate only if rate was actually set */
-	if (sai->mclk_rate) {
-		clk_rate_exclusive_put(sai->sai_mclk);
-		sai->mclk_rate = 0;
-	}
-
 	clk_disable_unprepare(sai->sai_ck);
 
 	spin_lock_irqsave(&sai->irq_lock, flags);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 33cd26763c0e..ff5ab24f3bd1 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -348,6 +348,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ep = 0x84;
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;
+	case USB_ID(0x0582, 0x01d8): /* BOSS Katana */
+		/* BOSS Katana amplifiers do not need quirks */
+		return 0;
 	}
 
 	if (attr == USB_ENDPOINT_SYNC_ASYNC &&
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index fbfde996fee7..0bbe1201a6ac 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1657,6 +1657,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	case 0x23ba:  /* Playback Designs */
 	case 0x25ce:  /* Mytek devices */
 	case 0x278b:  /* Rotel? */
+	case 0x292b:  /* Gustard/Ess based devices */
 	case 0x2ab6:  /* T+A devices */
 	case 0x3842:  /* EVGA */
 	case 0xc502:  /* HiBy devices */
diff --git a/sound/usb/validate.c b/sound/usb/validate.c
index 3c8f73a0eb12..a5e584b60dcd 100644
--- a/sound/usb/validate.c
+++ b/sound/usb/validate.c
@@ -75,7 +75,7 @@ static bool validate_processing_unit(const void *p,
 
 	if (d->bLength < sizeof(*d))
 		return false;
-	len = d->bLength < sizeof(*d) + d->bNrInPins;
+	len = sizeof(*d) + d->bNrInPins;
 	if (d->bLength < len)
 		return false;
 	switch (v->protocol) {