Merge tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This became a slightly big pull request, as the accumulated ASoC fixes
  are included here. Some highlights:

   - Revert of ASoC DAI startup changes that caused regression on some
     x86 platforms

   - Regression fix in HD-audio power management and driver blacklist

   - A collection of ASoC DAPM and topology fixes

   - Continued USB-audio fixes and quirks

   - Lots of small device-specific fixes

   - Rockchip S/PDIF DT stuff update for validation issues"

* tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (51 commits)
  ALSA: hda: Always use jackpoll helper for jack update after resume
  ALSA: hda/realtek - Add new codec supported for ALC245
  ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
  ALSA: usb-audio: Add connector notifier delegation
  ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen
  ASoC: wm8960: Fix wrong clock after suspend & resume
  ALSA: usx2y: Fix potential NULL dereference
  ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2
  ASoC: wm89xx: Add missing dependency
  ASoC: dapm: fixup dapm kcontrol widget
  ASoC: rsnd: Fix "status check failed" spam for multi-SSI
  ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent
  ASoC: meson: gx-card: fix codec-to-codec link setup
  ASoC: meson: axg-card: fix codec-to-codec link setup
  ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
  ALSA: hda: Remove ASUS ROG Zenith from the blacklist
  ALSA: hda/realtek - Fix unexpected init_amp override
  ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
  ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
  ASoC: stm32: sai: fix sai probe
  ...

44 files changed, 712 insertions, 387 deletions

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..457a2c065485 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);
@@ -2092,7 +2079,6 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
  * 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),
 	{}
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/