diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
index 7cd0e278ed85..a3ba2186d6a1 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
@@ -56,6 +56,9 @@ properties:
       - const: tx
       - const: rx
 
+  power-domains:
+    maxItems: 1
+
   rockchip,capture-channels:
     allOf:
       - $ref: /schemas/types.yaml#/definitions/uint32
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
deleted file mode 100644
index ec20c1271e92..000000000000
--- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
+++ /dev/null
@@ -1,45 +0,0 @@
-* Rockchip SPDIF transceiver
-
-The S/PDIF audio block is a stereo transceiver that allows the
-processor to receive and transmit digital audio via an coaxial cable or
-a fibre cable.
-
-Required properties:
-
-- compatible: should be one of the following:
-   - "rockchip,rk3066-spdif"
-   - "rockchip,rk3188-spdif"
-   - "rockchip,rk3228-spdif"
-   - "rockchip,rk3288-spdif"
-   - "rockchip,rk3328-spdif"
-   - "rockchip,rk3366-spdif"
-   - "rockchip,rk3368-spdif"
-   - "rockchip,rk3399-spdif"
-- reg: physical base address of the controller and length of memory mapped
-  region.
-- interrupts: should contain the SPDIF interrupt.
-- dmas: DMA specifiers for tx dma. See the DMA client binding,
-  Documentation/devicetree/bindings/dma/dma.txt
-- dma-names: should be "tx"
-- clocks: a list of phandle + clock-specifier pairs, one for each entry
-  in clock-names.
-- clock-names: should contain following:
-   - "hclk": clock for SPDIF controller
-   - "mclk" : clock for SPDIF bus
-
-Required properties on RK3288:
-  - rockchip,grf: the phandle of the syscon node for the general register
-                   file (GRF)
-
-Example for the rk3188 SPDIF controller:
-
-spdif: spdif@1011e000 {
-	compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
-	reg = <0x1011e000 0x2000>;
-	interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
-	dmas = <&dmac1_s 8>;
-	dma-names = "tx";
-	clock-names = "hclk", "mclk";
-	clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>;
-	#sound-dai-cells = <0>;
-};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
new file mode 100644
index 000000000000..c467152656f7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
@@ -0,0 +1,101 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip SPDIF transceiver
+
+description:
+  The S/PDIF audio block is a stereo transceiver that allows the
+  processor to receive and transmit digital audio via a coaxial or
+  fibre cable.
+
+maintainers:
+  - Heiko Stuebner <heiko@sntech.de>
+
+properties:
+  compatible:
+    oneOf:
+      - const: rockchip,rk3066-spdif
+      - const: rockchip,rk3228-spdif
+      - const: rockchip,rk3328-spdif
+      - const: rockchip,rk3366-spdif
+      - const: rockchip,rk3368-spdif
+      - const: rockchip,rk3399-spdif
+      - items:
+          - enum:
+            - rockchip,rk3188-spdif
+            - rockchip,rk3288-spdif
+          - const: rockchip,rk3066-spdif
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: clock for SPDIF bus
+      - description: clock for SPDIF controller
+
+  clock-names:
+    items:
+      - const: mclk
+      - const: hclk
+
+  dmas:
+    maxItems: 1
+
+  dma-names:
+    const: tx
+
+  power-domains:
+    maxItems: 1
+
+  rockchip,grf:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      The phandle of the syscon node for the GRF register.
+      Required property on RK3288.
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - "#sound-dai-cells"
+
+if:
+  properties:
+    compatible:
+      contains:
+        const: rockchip,rk3288-spdif
+
+then:
+  required:
+    - rockchip,grf
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/rk3188-cru.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    spdif: spdif@1011e000 {
+      compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
+      reg = <0x1011e000 0x2000>;
+      interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
+      clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>;
+      clock-names = "mclk", "hclk";
+      dmas = <&dmac1_s 8>;
+      dma-names = "tx";
+      #sound-dai-cells = <0>;
+    };
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index d4825b82c7a3..b33abe93b905 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -351,7 +351,6 @@ struct snd_soc_dai {
 
 	/* bit field */
 	unsigned int probed:1;
-	unsigned int started[SNDRV_PCM_STREAM_LAST + 1];
 };
 
 static inline struct snd_soc_pcm_stream *
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 13458e4fbb13..946f88a6c63d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -790,6 +790,9 @@ struct snd_soc_dai_link {
 	const struct snd_soc_pcm_stream *params;
 	unsigned int num_params;
 
+	struct snd_soc_dapm_widget *playback_widget;
+	struct snd_soc_dapm_widget *capture_widget;
+
 	unsigned int dai_fmt;           /* format to set on init */
 
 	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 86a632bf4d50..7e3ae4534df9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work)
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, jackpoll_work.work);
 
-	snd_hda_jack_set_dirty_all(codec);
-	snd_hda_jack_poll_all(codec);
+	/* for non-polling trigger: we need nothing if already powered on */
+	if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core))
+		return;
+
+	/* the power-up/down sequence triggers the runtime resume */
+	snd_hda_power_up_pm(codec);
+	/* update jacks manually if polling is required, too */
+	if (codec->jackpoll_interval) {
+		snd_hda_jack_set_dirty_all(codec);
+		snd_hda_jack_poll_all(codec);
+	}
+	snd_hda_power_down_pm(codec);
 
 	if (!codec->jackpoll_interval)
 		return;
@@ -2951,18 +2961,14 @@ static int hda_codec_runtime_resume(struct device *dev)
 static int hda_codec_force_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
-	bool forced_resume = hda_codec_need_resume(codec);
 	int ret;
 
-	/* The get/put pair below enforces the runtime resume even if the
-	 * device hasn't been used at suspend time.  This trick is needed to
-	 * update the jack state change during the sleep.
-	 */
-	if (forced_resume)
-		pm_runtime_get_noresume(dev);
 	ret = pm_runtime_force_resume(dev);
-	if (forced_resume)
-		pm_runtime_put(dev);
+	/* schedule jackpoll work for jack detection update */
+	if (codec->jackpoll_interval ||
+	    (pm_runtime_suspended(dev) && hda_codec_need_resume(codec)))
+		schedule_delayed_work(&codec->jackpoll_work,
+				      codec->jackpoll_interval);
 	return ret;
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index a5fab12defde..0310193ea1bd 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1004,7 +1004,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt)
 
 	if (status && from_rt) {
 		list_for_each_codec(codec, &chip->bus)
-			if (status & (1 << codec->addr))
+			if (!codec->relaxed_resume &&
+			    (status & (1 << codec->addr)))
 				schedule_delayed_work(&codec->jackpoll_work,
 						      codec->jackpoll_interval);
 	}
@@ -1044,9 +1045,7 @@ static int azx_suspend(struct device *dev)
 static int azx_resume(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
-	struct hda_codec *codec;
 	struct azx *chip;
-	bool forced_resume = false;
 
 	if (!azx_is_pm_ready(card))
 		return 0;
@@ -1058,19 +1057,7 @@ static int azx_resume(struct device *dev)
 	if (azx_acquire_irq(chip, 1) < 0)
 		return -EIO;
 
-	/* check for the forced resume */
-	list_for_each_codec(codec, &chip->bus) {
-		if (hda_codec_need_resume(codec)) {
-			forced_resume = true;
-			break;
-		}
-	}
-
-	if (forced_resume)
-		pm_runtime_get_noresume(dev);
 	pm_runtime_force_resume(dev);
-	if (forced_resume)
-		pm_runtime_put(dev);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 
 	trace_azx_resume(chip);
@@ -2091,10 +2078,10 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
  * 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 pci_device_id driver_blacklist[] = {
+	{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */
+	{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */
+	{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */
 	{}
 };
 
@@ -2114,7 +2101,7 @@ static int azx_probe(struct pci_dev *pci,
 	bool schedule_probe;
 	int err;
 
-	if (snd_pci_quirk_lookup(pci, driver_blacklist)) {
+	if (pci_match_id(driver_blacklist, pci)) {
 		dev_info(&pci->dev, "Skipping the blacklisted device\n");
 		return -ENODEV;
 	}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bb287a916dae..4eff16053bd5 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -38,6 +38,10 @@ static bool static_hdmi_pcm;
 module_param(static_hdmi_pcm, bool, 0644);
 MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
 
+static bool enable_acomp = true;
+module_param(enable_acomp, bool, 0444);
+MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)");
+
 struct hdmi_spec_per_cvt {
 	hda_nid_t cvt_nid;
 	int assigned;
@@ -2505,6 +2509,11 @@ static void generic_acomp_init(struct hda_codec *codec,
 {
 	struct hdmi_spec *spec = codec->spec;
 
+	if (!enable_acomp) {
+		codec_info(codec, "audio component disabled by module option\n");
+		return;
+	}
+
 	spec->port2pin = port2pin;
 	setup_drm_audio_ops(codec, ops);
 	if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index dc5557d79c43..c1a85c8f7b69 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -377,6 +377,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0233:
 	case 0x10ec0235:
 	case 0x10ec0236:
+	case 0x10ec0245:
 	case 0x10ec0255:
 	case 0x10ec0256:
 	case 0x10ec0257:
@@ -797,9 +798,11 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports)
 {
 	if (!alc_subsystem_id(codec, ports)) {
 		struct alc_spec *spec = codec->spec;
-		codec_dbg(codec,
-			  "realtek: Enable default setup for auto mode as fallback\n");
-		spec->init_amp = ALC_INIT_DEFAULT;
+		if (spec->init_amp == ALC_INIT_UNDEFINED) {
+			codec_dbg(codec,
+				  "realtek: Enable default setup for auto mode as fallback\n");
+			spec->init_amp = ALC_INIT_DEFAULT;
+		}
 	}
 }
 
@@ -8196,6 +8199,7 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->gen.mixer_nid = 0;
 		break;
 	case 0x10ec0215:
+	case 0x10ec0245:
 	case 0x10ec0285:
 	case 0x10ec0289:
 		spec->codec_variant = ALC269_TYPE_ALC215;
@@ -9457,6 +9461,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269),
diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c
index 024a7ee54cd5..e499c00e0c66 100644
--- a/sound/soc/amd/acp3x-rt5682-max9836.c
+++ b/sound/soc/amd/acp3x-rt5682-max9836.c
@@ -89,9 +89,9 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd)
 	}
 
 	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
-	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
-	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
-	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
 
 	ret = snd_soc_component_set_jack(component, &pco_jack, NULL);
 	if (ret) {
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e6a0c5d05fa5..e60e0b6a689c 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1525,6 +1525,7 @@ config SND_SOC_WM8804_SPI
 
 config SND_SOC_WM8900
 	tristate
+	depends on SND_SOC_I2C_AND_SPI
 
 config SND_SOC_WM8903
 	tristate "Wolfson Microelectronics WM8903 CODEC"
@@ -1576,6 +1577,7 @@ config SND_SOC_WM8985
 
 config SND_SOC_WM8988
 	tristate
+	depends on SND_SOC_I2C_AND_SPI
 
 config SND_SOC_WM8990
 	tristate
@@ -1594,6 +1596,7 @@ config SND_SOC_WM8994
 
 config SND_SOC_WM8995
 	tristate
+	depends on SND_SOC_I2C_AND_SPI
 
 config SND_SOC_WM8996
 	tristate
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index fba9b749839d..f26b77faed59 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm *
 hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi,
 			   struct hdac_hdmi_cvt *cvt)
 {
-	struct hdac_hdmi_pcm *pcm = NULL;
+	struct hdac_hdmi_pcm *pcm;
 
 	list_for_each_entry(pcm, &hdmi->pcm_list, head) {
 		if (pcm->cvt == cvt)
-			break;
+			return pcm;
 	}
 
-	return pcm;
+	return NULL;
 }
 
 static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c
index 40de9d7811d1..a448d2a2918a 100644
--- a/sound/soc/codecs/madera.c
+++ b/sound/soc/codecs/madera.c
@@ -1903,7 +1903,6 @@ const struct soc_enum madera_isrc_fsh[] = {
 			      MADERA_ISRC4_FSH_SHIFT, 0xf,
 			      MADERA_RATE_ENUM_SIZE,
 			      madera_rate_text, madera_rate_val),
-
 };
 EXPORT_SYMBOL_GPL(madera_isrc_fsh);
 
@@ -1924,7 +1923,6 @@ const struct soc_enum madera_isrc_fsl[] = {
 			      MADERA_ISRC4_FSL_SHIFT, 0xf,
 			      MADERA_RATE_ENUM_SIZE,
 			      madera_rate_text, madera_rate_val),
-
 };
 EXPORT_SYMBOL_GPL(madera_isrc_fsl);
 
@@ -1938,7 +1936,6 @@ const struct soc_enum madera_asrc1_rate[] = {
 			      MADERA_ASYNC_RATE_ENUM_SIZE,
 			      madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE,
 			      madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE),
-
 };
 EXPORT_SYMBOL_GPL(madera_asrc1_rate);
 
@@ -1964,7 +1961,6 @@ const struct soc_enum madera_asrc2_rate[] = {
 			      MADERA_ASYNC_RATE_ENUM_SIZE,
 			      madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE,
 			      madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE),
-
 };
 EXPORT_SYMBOL_GPL(madera_asrc2_rate);
 
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d5130193b4a2..e8a8bf7b4ffe 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 		dev_err(&client->dev,
 			"Error %d initializing CHIP_CLK_CTRL\n", ret);
 
+	/* Mute everything to avoid pop from the following power-up */
+	ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL,
+			   SGTL5000_CHIP_ANA_CTRL_DEFAULT);
+	if (ret) {
+		dev_err(&client->dev,
+			"Error %d muting outputs via CHIP_ANA_CTRL\n", ret);
+		goto disable_clk;
+	}
+
+	/*
+	 * If VAG is powered-on (e.g. from previous boot), it would be disabled
+	 * by the write to ANA_POWER in later steps of the probe code. This
+	 * may create a loud pop even with all outputs muted. The proper way
+	 * to circumvent this is disabling the bit first and waiting the proper
+	 * cool-down time.
+	 */
+	ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value);
+	if (ret) {
+		dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret);
+		goto disable_clk;
+	}
+	if (value & SGTL5000_VAG_POWERUP) {
+		ret = regmap_update_bits(sgtl5000->regmap,
+					 SGTL5000_CHIP_ANA_POWER,
+					 SGTL5000_VAG_POWERUP,
+					 0);
+		if (ret) {
+			dev_err(&client->dev, "Error %d disabling VAG\n", ret);
+			goto disable_clk;
+		}
+
+		msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+	}
+
 	/* Follow section 2.2.1.1 of AN3663 */
 	ana_pwr = SGTL5000_ANA_POWER_DEFAULT;
 	if (sgtl5000->num_supplies <= VDDD) {
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index a4bf4bca95bf..56ec5863f250 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -233,6 +233,7 @@
 /*
  * SGTL5000_CHIP_ANA_CTRL
  */
+#define SGTL5000_CHIP_ANA_CTRL_DEFAULT		0x0133
 #define SGTL5000_LINE_OUT_MUTE			0x0100
 #define SGTL5000_HP_SEL_MASK			0x0040
 #define SGTL5000_HP_SEL_SHIFT			6
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 1554631cb397..5b7f9fcf6cbf 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 
 	priv->regmap = devm_regmap_init(dev, NULL, client,
 					priv->chip->regmap_config);
-	if (IS_ERR(priv->regmap))
-		return PTR_ERR(priv->regmap);
+	if (IS_ERR(priv->regmap)) {
+		ret = PTR_ERR(priv->regmap);
+		goto disable_regs;
+	}
 
 	priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW);
 	if (IS_ERR(priv->pdn_gpio)) {
@@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 
 	ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0);
 	if (ret)
-		return ret;
+		goto disable_regs;
 
 	usleep_range(50000, 60000);
 
@@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 		 */
 		ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0);
 		if (ret)
-			return ret;
+			goto disable_regs;
 	}
 
-	return devm_snd_soc_register_component(&client->dev,
+	ret = devm_snd_soc_register_component(&client->dev,
 				      &priv->component_driver,
 				      &tas571x_dai, 1);
+	if (ret)
+		goto disable_regs;
+
+	return ret;
+
+disable_regs:
+	regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies);
+	return ret;
 }
 
 static int tas571x_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 55112c1bba5e..6cf0f6612bda 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -860,8 +860,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 
 	wm8960->is_stream_in_use[tx] = true;
 
-	if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON &&
-	    !wm8960->is_stream_in_use[!tx])
+	if (!wm8960->is_stream_in_use[!tx])
 		return wm8960_configure_clocking(component);
 
 	return 0;
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index f2d6f2f81f14..d39d479e2378 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}, {
 		/* COMP */
 		.num = 2,
@@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}, {
 		/* BOOST */
 		.num = 3,
@@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}, {
 		/* VISENSE */
 		.num = 4,
@@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}
 };
 
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index bcedec6c6117..7d85bd5aff9f 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -113,14 +113,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
 	}
 };
 
-static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
-	{
-		.adr = 0x000210025D130800,
-		.num_endpoints = 1,
-		.endpoints = &single_endpoint,
-	}
-};
-
 static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
 	{
 		.adr = 0x000110025D130800,
diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
index ef8500349f2f..16ec9f382b0f 100644
--- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
@@ -87,14 +87,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
 	}
 };
 
-static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
-	{
-		.adr = 0x000210025D130800,
-		.num_endpoints = 1,
-		.endpoints = &single_endpoint,
-	}
-};
-
 static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
 	{
 		.adr = 0x000110025D130800,
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index af46845f4ef2..89f7f64747cd 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -338,8 +338,10 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
 
 	if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
 		ret = axg_card_parse_tdm(card, np, index);
-	else if (axg_card_cpu_is_codec(dai_link->cpus->of_node))
+	else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
 		dai_link->params = &codec_params;
+		dai_link->no_pcm = 0; /* link is not a DPCM BE */
+	}
 
 	return ret;
 }
diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c
index 7b01dcb73e5e..4abf7efb7eac 100644
--- a/sound/soc/meson/gx-card.c
+++ b/sound/soc/meson/gx-card.c
@@ -108,8 +108,10 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np,
 		ret = gx_card_parse_i2s(card, np, index);
 
 	/* Or apply codec to codec params if necessary */
-	else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL"))
+	else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
 		dai_link->params = &codec_params;
+		dai_link->no_pcm = 0; /* link is not a DPCM BE */
+	}
 
 	return ret;
 }
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index d55e3ad96716..287ad2aa27f3 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -116,10 +116,8 @@ static int apq8096_platform_probe(struct platform_device *pdev)
 	card->dev = dev;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
-	if (ret) {
-		dev_err(dev, "Error parsing OF data\n");
+	if (ret)
 		goto err;
-	}
 
 	apq8096_add_be_ops(card);
 	ret = snd_soc_register_card(card);
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index c1a7624eaf17..2a5302f1db98 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -902,6 +902,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -917,6 +919,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -931,6 +935,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -946,6 +952,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -960,6 +968,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -975,6 +985,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -989,6 +1001,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -1004,6 +1018,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index b2de65c7f95c..68e9388ff46f 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -559,10 +559,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
 	card->dev = dev;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
-	if (ret) {
-		dev_err(dev, "Error parsing OF data\n");
+	if (ret)
 		goto parse_dt_fail;
-	}
 
 	data->card = card;
 	snd_soc_card_set_drvdata(card, data);
diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 358887848293..5e95c30fb2ba 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai,
 }
 EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup);
 
-#ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
-{
-	struct s3c_i2sv2_info *i2s = to_info(dai);
-	u32 iismod;
-
-	if (dai->active) {
-		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
-		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
-		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
-
-		/* some basic suspend checks */
-
-		iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			pr_warn("%s: RXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			pr_warn("%s: TXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			pr_warn("%s: IIS active\n", __func__);
-	}
-
-	return 0;
-}
-
-static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
-{
-	struct s3c_i2sv2_info *i2s = to_info(dai);
-
-	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
-		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
-
-	if (dai->active) {
-		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
-		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
-		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
-
-		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
-		       i2s->regs + S3C2412_IISFIC);
-
-		ndelay(250);
-		writel(0x0, i2s->regs + S3C2412_IISFIC);
-	}
-
-	return 0;
-}
-#else
-#define s3c2412_i2s_suspend NULL
-#define s3c2412_i2s_resume  NULL
-#endif
-
 int s3c_i2sv2_register_component(struct device *dev, int id,
 			   const struct snd_soc_component_driver *cmp_drv,
 			   struct snd_soc_dai_driver *dai_drv)
@@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id,
 	if (!ops->delay)
 		ops->delay = s3c2412_i2s_delay;
 
-	dai_drv->suspend = s3c2412_i2s_suspend;
-	dai_drv->resume = s3c2412_i2s_resume;
-
 	return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1);
 }
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component);
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 787a3f6e9f24..b35d828c1cfe 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int s3c2412_i2s_suspend(struct snd_soc_component *component)
+{
+	struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component);
+	u32 iismod;
+
+	if (component->active) {
+		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
+		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
+		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
+
+		/* some basic suspend checks */
+
+		iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
+			pr_warn("%s: RXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
+			pr_warn("%s: TXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
+			pr_warn("%s: IIS active\n", __func__);
+	}
+
+	return 0;
+}
+
+static int s3c2412_i2s_resume(struct snd_soc_component *component)
+{
+	struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component);
+
+	pr_info("component_active %d, IISMOD %08x, IISCON %08x\n",
+		component->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+
+	if (component->active) {
+		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
+		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
+		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
+
+		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
+		       i2s->regs + S3C2412_IISFIC);
+
+		ndelay(250);
+		writel(0x0, i2s->regs + S3C2412_IISFIC);
+	}
+
+	return 0;
+}
+#else
+#define s3c2412_i2s_suspend NULL
+#define s3c2412_i2s_resume  NULL
+#endif
+
 #define S3C2412_I2S_RATES \
 	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
@@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = {
 
 static const struct snd_soc_component_driver s3c2412_i2s_component = {
 	.name		= "s3c2412-i2s",
+	.suspend	= s3c2412_i2s_suspend,
+	.resume		= s3c2412_i2s_resume,
 };
 
 static int s3c2412_iis_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fc5d089868df..4a7d3413917f 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -594,10 +594,16 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod,
 	 * Capture:  It might not receave data. Do nothing
 	 */
 	if (rsnd_io_is_play(io)) {
-		rsnd_mod_write(mod, SSICR, cr | EN);
+		rsnd_mod_write(mod, SSICR, cr | ssi->cr_en);
 		rsnd_ssi_status_check(mod, DIRQ);
 	}
 
+	/* In multi-SSI mode, stop is performed by setting ssi0129 in
+	 * SSI_CONTROL to 0 (in rsnd_ssio_stop_gen2). Do nothing here.
+	 */
+	if (rsnd_ssi_multi_slaves_runtime(io))
+		return 0;
+
 	/*
 	 * disable SSI,
 	 * and, wait idle state
@@ -737,6 +743,9 @@ static void rsnd_ssi_parent_attach(struct rsnd_mod *mod,
 	if (!rsnd_rdai_is_clk_master(rdai))
 		return;
 
+	if (rsnd_ssi_is_multi_slave(mod, io))
+		return;
+
 	switch (rsnd_mod_id(mod)) {
 	case 1:
 	case 2:
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index f35d88211887..9c7c3e7539c9 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -221,7 +221,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 			i;
 
 		for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) {
-			shift	= (i * 4) + 16;
+			shift	= (i * 4) + 20;
 			val	= (val & ~(0xF << shift)) |
 				rsnd_mod_id(pos) << shift;
 		}
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 8f3cad8db89a..31c41559034b 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -295,24 +295,17 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
 {
 	int ret = 0;
 
-	if (!dai->started[substream->stream] &&
-	    dai->driver->ops->startup)
+	if (dai->driver->ops->startup)
 		ret = dai->driver->ops->startup(substream, dai);
 
-	if (ret == 0)
-		dai->started[substream->stream] = 1;
-
 	return ret;
 }
 
 void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			 struct snd_pcm_substream *substream)
 {
-	if (dai->started[substream->stream] &&
-	    dai->driver->ops->shutdown)
+	if (dai->driver->ops->shutdown)
 		dai->driver->ops->shutdown(substream, dai);
-
-	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 679ed60d850e..e2632841b321 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -423,7 +423,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 
 			memset(&template, 0, sizeof(template));
 			template.reg = e->reg;
-			template.mask = e->mask << e->shift_l;
+			template.mask = e->mask;
 			template.shift = e->shift_l;
 			template.off_val = snd_soc_enum_item_to_val(e, 0);
 			template.on_val = template.off_val;
@@ -546,8 +546,22 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol,
 	if (data->value == value)
 		return false;
 
-	if (data->widget)
-		data->widget->on_val = value;
+	if (data->widget) {
+		switch (dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->id) {
+		case snd_soc_dapm_switch:
+		case snd_soc_dapm_mixer:
+		case snd_soc_dapm_mixer_named_ctl:
+			data->widget->on_val = value & data->widget->mask;
+			break;
+		case snd_soc_dapm_demux:
+		case snd_soc_dapm_mux:
+			data->widget->on_val = value >> data->widget->shift;
+			break;
+		default:
+			data->widget->on_val = value;
+			break;
+		}
+	}
 
 	data->value = value;
 
@@ -4165,6 +4179,8 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card,
 	w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template);
 	if (IS_ERR(w)) {
 		ret = PTR_ERR(w);
+		dev_err(rtd->dev, "ASoC: Failed to create %s widget: %d\n",
+			link_name, ret);
 		goto outfree_kcontrol_news;
 	}
 
@@ -4283,52 +4299,58 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 	return 0;
 }
 
-static void dapm_add_valid_dai_widget(struct snd_soc_card *card,
-				      struct snd_soc_pcm_runtime *rtd,
-				      struct snd_soc_dai *codec_dai,
-				      struct snd_soc_dai *cpu_dai)
+static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm,
+				    struct snd_soc_dai *src_dai,
+				    struct snd_soc_dapm_widget *src,
+				    struct snd_soc_dapm_widget *dai,
+				    struct snd_soc_dai *sink_dai,
+				    struct snd_soc_dapm_widget *sink)
 {
-	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
-	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
+	dev_dbg(dapm->dev, "connected DAI link %s:%s -> %s:%s\n",
+		src_dai->component->name, src->name,
+		sink_dai->component->name, sink->name);
+
+	if (dai) {
+		snd_soc_dapm_add_path(dapm, src, dai, NULL, NULL);
+		src = dai;
+	}
+
+	snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL);
+}
+
+static void dapm_connect_dai_pair(struct snd_soc_card *card,
+				  struct snd_soc_pcm_runtime *rtd,
+				  struct snd_soc_dai *codec_dai,
+				  struct snd_soc_dai *cpu_dai)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_dapm_widget *dai, *codec, *playback_cpu, *capture_cpu;
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_str *streams = rtd->pcm->streams;
 
-	if (rtd->dai_link->params) {
+	if (dai_link->params) {
 		playback_cpu = cpu_dai->capture_widget;
 		capture_cpu = cpu_dai->playback_widget;
 	} else {
-		playback = cpu_dai->playback_widget;
-		capture = cpu_dai->capture_widget;
-		playback_cpu = playback;
-		capture_cpu = capture;
+		playback_cpu = cpu_dai->playback_widget;
+		capture_cpu = cpu_dai->capture_widget;
 	}
 
 	/* connect BE DAI playback if widgets are valid */
 	codec = codec_dai->playback_widget;
 
 	if (playback_cpu && codec) {
-		if (!playback) {
+		if (dai_link->params && !dai_link->playback_widget) {
 			substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-			playback = snd_soc_dapm_new_dai(card, substream,
-							"playback");
-			if (IS_ERR(playback)) {
-				dev_err(rtd->dev,
-					"ASoC: Failed to create DAI %s: %ld\n",
-					codec_dai->name,
-					PTR_ERR(playback));
+			dai = snd_soc_dapm_new_dai(card, substream, "playback");
+			if (IS_ERR(dai))
 				goto capture;
-			}
-
-			snd_soc_dapm_add_path(&card->dapm, playback_cpu,
-					      playback, NULL, NULL);
+			dai_link->playback_widget = dai;
 		}
 
-		dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-			cpu_dai->component->name, playback_cpu->name,
-			codec_dai->component->name, codec->name);
-
-		snd_soc_dapm_add_path(&card->dapm, playback, codec,
-				      NULL, NULL);
+		dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu,
+					dai_link->playback_widget,
+					codec_dai, codec);
 	}
 
 capture:
@@ -4336,52 +4358,20 @@ capture:
 	codec = codec_dai->capture_widget;
 
 	if (codec && capture_cpu) {
-		if (!capture) {
+		if (dai_link->params && !dai_link->capture_widget) {
 			substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
-			capture = snd_soc_dapm_new_dai(card, substream,
-						       "capture");
-			if (IS_ERR(capture)) {
-				dev_err(rtd->dev,
-					"ASoC: Failed to create DAI %s: %ld\n",
-					codec_dai->name,
-					PTR_ERR(capture));
+			dai = snd_soc_dapm_new_dai(card, substream, "capture");
+			if (IS_ERR(dai))
 				return;
-			}
-
-			snd_soc_dapm_add_path(&card->dapm, capture,
-					      capture_cpu, NULL, NULL);
+			dai_link->capture_widget = dai;
 		}
 
-		dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-			codec_dai->component->name, codec->name,
-			cpu_dai->component->name, capture_cpu->name);
-
-		snd_soc_dapm_add_path(&card->dapm, codec, capture,
-				      NULL, NULL);
+		dapm_connect_dai_routes(&card->dapm, codec_dai, codec,
+					dai_link->capture_widget,
+					cpu_dai, capture_cpu);
 	}
 }
 
-static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
-					  struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai *codec_dai;
-	int i;
-
-	if (rtd->num_cpus == 1) {
-		for_each_rtd_codec_dais(rtd, i, codec_dai)
-			dapm_add_valid_dai_widget(card, rtd, codec_dai,
-						  rtd->cpu_dais[0]);
-	} else if (rtd->num_codecs == rtd->num_cpus) {
-		for_each_rtd_codec_dais(rtd, i, codec_dai)
-			dapm_add_valid_dai_widget(card, rtd, codec_dai,
-						  rtd->cpu_dais[i]);
-	} else {
-		dev_err(card->dev,
-			"N cpus to M codecs link is not supported yet\n");
-	}
-
-}
-
 static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
 	int event)
 {
@@ -4422,6 +4412,8 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
 void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai *codec_dai;
+	int i;
 
 	/* for each BE DAI link... */
 	for_each_card_rtds(card, rtd)  {
@@ -4432,7 +4424,18 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 		if (rtd->dai_link->dynamic)
 			continue;
 
-		dapm_connect_dai_link_widgets(card, rtd);
+		if (rtd->num_cpus == 1) {
+			for_each_rtd_codec_dais(rtd, i, codec_dai)
+				dapm_connect_dai_pair(card, rtd, codec_dai,
+						      rtd->cpu_dais[0]);
+		} else if (rtd->num_codecs == rtd->num_cpus) {
+			for_each_rtd_codec_dais(rtd, i, codec_dai)
+				dapm_connect_dai_pair(card, rtd, codec_dai,
+						      rtd->cpu_dais[i]);
+		} else {
+			dev_err(card->dev,
+				"N cpus to M codecs link is not supported yet\n");
+		}
 	}
 }
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 289aebc15529..1f302de44052 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	int i;
 
 	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
-		playback = rtd->dai_link->dpcm_playback;
-		capture = rtd->dai_link->dpcm_capture;
+		cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+		if (rtd->num_cpus > 1) {
+			dev_err(rtd->dev,
+				"DPCM doesn't support Multi CPU yet\n");
+			return -EINVAL;
+		}
+
+		playback = rtd->dai_link->dpcm_playback &&
+			   snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK);
+		capture = rtd->dai_link->dpcm_capture &&
+			  snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE);
 	} else {
 		/* Adapt stream for codec2codec links */
 		int cpu_capture = rtd->dai_link->params ?
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 87f75edba3dc..6df3b0d12d87 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -894,7 +894,13 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count,
 		}
 
 		/* create any TLV data */
-		soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+		err = soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+		if (err < 0) {
+			dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+				mc->hdr.name);
+			kfree(sm);
+			continue;
+		}
 
 		/* pass control to driver for optional further init */
 		err = soc_tplg_init_kcontrol(tplg, &kc,
@@ -1118,6 +1124,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 	struct snd_soc_tplg_hdr *hdr)
 {
 	struct snd_soc_tplg_ctl_hdr *control_hdr;
+	int ret;
 	int i;
 
 	if (tplg->pass != SOC_TPLG_PASS_MIXER) {
@@ -1146,25 +1153,30 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 		case SND_SOC_TPLG_CTL_RANGE:
 		case SND_SOC_TPLG_DAPM_CTL_VOLSW:
 		case SND_SOC_TPLG_DAPM_CTL_PIN:
-			soc_tplg_dmixer_create(tplg, 1,
-					       le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_dmixer_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_CTL_ENUM_VALUE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE:
-			soc_tplg_denum_create(tplg, 1,
-					      le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_denum_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		case SND_SOC_TPLG_CTL_BYTES:
-			soc_tplg_dbytes_create(tplg, 1,
-					       le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_dbytes_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		default:
 			soc_bind_err(tplg, control_hdr, i);
 			return -EINVAL;
 		}
+		if (ret < 0) {
+			dev_err(tplg->dev, "ASoC: invalid control\n");
+			return ret;
+		}
+
 	}
 
 	return 0;
@@ -1272,7 +1284,9 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		routes[i]->dobj.index = tplg->index;
 		list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
 
-		soc_tplg_add_route(tplg, routes[i]);
+		ret = soc_tplg_add_route(tplg, routes[i]);
+		if (ret < 0)
+			break;
 
 		/* add route, but keep going if some fail */
 		snd_soc_dapm_add_routes(dapm, routes[i], 1);
@@ -1355,7 +1369,13 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
 		}
 
 		/* create any TLV data */
-		soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+		err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+		if (err < 0) {
+			dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+				mc->hdr.name);
+			kfree(sm);
+			continue;
+		}
 
 		/* pass control to driver for optional further init */
 		err = soc_tplg_init_kcontrol(tplg, &kc[i],
@@ -1766,10 +1786,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg)
 	return 0;
 }
 
-static void set_stream_info(struct snd_soc_pcm_stream *stream,
+static int set_stream_info(struct snd_soc_pcm_stream *stream,
 	struct snd_soc_tplg_stream_caps *caps)
 {
 	stream->stream_name = kstrdup(caps->name, GFP_KERNEL);
+	if (!stream->stream_name)
+		return -ENOMEM;
+
 	stream->channels_min = le32_to_cpu(caps->channels_min);
 	stream->channels_max = le32_to_cpu(caps->channels_max);
 	stream->rates = le32_to_cpu(caps->rates);
@@ -1777,6 +1800,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream,
 	stream->rate_max = le32_to_cpu(caps->rate_max);
 	stream->formats = le64_to_cpu(caps->formats);
 	stream->sig_bits = le32_to_cpu(caps->sig_bits);
+
+	return 0;
 }
 
 static void set_dai_flags(struct snd_soc_dai_driver *dai_drv,
@@ -1812,20 +1837,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	if (dai_drv == NULL)
 		return -ENOMEM;
 
-	if (strlen(pcm->dai_name))
+	if (strlen(pcm->dai_name)) {
 		dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL);
+		if (!dai_drv->name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+	}
 	dai_drv->id = le32_to_cpu(pcm->dai_id);
 
 	if (pcm->playback) {
 		stream = &dai_drv->playback;
 		caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (pcm->capture) {
 		stream = &dai_drv->capture;
 		caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (pcm->compress)
@@ -1835,11 +1869,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
-		kfree(dai_drv->playback.stream_name);
-		kfree(dai_drv->capture.stream_name);
-		kfree(dai_drv->name);
-		kfree(dai_drv);
-		return ret;
+		goto err;
 	}
 
 	dai_drv->dobj.index = tplg->index;
@@ -1860,6 +1890,14 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 		return ret;
 	}
 
+	return 0;
+
+err:
+	kfree(dai_drv->playback.stream_name);
+	kfree(dai_drv->capture.stream_name);
+	kfree(dai_drv->name);
+	kfree(dai_drv);
+
 	return ret;
 }
 
@@ -1916,11 +1954,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
 	if (strlen(pcm->pcm_name)) {
 		link->name = kstrdup(pcm->pcm_name, GFP_KERNEL);
 		link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL);
+		if (!link->name || !link->stream_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 	}
 	link->id = le32_to_cpu(pcm->pcm_id);
 
-	if (strlen(pcm->dai_name))
+	if (strlen(pcm->dai_name)) {
 		link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL);
+		if (!link->cpus->dai_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+	}
 
 	link->codecs->name = "snd-soc-dummy";
 	link->codecs->dai_name = "snd-soc-dummy-dai";
@@ -2088,7 +2135,9 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 			_pcm = pcm;
 		} else {
 			abi_match = false;
-			pcm_new_ver(tplg, pcm, &_pcm);
+			ret = pcm_new_ver(tplg, pcm, &_pcm);
+			if (ret < 0)
+				return ret;
 		}
 
 		/* create the FE DAIs and DAI links */
@@ -2436,13 +2485,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 	if (d->playback) {
 		stream = &dai_drv->playback;
 		caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (d->capture) {
 		stream = &dai_drv->capture;
 		caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (d->flag_mask)
@@ -2454,10 +2507,15 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 	ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
-		return ret;
+		goto err;
 	}
 
 	return 0;
+
+err:
+	kfree(dai_drv->playback.stream_name);
+	kfree(dai_drv->capture.stream_name);
+	return ret;
 }
 
 /* load physical DAI elements */
@@ -2466,7 +2524,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
 {
 	struct snd_soc_tplg_dai *dai;
 	int count;
-	int i;
+	int i, ret;
 
 	count = le32_to_cpu(hdr->count);
 
@@ -2481,7 +2539,12 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
 			return -EINVAL;
 		}
 
-		soc_tplg_dai_config(tplg, dai);
+		ret = soc_tplg_dai_config(tplg, dai);
+		if (ret < 0) {
+			dev_err(tplg->dev, "ASoC: failed to configure DAI\n");
+			return ret;
+		}
+
 		tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size));
 	}
 
@@ -2589,7 +2652,7 @@ static int soc_valid_header(struct soc_tplg *tplg,
 	}
 
 	/* big endian firmware objects not supported atm */
-	if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) {
+	if (le32_to_cpu(hdr->magic) == SOC_TPLG_MAGIC_BIG_ENDIAN) {
 		dev_err(tplg->dev,
 			"ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n",
 			tplg->pass, hdr->magic,
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 6c23c5769330..a32a3ef78ec5 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach,
 static struct snd_soc_dai_driver bdw_dai[] = {
 {
 	.name = "ssp0-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp1-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 };
 
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index f84391294f12..29fd1d86156c 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach,
 static struct snd_soc_dai_driver byt_dai[] = {
 {
 	.name = "ssp0-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp1-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp2-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	}
 },
 {
 	.name = "ssp3-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp4-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "ssp5-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 };
 
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 0d0c9afd8791..41f01c3e639e 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -837,7 +837,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai,
 		cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_32);
 		break;
 	default:
-		dev_err(cpu_dai->dev, "Data format not supported");
+		dev_err(cpu_dai->dev, "Data format not supported\n");
 		return -EINVAL;
 	}
 
@@ -1547,6 +1547,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
+		conf = &stm32_sai_pcm_config_spdif;
+
 	ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
 	if (ret) {
 		if (ret != -EPROBE_DEFER)
@@ -1556,15 +1559,10 @@ 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;
-
-	return 0;
+	return ret;
 }
 
 static int stm32_sai_sub_remove(struct platform_device *pdev)
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 50e1874c847c..5ffb457cc88c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp,
 	return false;
 }
 
+/*
+ * Many Focusrite devices supports a limited set of sampling rates per
+ * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type
+ * descriptor which has a non-standard bLength = 10.
+ */
+static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip,
+					struct audioformat *fp,
+					unsigned int rate)
+{
+	struct usb_interface *iface;
+	struct usb_host_interface *alts;
+	unsigned char *fmt;
+	unsigned int max_rate;
+
+	iface = usb_ifnum_to_if(chip->dev, fp->iface);
+	if (!iface)
+		return true;
+
+	alts = &iface->altsetting[fp->altset_idx];
+	fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen,
+				      NULL, UAC_FORMAT_TYPE);
+	if (!fmt)
+		return true;
+
+	if (fmt[0] == 10) { /* bLength */
+		max_rate = combine_quad(&fmt[6]);
+
+		/* Validate max rate */
+		if (max_rate != 48000 &&
+		    max_rate != 96000 &&
+		    max_rate != 192000 &&
+		    max_rate != 384000) {
+
+			usb_audio_info(chip,
+				"%u:%d : unexpected max rate: %u\n",
+				fp->iface, fp->altsetting, max_rate);
+
+			return true;
+		}
+
+		return rate <= max_rate;
+	}
+
+	return true;
+}
+
 /*
  * Helper function to walk the array of sample rate triplets reported by
  * the device. The problem is that we need to parse whole array first to
@@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
 			    !s1810c_valid_sample_rate(fp, rate))
 				goto skip_rate;
 
+			/* Filter out invalid rates on Focusrite devices */
+			if (USB_ID_VENDOR(chip->usb_id) == 0x1235 &&
+			    !focusrite_valid_sample_rate(chip, fp, rate))
+				goto skip_rate;
+
 			if (fp->rate_table)
 				fp->rate_table[nr_rates] = rate;
 			if (!fp->rate_min || rate < fp->rate_min)
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index e7b9040a54e6..a88d7854513b 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1776,8 +1776,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer,
 {
 	struct snd_kcontrol *kctl;
 	struct usb_mixer_elem_info *cval;
+	const struct usbmix_name_map *map;
 
-	if (check_ignored_ctl(find_map(imap, term->id, 0)))
+	map = find_map(imap, term->id, 0);
+	if (check_ignored_ctl(map))
 		return;
 
 	cval = kzalloc(sizeof(*cval), GFP_KERNEL);
@@ -1809,8 +1811,12 @@ static void build_connector_control(struct usb_mixer_interface *mixer,
 		usb_mixer_elem_info_free(cval);
 		return;
 	}
-	get_connector_control_name(mixer, term, is_input, kctl->id.name,
-				   sizeof(kctl->id.name));
+
+	if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)))
+		strlcat(kctl->id.name, " Jack", sizeof(kctl->id.name));
+	else
+		get_connector_control_name(mixer, term, is_input, kctl->id.name,
+					   sizeof(kctl->id.name));
 	kctl->private_free = snd_usb_mixer_elem_free;
 	snd_usb_mixer_add_control(&cval->head, kctl);
 }
@@ -3111,6 +3117,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		if (map->id == state.chip->usb_id) {
 			state.map = map->map;
 			state.selector_map = map->selector_map;
+			mixer->connector_map = map->connector_map;
 			mixer->ignore_ctl_error |= map->ignore_ctl_error;
 			break;
 		}
@@ -3192,10 +3199,32 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
+static int delegate_notify(struct usb_mixer_interface *mixer, int unitid,
+			   u8 *control, u8 *channel)
+{
+	const struct usbmix_connector_map *map = mixer->connector_map;
+
+	if (!map)
+		return unitid;
+
+	for (; map->id; map++) {
+		if (map->id == unitid) {
+			if (control && map->control)
+				*control = map->control;
+			if (channel && map->channel)
+				*channel = map->channel;
+			return map->delegated_id;
+		}
+	}
+	return unitid;
+}
+
 void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
 {
 	struct usb_mixer_elem_list *list;
 
+	unitid = delegate_notify(mixer, unitid, NULL, NULL);
+
 	for_each_mixer_elem(list, mixer, unitid) {
 		struct usb_mixer_elem_info *info =
 			mixer_elem_list_to_info(list);
@@ -3265,6 +3294,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 		return;
 	}
 
+	unitid = delegate_notify(mixer, unitid, &control, &channel);
+
 	for_each_mixer_elem(list, mixer, unitid)
 		count++;
 
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 65d6d08c96f5..41ec9dc4139b 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -6,6 +6,13 @@
 
 struct media_mixer_ctl;
 
+struct usbmix_connector_map {
+	u8 id;
+	u8 delegated_id;
+	u8 control;
+	u8 channel;
+};
+
 struct usb_mixer_interface {
 	struct snd_usb_audio *chip;
 	struct usb_host_interface *hostif;
@@ -18,6 +25,9 @@ struct usb_mixer_interface {
 	/* the usb audio specification version this interface complies to */
 	int protocol;
 
+	/* optional connector delegation map */
+	const struct usbmix_connector_map *connector_map;
+
 	/* Sound Blaster remote control stuff */
 	const struct rc_config *rc_cfg;
 	u32 rc_code;
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index b4e77000f441..0260c750e156 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -27,6 +27,7 @@ struct usbmix_ctl_map {
 	u32 id;
 	const struct usbmix_name_map *map;
 	const struct usbmix_selector_map *selector_map;
+	const struct usbmix_connector_map *connector_map;
 	int ignore_ctl_error;
 };
 
@@ -369,6 +370,33 @@ static const struct usbmix_name_map asus_rog_map[] = {
 	{}
 };
 
+/* TRX40 mobos with Realtek ALC1220-VB */
+static const struct usbmix_name_map trx40_mobo_map[] = {
+	{ 18, NULL }, /* OT, IEC958 - broken response, disabled */
+	{ 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */
+	{ 16, "Speaker" },		/* OT */
+	{ 22, "Speaker Playback" },	/* FU */
+	{ 7, "Line" },			/* IT */
+	{ 19, "Line Capture" },		/* FU */
+	{ 17, "Front Headphone" },	/* OT */
+	{ 23, "Front Headphone Playback" },	/* FU */
+	{ 8, "Mic" },			/* IT */
+	{ 20, "Mic Capture" },		/* FU */
+	{ 9, "Front Mic" },		/* IT */
+	{ 21, "Front Mic Capture" },	/* FU */
+	{ 24, "IEC958 Playback" },	/* FU */
+	{}
+};
+
+static const struct usbmix_connector_map trx40_mobo_connector_map[] = {
+	{ 10, 16 },	/* (Back) Speaker */
+	{ 11, 17 },	/* Front Headphone */
+	{ 13, 7 },	/* Line */
+	{ 14, 8 },	/* Mic */
+	{ 15, 9 },	/* Front Mic */
+	{}
+};
+
 /*
  * Control map entries
  */
@@ -500,7 +528,8 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
 	},
 	{	/* Gigabyte TRX40 Aorus Pro WiFi */
 		.id = USB_ID(0x0414, 0xa002),
-		.map = asus_rog_map,
+		.map = trx40_mobo_map,
+		.connector_map = trx40_mobo_connector_map,
 	},
 	{	/* ASUS ROG Zenith II */
 		.id = USB_ID(0x0b05, 0x1916),
@@ -512,11 +541,13 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
 	},
 	{	/* MSI TRX40 Creator */
 		.id = USB_ID(0x0db0, 0x0d64),
-		.map = asus_rog_map,
+		.map = trx40_mobo_map,
+		.connector_map = trx40_mobo_connector_map,
 	},
 	{	/* MSI TRX40 */
 		.id = USB_ID(0x0db0, 0x543d),
-		.map = asus_rog_map,
+		.map = trx40_mobo_map,
+		.connector_map = trx40_mobo_connector_map,
 	},
 	{ 0 } /* terminator */
 };
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index f0302ad7f5a0..bdff8674942e 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1509,11 +1509,15 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol,
 
 	/* use known values for that card: interface#1 altsetting#1 */
 	iface = usb_ifnum_to_if(chip->dev, 1);
-	if (!iface || iface->num_altsetting < 2)
-		return -EINVAL;
+	if (!iface || iface->num_altsetting < 2) {
+		err = -EINVAL;
+		goto end;
+	}
 	alts = &iface->altsetting[1];
-	if (get_iface_desc(alts)->bNumEndpoints < 1)
-		return -EINVAL;
+	if (get_iface_desc(alts)->bNumEndpoints < 1) {
+		err = -EINVAL;
+		goto end;
+	}
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
 	err = snd_usb_ctl_msg(chip->dev,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e009d584e7d0..a1df4c5b4f8c 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2756,90 +2756,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.type = QUIRK_MIDI_NOVATION
 	}
 },
-{
-	/*
-	 * Focusrite Scarlett Solo 2nd generation
-	 * Reports that playback should use Synch: Synchronous
-	 * while still providing a feedback endpoint. Synchronous causes
-	 * snapping on some sample rates.
-	 * Force it to use Synch: Asynchronous.
-	 */
-	USB_DEVICE(0x1235, 0x8205),
-	.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 = 1,
-				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = & (const struct audioformat) {
-					.formats = SNDRV_PCM_FMTBIT_S32_LE,
-					.channels = 2,
-					.iface = 1,
-					.altsetting = 1,
-					.altset_idx = 1,
-					.attributes = 0,
-					.endpoint = 0x01,
-					.ep_attr = USB_ENDPOINT_XFER_ISOC |
-						   USB_ENDPOINT_SYNC_ASYNC,
-					.protocol = UAC_VERSION_2,
-					.rates = SNDRV_PCM_RATE_44100 |
-						 SNDRV_PCM_RATE_48000 |
-						 SNDRV_PCM_RATE_88200 |
-						 SNDRV_PCM_RATE_96000 |
-						 SNDRV_PCM_RATE_176400 |
-						 SNDRV_PCM_RATE_192000,
-					.rate_min = 44100,
-					.rate_max = 192000,
-					.nr_rates = 6,
-					.rate_table = (unsigned int[]) {
-						44100, 48000, 88200,
-						96000, 176400, 192000
-					},
-					.clock = 41
-				}
-			},
-			{
-				.ifnum = 2,
-				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = & (const struct audioformat) {
-					.formats = SNDRV_PCM_FMTBIT_S32_LE,
-					.channels = 2,
-					.iface = 2,
-					.altsetting = 1,
-					.altset_idx = 1,
-					.attributes = 0,
-					.endpoint = 0x82,
-					.ep_attr = USB_ENDPOINT_XFER_ISOC |
-						   USB_ENDPOINT_SYNC_ASYNC |
-						   USB_ENDPOINT_USAGE_IMPLICIT_FB,
-					.protocol = UAC_VERSION_2,
-					.rates = SNDRV_PCM_RATE_44100 |
-						 SNDRV_PCM_RATE_48000 |
-						 SNDRV_PCM_RATE_88200 |
-						 SNDRV_PCM_RATE_96000 |
-						 SNDRV_PCM_RATE_176400 |
-						 SNDRV_PCM_RATE_192000,
-					.rate_min = 44100,
-					.rate_max = 192000,
-					.nr_rates = 6,
-					.rate_table = (unsigned int[]) {
-						44100, 48000, 88200,
-						96000, 176400, 192000
-					},
-					.clock = 41
-				}
-			},
-			{
-				.ifnum = 3,
-				.type = QUIRK_IGNORE_INTERFACE
-			},
-			{
-				.ifnum = -1
-			}
-		}
-	}
-},
 
 /* Access Music devices */
 {
@@ -3635,4 +3551,18 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 	}
 },
 
+#define ALC1220_VB_DESKTOP(vend, prod) { \
+	USB_DEVICE(vend, prod),	\
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
+		.vendor_name = "Realtek", \
+		.product_name = "ALC1220-VB-DT", \
+		.profile_name = "Realtek-ALC1220-VB-Desktop", \
+		.ifnum = QUIRK_NO_INTERFACE \
+	} \
+}
+ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */
+ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */
+ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */
+#undef ALC1220_VB_DESKTOP
+
 #undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a8ece1701068..351ba214a9d3 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1806,6 +1806,20 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
 		 */
 		fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
 		break;
+	case USB_ID(0x1235, 0x8200):  /* Focusrite Scarlett 2i4 2nd gen */
+	case USB_ID(0x1235, 0x8202):  /* Focusrite Scarlett 2i2 2nd gen */
+	case USB_ID(0x1235, 0x8205):  /* Focusrite Scarlett Solo 2nd gen */
+		/*
+		 * Reports that playback should use Synch: Synchronous
+		 * while still providing a feedback endpoint.
+		 * Synchronous causes snapping on some sample rates.
+		 * Force it to use Synch: Asynchronous.
+		 */
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
+			fp->ep_attr |= USB_ENDPOINT_SYNC_ASYNC;
+		}
+		break;
 	}
 }
 
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 37d290fe9d43..ecaf41265dcd 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -681,6 +681,8 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate)
 			us->submitted =	2*NOOF_SETRATE_URBS;
 			for (i = 0; i < NOOF_SETRATE_URBS; ++i) {
 				struct urb *urb = us->urb[i];
+				if (!urb)
+					continue;
 				if (urb->status) {
 					if (!err)
 						err = -ENODEV;